alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ ping 192.168.10.122 connect: Network is unreachable alexandr.kurkin@ws-053:~$ ^C alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ alexandr.kurkin@ws-053:~$ sudo minicom [sudo] password for alexandr.kurkin: Welcome to minicom 2.5 OPTIONS: I18n Compiled on May 2 2011, 00:39:27. Port /dev/ttyS0 Press CTRL-A Z for help on special keys root@CPE:/etc# ttyS0: 1 input overrun(s) AT S7=45 S0=0 L1 /bin/ash: AT: not found root@CPE:/etc# root@CPE:/etc# root@CPE:/etc# root@CPE:/etc# root@CPE:/etc# ifconfig br-lan Link encap:Ethernet HWaddr 00:E1:75:FF:00:5C inet addr:192.168.1.1 Bcast:192.168.1.255 Mask:255.255.255.0 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:0 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:0 (0.0 B) TX bytes:0 (0.0 B) eth0 Link encap:Ethernet HWaddr 00:E1:75:FF:00:5C UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:0 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:532 RX bytes:0 (0.0 B) TX bytes:0 (0.0 B) Interrupt:11 eth1 Link encap:Ethernet HWaddr 00:E1:75:FF:00:5D inet addr:192.168.10.122 Bcast:192.168.10.255 Mask:255.255.255.0 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:223033 errors:0 dropped:0 overruns:0 frame:0 TX packets:4124 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:532 RX bytes:16251398 (15.4 MiB) TX bytes:1945050 (1.8 MiB) Interrupt:11 lo Link encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:291 errors:0 dropped:0 overruns:0 frame:0 TX packets:291 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:23830 (23.2 KiB) TX bytes:23830 (23.2 KiB) root@CPE:/etc# root@CPE:/etc# root@CPE:/etc# root@CPE:/etc# __ __ _ _ | \/ | __ _ _ ____ _____| | | | |\/| |/ _` | '__\ \ / / _ \ | | | | | | (_| | | \ V / __/ | | |_| |_|\__,_|_| \_/ \___|_|_| _ _ ____ _ | | | | | __ ) ___ ___ | |_ | | | |___| _ \ / _ \ / _ \| __| | |_| |___| |_) | (_) | (_) | |_ \___/ |____/ \___/ \___/ \__| ** MARVELL BOARD: PROMSVYAZ CPE LE U-Boot 1.1.4 (Oct 22 2012 - 12:13:22) Marvell version: 3.5.9 U-Boot code: 00600000 -> 0067FFF0 BSS: -> 006D1220 Soc: 88F6282 A1CPU running @ 1000Mhz L2 running @ 333Mhz SysClock = 400Mhz , TClock = 200Mhz DRAM (DDR2) CAS Latency = 5 tRP = 5 tRAS = 18 tRCD=6 DRAM CS[0] base 0x00000000 size 128MB DRAM CS[1] base 0x08000000 size 128MB DRAM Total size 256MB 16bit width Addresses 8M - 0M are saved for the U-Boot usage. Mem malloc Initialization (8M - 7M): Done NAND:512 MB Flash: 0 kB CPU : Marvell Feroceon (Rev 1) Streaming disabled Write allocate disabled Module 0 is RGMII Initializing driver for 16 bit SPI gpio expander Writing 0x0004 to SPI gpio expander USB 0: host mode Writing 0x0006 to SPI gpio expander Writing 0x000e to SPI gpio expander PEX 0: interface detected no Link. PEX 1: interface detected no Link. Writing 0x000f to SPI gpio expander Net: egiga0 [PRIME], egiga1 Initial switch. Hit any key to stop autoboot: 0 NAND read: device 0 offset 0x100000, size 0x400000 Reading data from 0x4ff800 -- 100% complete. 4194304 bytes read: OK ## Booting image at 06400000 ... Image Name: Linux-2.6.31.8 Created: 2012-10-22 10:11:19 UTC Image Type: ARM Linux Kernel Image (uncompressed) Data Size: 2032800 Bytes = 1.9 MB Load Address: 00008000 Entry Point: 00008000 Verifying Checksum ... OK OK Starting kernel ... Uncompressing Linux.................................................................................................................. Linux version 2.6.31.8 (jenkins@sp-build02-lo1) (gcc version 4.5.4 20110808 (prerelease) (Linaro GCC 4.5-2011.08) ) #1 Mon Oct 22 132 CPU: Feroceon 88FR131 [56251311] revision 1 (ARMv5TE), cr=00053977 CPU: VIVT data cache, VIVT instruction cache Machine: Feroceon-KW Using UBoot passing parameters structure Memory policy: ECC disabled, Data cache writeback Built 1 zonelists in Zone order, mobility grouping off. Total pages: 65024 Kernel command line: console=ttyS0,115200 init=/etc/preinit root=/dev/mtdblock2 ro rootfstype=jffs2 mv_net_config=(00:E1:75:FF:00:5C0 PID hash table entries: 1024 (order: 10, 4096 bytes) Dentry cache hash table entries: 32768 (order: 5, 131072 bytes) Inode-cache hash table entries: 16384 (order: 4, 65536 bytes) Memory: 128MB 128MB = 256MB total Memory: 249344KB available (3820K code, 274K data, 124K init, 0K highmem) Hierarchical RCU implementation. NR_IRQS:128 Console: colour dummy device 80x30 Calibrating delay loop... 992.87 BogoMIPS (lpj=4964352) Mount-cache hash table entries: 512 CPU: Testing write buffer coherency: ok NET: Registered protocol family 16 Feroceon L2: Enabling L2 Feroceon L2: Cache support initialised. CPU Interface ------------- SDRAM_CS0 ....base 00000000, size 128MB SDRAM_CS1 ....base 08000000, size 128MB SDRAM_CS2 ....disable SDRAM_CS3 ....disable PEX0_MEM ....base e0000000, size 128MB PEX0_IO ....base f2000000, size 1MB PEX1_MEM ....base e8000000, size 128MB PEX1_IO ....base f2100000, size 1MB INTER_REGS ....base f1000000, size 1MB NFLASH_CS ....base fa000000, size 2MB SPI_CS ....base f4000000, size 16MB BOOT_ROM_CS ....no such DEV_BOOTCS ....no such CRYPT_ENG ....base f0000000, size 2MB Marvell Development Board (LSP Version KW_LSP_5.1.3_patch32)-- PROMSVYAZ CPE Soc: 88F6282 A1 LE Detected Tclk 200000000 and SysClk 400000000 No Hub pins get! MV Buttons Device Load Marvell USB EHCI Host controller #0: 88040440 PEX0 interface detected no Link. PCI: bus0: Fast back to back transfers enabled mvPexLocalBusNumSet: ERR. Invalid PEX interface 1 bio: create slab at 0 SCSI subsystem initialized usbcore: registered new interface driver usbfs usbcore: registered new interface driver hub usbcore: registered new device driver usb cfg80211: Using static regulatory domain info cfg80211: Regulatory domain: US (start_freq - end_freq @ bandwidth), (max_antenna_gain, max_eirp) (2402000 KHz - 2472000 KHz @ 40000 KHz), (600 mBi, 2700 mBm) (5170000 KHz - 5190000 KHz @ 40000 KHz), (600 mBi, 2300 mBm) (5190000 KHz - 5210000 KHz @ 40000 KHz), (600 mBi, 2300 mBm) (5210000 KHz - 5230000 KHz @ 40000 KHz), (600 mBi, 2300 mBm) (5230000 KHz - 5330000 KHz @ 40000 KHz), (600 mBi, 2300 mBm) (5735000 KHz - 5835000 KHz @ 40000 KHz), (600 mBi, 3000 mBm) cfg80211: Calling CRDA for country: US NET: Registered protocol family 2 IP route cache hash table entries: 2048 (order: 1, 8192 bytes) TCP established hash table entries: 8192 (order: 4, 65536 bytes) TCP bind hash table entries: 8192 (order: 3, 32768 bytes) TCP: Hash tables configured (established 8192 bind 8192) TCP reno registered NET: Registered protocol family 1 RTC has been updated!!! rtc mv_rtc: rtc core: registered kw-rtc as rtc0 RTC registered Switched to NOHz mode on CPU #0 Kirkwood hwmon thermal sensor initialized. XOR registered 4 channels XOR 2nd invalidate WA enabled cesadev_init(8000e358) mvCesaInit: sessions=640, queue=64, pSram=f0000000 MV Buttons Driver Load Registering mini_fo version $Id$ JFFS2 version 2.2. (NAND) (SUMMARY) �© 2001-2006 Red Hat, Inc. msgmni has been set to 487 alg: No test for cipher_null (cipher_null-generic) alg: No test for ecb(cipher_null) (ecb-cipher_null) alg: No test for digest_null (digest_null-generic) alg: No test for compress_null (compress_null-generic) alg: No test for stdrng (krng) alg: No test for hmac(digest_null) (hmac(digest_null-generic)) Block layer SCSI generic (bsg) driver version 0.4 loaded (major 253) io scheduler noop registered io scheduler anticipatory registered (default) Serial: 8250/16550 driver, 4 ports, IRQ sharing disabled serial8250.0: ttyS0 at MMIO 0xf1012000 (irq = 33) is a 16550A console [ttyS0] enabled serial8250.1: ttyS1 at MMIO 0xf1012100 (irq = 34) is a 16550A brd: module loaded Loading Marvell Ethernet Driver: o Cached descriptors in DRAM o DRAM SW cache-coherency o 1 Giga ports supported o Multi RX Queue support - 8 RX queues o Multi TX Queue support - 8 TX Queues o TCP segmentation offload (TSO) supported o UDP fragmentation offload (UFO) supported o Large Receive offload (LRO) supported o Receive checksum offload supported o Transmit checksum offload supported o Network Fast Processing (Routing) supported - (Disabled) o Network Fast Processing (NAT) supported o Driver ERROR statistics enabled o Proc tool API enabled o SKB Reuse supported - (Enabled) o SKB Recycle supported - (Enabled) o Gateway support enabled o Using Marvell Header Mode o L2 IGMP support o Rx descripors: q0=128 q1=128 q2=128 q3=128 q4=128 q5=128 q6=128 q7=128 o Tx descripors: q0=532 q1=532 q2=532 q3=532 q4=532 q5=532 q6=532 q7=532 o Loading network interface(s): o Loading Switch QuarterDeck driver Initial MV88E6176 switch success. qdLoadDriver OK in mv_switch_load o Loading Gateway interface(s): o Using command line network interface configuration o MTU set to 1500. o mac_addr 00:e1:75:ff:00:5c, VID 0x100, port list: port-0 port-1 port-2 port-3 o mac_addr 00:e1:75:ff:00:5d, VID 0x200, port list: port-4 o register under mv88fx_eth platform eth0: mixed HW and IP checksum settings. o eth0, ifindex = 2, GbE port = 0 o register under mv88fx_eth platform eth1: mixed HW and IP checksum settings. o eth1, ifindex = 3, GbE port = 0 init switch layer... mvFpRuleDb (8955c000): 2048 entries, 8192 bytes Enable NFP Using Hamming 1-bit ECC for NAND device NAND device: Manufacturer ID: 0xec, Chip ID: 0xdc (Samsung NAND 512MiB 3,3V 8-bit) Scanning device for bad blocks 6 cmdlinepart partitions found on MTD device nand_mtd Using built in partition definition Creating 6 MTD partitions on "nand_mtd": 0x000000000000-0x000000100000 : "uboot" 0x000000100000-0x000000500000 : "kernel" 0x000000500000-0x000002500000 : "rootfs" 0x000002500000-0x000002900000 : "kernel2" 0x000002900000-0x000004900000 : "rootfs2" 0x000004900000-0x000006900000 : "rootfs_data" tdm_match_controller_to_boardinfo tdm->bus_num == bi->bus_num tdm_new_device 1 tdm_new_device 2 tdm_new_device 3 tdm_new_device 4 tdm0.2 tdm_match_controller_to_boardinfo tdm->bus_num == bi->bus_num tdm_new_device 1 tdm_new_device 2 tdm_new_device 3 tdm_new_device 4 tdm0.4 ehci_hcd: USB 2.0 'Enhanced' Host Controller (EHCI) Driver ehci_marvell ehci_marvell.70059: Marvell Orion EHCI ehci_marvell ehci_marvell.70059: new USB bus registered, assigned bus number 1 ehci_marvell ehci_marvell.70059: irq 19, io base 0xf1050100 ehci_marvell ehci_marvell.70059: USB 2.0 started, EHCI 1.00 usb usb1: New USB device found, idVendor=1d6b, idProduct=0002 usb usb1: New USB device strings: Mfr=3, Product=2, SerialNumber=1 usb usb1: Product: Marvell Orion EHCI usb usb1: Manufacturer: Linux 2.6.31.8 ehci_hcd usb usb1: SerialNumber: ehci_marvell.70059 usb usb1: configuration #1 chosen from 1 choice hub 1-0:1.0: USB hub found hub 1-0:1.0: 1 port detected ohci_hcd: USB 1.1 'Open' Host Controller (OHCI) Driver uhci_hcd: USB Universal Host Controller Interface driver Initializing USB Mass Storage driver... usbcore: registered new interface driver usb-storage USB Mass Storage support registered. mice: PS/2 mouse device common for all mice i2c /dev entries driver usb251x-hub 0-002c: chip found, driver version 0.1 Orion Watchdog Timer: Initial timeout 21 sec, nowayout Registered led device: pscpe:green:wps Registered led device: pscpe:red:wps Registered led device: pscpe:red:wifi Registered led device: pscpe:red:register Registered led device: pscpe:green:register Registered led device: pscpe:red:alarm Registered led device: pscpe:green:alarm Registered led device: pscpe:green:phone1 Registered led device: pscpe:green:phone2 Registered led device: pscpe:red:phone spi_dev = spi0.0 si3226x si3226x.0: run Initialization slic driver si3226x si3226x.0: GPIO requested si3226x si3226x.0: slic->int_gpio = 43 si3226x si3226x.0: slic->irq = 107 si3226x si3226x.0: run slic initialization usb 1-1: new high speed USB device using ehci_marvell and address 2 si3226x si3226x.0: line: 0, upload patch - ok usb 1-1: New USB device found, idVendor=0424, idProduct=2512 usb 1-1: New USB device strings: Mfr=0, Product=0, SerialNumber=0 usb 1-1: configuration #1 chosen from 1 choice si3226x si3226x.0: line: 1, upload patch - ok hub 1-1:1.0: USB hub found hub 1-1:1.0: 2 ports detected Port 4: Link-up, Full-duplex, Speed-100Mbps. si3226x si3226x.0: settings loaded si3226x tdm0.2: run init line si3226x tdm0.2: init line done si3226x tdm0.4: run init line si3226x tdm0.4: init line done si3226x si3226x.0: request irq si3226x si3226x.0: success initialization slic driver nf_conntrack version 0.5.0 (4096 buckets, 16384 max) CONFIG_NF_CT_ACCT is deprecated and will be removed soon. Please use nf_conntrack.acct=1 kernel parameter, acct=1 nf_conntrack module option or sysctl net.netfilter.nf_conntrack_acct=1 to enable it. mvFpNatDb (895a8000): 8192 entries, 32768 bytes TCP cubic registered NET: Registered protocol family 17 NFP (fdb) init 256 entries, 1024 bytes Bridge firewalling registered 802.1Q VLAN Support v1.8 Ben Greear All bugs added by David S. Miller rtc mv_rtc: setting system clock to 2000-01-01 00:00:00 UTC (946684800) JFFS2 notice: (1) jffs2_build_xattr_subsystem: complete building xattr subsystem, 0 of xdatum (0 unchecked, 0 orphan) and 0 of xref . VFS: Mounted root (jffs2 filesystem) readonly on device 31:2. Freeing init memory: 124K Please be patient, while OpenWrt loads ... - preinit - Press the [f] key and hit [enter] to enter failsafe mode - regular preinit - Empty flash at 0x00ba1a84 ends at 0x00ba2000 Empty flash at 0x00baa770 ends at 0x00baa800 JFFS2 notice: (853) jffs2_build_xattr_subsystem: complete building xattr subsystem, 0 of xdatum (0 unchecked, 0 orphan) and 0 of xre. switching to jffs2 mini_fo: using base directory: / mini_fo: using storage directory: /overlay - init - Please press Enter to activate this console. PPP generic driver version 2.4.2 tun: Universal TUN/TAP device driver, 1.6 tun: (C) 1999-2004 Max Krasnyansky PPP MPPE Compression module registered ip_tables: (C) 2000-2006 Netfilter Core Team NET: Registered protocol family 24 NFP (pppoe) init 128 entries, 16384 bytes PPTP driver version 0.8.5 xt_time: kernel timezone is -0000 nf_conntrack_rtsp v0.6.21 loading nf_nat_rtsp v0.6.21 loading Netfilter messages via NETLINK v0.30. ctnetlink v0.93: registering with nfnetlink. usbcore: registered new interface driver usbserial USB Serial support registered for generic usbcore: registered new interface driver usbserial_generic usbserial: USB Serial Driver core USB Serial support registered for GSM modem (1-port) usbcore: registered new interface driver option option: v0.7.2:USB Driver for GSM modems usbcore: registered new interface driver hiddev usbcore: registered new interface driver usbhid usbhid: v2.6:USB HID core driver mv_gateway: starting eth0 device eth0 entered promiscuous mode br-lan: port 1(eth0) entering forwarding state mv_gateway: eth0 change mac address to 00:e1:75:ff:00:5c mv_gateway: starting eth1 switch register access: type=2, port=0, reg=10 - SUCCESS, val=0x0000 switch register access: type=2, port=1, reg=10 - SUCCESS, val=0x0000 switch register access: type=2, port=2, reg=10 - SUCCESS, val=0x0000 switch register access: type=2, port=3, reg=10 - SUCCESS, val=0x0000 switch register access: type=2, port=4, reg=10 - SUCCESS, val=0x0000 switch register access: type=2, port=5, reg=10 - SUCCESS, val=0x0138 switch register access: type=3, port=0, reg=16 - SUCCESS, val=0x0c80 switch register access: type=3, port=0, reg=17 - SUCCESS, val=0x0000 switch register access: type=3, port=0, reg=18 - SUCCESS, val=0x0000 switch register access: type=3, port=0, reg=19 - SUCCESS, val=0x0000 switch register access: type=3, port=0, reg=20 - SUCCESS, val=0x0000 switch register access: type=3, port=0, reg=21 - SUCCESS, val=0x0000 switch register access: type=3, port=0, reg=22 - SUCCESS, val=0x0000 switch register access: type=3, port=0, reg=23 - SUCCESS, val=0x0000 switch register access: type=3, port=0, reg=24 - SUCCESS, val=0x0000 slic_enable_echo 0 slic_enable_echo 1 JFFS2 notice: (854) check_node_data: wrong data CRC in data node at 0x00ba9f30: read 0xd4bf6c29, calculated 0xa08c643e. BusyBox v1.19.3 (2012-10-22 11:57:12 FET) built-in shell (ash) Enter 'help' for a list of built-in commands. ____ | _ \ _ __ ___ _ __ ___ _____ ___ _ __ _ ____ | |_) | '__/ _ \| '_ ` _ \/ __\ \ / / | | |/ _` |_ / | __/| | | (_) | | | | | \__ \\ V /| |_| | (_| |/ / |_| |_| \___/|_| |_| |_|___/ \_/ \__, |\__,_/___| |___/ (svn r1326) ----------------------------------------------------- root@CPE:/# root@CPE:/# root@CPE:/# root@CPE:/# ifconfig br-lan Link encap:Ethernet HWaddr 00:E1:75:FF:00:5C inet addr:192.168.1.1 Bcast:192.168.1.255 Mask:255.255.255.0 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:0 errors:0 dropped:0 overruns:0 carrier:0 ; Set per parking lot. ;transferdigittimeout => 3 ; Number of seconds to wait between digits when transferring a call ; (default is 3 seconds) ;xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr ; to indicate a failed transfer ;pickupexten = *8 ; Configure the pickup extension. (default is *8) [directories] astetcdir => /etc/asterisk astmoddir => /usr/lib/asterisk/modules astvarlibdir => /usr/lib/asterisk astdbdir => /usr/lib/asterisk ; '/'. ; ; Special level name "*" means all levels, even dynamic levels registered ; by modules after the logger has been initialized (this means that loading ; and unloading modules that create/remove dynamic logger levels will result ; in these levels being included on filenames that have a level name of "*", ; without any need to perform a 'logger reload' or similar operation). Note ; that there is no value in specifying both "*" and specific level names for ; a filename; the "*" level means all levels, and the remaining level names ; will be ignored. ; ; We highly recommend that you DO NOT turn on debug mode if you are simply ; running a production system. Debug mode turns on a LOT of extra messages, ; most of which you are unlikely to understand without an understanding of ; the underlying code. Do NOT report debug messages as code issues, unless ; you have a specific issue that you are attempting to debug. They are ; messages for just that -- debugging -- and do not rise to the level of ; something that merit your attention as an Asterisk administrator. Debug ; messages are also very verbose and can and do fill up logfiles quickly; ; this is another reason not to have debug mode on a production system unless ; you are in the process of debugging a specific issue. ; debug => debug ;console => notice,warning,error,dtmf console => notice,warning,error,debug messages => notice,warning,error full => notice,warning,error,debug,verbose,dtmf,fax ;syslog keyword : This special keyword logs to syslog facility ; ;syslog.local0 => notice,warning,error ; ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ ~ root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# /etc/init.d/asterisk restart root@CPE:/etc/asterisk# slic_enable_echo 0 slic_enable_echo 1 root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# ifconfig br-lan Link encap:Ethernet HWaddr 00:E1:75:FF:00:5C inet addr:192.168.1.1 Bcast:192.168.1.255 Mask:255.255.255.0 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:0 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:0 (0.0 B) TX bytes:0 (0.0 B) eth0 Link encap:Ethernet HWaddr 00:E1:75:FF:00:5C UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:0 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:532 RX bytes:0 (0.0 B) TX bytes:0 (0.0 B) Interrupt:11 eth1 Link encap:Ethernet HWaddr 00:E1:75:FF:00:5D inet addr:192.168.10.192 Bcast:192.168.10.255 Mask:255.255.255.0 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:183655 errors:0 dropped:0 overruns:0 frame:0 TX packets:1661 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:532 RX bytes:15895056 (15.1 MiB) TX bytes:166859 (162.9 KiB) Interrupt:11 lo Link encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:64 errors:0 dropped:0 overruns:0 frame:0 TX packets:64 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:4928 (4.8 KiB) TX bytes:4928 (4.8 KiB) root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# cd /var/lib/asterisk/ root@CPE:/tmp/lib/asterisk# ls root@CPE:/tmp/lib/asterisk# ls root@CPE:/tmp/lib/asterisk# mkdir sounds root@CPE:/tmp/lib/asterisk# ls sounds root@CPE:/tmp/lib/asterisk# cd sounds/ root@CPE:/tmp/lib/asterisk/sounds# cp /usr/lib/asterisk/sounds/*.