Voicemail Greeting/Message does not play

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Max Clark

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Apr 13, 2015, 6:43:24 PM4/13/15
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Hello,

After being deposited to voicemail the greeting/message does not play for the caller. What's strange is that the "if you are happy with this message" prompt does play. Where should I look to troubleshoot this?

Thanks,
Max

George Niculae

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Apr 13, 2015, 6:45:14 PM4/13/15
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On Tue, Apr 14, 2015 at 12:43 AM, Max Clark <max....@gmail.com> wrote:
Hello,

After being deposited to voicemail the greeting/message does not play for the caller. What's strange is that the "if you are happy with this message" prompt does play. Where should I look to troubleshoot this?


Typically in /var/log/sipxpbx/sipxivr.log file and freeswitch.log
Is the custom greeting what you expect to be played and is not?

George 

Max Clark

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Apr 13, 2015, 6:50:16 PM4/13/15
to George Niculae, sipxco...@googlegroups.com
No greeting is being played - just silence.

George Niculae

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Apr 13, 2015, 6:51:21 PM4/13/15
to Max Clark, sipxco...@googlegroups.com
On Tue, Apr 14, 2015 at 12:50 AM, Max Clark <max....@gmail.com> wrote:
No greeting is being played - just silence.


Does it happen for all users or only for some of them? 

Max Clark

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Apr 13, 2015, 6:52:13 PM4/13/15
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3/30 for certain - haven't dialed the rest to test yet.

George Niculae

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Apr 13, 2015, 6:54:56 PM4/13/15
to Max Clark, sipxco...@googlegroups.com
On Tue, Apr 14, 2015 at 12:52 AM, Max Clark <max....@gmail.com> wrote:
3/30 for certain - haven't dialed the rest to test yet.


OK, probably to all of them then .. check those logs and also file permissions on user mailboxes. You could also try recording a new greeting for one of those users and see if it is played 

Max Clark

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Apr 15, 2015, 3:54:14 PM4/15/15
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Some more information on this issue...

Calls from PSTN to a DID gives ringing, then voicemail greeting and recording.

Calls from PSTN to a DID that is answered, and is then blind transferred to another extension, the caller hears hold music, then nothing, then "if you are satisfied with your message...".

Calls from PSTN to a DID that is answered, and is then blind transferred to 8 + ext, the caller hears brief hold music and then the the voicemail greeting.

I'm pretty sure this isn't a file permission issue, and the extension definitely has a greeting recorded. Where do I go from here?

George Niculae

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Apr 15, 2015, 3:56:19 PM4/15/15
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Did you spot anything in those logs? Maybe codec issue?

George 

Max Clark

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Apr 15, 2015, 5:13:00 PM4/15/15
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Here's the sipxivr.log for a blind transfer to another user where the voicemail greeting was not played.

[root@pbx sipxpbx]# grep Thread-1794 sipxivr.log
"2015-04-15T21:00:26.742000Z":19296:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"SipXivr::run Accepting call-id 485d2ca0-5e55-1233-3997-c03fd56ce398 from 2...@192.168.1.10 to I...@192.168.1.10:15060"
"2015-04-15T21:00:27.809000Z":19297:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Starting voicemail for mailbox \"207\" action=\"deposit"
"2015-04-15T21:00:27.810000Z":19298:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Mailbox 207 Standard Greeting"
"2015-04-15T21:00:27.810000Z":19299:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Mailbox org.sipfoundry.commons.userdb.User@53ef7ba0 Deposit Voicemail from \"Kara\" <sip:2...@192.168.1.10>"
"2015-04-15T21:00:27.810000Z":19300:sipXivr:WARNING:pbx:Thread-1794:00000000:ResourceBundleMessageSource:"ResourceBundle [EmailFormats] not found for MessageSource: Can't find bundle for base name EmailFormats, locale en"
"2015-04-15T21:00:38.287000Z":19302:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Collect::start 1 100/0/0 mask 1234567890ABCD#*i"
"2015-04-15T21:00:38.408000Z":19303:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"depositVoicemail Collected digits="
"2015-04-15T21:00:49.102000Z":19316:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Mailbox 207 Deposit Voicemail recorded message"
"2015-04-15T21:00:55.743000Z":19323:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Collect::start 1 7000/0/0 mask 1234567890ABCD#*i"
"2015-04-15T21:00:55.744000Z":19324:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"FSESI::invoke throw DisconnectException"
"2015-04-15T21:00:56.744000Z":19327:sipXivr:WARNING:pbx:Thread-1794:00000000:ResourceBundleMessageSource:"ResourceBundle [EmailFormats] not found for MessageSource: Can't find bundle for base name EmailFormats, locale en"
"2015-04-15T21:00:56.745000Z":19328:sipXivr:WARNING:pbx:Thread-1794:00000000:ResourceBundleMessageSource:"ResourceBundle [EmailFormats] not found for MessageSource: Can't find bundle for base name EmailFormats, locale en"
"2015-04-15T21:00:56.748000Z":19330:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"VmMessage::newMessage created message /var/sipxdata/mediaserver/data/mailstore/207/inbox/100001840-00.xml"
"2015-04-15T21:00:56.749000Z":19331:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Emailer::queueVm2Email queuing e-mail for 2...@foo.com"
"2015-04-15T21:00:56.749000Z":19332:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Ending voicemail

Here is the freeswitch.log from the same period of time. The uuid doesn't change in the log, so I don't see a way to to filter this easily.

