Summary: | ASTERISK-07994: Calling from SIP to ISDN via ZAP - hangup problem | ||
Reporter: | radon (radon) | Labels: | |
Date Opened: | 2006-10-24 11:32:13 | Date Closed: | 2006-10-24 15:09:13 |
Priority: | Blocker | Regression? | No |
Status: | Closed/Complete | Components: | Core/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | When I call from SIP phone through ISDN card to PSTN and I hangup SIP phone, called side is continuing to ring and ring and ring ... I must manually press cancel on called site to cancel ringing. --- Same problem tested on systems with software: Gentoo - kernel 2.6.17-gentoo-r8 and vanilla kernel 2.6.15 Debian testing Asterisk v. 1.2.10(debian), 1.2.12.1(debian, gentoo), 1.2.13(debian, gentoo) Zaptel v. 1.2.9.1 (gentoo), 1.2.9.1.dfsg-2 (debian) Libpri v. 1.2.3-1 (debian), 1.2.3-r1(gentoo) Everything with BRIstuff patch. --- HW isdn card: 02:0a.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) Subsystem: Cologne Chip Designs GmbH ISDN Board Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 16 (4000ns max) Interrupt: pin A routed to IRQ 3 Region 0: I/O ports at dff0 [disabled] [size=8] Region 1: Memory at feafec00 (32-bit, non-prefetchable) [size=256] Capabilities: [40] Power Management version 1 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME+ --- Tested with hardware ip phone (Grandstream GXP-2000) and with software sip phone (x-lite). --- I think it isn't hw problem, with mISDN it works ok. ****** ADDITIONAL INFORMATION ****** =============================================== Zaptel configuration =============================================== iprouter ~ # modprobe zaptel iprouter ~ # modprobe zaphfc iprouter ~ # ztcfg -vv SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. Changing signalling on channel 1 from Unused to Clear channel Changing signalling on channel 2 from Unused to Clear channel Changing signalling on channel 3 from Unused to HDLC with FCS check --- From dmesg: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.9.1 Echo Canceller: MG2 ACPI: PCI Interrupt Link [LNKG] enabled at IRQ 3 PCI: setting IRQ 3 as level-triggered ACPI: PCI Interrupt 0000:02:0a.0[A] -> Link [LNKG] -> GSI 3 (level, low) -> IRQ 3 zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem e0d02c00 fifo de8e8000(0x1e8e8000) IRQ 3 HZ 250 zaphfc: Card 0 configured for TE mode zaphfc: 1 hfc-pci card(s) in this box. Registered tone zone 3 (Netherlands) Note: I tried set loadzone/defaultzone in zaptel.conf with nl and cz (many ppl in cz runs it with nl ok). -------------------------------------------------------------------- Configuration of zaptel.conf/zapata.conf cat zapata.conf ; ; Zapata telephony interface ; ; Configuration file [channels] ; ; Default language ; language=cz ; ; Default context ; ; switchtype = euroisdn ; p2mp TE mode signalling = bri_cpe_ptmp ; p2p TE mode ;signalling = bri_cpe ; p2mp NT mode ;signalling = bri_net_ptmp ; p2p NT mode ;signalling = bri_net pridialplan = dynamic prilocaldialplan = local nationalprefix = internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=no immediate=no group = 1 context=isdn channel => 1-2 cat zaptel.conf # hfc-s pci a span definition # most of the values should be bogus because we are not really zaptel loadzone=nl defaultzone=nl #span=1,1,3,ccs,hdb3,crc4 span=1,1,3,ccs,ami bchan=1-2 dchan=3 ------------------------------------------------------------------- Debug log: iprouter*CLI> set verbose 5 iprouter*CLI> set debug 5 iprouter*CLI> sip debug iprouter*CLI> pri debug span 1 --------- We are starting call from sip phone to ISDN iprouter*CLI> <-- SIP read from 89.76.27.25:5060: INVITE sip:1005@83.10.21.42 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.99:5060;branch=z9hG4bK71c64ba884b528f0 From: "peta" <sip:peta@83.10.21.42>;tag=d0beb7cc0faad732 To: <sip:1005@83.10.21.42> Contact: <sip:peta@192.168.2.99:5060> Supported: replaces, timer Call-ID: d65e78c8ec10bfd3@192.168.2.99 CSeq: 20716 INVITE User-Agent: Grandstream GXP2000 1.1.1.