Summary: | ASTERISK-11864: Unistim call to VoiceMailMain application fails | ||
Reporter: | Perry Browning (perryb) | Labels: | |
Date Opened: | 2008-04-16 16:40:58 | Date Closed: | 2008-04-18 14:31:16 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_unistim |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | Unistim calls placed to VoiceMailMain extension ring once and drop. All other calls to and from Unistim set complete normally. See additional information for actual log and debug outputs. ****** STEPS TO REPRODUCE ****** 1. Define unistim set in unistim.conf 2. Define VoiceMailMain extension 3. Call VoiceMailMain extension ****** ADDITIONAL INFORMATION ****** ***** asterisk log ******** -- Device 'campus' successfuly registered -- Starting switch on '40085@campus-0' to 40089 -- Executing [40089@from-unistim-clients:1] Ringing("USTM/40085@campus-0", "") in new stack -- Executing [40089@from-unistim-clients:2] Wait("USTM/40085@campus-0", "2") in new stack -- Executing [40089@from-unistim-clients:3] VoiceMailMain("USTM/40085@campus-0", "") in new stack [Apr 16 13:18:50] WARNING[16968]: chan_unistim.c:3974 unistim_write: Asked to transmit frame type slin (64), while native formats is ulaw (4) (read/write = ulaw (4)/4) [Apr 16 13:18:50] WARNING[16968]: res_adsi.c:181 adsi_careful_send: Failed to carefully write frame [Apr 16 13:18:50] WARNING[16968]: res_adsi.c:214 __adsi_transmit_messages: Unable to send CAS [Apr 16 13:18:50] WARNING[16968]: file.c:662 ast_readaudio_callback: Failed to write frame -- <USTM/40085@campus-0> Playing 'vm-login.ulaw' (language 'en') [Apr 16 13:18:50] WARNING[16968]: app_voicemail.c:6858 vm_authenticate: Couldn't stream login file [Apr 16 13:18:50] WARNING[16968]: res_adsi.c:214 __adsi_transmit_messages: Unable to send CAS [Apr 16 13:18:50] WARNING[16968]: res_adsi.c:214 __adsi_transmit_messages: Unable to send CAS USTM(40085@campus-0) channel already destroyed ****** unistim debug ******* Request received Key pressed : keycode = 0x61 - current state : 2 Sending select output packet output=c2 volume=1 mute=0 > Sending datas with seq #0x010d Using slot #0 : Sending led_update (18) > Sending datas with seq #0x010e Using slot #1 : Sending led_update (10) > Sending datas with seq #0x010f Using slot #2 : Sending led_update (9) > Sending datas with seq #0x0110 Using slot #3 : Sending favorite pos 0 with status 0x28 > Sending datas with seq #0x0111 Using slot #4 : unistim_new sub=0 (0xb7b42318) chan=0x93bae30 Best codec = 4 from nativeformats 12 (line cap=12 global=12) Starting RTP. Bind on 128.xxx.xxx.xx RTP started : Our IP/port is : 128.xxx.xxx.xx:19300 with codec ulaw (4) Starting phone RTP stack. Our public IP is 128.xxx.xxx.xx Sending packet_send_rtp_packet_size for codec 0 > Sending datas with seq #0x0112 Using slot ASTERISK-1 : Sending Jitter Buffer Parameters Configuration > Sending datas with seq #0x0113 Using slot ASTERISK-2 : Sending packet_send_call default method > Sending datas with seq #0x0114 Using slot ASTERISK-3 : Sending select output packet output=c2 volume=1 mute=0 > Sending datas with seq #0x0115 Using slot ASTERISK-4 : Sending led_update (18) > Sending datas with seq #0x0116 Using slot ASTERISK-5 : Sending led_update (10) > Sending datas with seq #0x0117 Using slot ASTERISK-6 : Sending led_update (9) > Sending datas with seq #0x0118 Using slot ASTERISK-7 : Sending favorite pos 0 with status 0x28 > Sending datas with seq #0x0119 Using slot ASTERISK-8 : Sending text at pos 0, inverse flag 5 > Sending datas with seq #0x011a Using slot ASTERISK-9 : Sending text at pos 32, inverse flag 5 > Sending datas with