69a17cf6-e3b2-11e4-b37d-1fc9933e9a27 2015-04-15 14:00:41.752172 [NOTICE] mod_dptools.c:1232 Hangup sofia/foo.com/+18189...@192.168.1.10 [CS_EXECUTE] [NORMAL_CLEARING]
69a14fb0-e3b2-11e4-b379-1fc9933e9a27 2015-04-15 14:00:41.752172 [NOTICE] sofia.c:926 Hangup sofia/foo.com/sip:I...@192.168.1.10:15060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
69a14fb0-e3b2-11e4-b379-1fc9933e9a27 2015-04-15 14:00:41.772187 [NOTICE] switch_core_session.c:1632 Session 2689 (sofia/foo.com/sip:I...@192.168.1.10:15060) Ended
69a14fb0-e3b2-11e4-b379-1fc9933e9a27 2015-04-15 14:00:41.772187 [NOTICE] switch_core_session.c:1636 Close Channel sofia/foo.com/sip:I...@192.168.1.10:15060 [CS_DESTROY]
69a17cf6-e3b2-11e4-b37d-1fc9933e9a27 2015-04-15 14:00:41.772187 [NOTICE] switch_core_session.c:1632 Session 2690 (sofia/foo.com/+18189...@192.168.1.10) Ended
69a17cf6-e3b2-11e4-b37d-1fc9933e9a27 2015-04-15 14:00:41.772187 [NOTICE] switch_core_session.c:1636 Close Channel sofia/foo.com/+18189...@192.168.1.10 [CS_DESTROY]
7a6f8514-e3b2-11e4-b3d7-1fc9933e9a27 2015-04-15 14:00:42.572177 [NOTICE] switch_channel.c:1053 New Channel sofia/foo.com/~~id~bri...@foo.com [7a6f8514-e3b2-11e4-b3d7-1fc9933e9a27]
7a73276e-e3b2-11e4-b3df-1fc9933e9a27 2015-04-15 14:00:42.592206 [NOTICE] switch_channel.c:1053 New Channel sofia/foo.com/sip:I...@192.168.1.10:15060 [7a73276e-e3b2-11e4-b3df-1fc9933e9a27]
7a735298-e3b2-11e4-b3e3-1fc9933e9a27 2015-04-15 14:00:42.592206 [NOTICE] switch_channel.c:1053 New Channel sofia/foo.com/~~id~bri...@192.168.1.10 [7a735298-e3b2-11e4-b3e3-1fc9933e9a27]
7a735298-e3b2-11e4-b3e3-1fc9933e9a27 2015-04-15 14:00:42.612228 [NOTICE] sofia_media.c:92 Pre-Answer sofia/foo.com/~~id~bri...@192.168.1.10!
7a735298-e3b2-11e4-b3e3-1fc9933e9a27 2015-04-15 14:00:42.612228 [NOTICE] mod_dptools.c:1258 Channel [sofia/foo.com/~~id~bri...@192.168.1.10] has been answered
7a73276e-e3b2-11e4-b3df-1fc9933e9a27 2015-04-15 14:00:42.612228 [NOTICE] sofia.c:7057 Channel [sofia/foo.com/sip:I...@192.168.1.10:15060] has been answered
7a6f8514-e3b2-11e4-b3d7-1fc9933e9a27 2015-04-15 14:00:42.632201 [NOTICE] sofia_media.c:92 Pre-Answer sofia/foo.com/~~id~bri...@foo.com!
7a6f8514-e3b2-11e4-b3d7-1fc9933e9a27 2015-04-15 14:00:42.632201 [NOTICE] switch_ivr_originate.c:3493 Channel [sofia/foo.com/~~id~bri...@foo.com] has been answered
7cdbf59e-e3b2-11e4-b3ec-1fc9933e9a27 2015-04-15 14:00:46.632185 [NOTICE] switch_channel.c:1053 New Channel sofia/foo.com/2...@192.168.1.10 [7cdbf59e-e3b2-11e4-b3ec-1fc9933e9a27]
7cdbf59e-e3b2-11e4-b3ec-1fc9933e9a27 2015-04-15 14:00:46.632185 [NOTICE] sofia_media.c:92 Pre-Answer sofia/foo.com/2...@192.168.1.10!
7cdbf59e-e3b2-11e4-b3ec-1fc9933e9a27 2015-04-15 14:00:46.632185 [NOTICE] mod_dptools.c:1258 Channel [sofia/foo.com/2...@192.168.1.10] has been answered
70fb54a4-e3b2-11e4-b3b0-1fc9933e9a27 2015-04-15 14:00:46.632185 [NOTICE] switch_ivr_bridge.c:1979 Hangup sofia/foo.com/2...@192.168.1.10 [CS_PARK] [PICKED_OFF]
70fb54a4-e3b2-11e4-b3b0-1fc9933e9a27 2015-04-15 14:00:46.652189 [NOTICE] switch_core_session.c:1632 Session 2697 (sofia/foo.com/2...@192.168.1.10) Ended
70fb54a4-e3b2-11e4-b3b0-1fc9933e9a27 2015-04-15 14:00:46.652189 [NOTICE] switch_core_session.c:1636 Close Channel sofia/foo.com/2...@192.168.1.10 [CS_DESTROY]