* ./ root@CPE:/tmp/lib/asterisk/sounds# ls CHANGES-asterisk-core-en-1.4.22 spy-misdn.gsm CREDITS-asterisk-core-en-1.4.22 spy-mobile.gsm LICENSE-asterisk-core-en-1.4.22 spy-nbs.gsm agent-alreadyon.gsm spy-sip.gsm agent-incorrect.gsm spy-skinny.gsm agent-loggedoff.gsm spy-unistim.gsm agent-loginok.gsm spy-usbradio.gsm agent-newlocation.gsm spy-zap.gsm agent-pass.gsm ss-noservice.gsm agent-user.gsm transfer.gsm ascending-2tone.gsm tt-allbusy.gsm auth-incorrect.gsm tt-monkeys.gsm auth-thankyou.gsm tt-monkeysintro.gsm beep.gsm tt-somethingwrong.gsm beeperr.gsm tt-weasels.gsm confbridge-begin-glorious-a.gsm vm-Cust1.gsm confbridge-begin-glorious-b.gsm vm-Cust2.gsm confbridge-begin-glorious-c.gsm vm-Cust3.gsm confbridge-begin-leader.gsm vm-Cust4.gsm confbridge-conf-begin.gsm vm-Cust5.gsm confbridge-conf-end.gsm vm-Family.gsm confbridge-dec-list-vol-in.gsm vm-Friends.gsm confbridge-dec-list-vol-out.gsm vm-INBOX.gsm confbridge-dec-talk-vol-in.gsm vm-Old.gsm confbridge-dec-talk-vol-out.gsm vm-Urgent.gsm confbridge-has-joined.gsm vm-Work.gsm confbridge-has-left.gsm vm-advopts.gsm confbridge-inc-list-vol-in.gsm vm-and.gsm confbridge-inc-list-vol-out.gsm vm-calldiffnum.gsm confbridge-inc-talk-vol-in.gsm vm-changeto.gsm confbridge-inc-talk-vol-out.gsm vm-delete.gsm confbridge-invalid.gsm vm-deleted.gsm confbridge-join.gsm vm-dialout.gsm confbridge-leave-in.gsm vm-duration.gsm confbridge-leave-out.gsm vm-enter-num-to-call.gsm confbridge-leave.gsm vm-extension.gsm confbridge-lock-extended.gsm vm-first.gsm confbridge-lock-in.gsm vm-for.gsm confbridge-lock-no-join.gsm vm-forward-multiple.gsm confbridge-lock-out.gsm vm-forward.gsm confbridge-locked.gsm vm-forwardoptions.gsm confbridge-menu-exit-in.gsm vm-from-extension.gsm confbridge-menu-exit-out.gsm vm-from-phonenumber.gsm confbridge-mute-extended.gsm vm-from.gsm confbridge-mute-in.gsm vm-goodbye.gsm confbridge-mute-out.gsm vm-helpexit.gsm confbridge-muted.gsm vm-incorrect-mailbox.gsm confbridge-only-one.gsm vm-incorrect.gsm confbridge-only-participant.gsm vm-instructions.gsm confbridge-participants.gsm vm-intro.gsm confbridge-pin-bad.gsm vm-invalid-password.gsm confbridge-pin.gsm vm-invalidpassword.gsm confbridge-remove-last-in.gsm vm-isonphone.gsm confbridge-remove-last-out.gsm vm-isunavail.gsm confbridge-removed.gsm vm-last.gsm confbridge-rest-list-vol-in.gsm vm-leavemsg.gsm confbridge-rest-list-vol-out.gsm vm-login.gsm confbridge-rest-talk-vol-in.gsm vm-mailboxfull.gsm confbridge-rest-talk-vol-out.gsm vm-marked-nonurgent.gsm confbridge-there-are.gsm vm-marked-urgent.gsm confbridge-unlocked.gsm vm-message.gsm confbridge-unmuted.gsm vm-messages.gsm core-sounds-en.txt vm-minutes.gsm demo-abouttotry.gsm vm-mismatch.gsm demo-congrats.gsm vm-msginstruct.gsm demo-echodone.gsm vm-msgsaved.gsm demo-echotest.gsm vm-newpassword.gsm demo-enterkeywords.gsm vm-newuser.gsm demo-instruct.gsm vm-next.gsm demo-moreinfo.gsm vm-no.gsm demo-nogo.gsm vm-nobodyavail.gsm demo-nomatch.gsm vm-nobox.gsm demo-thanks.gsm vm-nomore.gsm descending-2tone.gsm vm-nonumber.gsm dir-first.gsm vm-num-i-have.gsm dir-firstlast.gsm vm-onefor-full.gsm dir-instr.gsm vm-onefor.gsm dir-intro-fn.gsm vm-options.gsm dir-intro.gsm vm-opts-full.gsm dir-last.gsm vm-opts.gsm dir-multi1.gsm vm-passchanged.gsm dir-multi2.gsm vm-password.gsm dir-multi3.gsm vm-pls-try-again.gsm dir-multi9.gsm vm-press.gsm dir-nomatch.gsm vm-prev.gsm dir-nomore.gsm vm-reachoper.gsm dir-pls-enter.gsm vm-rec-busy.gsm dir-usingkeypad.gsm vm-rec-name.gsm dir-welcome.gsm vm-rec-temp.gsm hello-world.gsm vm-rec-unv.gsm hours.gsm vm-received.gsm invalid.gsm vm-record-prepend.gsm minutes.gsm vm-reenterpassword.gsm pbx-invalid.gsm vm-repeat.gsm pbx-invalidpark.gsm vm-review-nonurgent.gsm pbx-parkingfailed.gsm vm-review-urgent.gsm pbx-transfer.gsm vm-review.gsm priv-callee-options.gsm vm-saved.gsm priv-callpending.gsm vm-savedto.gsm priv-introsaved.gsm vm-savefolder.gsm priv-recordintro.gsm vm-savemessage.gsm privacy-incorrect.gsm vm-saveoper.gsm privacy-prompt.gsm vm-sorry.gsm privacy-thankyou.gsm vm-star-cancel.gsm privacy-unident.gsm vm-starmain.gsm queue-callswaiting.gsm vm-tempgreetactive.gsm queue-holdtime.gsm vm-tempgreeting.gsm queue-less-than.gsm vm-tempgreeting2.gsm queue-minute.gsm vm-tempremoved.gsm queue-minutes.gsm vm-then-pound.gsm queue-periodic-announce.gsm vm-theperson.gsm queue-quantity1.gsm vm-tmpexists.gsm queue-quantity2.gsm vm-tocallback.gsm queue-reporthold.gsm vm-tocallnum.gsm queue-seconds.gsm vm-tocancel.gsm queue-thankyou.gsm vm-tocancelmsg.gsm queue-thereare.gsm vm-toenternumber.gsm queue-youarenext.gsm vm-toforward.gsm screen-callee-options.gsm vm-tohearenv.gsm seconds.gsm vm-tomakecall.gsm spy-agent.gsm vm-tooshort.gsm spy-console.gsm vm-toreply.gsm spy-dahdi.gsm vm-torerecord.gsm spy-h323.gsm vm-undelete.gsm spy-iax.gsm vm-undeleted.gsm spy-iax2.gsm vm-unknown-caller.gsm spy-jingle.gsm vm-whichbox.gsm spy-local.gsm vm-youhave.gsm spy-mgcp.gsm root@CPE:/tmp/lib/asterisk/sounds# root@CPE:/tmp/lib/asterisk/sounds# root@CPE:/tmp/lib/asterisk/sounds# root@CPE:/tmp/lib/asterisk/sounds# root@CPE:/tmp/lib/asterisk/sounds# root@CPE:/tmp/lib/asterisk/sounds# root@CPE:/tmp/lib/asterisk/sounds# root@CPE:/tmp/lib/asterisk/sounds# root@CPE:/tmp/lib/asterisk/sounds# ls CHANGES-asterisk-core-en-1.4.22 spy-misdn.gsm CREDITS-asterisk-core-en-1.4.22 spy-mobile.gsm LICENSE-asterisk-core-en-1.4.22 spy-nbs.gsm agent-alreadyon.gsm spy-sip.gsm agent-incorrect.gsm spy-skinny.gsm agent-loggedoff.gsm spy-unistim.gsm agent-loginok.gsm spy-usbradio.gsm agent-newlocation.gsm spy-zap.gsm agent-pass.gsm ss-noservice.gsm agent-user.gsm transfer.gsm ascending-2tone.gsm tt-allbusy.gsm auth-incorrect.gsm tt-monkeys.gsm auth-thankyou.gsm tt-monkeysintro.gsm beep.gsm tt-somethingwrong.gsm beeperr.gsm tt-weasels.gsm confbridge-begin-glorious-a.gsm vm-Cust1.gsm confbridge-begin-glorious-b.gsm vm-Cust2.gsm confbridge-begin-glorious-c.gsm vm-Cust3.gsm confbridge-begin-leader.gsm vm-Cust4.gsm confbridge-conf-begin.gsm vm-Cust5.gsm confbridge-conf-end.gsm vm-Family.gsm confbridge-dec-list-vol-in.gsm vm-Friends.gsm confbridge-dec-list-vol-out.gsm vm-INBOX.gsm confbridge-dec-talk-vol-in.gsm vm-Old.gsm confbridge-dec-talk-vol-out.gsm vm-Urgent.gsm confbridge-has-joined.gsm vm-Work.gsm confbridge-has-left.gsm vm-advopts.gsm confbridge-inc-list-vol-in.gsm vm-and.gsm confbridge-inc-list-vol-out.gsm vm-calldiffnum.gsm confbridge-inc-talk-vol-in.gsm vm-changeto.gsm confbridge-inc-talk-vol-out.gsm vm-delete.gsm confbridge-invalid.gsm vm-deleted.gsm confbridge-join.gsm vm-dialout.gsm confbridge-leave-in.gsm vm-duration.gsm confbridge-leave-out.gsm vm-enter-num-to-call.gsm confbridge-leave.gsm vm-extension.gsm confbridge-lock-extended.gsm vm-first.gsm confbridge-lock-in.gsm vm-for.gsm confbridge-lock-no-join.gsm vm-forward-multiple.gsm confbridge-lock-out.gsm vm-forward.gsm confbridge-locked.gsm vm-forwardoptions.gsm confbridge-menu-exit-in.gsm vm-from-extension.gsm confbridge-menu-exit-out.gsm vm-from-phonenumber.gsm confbridge-mute-extended.gsm vm-from.gsm confbridge-mute-in.gsm vm-goodbye.gsm confbridge-mute-out.gsm vm-helpexit.gsm confbridge-muted.gsm vm-incorrect-mailbox.gsm confbridge-only-one.gsm vm-incorrect.gsm confbridge-only-participant.gsm vm-instructions.gsm confbridge-participants.gsm vm-intro.gsm confbridge-pin-bad.gsm vm-invalid-password.gsm confbridge-pin.gsm vm-invalidpassword.gsm confbridge-remove-last-in.gsm vm-isonphone.gsm confbridge-remove-last-out.gsm vm-isunavail.gsm confbridge-removed.gsm vm-last.gsm confbridge-rest-list-vol-in.gsm vm-leavemsg.gsm confbridge-rest-list-vol-out.gsm vm-login.gsm confbridge-rest-talk-vol-in.gsm vm-mailboxfull.gsm confbridge-rest-talk-vol-out.gsm vm-marked-nonurgent.gsm confbridge-there-are.gsm vm-marked-urgent.gsm confbridge-unlocked.gsm vm-message.gsm confbridge-unmuted.gsm vm-messages.gsm core-sounds-en.txt vm-minutes.gsm demo-abouttotry.gsm vm-mismatch.gsm demo-congrats.gsm vm-msginstruct.gsm demo-echodone.gsm vm-msgsaved.gsm demo-echotest.gsm vm-newpassword.gsm demo-enterkeywords.gsm vm-newuser.gsm demo-instruct.gsm vm-next.gsm demo-moreinfo.gsm vm-no.gsm demo-nogo.gsm vm-nobodyavail.gsm demo-nomatch.gsm vm-nobox.gsm demo-thanks.gsm vm-nomore.gsm descending-2tone.gsm vm-nonumber.gsm dir-first.gsm vm-num-i-have.gsm dir-firstlast.gsm vm-onefor-full.gsm dir-instr.gsm vm-onefor.gsm dir-intro-fn.gsm vm-options.gsm dir-intro.gsm vm-opts-full.gsm dir-last.gsm vm-opts.gsm dir-multi1.gsm vm-passchanged.gsm dir-multi2.gsm vm-password.gsm dir-multi3.gsm vm-pls-try-again.gsm dir-multi9.gsm vm-press.gsm dir-nomatch.gsm vm-prev.gsm dir-nomore.gsm vm-reachoper.gsm dir-pls-enter.gsm vm-rec-busy.gsm dir-usingkeypad.gsm vm-rec-name.gsm dir-welcome.gsm vm-rec-temp.gsm hello-world.gsm vm-rec-unv.gsm hours.gsm vm-received.gsm invalid.gsm vm-record-prepend.gsm minutes.gsm vm-reenterpassword.gsm pbx-invalid.gsm vm-repeat.gsm pbx-invalidpark.gsm vm-review-nonurgent.gsm