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE ontent-Type: application/sdp Content-Length: 252 v=0 o=peta 8000 8000 IN IP4 192.168.2.99 s=SIP Call c=IN IP4 192.168.2.99 t=0 0 m=audio 5004 RTP/AVP 3 8 18 4 0 a=sendrecv a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 --- (13 headers 13 lines) --- Using INVITE request as basis request - d65e78c8ec10bfd3@192.168.2.99 Sending to 192.168.2.99 : 5060 (NAT) Reliably Transmitting (NAT) to 89.76.27.25:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.99:5060;branch=z9hG4bK71c64ba884b528f0;received=89.76.27.25 From: "peta" <sip:peta@83.10.21.42>;tag=d0beb7cc0faad732 To: <sip:1005@83.10.21.42>;tag=as3450040d Call-ID: d65e78c8ec10bfd3@192.168.2.99 CSeq: 20716 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0e74039f" Content-Length: 0 --- Scheduling destruction of call 'd65e78c8ec10bfd3@192.168.2.99' in 15000 ms Found user 'peta' iprouter*CLI> <-- SIP read from 89.76.27.25:5060: ACK sip:1005@83.10.21.42 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.99:5060;branch=z9hG4bK71c64ba884b528f0 From: "peta" <sip:peta@83.10.21.42>;tag=d0beb7cc0faad732 To: <sip:1005@83.10.21.42>;tag=as3450040d Contact: <sip:peta@192.168.2.99:5060> Call-ID: d65e78c8ec10bfd3@192.168.2.99 CSeq: 20716 ACK User-Agent: Grandstream GXP2000 1.1.1.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (11 headers 0 lines) --- iprouter*CLI> <-- SIP read from 89.76.27.25:5060: INVITE sip:1005@83.10.21.42 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.99:5060;branch=z9hG4bKbe09fc25bf3c4bf9 From: "peta" <sip:peta@83.10.21.42>;tag=d0beb7cc0faad732 To: <sip:1005@83.10.21.42> Contact: <sip:peta@192.168.2.99:5060> Supported: replaces, timer Proxy-Authorization: Digest username="peta", realm="asterisk", algorithm=MD5, uri="sip:1005@83.10.21.42", nonce="0e74039f", response="dd0ce75d46647fa8163f8a7b57127eee" Call-ID: d65e78c8ec10bfd3@192.168.2.99 CSeq: 20717 INVITE User-Agent: Grandstream GXP2000 1.1.1.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 252 v=0 o=peta 8000 8001 IN IP4 192.168.2.99 s=SIP Call c=IN IP4 192.168.2.99 t=0 0 m=audio 5004 RTP/AVP 3 8 18 4 0 a=sendrecv a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 --- (14 headers 13 lines) --- Using INVITE request as basis request - d65e78c8ec10bfd3@192.168.2.99 Sending to 192.168.2.99 : 5060 (NAT) Found user 'peta' Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 0 Peer audio RTP is at port 192.168.2.99:5004 Found description format GSM Found description format PCMA Found description format G729 Found description format G723 Found description format PCMU Capabilities: us - 0x11e (gsm|ulaw|alaw|g726|g729), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 1005 in internal (domain 83.10.21.42) list_route: hop: <sip:peta@192.168.2.99:5060> Transmitting (NAT) to 89.76.27.25:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.99:5060;branch=z9hG4bKbe09fc25bf3c4bf9;received=89.76.27.25 From: "peta" <sip:peta@83.10.21.42>;tag=d0beb7cc0faad732 To: <sip:1005@83.10.21.42> Call-ID: d65e78c8ec10bfd3@192.168.2.99 CSeq: 20717 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:1005@83.10.21.42> Content-Length: 0 --- -- Executing Dial("SIP/peta-0817f350", "Zap/g1/603961234||r") in new stack 1 -- Making new call for cr 131 -- Requested transfer capability: 0x00 - SPEECH 1 > Protocol Discriminator: Q.931 (8) len=33 1 > Call Ref: len= 1 (reference 3/0x3) (Originator) 1 > Message type: SETUP (5) 1 > [1 041 031 801 901 a31 ] 1 > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) 1 > Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 > Ext: 1 User information layer 1: A-Law (35) 1 > [1 181 011 811 ] 1 > Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Preferred Dchan: 0 1 > ChanSel: B1 channel 1 ] 1 > [1 6c1 061 411 811 701 651 741 611 ] 1 > Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1 > Presentation: Presentation permitted, user number passed network screening (1) 'peta' ] 1 > [1 701 0a1 a11 361 301 321 311 391 361 351 361 301 ] 1 > Called Number (len=12) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '603961234' ] 1 > [1 a11 ] 1 > Sending Complete (len= 1) -- Called g1/603961234 Transmitting (NAT) to 89.76.27.25:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.2.99:5060;branch=z9hG4bKbe09fc25bf3c4bf9;received=89.76.27.25 