6f30c0d2-e3b2-11e4-b39b-1fc9933e9a27 2015-04-15 14:00:46.692174 [NOTICE] sofia.c:926 Hangup sofia/foo.com/~~id~bri...@foo.com [CS_EXECUTE] [NORMAL_CLEARING]
6f34446e-e3b2-11e4-b3a3-1fc9933e9a27 2015-04-15 14:00:46.692174 [NOTICE] switch_ivr_bridge.c:753 Hangup sofia/foo.com/sip:I...@192.168.1.10:15060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
6f34446e-e3b2-11e4-b3a3-1fc9933e9a27 2015-04-15 14:00:46.692174 [NOTICE] switch_core_session.c:1632 Session 2695 (sofia/foo.com/sip:I...@192.168.1.10:15060) Ended
6f34446e-e3b2-11e4-b3a3-1fc9933e9a27 2015-04-15 14:00:46.692174 [NOTICE] switch_core_session.c:1636 Close Channel sofia/foo.com/sip:I...@192.168.1.10:15060 [CS_DESTROY]
6f30c0d2-e3b2-11e4-b39b-1fc9933e9a27 2015-04-15 14:00:46.692174 [NOTICE] switch_core_session.c:1632 Session 2694 (sofia/foo.com/~~id~bri...@foo.com) Ended
6f30c0d2-e3b2-11e4-b39b-1fc9933e9a27 2015-04-15 14:00:46.692174 [NOTICE] switch_core_session.c:1636 Close Channel sofia/foo.com/~~id~bri...@foo.com [CS_DESTROY]
6f346bce-e3b2-11e4-b3a7-1fc9933e9a27 2015-04-15 14:00:46.712181 [NOTICE] sofia.c:926 Hangup sofia/foo.com/~~id~bri...@192.168.1.10 [CS_EXECUTE] [NORMAL_CLEARING]
6f346bce-e3b2-11e4-b3a7-1fc9933e9a27 2015-04-15 14:00:46.712181 [NOTICE] switch_core_session.c:1632 Session 2696 (sofia/foo.com/~~id~bri...@192.168.1.10) Ended
6f346bce-e3b2-11e4-b3a7-1fc9933e9a27 2015-04-15 14:00:46.712181 [NOTICE] switch_core_session.c:1636 Close Channel sofia/foo.com/~~id~bri...@192.168.1.10 [CS_DESTROY]
7eb424f4-e3b2-11e4-b3f7-1fc9933e9a27 2015-04-15 14:00:49.732183 [NOTICE] switch_channel.c:1053 New Channel sofia/foo.com/+16619...@192.168.13.114 [7eb424f4-e3b2-11e4-b3f7-1fc9933e9a27]
7eb7ae94-e3b2-11e4-b3ff-1fc9933e9a27 2015-04-15 14:00:49.752234 [NOTICE] switch_channel.c:1053 New Channel sofia/foo.com/sip:I...@192.168.1.10:15060 [7eb7ae94-e3b2-11e4-b3ff-1fc9933e9a27]
7eb7d7ac-e3b2-11e4-b403-1fc9933e9a27 2015-04-15 14:00:49.752234 [NOTICE] switch_channel.c:1053 New Channel sofia/foo.com/+16619...@192.168.1.10 [7eb7d7ac-e3b2-11e4-b403-1fc9933e9a27]
7eb7d7ac-e3b2-11e4-b403-1fc9933e9a27 2015-04-15 14:00:49.772175 [NOTICE] sofia_media.c:92 Pre-Answer sofia/foo.com/+16619...@192.168.1.10!
7eb7d7ac-e3b2-11e4-b403-1fc9933e9a27 2015-04-15 14:00:49.772175 [NOTICE] mod_dptools.c:1258 Channel [sofia/foo.com/+16619...@192.168.1.10] has been answered
7eb7ae94-e3b2-11e4-b3ff-1fc9933e9a27 2015-04-15 14:00:49.772175 [NOTICE] sofia.c:7057 Channel [sofia/foo.com/sip:I...@192.168.1.10:15060] has been answered
7eb424f4-e3b2-11e4-b3f7-1fc9933e9a27 2015-04-15 14:00:49.792204 [NOTICE] sofia_media.c:92 Pre-Answer sofia/foo.com/+16619...@192.168.13.114!
7eb424f4-e3b2-11e4-b3f7-1fc9933e9a27 2015-04-15 14:00:49.792204 [NOTICE] switch_ivr_originate.c:3493 Channel [sofia/foo.com/+16619...@192.168.13.114] has been answered
699dc692-e3b2-11e4-b371-1fc9933e9a27 2015-04-15 14:00:51.072200 [NOTICE] mod_sofia.c:1890 Hangup sofia/foo.com/+18189...@192.168.13.114 [CS_EXECUTE] [BLIND_TRANSFER]
699dc692-e3b2-11e4-b371-1fc9933e9a27 2015-04-15 14:00:51.072200 [NOTICE] switch_core_session.c:1632 Session 2688 (sofia/foo.com/+18189...@192.168.13.114) Ended
699dc692-e3b2-11e4-b371-1fc9933e9a27 2015-04-15 14:00:51.072200 [NOTICE] switch_core_session.c:1636 Close Channel sofia/foo.com/+18189...@192.168.13.114 [CS_DESTROY]