pbx-parkingfailed.gsm vm-review-urgent.gsm pbx-transfer.gsm vm-review.gsm priv-callee-options.gsm vm-saved.gsm priv-callpending.gsm vm-savedto.gsm priv-introsaved.gsm vm-savefolder.gsm priv-recordintro.gsm vm-savemessage.gsm privacy-incorrect.gsm vm-saveoper.gsm privacy-prompt.gsm vm-sorry.gsm privacy-thankyou.gsm vm-star-cancel.gsm privacy-unident.gsm vm-starmain.gsm queue-callswaiting.gsm vm-tempgreetactive.gsm queue-holdtime.gsm vm-tempgreeting.gsm queue-less-than.gsm vm-tempgreeting2.gsm queue-minute.gsm vm-tempremoved.gsm queue-minutes.gsm vm-then-pound.gsm queue-periodic-announce.gsm vm-theperson.gsm queue-quantity1.gsm vm-tmpexists.gsm queue-quantity2.gsm vm-tocallback.gsm queue-reporthold.gsm vm-tocallnum.gsm queue-seconds.gsm vm-tocancel.gsm queue-thankyou.gsm vm-tocancelmsg.gsm queue-thereare.gsm vm-toenternumber.gsm queue-youarenext.gsm vm-toforward.gsm screen-callee-options.gsm vm-tohearenv.gsm seconds.gsm vm-tomakecall.gsm spy-agent.gsm vm-tooshort.gsm spy-console.gsm vm-toreply.gsm spy-dahdi.gsm vm-torerecord.gsm spy-h323.gsm vm-undelete.gsm spy-iax.gsm vm-undeleted.gsm spy-iax2.gsm vm-unknown-caller.gsm spy-jingle.gsm vm-whichbox.gsm spy-local.gsm vm-youhave.gsm spy-mgcp.gsm root@CPE:/tmp/lib/asterisk/sounds# root@CPE:/tmp/lib/asterisk/sounds# root@CPE:/tmp/lib/asterisk/sounds# root@CPE:/tmp/lib/asterisk/sounds# root@CPE:/tmp/lib/asterisk/sounds# root@CPE:/tmp/lib/asterisk/sounds# cd .. root@CPE:/tmp/lib/asterisk# ls sounds root@CPE:/tmp/lib/asterisk# cd /etc root@CPE:/etc# as /bin/ash: as: not found root@CPE:/etc# cd /etc/asterisk/ root@CPE:/etc/asterisk# ls asterisk.conf manager.conf cdr.conf manager.d cdr_custom.conf modules.conf codecs.conf musiconhold.conf extensions.conf res_fax.conf extensions_audiorecord.conf rtp.conf extensions_fxs.conf si3226x.conf extensions_hotlines.conf si3226x_blacklist.conf extensions_peers.conf si3226x_users.conf extensions_users.conf sip.conf extensions_warmlines.conf sip_registrations.conf features.conf sip_users.conf indications.conf udptl.conf logger.conf users.conf root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# ps PID USER VSZ STAT COMMAND 1 root 1128 S init 2 root 0 SW< [kthreadd] 3 root 0 SW< [ksoftirqd/0] 4 root 0 SW< [events/0] 5 root 0 SW< [khelper] 8 root 0 SW< [async/mgr] 112 root 0 SW< [kblockd/0] 124 root 0 SW< [khubd] 127 root 0 SW< [kseriod] 156 root 0 SW [crypto] 157 root 0 SW [crypto_ret] 164 root 0 SW [pdflush] 165 root 0 SW [pdflush] 166 root 0 SW< [kswapd0] 167 root 0 SW< [aio/0] 168 root 0 SW< [crypto/0] 472 root 0 SW< [mtdblockd] 473 root 0 SW< [nftld] 495 root 0 SW< [spi_gpio.0] 854 root 0 SWN [jffs2_gcd_mtd5] 867 root 1168 S {rcS} /bin/sh /etc/init.d/rcS S boot 870 root 1120 S logger -s -p 6 -t sysinit 871 root 1140 S /bin/ash --login 872 root 1128 S init 972 root 0 SW< [usbhid_resumer] 985 root 1132 S /sbin/syslogd -C16 987 root 1112 S /sbin/klogd 989 root 588 S /sbin/hotplug2 --override --persistent --set-rules-f 1832 root 772 S /usr/sbin/ntpclient -i 600 -s -l -D -p 123 -h 0.open 1908 root 1132 S /sbin/udhcpc -t 0 -i eth1 -C promsvyaz-cpe -V promsv 1981 root 848 S /usr/sbin/dropbear -P /var/run/dropbear.1.pid -p 22 1994 root 2164 S /usr/sbin/snmpd -Lf /dev/null -p /var/run/snmpd.pid 1997 root 1116 S /usr/sbin/telnetd -l /bin/login 2003 root 1036 S /usr/sbin/uhttpd -f -h /www -r CPE -x /cgi-bin -t 60 2007 root 556 S < /sbin/btnsd 2024 nobody 708 S /usr/sbin/dnsmasq -K -D -y -Z -b -E -s lan -S /lan/ 2038 root 1548 S lua /usr/lib/lua/tr069/tr069_client.lua 2254 root 12384 S /usr/sbin/asterisk 2376 root 1120 R ps root@CPE:/etc/asterisk# /etc/init.d/asterisk stop root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# ps PID USER VSZ STAT COMMAND 1 root 1128 S init 2 root 0 SW< [kthreadd] 3 root 0 SW< [ksoftirqd/0] 4 root 0 SW< [events/0] 5 root 0 SW< [khelper] 8 root 0 SW< [async/mgr] 112 root 0 SW< [kblockd/0] 124 root 0 SW< [khubd] 127 root 0 SW< [kseriod] 156 root 0 SW [crypto] 157 root 0 SW [crypto_ret] 164 root 0 SW [pdflush] 165 root 0 SW [pdflush] 166 root 0 SW< [kswapd0] 167 root 0 SW< [aio/0] 168 root 0 SW< [crypto/0] 472 root 0 SW< [mtdblockd] 473 root 0 SW< [nftld] 495 root 0 SW< [spi_gpio.0] 854 root 0 SWN [jffs2_gcd_mtd5] 867 root 1168 S {rcS} /bin/sh /etc/init.d/rcS S boot 870 root 1120 S logger -s -p 6 -t sysinit 871 root 1140 R /bin/ash --login 872 root 1128 S init 972 root 0 SW< [usbhid_resumer] 985 root 1132 S /sbin/syslogd -C16 987 root 1112 S /sbin/klogd 989 root 588 S /sbin/hotplug2 --override --persistent --set-rules-f 1832 root 772 S /usr/sbin/ntpclient -i 600 -s -l -D -p 123 -h 0.open 1908 root 1132 S /sbin/udhcpc -t 0 -i eth1 -C promsvyaz-cpe -V promsv 1981 root 848 S /usr/sbin/dropbear -P /var/run/dropbear.1.pid -p 22 1994 root 2164 S /usr/sbin/snmpd -Lf /dev/null -p /var/run/snmpd.pid 1997 root 1116 S /usr/sbin/telnetd -l /bin/login 2003 root 1036 S /usr/sbin/uhttpd -f -h /www -r CPE -x /cgi-bin -t 60 2007 root 556 S < /sbin/btnsd 2024 nobody 708 S /usr/sbin/dnsmasq -K -D -y -Z -b -E -s lan -S /lan/ 2038 root 1548 S lua /usr/lib/lua/tr069/tr069_client.lua 2380 root 1120 R ps root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# /etc/init.d/asterisk start root@CPE:/etc/asterisk# slic_enable_echo 0 slic_enable_echo 1 root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# root@CPE:/etc/asterisk# asterisk -rvvvvvv Asterisk 1.8.16.0, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found Connected to Asterisk 1.8.16.0 currently running on CPE (pid = 2383) Verbosity is at least 8 Core debug is at least 9 -- Remote UNIX connection CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> CPE*CLI> sip show peerd No such command 'sip show peerd' (type 'core show help sip show peerd' for other possible commands) CPE*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status 6000/6000 192.168.10.207 D A 5060 Unmonitored 6001/6001 192.168.10.208 D A 5060 Unmonitored 6003/6003 192.168.10.211 D A 5060 Unmonitored 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline] [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:8165 find_call: = Looking for Call ID: NjYzZmY4MmU3YzQzMDZjNmExMTljZDlhZDJjNjc0Y2M. (Chec [Nov 13 11:53:48] DEBUG[2397]: acl.c:710 ast_ouraddrfor: Not an IPv4 nor IPv6 address, cannot get port. [Nov 13 11:53:48] DEBUG[2397]: acl.c:736 ast_ouraddrfor: For destination '192.168.10.207', our source address is '192.168.10.192'. [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:3544 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.10.192:5060 [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:7845 sip_alloc: Allocating new SIP dialog for NjYzZmY4MmU3YzQzMDZjNmExMTljZDlhZDJjNjc0Y2M.) [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:25303 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [Nov 13 11:53:48] DEBUG[2397]: sip/reqresp_parser.c:1550 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Nov 13 11:53:48] DEBUG[2397]: sip/reqresp_parser.c:1566 parse_sip_options: Found SIP option: -replaces- [Nov 13 11:53:48] DEBUG[2397]: sip/reqresp_parser.c:1574 parse_sip_options: Matched SIP option: replaces [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.207:5060' into... [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.207' and port '5060'. [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.192' into... [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.192' and port ''. [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:3392 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.10.207:50 [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:8165 find_call: = Looking for Call ID: NjYzZmY4MmU3YzQzMDZjNmExMTljZDlhZDJjNjc0Y2M. (Chec [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:25303 handle_incoming: **** Received ACK (6) - Command in SIP ACK [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:4076 __sip_ack: Stopping retransmission on 'NjYzZmY4MmU3YzQzMDZjNmExMTljZDlhZDJjNjc0Y2M.' d [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:8165 find_call: = Looking for Call ID: NjYzZmY4MmU3YzQzMDZjNmExMTljZDlhZDJjNjc0Y2M. (Chec [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.192' into... [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.192' and port ''. [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.192' into... [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.192' and port ''. [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:25303 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.207:5060' into... [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.207' and port '5060'. [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.192' into... [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.192' and port ''. [Nov 13 11:53:48] DEBUG[2397]: rtp_engine.c:350 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x2a2fb0' [Nov 13 11:53:48] DEBUG[2397]: res_rtp_asterisk.c:557 ast_rtp_new: Allocated port 19854 for RTP instance '0x2a2fb0' [Nov 13 11:53:48] DEBUG[2397]: rtp_engine.c:359 ast_rtp_instance_new: RTP instance '0x2a2fb0' is setup and ready to go [Nov 13 11:53:48] DEBUG[2397]: res_rtp_asterisk.c:2536 ast_rtp_prop_set: Setup RTCP on RTP instance '0x2a2fb0' == Using SIP RTP CoS mark 5 [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:5142 do_setnat: Setting NAT on RTP to Off [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP o=- 12997281225368277 1 IN IP4 192.168.10.2. [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP s=CounterPath X-Lite 5.0.0... UNSUPPORTED O. [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.207' into... [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.207' and port ''. [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP c=IN IP4 192.168.10.207... OK. [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP b=AS:4096... UNSUPPORTED OR FAILED. [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Nov 13 11:53:48] DEBUG[2397]: rtp_engine.c:541 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x2b3d8e48 [Nov 13 11:53:48] DEBUG[2397]: rtp_engine.c:541 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x2b3d8e48 [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:9229 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... . [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:9229 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAI. [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:9229 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 13 11:53:48] DEBUG[2397]: rtp_engine.c:644 ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0x2b3d8e48 [Nov 13 11:53:48] DEBUG[2397]: rtp_engine.c:644 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x2b3d8e48 [Nov 13 11:53:48] DEBUG[2397]: res_rtp_asterisk.c:2576 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x2a2fb0' [Nov 13 11:53:48] DEBUG[2397]: rtp_engine.c:522 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0x2b3d8e48 to 0x2a3160 [Nov 13 11:53:48] DEBUG[2397]: rtp_engine.c:522 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x2b3d8e48 to 0x2a3160 [Nov 13 11:53:48] DEBUG[2397]: res_rtp_asterisk.c:2502 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x2a2fb0' [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:9474 process_sdp: We're settling with these formats: 0x4 (ulaw) [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:22829 handle_request_invite: Checking SIP call limits for device 6000 [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:5955 update_call_counter: Updating call counter for incoming call [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.192' into... [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.192' and port ''. [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.192' into... [Nov 13 11:53:48] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.192' and port ''. [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:7141 sip_new: *** Our native formats are 0x4 (ulaw) [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:7142 sip_new: *** Joint capabilities are 0x4 (ulaw) [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:7143 sip_new: *** Our capabilities are 0x90d (g723|ulaw|alaw|g726|g729) [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:7144 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:7174 sip_new: This channel will not be able to handle video. [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:14386 build_route: build_route: Contact hop: [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:23130 handle_request_invite: SIP/6000-00000000: New call is still down.... Trying... [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:3392 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.207:50 [Nov 13 11:53:48] DEBUG[2388]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - 6000 [Nov 13 11:53:48] DEBUG[2388]: chan_sip.c:26608 sip_devicestate: Checking device state for peer 6000 [Nov 13 11:53:48] DEBUG[2388]: devicestate.c:460 do_state_change: Changing state for SIP/6000 - state 1 (Not in use) [Nov 13 11:53:48] DEBUG[2388]: devicestate.c:440 devstate_event: device 'SIP/6000' state '1' [Nov 13 11:53:48] DEBUG[2402]: db.c:297 ast_db_get: Unable to find key '6000' in family 'blacklist' [Nov 13 11:53:48] DEBUG[2402]: db.c:297 ast_db_get: Unable to find key '6000' in family 'blacklist' [Nov 13 11:53:48] DEBUG[2402]: pbx.c:4075 pbx_substitute_variables_helper_full: Function result is '0' [Nov 13 11:53:48] DEBUG[2402]: pbx.c:4143 pbx_substitute_variables_helper_full: Expression result is '0' [Nov 13 11:53:48] DEBUG[2402]: pbx.c:4247 pbx_extension_helper: Launching 'GotoIf' -- Executing [6001@sip-6000:1] GotoIf("SIP/6000-00000000", "0?blacklisted:dial") in new stack -- Goto (sip-6000,6001,6) [Nov 13 11:53:48] DEBUG[2402]: pbx.c:3256 ast_str_retrieve_variable: Result of 'EXTEN' is '6001' [Nov 13 11:53:48] DEBUG[2402]: pbx.c:4247 pbx_extension_helper: Launching 'Goto' -- Executing [6001@sip-6000:6] Goto("SIP/6000-00000000", "sip-6000-dial,6001,1") in new stack -- Goto (sip-6000-dial,6001,1) [Nov 13 11:53:48] DEBUG[2402]: pbx.c:4247 pbx_extension_helper: Launching 'NoOp' -- Executing [6001@sip-6000-dial:1] NoOp("SIP/6000-00000000", "") in new stack [Nov 13 11:53:48] DEBUG[2402]: pbx.c:3256 ast_str_retrieve_variable: Result of 'RINGTIME' is '30' [Nov 13 11:53:48] DEBUG[2402]: pbx.c:4247 pbx_extension_helper: Launching 'Dial' -- Executing [6001@sip-6000-dial:2] Dial("SIP/6000-00000000", "SIP/6001, 30, kKtT") in new stack [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:26708 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:7845 sip_alloc: Allocating new SIP dialog for 1b57b8b009d3cdec7aa15e55441cdbd5@(null) - IN) [Nov 13 11:53:48] DEBUG[2402]: rtp_engine.c:350 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x2acaf0' [Nov 13 11:53:48] DEBUG[2402]: res_rtp_asterisk.c:557 ast_rtp_new: Allocated port 19886 for RTP instance '0x2acaf0' [Nov 13 11:53:48] DEBUG[2402]: rtp_engine.c:359 ast_rtp_instance_new: RTP instance '0x2acaf0' is setup and ready to go [Nov 13 11:53:48] DEBUG[2402]: res_rtp_asterisk.c:2536 ast_rtp_prop_set: Setup RTCP on RTP instance '0x2acaf0' == Using SIP RTP CoS mark 5 [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:5142 do_setnat: Setting NAT on RTP to Off [Nov 13 11:53:48] DEBUG[2402]: acl.c:710 ast_ouraddrfor: Not an IPv4 nor IPv6 address, cannot get port. [Nov 13 11:53:48] DEBUG[2402]: acl.c:736 ast_ouraddrfor: For destination '192.168.10.208', our source address is '192.168.10.192'. [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:3544 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.10.192:5060 [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:7141 sip_new: *** Our native formats are 0x4 (ulaw) [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:7142 sip_new: *** Joint capabilities are 0x4 (ulaw) [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:7143 sip_new: *** Our capabilities are 0x90d (g723|ulaw|alaw|g726|g729) [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:7144 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:7146 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:7174 sip_new: This channel will not be able to handle video. [Nov 13 11:53:48] DEBUG[2402]: rtp_engine.c:1473 ast_rtp_instance_early_bridge_make_compatible: Seeded SDP of 'SIP/6001-00000001' wi' [Nov 13 11:53:48] DEBUG[2402]: channel.c:6169 ast_channel_inherit_variables: Not copying variable DIALEDTIME. [Nov 13 11:53:48] DEBUG[2402]: channel.c:6169 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. [Nov 13 11:53:48] DEBUG[2402]: channel.c:6169 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. [Nov 13 11:53:48] DEBUG[2402]: channel.c:6169 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. [Nov 13 11:53:48] DEBUG[2402]: channel.c:6169 ast_channel_inherit_variables: Not copying variable DIALSTATUS. [Nov 13 11:53:48] DEBUG[2402]: channel.c:6169 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Nov 13 11:53:48] DEBUG[2402]: channel.c:6169 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Nov 13 11:53:48] DEBUG[2402]: channel.c:6169 ast_channel_inherit_variables: Not copying variable SIPURI. [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:5688 sip_call: Outgoing Call for 6001 [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:5955 update_call_counter: Updating call counter for outgoing call [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:11463 add_sdp: ** Our capability: 0x90d (g723|ulaw|alaw|g726|g729) Video flag: False Text e [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:11464 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:11573 add_sdp: -- Done with adding codecs to SDP [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:11759 add_sdp: Done building SDP. Settling with this capability: 0x90d (g723|ulaw|alaw|g72) [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:3075 initialize_initreq: Initializing initreq for method INVITE - callid 3a617f3264cc2f6100 [Nov 13 11:53:48] DEBUG[2402]: chan_sip.c:3392 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.208:50 -- Called SIP/6001 [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:3651 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retr [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:3392 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.208:50 [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:8165 find_call: = Looking for Call ID: 3a617f3264cc2f6104a3dc5f7ad2a78c@192.168.10.192:50 [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:4117 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '3a61d [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:20152 handle_response_invite: SIP response 100 to standard invite [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:8165 find_call: = Looking for Call ID: 3a617f3264cc2f6104a3dc5f7ad2a78c@192.168.10.192:50 [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:4117 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '3a61d [Nov 13 11:53:48] DEBUG[2397]: chan_sip.c:20152 handle_response_invite: SIP response 100 to standard invite [Nov 13 11:53:49] DEBUG[2397]: chan_sip.c:8165 find_call: = Looking for Call ID: 3a617f3264cc2f6104a3dc5f7ad2a78c@192.168.10.192:50 [Nov 13 11:53:49] DEBUG[2397]: chan_sip.c:4117 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '3a61d [Nov 13 11:53:49] DEBUG[2397]: chan_sip.c:20152 handle_response_invite: SIP response 180 to standard invite [Nov 13 11:53:49] DEBUG[2397]: chan_sip.c:14386 build_route: build_route: Contact hop: [Nov 13 11:53:49] DEBUG[2388]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - 6001 [Nov 13 11:53:49] DEBUG[2388]: chan_sip.c:26608 sip_devicestate: Checking device state for peer 6001 [Nov 13 11:53:49] DEBUG[2388]: devicestate.c:460 do_state_change: Changing state for SIP/6001 - state 1 (Not in use) [Nov 13 11:53:49] DEBUG[2388]: devicestate.c:440 devstate_event: device 'SIP/6001' state '1' -- SIP/6001-00000001 is ringing [Nov 13 11:53:49] DEBUG[2402]: rtp_engine.c:1556 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/6000-00000000' with' [Nov 13 11:53:49] DEBUG[2402]: chan_sip.c:3392 __sip_xmit: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.10.207:50 [Nov 13 11:53:49] DEBUG[2402]: channel.c:4536 ast_indicate_data: Driver for channel 'SIP/6000-00000000' does not support indication t [Nov 13 11:53:49] DEBUG[2402]: channel.c:5192 set_format: Set channel SIP/6000-00000000 to write format slin [Nov 13 11:53:49] DEBUG[2402]: channel.c:3523 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Nov 13 11:53:49] DEBUG[2402]: channel.c:4686 ast_prod: Prodding channel 'SIP/6000-00000000' [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:1331 ast_rtp_write: Received frame with no data for RTP instance '0x2a2fb0' so droe [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:1360 ast_rtp_write: Ooh, format changed from unknown to ulaw [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:1391 ast_rtp_write: Created smoother: format: ulaw ms: 20 len: 160 [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:1256 ast_rtp_raw_write: Starting RTCP transmission on RTP instance '0x2a2fb0' [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:49] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:50] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:51] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:1806 ast_rtcp_read: Got RTCP report of 132 bytes [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:8165 find_call: = Looking for Call ID: 3a617f3264cc2f6104a3dc5f7ad2a78c@192.168.10.192:50 [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:4038 __sip_ack: Acked pending invite 102 [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:4076 __sip_ack: Stopping retransmission on '3a617f3264cc2f6104a3dc5f7ad2a78c@192.168.10.19d [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:20152 handle_response_invite: SIP response 200 to standard invite [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP o=- 12997281231407515 1 IN IP4 192.168.10.2. [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP s=CounterPath X-Lite 5.0.0... UNSUPPORTED O. [Nov 13 11:53:52] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.208' into... [Nov 13 11:53:52] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.208' and port ''. [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP c=IN IP4 192.168.10.208... OK. [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP b=AS:1638... UNSUPPORTED OR FAILED. [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Nov 13 11:53:52] DEBUG[2397]: rtp_engine.c:541 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x2b3d8f60 [Nov 13 11:53:52] DEBUG[2397]: rtp_engine.c:541 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x2b3d8f60 [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:9229 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... . [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:9229 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAI. [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:9229 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 13 11:53:52] DEBUG[2397]: rtp_engine.c:644 ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0x2b3d8f60 [Nov 13 11:53:52] DEBUG[2397]: rtp_engine.c:644 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x2b3d8f60 [Nov 13 11:53:52] DEBUG[2397]: res_rtp_asterisk.c:2576 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x2acaf0' [Nov 13 11:53:52] DEBUG[2397]: rtp_engine.c:522 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0x2b3d8f60 to 0x2acca0 [Nov 13 11:53:52] DEBUG[2397]: rtp_engine.c:522 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x2b3d8f60 to 0x2acca0 [Nov 13 11:53:52] DEBUG[2397]: res_rtp_asterisk.c:2502 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x2acaf0' [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:9474 process_sdp: We're settling with these formats: 0x4 (ulaw) [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:9479 process_sdp: We have an owner, now see if we need to change this call [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:5955 update_call_counter: Updating call counter for outgoing call [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:14386 build_route: build_route: Contact hop: [Nov 13 11:53:52] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.208:5060' into... [Nov 13 11:53:52] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.208' and port '5060'. [Nov 13 11:53:52] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.208:5060' into... [Nov 13 11:53:52] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.208' and port '5060'. [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:3392 __sip_xmit: Trying to put 'ACK sip:600' onto UDP socket destined for 192.168.10.208:50 [Nov 13 11:53:52] DEBUG[2388]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - 6001 [Nov 13 11:53:52] DEBUG[2388]: chan_sip.c:26608 sip_devicestate: Checking device state for peer 6001 [Nov 13 11:53:52] DEBUG[2388]: devicestate.c:460 do_state_change: Changing state for SIP/6001 - state 1 (Not in use) [Nov 13 11:53:52] DEBUG[2388]: devicestate.c:440 devstate_event: device 'SIP/6001' state '1' -- SIP/6001-00000001 answered SIP/6000-00000000 [Nov 13 11:53:52] DEBUG[2402]: channel.c:5192 set_format: Set channel SIP/6000-00000000 to write format ulaw [Nov 13 11:53:52] DEBUG[2402]: channel.c:3523 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Nov 13 11:53:52] DEBUG[2388]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - 6000 [Nov 13 11:53:52] DEBUG[2388]: chan_sip.c:26608 sip_devicestate: Checking device state for peer 6000 [Nov 13 11:53:52] DEBUG[2388]: devicestate.c:460 do_state_change: Changing state for SIP/6000 - state 1 (Not in use) [Nov 13 11:53:52] DEBUG[2388]: devicestate.c:440 devstate_event: device 'SIP/6000' state '1' [Nov 13 11:53:52] DEBUG[2402]: chan_sip.c:6542 sip_answer: SIP answering channel: SIP/6000-00000000 [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: chan_sip.c:11864 transmit_response_with_sdp: Setting framing from config on incoming call [Nov 13 11:53:52] DEBUG[2402]: chan_sip.c:11463 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Nov 13 11:53:52] DEBUG[2402]: chan_sip.c:11464 add_sdp: ** Our prefcodec: 0x0 (nothing) [Nov 13 11:53:52] DEBUG[2402]: chan_sip.c:11573 add_sdp: -- Done with adding codecs to SDP [Nov 13 11:53:52] DEBUG[2402]: chan_sip.c:11759 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Nov 13 11:53:52] DEBUG[2402]: chan_sip.c:3392 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.207:50 [Nov 13 11:53:52] DEBUG[2402]: features.c:4043 ast_bridge_call: bridge answer set, chan answer set [Nov 13 11:53:52] DEBUG[2402]: features.c:3885 clear_dialed_interfaces: Removing dialed interfaces datastore on SIP/6001-00000001 sig [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:1806 ast_rtcp_read: Got RTCP report of 132 bytes [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:8165 find_call: = Looking for Call ID: NjYzZmY4MmU3YzQzMDZjNmExMTljZDlhZDJjNjc0Y2M. (Chec [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:25303 handle_incoming: **** Received ACK (6) - Command in SIP ACK [Nov 13 11:53:52] DEBUG[2397]: chan_sip.c:4076 __sip_ack: Stopping retransmission on 'NjYzZmY4MmU3YzQzMDZjNmExMTljZDlhZDJjNjc0Y2M.' d [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:1360 ast_rtp_write: Ooh, format changed from unknown to ulaw [Nov 13 11:53:52] DEBUG[2402]: res_rtp_asterisk.c:1391 ast_rtp_write: Created smoother: format: ulaw ms: 20 len: 160 [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:1501 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 192.168.10.208:5062 [Nov 13 11:53:55] DEBUG[2402]: channel.c:7218 ast_generic_bridge: Got DTMF begin on channel (SIP/6001-00000001) [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:55] DEBUG[2402]: channel.c:7612 ast_channel_bridge: Bridge stops bridging channels SIP/6000-00000000 and SIP/6001-00001 [Nov 13 11:53:55] DEBUG[2402]: features.c:4269 ast_bridge_call: Not passing DTMF through, since it may be a feature code [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:1806 ast_rtcp_read: Got RTCP report of 176 bytes [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:1501 create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 192.168.10.208:5062 [Nov 13 11:53:55] DEBUG[2402]: channel.c:7218 ast_generic_bridge: Got DTMF end on channel (SIP/6001-00000001) [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:55] DEBUG[2402]: channel.c:7612 ast_channel_bridge: Bridge stops bridging channels SIP/6000-00000000 and SIP/6001-00001 [Nov 13 11:53:55] DEBUG[2402]: features.c:3395 feature_interpret: Feature interpret: chan=SIP/6000-00000000, peer=SIP/6001-00000001,# [Nov 13 11:53:55] DEBUG[2402]: features.c:4330 ast_bridge_call: Set feature timer to 1000 ms [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [Nov 13 11:53:55] DEBUG[2402]: res_rtp_asterisk.c:1806 ast_rtcp_read: Got RTCP report of 176 bytes [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:1501 create_dtmf_frame: Creating BEGIN DTMF Frame: 50 (2), at 192.168.10.208:5062 [Nov 13 11:53:56] DEBUG[2402]: channel.c:7218 ast_generic_bridge: Got DTMF begin on channel (SIP/6001-00000001) [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:56] DEBUG[2402]: channel.c:7612 ast_channel_bridge: Bridge stops bridging channels SIP/6000-00000000 and SIP/6001-00001 [Nov 13 11:53:56] DEBUG[2402]: features.c:4269 ast_bridge_call: Not passing DTMF through, since it may be a feature code [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:1501 create_dtmf_frame: Creating END DTMF Frame: 50 (2), at 192.168.10.208:5062 [Nov 13 11:53:56] DEBUG[2402]: channel.c:7218 ast_generic_bridge: Got DTMF end on channel (SIP/6001-00000001) [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update [Nov 13 11:53:56] DEBUG[2402]: channel.c:7612 ast_channel_bridge: Bridge stops bridging channels SIP/6000-00000000 and SIP/6001-00001 [Nov 13 11:53:56] DEBUG[2402]: features.c:3395 feature_interpret: Feature interpret: chan=SIP/6000-00000000, peer=SIP/6001-00000001,# [Nov 13 11:53:56] DEBUG[2402]: features.c:3275 feature_interpret_helper: Feature detected: fname=Attended Transfer sname=atxfer exte2 [Nov 13 11:53:56] DEBUG[2402]: features.c:2524 builtin_atxfer: Executing Attended Transfer SIP/6000-00000000, SIP/6001-00000001 (sen [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:829 ast_rtp_update_source: Setting the marker bit due to a source update -- Started music on hold, class 'default', on SIP/6000-00000000 [Nov 13 11:53:56] DEBUG[2402]: channel.c:3523 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Nov 13 11:53:56] DEBUG[2402]: channel.c:5192 set_format: Set channel SIP/6001-00000001 to write format gsm [Nov 13 11:53:56] DEBUG[2403]: channel.c:3663 ast_read_generator_actions: Generator got voice, switching to phase locked mode [Nov 13 11:53:56] DEBUG[2403]: channel.c:3523 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Nov 13 11:53:56] DEBUG[2403]: channel.c:5192 set_format: Set channel SIP/6000-00000000 to write format gsm [Nov 13 11:53:56] DEBUG[2403]: res_musiconhold.c:341 ast_moh_files_next: SIP/6000-00000000 Opened file 0 '/usr/lib/asterisk/sounds/m' [Nov 13 11:53:56] DEBUG[2403]: res_rtp_asterisk.c:1177 ast_rtp_raw_write: Difference is 1112, ms is 159 [Nov 13 11:53:56] DEBUG[2402]: channel.c:3523 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second -- Playing 'pbx-transfer.gsm' (language 'en') [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 13 11:53:56] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 13 11:53:57] DEBUG[2402]: channel.c:3523 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Nov 13 11:53:57] DEBUG[2402]: channel.c:3523 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Nov 13 11:53:57] DEBUG[2402]: channel.c:3523 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Nov 13 11:53:57] DEBUG[2402]: channel.c:5192 set_format: Set channel SIP/6001-00000001 to write format ulaw [Nov 13 11:53:57] DEBUG[2402]: channel.c:5192 set_format: Set channel SIP/6001-00000001 to write format slin [Nov 13 11:53:57] DEBUG[2402]: channel.c:3523 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Nov 13 11:53:57] DEBUG[2402]: channel.c:3663 ast_read_generator_actions: Generator got voice, switching to phase locked mode [Nov 13 11:53:57] DEBUG[2402]: channel.c:3523 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Nov 13 11:53:58] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [Nov 13 11:53:58] DEBUG[2402]: res_rtp_asterisk.c:1501 create_dtmf_frame: Creating BEGIN DTMF Frame: 54 (6), at 192.168.10.208:5062 [Nov 13 11:53:58] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [Nov 13 11:53:58] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [Nov 13 11:53:58] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [Nov 13 11:53:58] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [Nov 13 11:53:58] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [Nov 13 11:53:58] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [Nov 13 11:53:58] DEBUG[2402]: res_rtp_asterisk.c:1501 create_dtmf_frame: Creating END DTMF Frame: 54 (6), at 192.168.10.208:5062 [Nov 13 11:53:58] DEBUG[2402]: channel.c:5192 set_format: Set channel SIP/6001-00000001 to write format ulaw [Nov 13 11:53:58] DEBUG[2402]: channel.c:3523 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Nov 13 11:53:58] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [Nov 13 11:53:58] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [Nov 13 11:53:58] DEBUG[2402]: res_rtp_asterisk.c:1806 ast_rtcp_read: Got RTCP report of 176 bytes [Nov 13 11:53:58] DEBUG[2403]: res_rtp_asterisk.c:1806 ast_rtcp_read: Got RTCP report of 176 bytes [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1501 create_dtmf_frame: Creating BEGIN DTMF Frame: 48 (0), at 192.168.10.208:5062 [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1501 create_dtmf_frame: Creating END DTMF Frame: 48 (0), at 192.168.10.208:5062 [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1501 create_dtmf_frame: Creating BEGIN DTMF Frame: 48 (0), at 192.168.10.208:5062 [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1501 create_dtmf_frame: Creating END DTMF Frame: 48 (0), at 192.168.10.208:5062 [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000000 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1501 create_dtmf_frame: Creating BEGIN DTMF Frame: 51 (3), at 192.168.10.208:5062 [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1501 create_dtmf_frame: Creating END DTMF Frame: 51 (3), at 192.168.10.208:5062 [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 13 11:53:59] DEBUG[2402]: res_rtp_asterisk.c:1545 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 13 11:54:01] DEBUG[2402]: res_rtp_asterisk.c:1806 ast_rtcp_read: Got RTCP report of 176 bytes [Nov 13 11:54:02] DEBUG[2403]: res_rtp_asterisk.c:1806 ast_rtcp_read: Got RTCP report of 176 bytes [Nov 13 11:54:03] DEBUG[2402]: features.c:825 get_parking_exten: Checking if 6003@sip-6001 is a parking exten [Nov 13 11:54:03] DEBUG[2402]: channel.c:6169 ast_channel_inherit_variables: Not copying variable BRIDGEPVTCALLID. [Nov 13 11:54:03] DEBUG[2402]: channel.c:6169 ast_channel_inherit_variables: Not copying variable BRIDGEPEER. [Nov 13 11:54:03] DEBUG[2402]: channel.c:6169 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. [Nov 13 11:54:03] DEBUG[2402]: channel.c:6169 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Nov 13 11:54:03] DEBUG[2404]: db.c:297 ast_db_get: Unable to find key '6001' in family 'blacklist' [Nov 13 11:54:03] DEBUG[2404]: db.c:297 ast_db_get: Unable to find key '' in family 'blacklist' [Nov 13 11:54:03] DEBUG[2404]: pbx.c:4075 pbx_substitute_variables_helper_full: Function result is '0' [Nov 13 11:54:03] DEBUG[2404]: pbx.c:4143 pbx_substitute_variables_helper_full: Expression result is '0' [Nov 13 11:54:03] DEBUG[2404]: pbx.c:4247 pbx_extension_helper: Launching 'GotoIf' -- Executing [6003@sip-6001:1] GotoIf("Local/6003@sip-6001-d94b;2", "0?blacklisted:dial") in new stack -- Goto (sip-6001,6003,6) [Nov 13 11:54:03] DEBUG[2404]: pbx.c:3256 ast_str_retrieve_variable: Result of 'EXTEN' is '6003' [Nov 13 11:54:03] DEBUG[2404]: pbx.c:4247 pbx_extension_helper: Launching 'Goto' -- Executing [6003@sip-6001:6] Goto("Local/6003@sip-6001-d94b;2", "sip-6001-dial,6003,1") in new stack -- Goto (sip-6001-dial,6003,1) [Nov 13 11:54:03] DEBUG[2404]: pbx.c:4247 pbx_extension_helper: Launching 'NoOp' -- Executing [6003@sip-6001-dial:1] NoOp("Local/6003@sip-6001-d94b;2", "") in new stack [Nov 13 11:54:03] DEBUG[2404]: pbx.c:3256 ast_str_retrieve_variable: Result of 'RINGTIME' is '30' [Nov 13 11:54:03] DEBUG[2404]: pbx.c:4247 pbx_extension_helper: Launching 'Dial' -- Executing [6003@sip-6001-dial:2] Dial("Local/6003@sip-6001-d94b;2", "SIP/6003, 30, kKtT") in new stack [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:26708 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:7845 sip_alloc: Allocating new SIP dialog for 176438023f9580211f6b622d20afd095@(null) - IN) [Nov 13 11:54:03] DEBUG[2404]: rtp_engine.c:350 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x2c22a0' [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:557 ast_rtp_new: Allocated port 19866 for RTP instance '0x2c22a0' [Nov 13 11:54:03] DEBUG[2404]: rtp_engine.c:359 ast_rtp_instance_new: RTP instance '0x2c22a0' is setup and ready to go [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:2536 ast_rtp_prop_set: Setup RTCP on RTP instance '0x2c22a0' == Using SIP RTP CoS mark 5 [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:5142 do_setnat: Setting NAT on RTP to Off [Nov 13 11:54:03] DEBUG[2404]: acl.c:710 ast_ouraddrfor: Not an IPv4 nor IPv6 address, cannot get port. [Nov 13 11:54:03] DEBUG[2404]: acl.c:736 ast_ouraddrfor: For destination '192.168.10.211', our source address is '192.168.10.192'. [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:3544 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.10.192:5060 [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:7141 sip_new: *** Our native formats are 0x4 (ulaw) [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:7142 sip_new: *** Joint capabilities are 0x4 (ulaw) [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:7143 sip_new: *** Our capabilities are 0x90d (g723|ulaw|alaw|g726|g729) [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:7144 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:7146 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:7174 sip_new: This channel will not be able to handle video. [Nov 13 11:54:03] DEBUG[2404]: rtp_engine.c:1412 ast_rtp_instance_early_bridge_make_compatible: Can't find native functions for chan' [Nov 13 11:54:03] DEBUG[2404]: rtp_engine.c:1473 ast_rtp_instance_early_bridge_make_compatible: Seeded SDP of 'SIP/6003-00000002' wi' [Nov 13 11:54:03] DEBUG[2404]: channel.c:6169 