Summary: | ASTERISK-09831: if asterisk is received get perpetual dtmf | ||
Reporter: | John Covici (covici) | Labels: | |
Date Opened: | 2007-07-07 09:55:08 | Date Closed: | 2011-06-07 14:02:53 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_zap |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ( 0) rtp-debug.log | |
Description: | if I call my cell phone using asterisk, and if te cell phone sends * a perpetual dtmf exists till the call is hung up. ****** ADDITIONAL INFORMATION ****** I am using a provider with SIP inband dtmf. If I call using pstn this does not happen. I am using a Digium 400p -- fxs port for the handset and fxo to pstn. | ||
Comments: | By: John Covici (covici) 2007-07-07 10:03:31 Here is the log output. [Jul 3 03:23:07] VERBOSE[26933] logger.c: -- Executing [s@macro-dialout-trunk:14] Dial("Zap/1-1", "SIP/galaxyvoice/7034314045||TW") in new stack [Jul 3 03:23:07] VERBOSE[26933] logger.c: -- Called galaxyvoice/7034314045 [Jul 3 03:23:07] VERBOSE[26933] logger.c: -- SIP/galaxyvoice-0820cdc0 is making progress passing it to Zap/1-1 [Jul 3 03:23:07] DEBUG[26933] chan_zap.c: Received AST_CONTROL_PROGRESS on Zap/1-1 [Jul 3 03:23:23] NOTICE[26851] chan_iax2.c: Peer 'covici' is not dynamic (from 140.239.173.226) [Jul 3 03:23:25] VERBOSE[26933] logger.c: -- SIP/galaxyvoice-0820cdc0 answered Zap/1-1 [Jul 3 03:23:25] DEBUG[26933] chan_zap.c: Took Zap/1-1 off hook [Jul 3 03:23:35] DTMF[26933] channel.c: DTMF end '*' received on SIP/galaxyvoice-0820cdc0, duration 0 ms [Jul 3 03:23:35] DTMF[26933] channel.c: DTMF begin emulation of '*' with duration 100 queued on SIP/galaxyvoice-0820cdc0 [Jul 3 03:23:35] DEBUG[26933] chan_zap.c: Started VLDTMF digit '*' [Jul 3 03:23:46] VERBOSE[26933] logger.c: == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'Zap/1-1' in macro 'dialout-trunk' [Jul 3 03:23:46] VERBOSE[26933] logger.c: == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'Zap/1-1' [Jul 3 03:23:46] VERBOSE[26933] logger.c: -- Executing [h@macro-dialout-trunk:1] Macro("Zap/1-1", "hangupcall") in new stack [Jul 3 03:23:46] VERBOSE[26933] logger.c: -- Executing [s@macro-hangupcall:1] ResetCDR("Zap/1-1", "w") in new stack [Jul 3 03:23:46] DEBUG[26933] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record. [Jul 3 03:23:46] DEBUG[26933] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2007-07-03 03:23:01','\"John Covici\" <7037775986>','7037775986','7034314045','from-internal', 'Zap/1-1','SIP/galaxyvoice-0820cdc0','ResetCDR','w',45,21,'ANSWERED',3,'') [Jul 3 03:23:46] DEBUG[26933] app_macro.c: Executed application: ResetCDR [Jul 3 03:23:46] VERBOSE[26933] logger.c: -- Executing [s@macro-hangupcall:2] NoCDR("Zap/1-1", "") in new stack [Jul 3 03:23:46] DEBUG[26933] app_macro.c: Executed application: NoCDR [Jul 3 03:23:46] VERBOSE[26933] logger.c: -- Executing [s@macro-hangupcall:3] GotoIf("Zap/1-1", "1?skiprg") in new stack [Jul 3 03:23:46] VERBOSE[26933] logger.c: -- Goto (macro-hangupcall,s,6) [Jul 3 03:23:46] DEBUG[26933] app_macro.c: Executed application: GotoIf [Jul 3 03:23:46] VERBOSE[26933] logger.c: -- Executing [s@macro-hangupcall:6] GotoIf("Zap/1-1", "1?theend") in new stack [Jul 3 03:23:46] VERBOSE[26933] logger.c: -- Goto (macro-hangupcall,s,9) [Jul 3 03:23:46] DEBUG[26933] app_macro.c: Executed application: GotoIf [Jul 3 03:23:46] VERBOSE[26933] logger.c: -- Executing [s@macro-hangupcall:9] Wait("Zap/1-1", "5") in new stack [Jul 3 03:23:51] DEBUG[26933] app_macro.c: Executed application: Wait [Jul 3 03:23:51] VERBOSE[26933] logger.c: -- Executing [s@macro-hangupcall:10] Hangup("Zap/1-1", "") in new stack [Jul 3 03:23:51] VERBOSE[26933] logger.c: == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'Zap/1-1' in macro 'hangupcall' [Jul 3 03:23:51] VERBOSE[26933] logger.c: == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'Zap/1-1' [Jul 3 03:23:51] VERBOSE[26933] logger.c: -- Hungup 'Zap/1-1' [Jul 3 03:23:55] DEBUG[26847] chan_zap.c: Message status for 200 changed from -1 to 0 on 1 [Jul 3 03:24:13] NOTICE[26856] chan_iax2.c: Peer 'covici' is not dynamic (from 140.239.173.226) Now what I tried just for grins, was to route the call out my fxo rather than my sip provider and I did not have the dtmf problem when doing that. Here is the log output for that one. [Jul 3 03:32:10] VERBOSE[27683] logger.c: -- Executing [s@macro-dialout-trunk:14] Dial("Zap/1-1", "ZAP/4/www7034314045||TW") in new stack [Jul 3 03:32:10] DEBUG[27683] chan_zap.c: Dialing 'www7034314045' [Jul 3 03:32:10] DEBUG[27683] chan_zap.c: Deferring dialing... [Jul 3 03:32:10] VERBOSE[27683] logger.c: -- Called 4/www7034314045 [Jul 3 03:32:14] DEBUG[27683] chan_zap.c: Engaged echo training on channel 4 [Jul 3 03:32:15] DEBUG[27683] chan_zap.c: Echo cancellation already on [Jul 3 03:32:15] VERBOSE[27683] logger.c: -- Zap/4-1 answered Zap/1-1 [Jul 3 03:32:15] DEBUG[27683] chan_zap.c: Took Zap/1-1 off hook [Jul 3 03:32:34] NOTICE[26852] chan_iax2.c: Peer 'covici' is not dynamic (from 140.239.173.226) [Jul 3 03:32:37] DTMF[27683] channel.c: DTMF end '*' received on Zap/4-1, duration 0 ms [Jul 3 03:32:37] DTMF[27683] channel.c: DTMF begin emulation of '*' with duration 100 queued on Zap/4-1 [Jul 3 03:32:37] DEBUG[27683] chan_zap.c: Started VLDTMF digit '*' [Jul 3 03:32:37] DTMF[27683] channel.c: DTMF end emulation of '*' queued on Zap/4-1 [Jul 3 03:32:37] DEBUG[27683] chan_zap.c: Ending VLDTMF digit '*' [Jul 3 03:32:40] DTMF[27683] channel.c: DTMF end '*' received on Zap/4-1, duration 0 ms [Jul 3 03:32:40] DTMF[27683] channel.c: DTMF begin emulation of '*' with duration 100 queued on Zap/4-1 [Jul 3 03:32:40] DEBUG[27683] chan_zap.c: Started VLDTMF digit '*' [Jul 3 03:32:40] DTMF[27683] channel.c: DTMF end emulation of '*' queued on Zap/4-1 [Jul 3 03:32:40] DEBUG[27683] chan_zap.c: Ending VLDTMF digit '*' [Jul 3 03:32:55] VERBOSE[27683] logger.c: -- Hungup 'Zap/4-1' By: Joshua C. Colp (jcolp) 2007-07-08 20:18:08 Are you using internal timing? (it would be enabled in asterisk.conf) Can you attach an rtp debug so I can see the stream from your provider? By: John Covici (covici) 2007-07-10 05:58:52 Timing is not mentioned in asterisk.conf and I am using a zaptel card, so I imagine the timing would come from there. I am attaching the debug output for rtp from the time the call is answered till the hangup. By: Joshua C. Colp (jcolp) 2007-07-10 08:16:13 Please turn on internal_timing in asterisk.conf and try again. By: John Covici (covici) 2007-07-11 02:40:16 I turned on internal timing by putting an [options] section in asterisk.conf and the line internal_timing = yes but it did not make a difference. By: Mike Loebl (mloebl) 2007-09-12 20:20:57 I'm seeing a very similiar problem here with a 400P card and GalaxyVoice as well. If I dial in or receive a call from one of the analog lines on the 400P, it will randomly get stuck transmitting a tone until I hang up. If I use a SIP phone, the problem does not occur. By: John Covici (covici) 2007-09-12 22:52:52 With me, using an analog line to the cell phone directly works, using Galaxyvoice does not! I may sign up with another provider to see if that makes a difference. I will let you know if it does. By: John Covici (covici) 2007-09-16 16:03:21 I have discovered something interesting concerning this bug -- I have been using another SIP provider and I am not seeing the problem with the new provider. Very strange! By: deeperror (deeperror) 2007-10-03 14:26:49 Also experiencing same issue with 1.4.11 and rhino r4t1 card. [Oct 3 14:02:12] DEBUG[16892] chan_zap.c: Started VLDTMF digit '8' By: Joshua C. Colp (jcolp) 2007-12-18 10:09:48.000-0600 covici: Quite strange indeed... I saw nothing to indicate why it's happening... deeperror: There have been a few DTMF improvements since that version, and without full information it is hard to determine things. By: Jason Parker (jparker) 2007-12-26 15:44:04.000-0600 Closing. This appears to have been a problem with a specific voip provider. deeperror, if you can reproduce your issue, please contact a bug marshal to get this bug reopened, or open a new issue. |