George Niculae

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Apr 15, 2015, 5:19:42 PM4/15/15
to Max Clark, sipxco...@googlegroups.com
You will have to put both logs on debug and recreate the issue / send logs, that is IVR in System > Voicemail > Logging Level and for FS in System > media Services, click on server name > Debug option.
Make sure you put them back on default values after reproducing the issue

George

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Max Clark

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Apr 15, 2015, 6:06:48 PM4/15/15
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George,

The logs are large and unfiltered:

freeswitch.log https://dpaste.de/oJJA

The outside caller called the autoattendant, transfered to extension 236 (talked to person), then was blind transfered to 225. On the transfer from 236 to 225 hold music was played, the 225 voicemail greeting was not, and then the "if you are satisfied with your message".

Thanks!
Max


On Wednesday, April 15, 2015 at 2:19:42 PM UTC-7, George Niculae wrote:
You will have to put both logs on debug and recreate the issue / send logs, that is IVR in System > Voicemail > Logging Level and for FS in System > media Services, click on server name > Debug option.
Make sure you put them back on default values after reproducing the issue

George
On Wed, Apr 15, 2015 at 11:13 PM, Max Clark <max....@gmail.com> wrote:
Here's the sipxivr.log for a blind transfer to another user where the voicemail greeting was not played.

[root@pbx sipxpbx]# grep Thread-1794 sipxivr.log
"2015-04-15T21:00:26.742000Z":19296:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"SipXivr::run Accepting call-id 485d2ca0-5e55-1233-3997-c03fd56ce398 from 2...@192.168.1.10 to I...@192.168.1.10:15060"
"2015-04-15T21:00:27.809000Z":19297:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Starting voicemail for mailbox \"207\" action=\"deposit"
"2015-04-15T21:00:27.810000Z":19298:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Mailbox 207 Standard Greeting"
"2015-04-15T21:00:27.810000Z":19299:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Mailbox org.sipfoundry.commons.userdb.User@53ef7ba0 Deposit Voicemail from \"Kara\" <sip...@192.168.1.10>"
"2015-04-15T21:00:27.810000Z":19300:sipXivr:WARNING:pbx:Thread-1794:00000000:ResourceBundleMessageSource:"ResourceBundle [EmailFormats] not found for MessageSource: Can't find bundle for base name EmailFormats, locale en"
"2015-04-15T21:00:38.287000Z":19302:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Collect::start 1 100/0/0 mask 1234567890ABCD#*i"
"2015-04-15T21:00:38.408000Z":19303:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"depositVoicemail Collected digits="
"2015-04-15T21:00:49.102000Z":19316:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Mailbox 207 Deposit Voicemail recorded message"
"2015-04-15T21:00:55.743000Z":19323:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Collect::start 1 7000/0/0 mask 1234567890ABCD#*i"
"2015-04-15T21:00:55.744000Z":19324:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"FSESI::invoke throw DisconnectException"
"2015-04-15T21:00:56.744000Z":19327:sipXivr:WARNING:pbx:Thread-1794:00000000:ResourceBundleMessageSource:"ResourceBundle [EmailFormats] not found for MessageSource: Can't find bundle for base name EmailFormats, locale en"
"2015-04-15T21:00:56.745000Z":19328:sipXivr:WARNING:pbx:Thread-1794:00000000:ResourceBundleMessageSource:"ResourceBundle [EmailFormats] not found for MessageSource: Can't find bundle for base name EmailFormats, locale en"
"2015-04-15T21:00:56.748000Z":19330:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"VmMessage::newMessage created message /var/sipxdata/mediaserver/data/mailstore/207/inbox/100001840-00.xml"
"2015-04-15T21:00:56.749000Z":19331:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Emailer::queueVm2Email queuing e-mail for 2...@foo.com"
"2015-04-15T21:00:56.749000Z":19332:sipXivr:INFO:pbx:Thread-1794:00000000:sipxivr:"Ending voicemail

Here is the freeswitch.log from the same period of time. The uuid doesn't change in the log, so I don't see a way to to filter this easily.