seq #0x011b Using slot ASTERISK-10 : Sending text at pos 64, inverse flag 5 > Sending datas with seq #0x011c Using slot ASTERISK-11 : Sending status text > Sending datas with seq #0x011d Using slot ASTERISK-12 : -- Starting switch on '40085@campus-0' to 40089 -- Executing [40089@from-unistim-clients:1] Ringing("USTM/40085@campus-0", "") in new stack -- Asked to indicate 'Remote end is ringing' condition on channel USTM/40085@campus-0 Sending text at pos 64, inverse flag 5 > Sending datas with seq #0x011e Using slot ASTERISK-13 : -- Executing [40089@from-unistim-clients:2] Wait("USTM/40085@campus-0", "2") in new stack > ACK received for packet #0x010d > We still have packets in our send queue > ACK received for packet #0x010e > We still have packets in our send queue > ACK received for packet #0x010f > We still have packets in our send queue > ACK received for packet #0x0110 > We still have packets in our send queue > ACK received for packet #0x0111 > We still have packets in our send queue > ACK received for packet #0x0112 > We still have packets in our send queue > ACK received for packet #0x0113 > We still have packets in our send queue > ACK received for packet #0x0114 > We still have packets in our send queue > ACK received for packet #0x0115 > We still have packets in our send queue > ACK received for packet #0x0116 > We still have packets in our send queue > ACK received for packet #0x0117 > We still have packets in our send queue > ACK received for packet #0x0118 > We still have packets in our send queue > ACK received for packet #0x0119 > We still have packets in our send queue > ACK received for packet #0x011a > We still have packets in our send queue > ACK received for packet #0x011b > We still have packets in our send queue > ACK received for packet #0x011c > We still have packets in our send queue > ACK received for packet #0x011d > We still have packets in our send queue > ACK received for packet #0x011e > Our send queue is completely ACKed. Request received -- Executing [40089@from-unistim-clients:3] VoiceMailMain("USTM/40085@campus-0", "") in new stack unistim_answer(USTM/40085@campus-0) on 40085@campus-0 Sending text at pos 64, inverse flag 5 > Sending datas with seq #0x011f Using slot #0 : Sending status text > Sending datas with seq #0x0120 Using slot #1 : Sending start timer > Sending datas with seq #0x0121 Using slot #2 : > ACK received for packet #0x011f > We still have packets in our send queue > ACK received for packet #0x0120 > We still have packets in our send queue > ACK received for packet #0x0121 > Our send queue is completely ACKed. [Apr 16 14:25:30] WARNING[16968]: chan_unistim.c:3974 unistim_write: Asked to transmit frame type slin (64), while native formats is ulaw (4) (read/write = ulaw (4)/4) [Apr 16 14:25:30] WARNING[16968]: res_adsi.c:181 adsi_careful_send: Failed to carefully write frame [Apr 16 14:25:30] WARNING[16968]: res_adsi.c:214 __adsi_transmit_messages: Unable to send CAS [Apr 16 14:25:30] WARNING[16968]: file.c:662 ast_readaudio_callback: Failed to write frame -- <USTM/40085@campus-0> Playing 'vm-login.ulaw' (language 'en') [Apr 16 14:25:30] WARNING[16968]: app_voicemail.c:6858 vm_authenticate: Couldn't stream login file [Apr 16 14:25:30] WARNING[16968]: res_adsi.c:214 __adsi_transmit_messages: Unable to send CAS [Apr 16 14:25:30] WARNING[16968]: res_adsi.c:214 __adsi_transmit_messages: Unable to send CAS unistim_hangup(USTM/40085@campus-0) on 40085@campus Sending no ring packet > Sending datas with seq #0x0122 Using slot #0 : Sending end call > Sending datas with seq #0x0123 Using slot #1 : Destroying RTP session