From: "peta" <sip:peta@83.10.21.42>;tag=d0beb7cc0faad732 To: <sip:1005@83.10.21.42>;tag=as727f0068 Call-ID: d65e78c8ec10bfd3@192.168.2.99 CSeq: 20717 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:1005@83.10.21.42> Content-Length: 0 --- 1 < Protocol Discriminator: Q.931 (8) len=12 1 < Call Ref: len= 1 (reference 131/0x83) (Terminator) 1 < Message type: STATUS (125) 1 < [1 081 031 821 e41 6c1 ] 1 < Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) 1 < Ext: 1 Cause: Unknown (100), class = Protocol Error (6) ] 1 < Cause data 1: 6c (108) 1 < [1 141 011 011 ] 1 < Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1) 1 -- Processing IE 8 (cs0, Cause) 1 -- Processing IE 20 (cs0, Call State) Oct 24 04:02:34 WARNING[6529]: chan_zap.c:8528 zt_pri_error: 1 updating callstate, peercallstate 2 to 1 1 < Protocol Discriminator: Q.931 (8) len=7 1 < Call Ref: len= 1 (reference 131/0x83) (Terminator) 1 < Message type: CALL PROCEEDING (2) 1 < [1 181 011 891 ] 1 < Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 1 < ChanSel: B1 channel 1 ] 1 -- Processing IE 24 (cs0, Channel Identification) -- Zap/1-1 is proceeding passing it to SIP/peta-0817f350 iprouter*CLI> 1 < Protocol Discriminator: Q.931 (8) len=4 1 < Call Ref: len= 1 (reference 131/0x83) (Terminator) 1 < Message type: ALERTING (1) -- Zap/1-1 is ringing iprouter*CLI> Destroying call '93c49078b92ea34a@192.168.2.99' ------------------------------------------------------------------------- Now we press cancel on SIP phone ------------------------------------------------------------------------- <-- SIP read from 89.76.27.25:5060: --- (0 headers 0 lines) Nat keepalive --- iprouter*CLI> <-- SIP read from 89.76.27.25:5060: CANCEL sip:1005@83.10.21.42 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.99:5060;branch=z9hG4bKbe09fc25bf3c4bf9 From: "peta" <sip:peta@83.10.21.42>;tag=d0beb7cc0faad732 To: <sip:1005@83.10.21.42> Call-ID: d65e78c8ec10bfd3@192.168.2.99 CSeq: 20717 CANCEL User-Agent: Grandstream GXP2000 1.1.1.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (10 headers 0 lines) --- Sending to 192.168.2.99 : 5060 (NAT) Reliably Transmitting (NAT) to 89.76.27.25:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.2.99:5060;branch=z9hG4bKbe09fc25bf3c4bf9;received=89.76.27.25 From: "peta" <sip:peta@83.10.21.42>;tag=d0beb7cc0faad732 To: <sip:1005@83.10.21.42>;tag=as727f0068 Call-ID: d65e78c8ec10bfd3@192.168.2.99 CSeq: 20717 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Transmitting (NAT) to 89.76.27.25:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.99:5060;branch=z9hG4bKbe09fc25bf3c4bf9;received=89.76.27.25 From: "peta" <sip:peta@83.10.21.42>;tag=d0beb7cc0faad732 To: <sip:1005@83.10.21.42>;tag=as727f0068 Call-ID: d65e78c8ec10bfd3@192.168.2.99 CSeq: 20717 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:1005@83.10.21.42> Content-Length: 0 --- 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Delivered, peerstate Call Received -- Hungup 'Zap/1-1' == Spawn extension (internal, 1005, 1) exited non-zero on 'SIP/peta-0817f350' iprouter*CLI> <-- SIP read from 89.76.27.25:5060: ACK sip:1005@83.10.21.42 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.99:5060;branch=z9hG4bKbe09fc25bf3c4bf9 From: "peta" <sip:peta@83.10.21.42>;tag=d0beb7cc0faad732 To: <sip:1005@83.10.21.42>;tag=as727f0068 Contact: <sip:peta@192.168.2.99:5060> Proxy-Authorization: Digest username="peta", realm="asterisk", algorithm=MD5, uri="sip:1005@83.10.21.42", nonce="0e74039f", response="fa004d4e666eb10a0605d9b4f4698905" Call-ID: d65e78c8ec10bfd3@192.168.2.99 CSeq: 20717 ACK User-Agent: Grandstream GXP2000 1.1.1.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (12 headers 0 lines) --- Destroying call 'd65e78c8ec10bfd3@192.168.2.99' Zap/1-1 didn't really hangup ... it continuous calling to called said. Called side is still ringing and ringing ... I must manually press cancel on called side to cancel ringing. ------------------------------------------------------------------------ | ||
Comments: | By: Russell Bryant (russell) 2006-10-24 15:09:13 Since this issue happens on a patched system, we can not help you. |