69a17cf6-e3b2-11e4-b37d-1fc9933e9a27 2015-04-15 14:00:41.752172 [NOTICE] mod_dptools.c:1232 Hangup sofia/foo.com/+18189632644@192.168.1.10 [CS_EXECUTE] [NORMAL_CLEARING]
69a14fb0-e3b2-11e4-b379-1fc9933e9a27 2015-04-15 14:00:41.752172 [NOTICE] sofia.c:926 Hangup sofia/foo.com/sip:I...@192.168.1.10:15060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
69a14fb0-e3b2-11e4-b379-1fc9933e9a27 2015-04-15 14:00:41.772187 [NOTICE] switch_core_session.c:1632 Session 2689 (sofia/foo.com/sip:IVR@192.168.1.10:15060) Ended
69a14fb0-e3b2-11e4-b379-1fc9933e9a27 2015-04-15 14:00:41.772187 [NOTICE] switch_core_session.c:1636 Close Channel sofia/foo.com/sip:I...@192.168.1.10:15060 [CS_DESTROY]
69a17cf6-e3b2-11e4-b37d-1fc9933e9a27 2015-04-15 14:00:41.772187 [NOTICE] switch_core_session.c:1632 Session 2690 (sofia/foo.com/+18189632644@192.168.1.10) Ended
69a17cf6-e3b2-11e4-b37d-1fc9933e9a27 2015-04-15 14:00:41.772187 [NOTICE] switch_core_session.c:1636 Close Channel sofia/foo.com/+18189632644@192.168.1.10 [CS_DESTROY]
7a6f8514-e3b2-11e4-b3d7-1fc9933e9a27 2015-04-15 14:00:42.572177 [NOTICE] switch_channel.c:1053 New Channel sofia/foo.com/~~id~bridge@foo.com [7a6f8514-e3b2-11e4-b3d7-1fc9933e9a27]
7a73276e-e3b2-11e4-b3df-1fc9933e9a27 2015-04-15 14:00:42.592206 [NOTICE] switch_channel.c:1053 New Channel sofia/foo.com/sip:I...@192.168.1.10:15060 [7a73276e-e3b2-11e4-b3df-1fc9933e9a27]
7a735298-e3b2-11e4-b3e3-1fc9933e9a27 2015-04-15 14:00:42.592206 [NOTICE] switch_channel.c:1053 New Channel sofia/foo.com/~~id~bridge@192.168.1.10 [7a735298-e3b2-11e4-b3e3-1fc9933e9a27]
7a735298-e3b2-11e4-b3e3-1fc9933e9a27 2015-04-15 14:00:42.612228 [NOTICE] sofia_media.c:92 Pre-Answer sofia/foo.com/~~id~bridge@192.168.1.10!
7a735298-e3b2-11e4-b3e3-1fc9933e9a27 2015-04-15 14:00:42.612228 [NOTICE] mod_dptools.c:1258 Channel [sofia/foo.com/~~id~bridge@192.168.1.10] has been answered
7a73276e-e3b2-11e4-b3df-1fc9933e9a27 2015-04-15 14:00:42.612228 [NOTICE] sofia.c:7057 Channel [sofia/foo.com/sip:IVR@192.168.1.10:15060] has been answered
7a6f8514-e3b2-11e4-b3d7-1fc9933e9a27 2015-04-15 14:00:42.632201 [NOTICE] sofia_media.c:92 Pre-Answer sofia/foo.com/~~id~bridge@foo.com!
7a6f8514-e3b2-11e4-b3d7-1fc9933e9a27 2015-04-15 14:00:42.632201 [NOTICE] switch_ivr_originate.c:3493 Channel [sofia/foo.com/~~id~bridge@foo.com] has been answered
7cdbf59e-e3b2-11e4-b3ec-1fc9933e9a27 2015-04-15 14:00:46.632185 [NOTICE] switch_channel.c:1053 New Channel sofia/foo.com/2...@192.168.1.10 [7cdbf59e-e3b2-11e4-b3ec-1fc9933e9a27]
7cdbf59e-e3b2-11e4-b3ec-1fc9933e9a27 2015-04-15 14:00:46.632185 [NOTICE] sofia_media.c:92 Pre-Answer sofia/foo.com/2...@192.168.1.10!
7cdbf59e-e3b2-11e4-b3ec-1fc9933e9a27 2015-04-15 14:00:46.632185 [NOTICE] mod_dptools.c:1258 Channel [sofia/foo.com/2...@192.168.1.10] has been answered
70fb54a4-e3b2-11e4-b3b0-1fc9933e9a27 2015-04-15 14:00:46.632185 [NOTICE] switch_ivr_bridge.c:1979 Hangup sofia/foo.com/2...@192.168.1.10 [CS_PARK] [PICKED_OFF]
70fb54a4-e3b2-11e4-b3b0-1fc9933e9a27 2015-04-15 14:00:46.652189 [NOTICE] switch_core_session.c:1632 Session 2697 (sofia/foo.com/2...@192.168.1.10) Ended
70fb54a4-e3b2-11e4-b3b0-1fc9933e9a27 2015-04-15 14:00:46.652189 [NOTICE] switch_core_session.c:1636 Close Channel sofia/foo.com/2...@192.168.1.10 [CS_DESTROY]