Sending stop timer > Sending datas with seq #0x0124 Using slot #2 : USTM(40085@campus-0) channel already destroyed Sending Stream Based Tone Off > Sending datas with seq #0x0125 Using slot #3 : Sending select output packet output=c2 volume=1 mute=ce > Sending datas with seq #0x0126 Using slot #4 : Sending led_update (10) > Sending datas with seq #0x0127 Using slot ASTERISK-1 : Sending led_update (9) > Sending datas with seq #0x0128 Using slot ASTERISK-2 : Sending favorite pos 0 with status 0x28 > Sending datas with seq #0x0129 Using slot ASTERISK-3 : Sending favorite pos 0 with status 0x20 > Sending datas with seq #0x012a Using slot ASTERISK-4 : Sending status text > Sending datas with seq #0x012b Using slot ASTERISK-5 : Sending text at pos 32, inverse flag 5 > Sending datas with seq #0x012c Using slot ASTERISK-6 : Sending text at pos 0, inverse flag 5 > Sending datas with seq #0x012d Using slot ASTERISK-7 : Sending text at pos 64, inverse flag 5 > Sending datas with seq #0x012e Using slot ASTERISK-8 : Sending title text > Sending datas with seq #0x012f Using slot ASTERISK-9 : Sending favorite pos 0 with status 0x20 > Sending datas with seq #0x0130 Using slot ASTERISK-10 : > ACK received for packet #0x0122 > We still have packets in our send queue > ACK received for packet #0x0125 169.237.213.15 ACK gap : Received ACK #0x0125, previous was #0x0122 > We still have packets in our send queue > ACK received for packet #0x0126 > We still have packets in our send queue > ACK received for packet #0x0127 > We still have packets in our send queue > ACK received for packet #0x0128 > We still have packets in our send queue > ACK received for packet #0x0129 > We still have packets in our send queue > ACK received for packet #0x012a > We still have packets in our send queue > ACK received for packet #0x012b > We still have packets in our send queue > ACK received for packet #0x012c > We still have packets in our send queue > ACK received for packet #0x012d > We still have packets in our send queue > ACK received for packet #0x012e > We still have packets in our send queue > ACK received for packet #0x012f > We still have packets in our send queue > ACK received for packet #0x0130 > Our send queue is completely ACKed. Request received > Sending ping > Sending datas with seq #0x0131 Using slot #0 : > ACK received for packet #0x0131 > Our single packet was ACKed. sip2*CLI> unistim set debug off UNISTIM Debugging Disabled sip2*CLI> | ||
Comments: | By: Digium Subversion (svnbot) 2008-04-18 14:30:22 Repository: asterisk Revision: 114271 U trunk/channels/chan_unistim.c ------------------------------------------------------------------------ r114271 | file | 2008-04-18 14:30:21 -0500 (Fri, 18 Apr 2008) | 4 lines Make sure ADSI is marked as unavailable on Unistim channels so voicemail does not try to do some ADSI jazz. (closes issue ASTERISK-11864) Reported by: PerryB ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=114271 By: Digium Subversion (svnbot) 2008-04-18 14:31:16 Repository: asterisk Revision: 114272 _U branches/1.6.0/ U branches/1.6.0/channels/chan_unistim.c ------------------------------------------------------------------------ r114272 | file | 2008-04-18 14:31:15 -0500 (Fri, 18 Apr 2008) | 12 lines Merged revisions 114271 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114271 | file | 2008-04-18 16:35:33 -0300 (Fri, 18 Apr 2008) | 4 lines Make sure ADSI is marked as unavailable on Unistim channels so voicemail does not try to do some ADSI jazz. (closes issue ASTERISK-11864) Reported by: PerryB ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=114272 |