ast_channel_inherit_variables: Not copying variable DIALEDTIME. [Nov 13 11:54:03] DEBUG[2404]: channel.c:6169 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. [Nov 13 11:54:03] DEBUG[2404]: channel.c:6169 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. [Nov 13 11:54:03] DEBUG[2404]: channel.c:6169 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. [Nov 13 11:54:03] DEBUG[2404]: channel.c:6169 ast_channel_inherit_variables: Not copying variable DIALSTATUS. [Nov 13 11:54:03] DEBUG[2404]: channel.c:6169 ast_channel_inherit_variables: Not copying variable TRANSFERERNAME. [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:5688 sip_call: Outgoing Call for 6003 [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:5955 update_call_counter: Updating call counter for outgoing call [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:11463 add_sdp: ** Our capability: 0x90d (g723|ulaw|alaw|g726|g729) Video flag: False Text e [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:11464 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:11573 add_sdp: -- Done with adding codecs to SDP [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:11759 add_sdp: Done building SDP. Settling with this capability: 0x90d (g723|ulaw|alaw|g72) [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:3075 initialize_initreq: Initializing initreq for method INVITE - callid 2759ea022649738400 [Nov 13 11:54:03] DEBUG[2404]: chan_sip.c:3392 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.211:50 -- Called SIP/6003 [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2397]: chan_sip.c:8165 find_call: = Looking for Call ID: 2759ea02264973840bcba4d64a844b39@192.168.10.192:50 [Nov 13 11:54:03] DEBUG[2397]: chan_sip.c:4117 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2759d [Nov 13 11:54:03] DEBUG[2397]: chan_sip.c:20152 handle_response_invite: SIP response 100 to standard invite [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:03] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:04] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:04] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:04] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:04] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:04] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:04] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:04] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:04] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:04] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:04] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:04] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:04] DEBUG[2404]: res_rtp_asterisk.c:1325 ast_rtp_write: No remote address on RTP instance '0x2c22a0' so dropping frame [Nov 13 11:54:04] DEBUG[2397]: chan_sip.c:8165 find_call: = Looking for Call ID: 2759ea02264973840bcba4d64a844b39@192.168.10.192:50 [Nov 13 11:54:04] DEBUG[2397]: chan_sip.c:4117 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2759d [Nov 13 11:54:04] DEBUG[2397]: chan_sip.c:20152 handle_response_invite: SIP response 180 to standard invite [Nov 13 11:54:04] DEBUG[2397]: chan_sip.c:14386 build_route: build_route: Contact hop: [Nov 13 11:54:04] DEBUG[2388]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - 6003 [Nov 13 11:54:04] DEBUG[2388]: chan_sip.c:26608 sip_devicestate: Checking device state for peer 6003 [Nov 13 11:54:04] DEBUG[2388]: devicestate.c:460 do_state_change: Changing state for SIP/6003 - state 1 (Not in use) [Nov 13 11:54:04] DEBUG[2388]: devicestate.c:440 devstate_event: device 'SIP/6003' state '1' -- SIP/6003-00000002 is ringing [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:8165 find_call: = Looking for Call ID: 2759ea02264973840bcba4d64a844b39@192.168.10.192:50 [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:4038 __sip_ack: Acked pending invite 102 [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:4076 __sip_ack: Stopping retransmission on '2759ea02264973840bcba4d64a844b39@192.168.10.19d [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:20152 handle_response_invite: SIP response 200 to standard invite [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP o=- 12997281249488996 1 IN IP4 192.168.10.2. [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP s=CounterPath X-Lite 5.0.0... UNSUPPORTED O. [Nov 13 11:54:09] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.211' into... [Nov 13 11:54:09] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.211' and port ''. [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP c=IN IP4 192.168.10.211... OK. [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP b=AS:1638... UNSUPPORTED OR FAILED. [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:8958 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Nov 13 11:54:09] DEBUG[2397]: rtp_engine.c:541 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x2b3d8f60 [Nov 13 11:54:09] DEBUG[2397]: rtp_engine.c:541 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x2b3d8f60 [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:9229 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... . [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:9229 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAI. [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:9229 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 13 11:54:09] DEBUG[2397]: rtp_engine.c:644 ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0x2b3d8f60 [Nov 13 11:54:09] DEBUG[2397]: rtp_engine.c:644 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x2b3d8f60 [Nov 13 11:54:09] DEBUG[2397]: res_rtp_asterisk.c:2576 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x2c22a0' [Nov 13 11:54:09] DEBUG[2397]: rtp_engine.c:522 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0x2b3d8f60 to 0x2c2450 [Nov 13 11:54:09] DEBUG[2397]: rtp_engine.c:522 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x2b3d8f60 to 0x2c2450 [Nov 13 11:54:09] DEBUG[2397]: res_rtp_asterisk.c:2502 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x2c22a0' [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:9474 process_sdp: We're settling with these formats: 0x4 (ulaw) [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:9479 process_sdp: We have an owner, now see if we need to change this call [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:5955 update_call_counter: Updating call counter for outgoing call [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:14386 build_route: build_route: Contact hop: [Nov 13 11:54:09] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.211:5060' into... [Nov 13 11:54:09] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.211' and port '5060'. [Nov 13 11:54:09] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.211:5060' into... [Nov 13 11:54:09] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.211' and port '5060'. [Nov 13 11:54:09] DEBUG[2397]: chan_sip.c:3392 __sip_xmit: Trying to put 'ACK sip:600' onto UDP socket destined for 192.168.10.211:50 [Nov 13 11:54:13] DEBUG[2397]: chan_sip.c:8165 find_call: = Looking for Call ID: 3a617f3264cc2f6104a3dc5f7ad2a78c@192.168.10.192:50 [Nov 13 11:54:13] DEBUG[2397]: chan_sip.c:25303 handle_incoming: **** Received BYE (8) - Command in SIP BYE [Nov 13 11:54:13] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.208:5060' into... [Nov 13 11:54:13] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.208' and port '5060'. [Nov 13 11:54:13] DEBUG[2397]: chan_sip.c:3088 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 3a617f3264cc2f6104a3dc5f7ad2a78c@10 [Nov 13 11:54:13] DEBUG[2397]: res_rtp_asterisk.c:2576 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x2acaf0' [Nov 13 11:54:13] DEBUG[2397]: chan_sip.c:24105 handle_request_bye: Received bye, issuing owner hangup [Nov 13 11:54:13] DEBUG[2397]: chan_sip.c:3392 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.208:50 [Nov 13 11:54:15] DEBUG[2397]: chan_sip.c:8165 find_call: = Looking for Call ID: 2759ea02264973840bcba4d64a844b39@192.168.10.192:50 [Nov 13 11:54:15] DEBUG[2397]: chan_sip.c:25303 handle_incoming: **** Received BYE (8) - Command in SIP BYE [Nov 13 11:54:15] DEBUG[2397]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.10.211:5060' into... [Nov 13 11:54:15] DEBUG[2397]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.10.211' and port '5060'. [Nov 13 11:54:15] DEBUG[2397]: chan_sip.c:3088 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 2759ea02264973840bcba4d64a844b39@10 [Nov 13 11:54:15] DEBUG[2397]: res_rtp_asterisk.c:2576 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x2c22a0' [Nov 13 11:54:15] DEBUG[2397]: chan_sip.c:24105 handle_request_bye: Received bye, issuing owner hangup [Nov 13 11:54:15] DEBUG[2397]: chan_sip.c:3392 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.211:50 [Nov 13 11:54:45] WARNING[2397]: chan_sip.c:3918 __sip_autodestruct: Autodestruct on dialog '3a617f3264cc2f6104a3dc5f7ad2a78c@192.16s [Nov 13 11:54:47] WARNING[2397]: chan_sip.c:3918 __sip_autodestruct: Autodestruct on dialog '2759ea02264973840bcba4d64a844b39@192.16s [Nov 13 11:55:17] WARNING[2397]: chan_sip.c:3918 __sip_autodestruct: Autodestruct on dialog '3a617f3264cc2f6104a3dc5f7ad2a78c@192.16s [Nov 13 11:55:19] WARNING[2397]: chan_sip.c:3918 __sip_autodestruct: Autodestruct on dialog '2759ea02264973840bcba4d64a844b39@192.16s [Nov 13 11:55:49] WARNING[2397]: chan_sip.c:3918 __sip_autodestruct: Autodestruct on dialog '3a617f3264cc2f6104a3dc5f7ad2a78c@192.16s [Nov 13 11:55:51] WARNING[2397]: chan_sip.c:3918 __sip_autodestruct: Autodestruct on dialog '2759ea02264973840bcba4d64a844b39@192.16s CPE*CLI> CTRL-A Z for help |115200 8N1 | NOR | Minicom 2.5 | VT102 | Offline