6f30c0d2-e3b2-11e4-b39b-1fc9933e9a27 2015-04-15 14:00:46.692174 [NOTICE] sofia.c:926 Hangup sofia/foo.com/~~id~bridge@foo.com [CS_EXECUTE] [NORMAL_CLEARING]
6f34446e-e3b2-11e4-b3a3-1fc9933e9a27 2015-04-15 14:00:46.692174 [NOTICE] switch_ivr_bridge.c:753 Hangup sofia/foo.com/sip:I...@192.168.1.10:15060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
6f34446e-e3b2-11e4-b3a3-1fc9933e9a27 2015-04-15 14:00:46.692174 [NOTICE] switch_core_session.c:1632 Session 2695 (sofia/foo.com/sip:IVR@192.168.1.10:15060) Ended
6f34446e-e3b2-11e4-b3a3-1fc9933e9a27 2015-04-15 14:00:46.692174 [NOTICE] switch_core_session.c:1636 Close Channel sofia/foo.com/sip:I...@192.168.1.10:15060 [CS_DESTROY]
6f30c0d2-e3b2-11e4-b39b-1fc9933e9a27 2015-04-15 14:00:46.692174 [NOTICE] switch_core_session.c:1632 Session 2694 (sofia/foo.com/~~id~bridge@foo.com) Ended
6f30c0d2-e3b2-11e4-b39b-1fc9933e9a27 2015-04-15 14:00:46.692174 [NOTICE] switch_core_session.c:1636 Close Channel sofia/foo.com/~~id~bridge@foo.com [CS_DESTROY]
6f346bce-e3b2-11e4-b3a7-1fc9933e9a27 2015-04-15 14:00:46.712181 [NOTICE] sofia.c:926 Hangup sofia/foo.com/~~id~bridge@192.168.1.10 [CS_EXECUTE] [NORMAL_CLEARING]
6f346bce-e3b2-11e4-b3a7-1fc9933e9a27 2015-04-15 14:00:46.712181 [NOTICE] switch_core_session.c:1632 Session 2696 (sofia/foo.com/~~id~bridge@192.168.1.10) Ended
6f346bce-e3b2-11e4-b3a7-1fc9933e9a27 2015-04-15 14:00:46.712181 [NOTICE] switch_core_session.c:1636 Close Channel sofia/foo.com/~~id~bridge@192.168.1.10 [CS_DESTROY]
7eb424f4-e3b2-11e4-b3f7-1fc9933e9a27 2015-04-15 14:00:49.732183 [NOTICE] switch_channel.c:1053 New Channel sofia/foo.com/+16619640544@192.168.13.114 [7eb424f4-e3b2-11e4-b3f7-1fc9933e9a27]
7eb7ae94-e3b2-11e4-b3ff-1fc9933e9a27 2015-04-15 14:00:49.752234 [NOTICE] switch_channel.c:1053 New Channel sofia/foo.com/sip:I...@192.168.1.10:15060 [7eb7ae94-e3b2-11e4-b3ff-1fc9933e9a27]
7eb7d7ac-e3b2-11e4-b403-1fc9933e9a27 2015-04-15 14:00:49.752234 [NOTICE] switch_channel.c:1053 New Channel sofia/foo.com/+16619640544@192.168.1.10 [7eb7d7ac-e3b2-11e4-b403-1fc9933e9a27]
7eb7d7ac-e3b2-11e4-b403-1fc9933e9a27 2015-04-15 14:00:49.772175 [NOTICE] sofia_media.c:92 Pre-Answer sofia/foo.com/+16619640544@192.168.1.10!
7eb7d7ac-e3b2-11e4-b403-1fc9933e9a27 2015-04-15 14:00:49.772175 [NOTICE] mod_dptools.c:1258 Channel [sofia/foo.com/+16619640544@192.168.1.10] has been answered
7eb7ae94-e3b2-11e4-b3ff-1fc9933e9a27 2015-04-15 14:00:49.772175 [NOTICE] sofia.c:7057 Channel [sofia/foo.com/sip:IVR@192.168.1.10:15060] has been answered
7eb424f4-e3b2-11e4-b3f7-1fc9933e9a27 2015-04-15 14:00:49.792204 [NOTICE] sofia_media.c:92 Pre-Answer sofia/foo.com/+16619640544@192.168.13.114!
7eb424f4-e3b2-11e4-b3f7-1fc9933e9a27 2015-04-15 14:00:49.792204 [NOTICE] switch_ivr_originate.c:3493 Channel [sofia/foo.com/+16619640544@192.168.13.114] has been answered
699dc692-e3b2-11e4-b371-1fc9933e9a27 2015-04-15 14:00:51.072200 [NOTICE] mod_sofia.c:1890 Hangup sofia/foo.com/+18189632644@192.168.13.114 [CS_EXECUTE] [BLIND_TRANSFER]
699dc692-e3b2-11e4-b371-1fc9933e9a27 2015-04-15 14:00:51.072200 [NOTICE] switch_core_session.c:1632 Session 2688 (sofia/foo.com/+18189632644@192.168.13.114) Ended
699dc692-e3b2-11e4-b371-1fc9933e9a27 2015-04-15 14:00:51.072200 [NOTICE] switch_core_session.c:1636 Close Channel sofia/foo.com/+18189632644@192.168.13.114 [CS_DESTROY]

George Niculae

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Apr 15, 2015, 6:35:28 PM4/15/15
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From what I see in the logs it doesn't even try to play 225 greeting but plays silence, does this happen with registered greetings only or is the same with standard greeting? I see it plays for user 210 for example, but not for 225

854a32ca-e3b9-11e4-bf2d-a9db99c24734 2015-04-15 14:51:08.332206 [DEBUG] switch_ivr.c:613 sofia/foo.com/2...@192.168.1.10 Command Execute playback(/var/sipxdata/mediaserver/data/mailstore/210/standard.wav)

George

Max Clark

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Apr 15, 2015, 6:44:20 PM4/15/15
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To my knowledge there are no registered greetings on the system. This only happens when one extension blind transfers to a second extension. If the first extension blind transfers to the voicemail deposit (8+ext) it works fine.

George Niculae

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Apr 15, 2015, 6:53:25 PM4/15/15
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Can you try this scenario as well and let me know the result: external call directly in 236, then blind transfer to 225 (ideally would be if you could move the DID number of AA as an alias for user 236)

George

On Thu, Apr 16, 2015 at 12:44 AM, Max Clark <max....@gmail.com> wrote:
To my knowledge there are no registered greetings on the system. This only happens when one extension blind transfers to a second extension. If the first extension blind transfers to the voicemail deposit (8+ext) it works fine.

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Max Clark

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Apr 15, 2015, 7:03:24 PM4/15/15
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George,

Here are the logs:


The 236 DID was called directly, they blind transferred to 238 with the same no greeting result.

Thanks,
Max

George Niculae

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Apr 15, 2015, 7:08:06 PM4/15/15
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OK, I can see now that greetings are played (logs below) so I suspect codec / ITSP related. Let's try one more time but now removing ITSP from flow, so call from an internal extension to 236 and then blind transfer to 238 and see the result

George

1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:40.872235 [DEBUG] mod_dptools.c:1435 sofia/foo.com/2...@192.168.1.10 SET [playback_terminators]=[0123456789#*]
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:40.892236 [DEBUG] switch_core_session.c:1187 Send signal sofia/foo.com/2...@192.168.1.10 [BREAK]
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:40.892236 [DEBUG] switch_ivr.c:613 sofia/foo.com/2...@192.168.1.10 Command Execute playback(/var/sipxdata/mediaserver/data/mailstore/238/name.wav)
1890a682-e3c3-11e4-a59a-336387e49d24 EXECUTE sofia/foo.com/2...@192.168.1.10 playback(/var/sipxdata/mediaserver/data/mailstore/238/name.wav)
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:40.892236 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@8000hz 1 channels 20ms
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:43.132235 [DEBUG] switch_ivr_play_say.c:1710 done playing file /var/sipxdata/mediaserver/data/mailstore/238/name.wav
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:43.152202 [DEBUG] switch_core_session.c:1187 Send signal sofia/foo.com/2...@192.168.1.10 [BREAK]
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:43.152202 [DEBUG] switch_ivr.c:613 sofia/foo.com/2...@192.168.1.10 Command Execute playback(/usr/share/www/doc/stdprompts_en/is_not_available.wav)
1890a682-e3c3-11e4-a59a-336387e49d24 EXECUTE sofia/foo.com/2...@192.168.1.10 playback(/usr/share/www/doc/stdprompts_en/is_not_available.wav)
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:43.152202 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@8000hz 1 channels 20ms
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:44.192202 [DEBUG] switch_ivr_play_say.c:1710 done playing file /usr/share/www/doc/stdprompts_en/is_not_available.wav
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:44.212223 [DEBUG] switch_core_session.c:1187 Send signal sofia/foo.com/2...@192.168.1.10 [BREAK]
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:44.212223 [DEBUG] switch_ivr.c:613 sofia/foo.com/2...@192.168.1.10 Command Execute playback(/usr/share/www/doc/stdprompts_en/please_leave_a_msg.wav)
1890a682-e3c3-11e4-a59a-336387e49d24 EXECUTE sofia/foo.com/2...@192.168.1.10 playback(/usr/share/www/doc/stdprompts_en/please_leave_a_msg.wav)
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:44.212223 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@8000hz 1 channels 20ms
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:45.432188 [DEBUG] switch_ivr_play_say.c:1710 done playing file /usr/share/www/doc/stdprompts_en/please_leave_a_msg.wav
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:45.452235 [DEBUG] switch_core_session.c:1187 Send signal sofia/foo.com/2...@192.168.1.10 [BREAK]
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:45.452235 [DEBUG] switch_ivr.c:613 sofia/foo.com/2...@192.168.1.10 Command Execute playback(/usr/share/www/doc/stdprompts_en/deposit_greeting_extn.wav)
1890a682-e3c3-11e4-a59a-336387e49d24 EXECUTE sofia/foo.com/2...@192.168.1.10 playback(/usr/share/www/doc/stdprompts_en/deposit_greeting_extn.wav)
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:45.452235 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@8000hz 1 channels 20ms
1890a682-e3c3-11e4-a59a-336387e49d24 2015-04-15 15:59:50.692236 [DEBUG] switch_ivr_play_say.c:1710 done playing file /usr/share/www/doc/stdprompts_en/deposit_greeting_extn.wav

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Max Clark

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Apr 15, 2015, 7:16:36 PM4/15/15
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Same result. Extension 303 dialed 236 which blind transferred to 238.


BTW L16 is not selected as an allowed Codec in the Media Services:

George Niculae

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Apr 15, 2015, 7:18:08 PM4/15/15
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You could try to add it and check if it makes a difference. What codecs do you have enabled?

George

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Max Clark

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Apr 15, 2015, 7:21:47 PM4/15/15
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PCMU
PCMA
G729
--
Max

George Niculae

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Apr 15, 2015, 7:22:42 PM4/15/15
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OK, try with L16 as well. Also, could you get locally these files and try listen them?

 /usr/share/www/doc/stdprompts_en/is_not_available.wav
/var/sipxdata/mediaserver/data/mailstore/238/name.wav
/usr/share/www/doc/stdprompts_en/please_leave_a_msg.wav

Max Clark

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Apr 15, 2015, 7:35:43 PM4/15/15
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George,

Reset the codec selected list to:

G722
PCMU
PCMA
speex
L16

Same result on the blind transfer:

https://dpaste.de/gCfk

Downloaded the wav files and they play fine in Quicktime. They're attached as well.

Max
is_not_available.wav
name.wav
please_leave_a_msg.wav

Max Clark

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Apr 15, 2015, 7:38:55 PM4/15/15
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I should have said that that log was PSTN -> 236 -> 218.

George Niculae

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Apr 15, 2015, 7:57:35 PM4/15/15
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Looking again in https://dpaste.de/Pvqy/raw it is clearly related to codec / PTIME, someone suggest here http://markmail.org/message/rvre7lbtiilyqx4h#query:+page:1+mid:yd5qoluasdkzuggd+state:results to use

<param name="rtp-autofix-timing" value="false"/>

so you could add thie line in /etc/sipxpbx/freeswitch/sofia.conf.xml.vm just after

<param name="disable-rtp-auto-adjust" value="true"/>

then send profiles to server and retry


3c4-11e4-85b0-4b9bc78c3823 2015-04-15 16:12:31.352208 [DEBUG] switch_ivr.c:613 sofia/foo.com/2...@192.168.1.10 Command Execute playback(/usr/share/www/doc/stdprompts_en/deposit_options.wav)
d77b8796-e3c4-11e4-85b0-4b9bc78c3823 EXECUTE sofia/foo.com/2...@192.168.1.10 playback(/usr/share/www/doc/stdprompts_en/deposit_options.wav)
d77b8796-e3c4-11e4-85b0-4b9bc78c3823 2015-04-15 16:12:31.352208 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@8000hz 1 channels 20ms
e34c42cc-e3c4-11e4-85c1-4b9bc78c3823 2015-04-15 16:12:31.392211 [DEBUG] switch_rtp.c:5574 Correct ip/port confirmed.
d77b8796-e3c4-11e4-85b0-4b9bc78c3823 2015-04-15 16:12:31.412236 [DEBUG] switch_rtp.c:5574 Correct ip/port confirmed.
e34c42cc-e3c4-11e4-85c1-4b9bc78c3823 2015-04-15 16:12:31.612254 [WARNING] switch_core_media.c:1882 Asynchronous PTIME not supported, changing our end from 20 to 30
e34c42cc-e3c4-11e4-85c1-4b9bc78c3823 2015-04-15 16:12:31.632189 [DEBUG] switch_core_media.c:2252 Changing Codec from PCMU@20ms@8000hz to PCMU@30ms@8000hz
e34c42cc-e3c4-11e4-85c1-4b9bc78c3823 2015-04-15 16:12:31.672211 [DEBUG] switch_rtp.c:3233 RE-Starting timer [soft] 240 bytes per 30ms
e34c42cc-e3c4-11e4-85c1-4b9bc78c3823 2015-04-15 16:12:31.672211 [DEBUG] switch_core_media.c:2343 Set Codec sofia/foo.com/2...@192.168.1.10 PCMU/8000 30 ms 240 samples 64000 bits
e34c42cc-e3c4-11e4-85c1-4b9bc78c3823 2015-04-15 16:12:31.672211 [DEBUG] switch_core_codec.c:123 sofia/foo.com/2...@192.168.1.10 Original read codec replaced with PCMU:0
e34c42cc-e3c4-11e4-85c1-4b9bc78c3823 2015-04-15 16:12:31.692226 [DEBUG] switch_core_io.c:1459 Engaging Write Buffer at 480 bytes to accommodate 320->480
d77b53c0-e3c4-11e4-85ac-4b9bc78c3823 2015-04-15 16:12:31.732212 [DEBUG] switch_core_io.c:1459 Engaging Write Buffer at 320 bytes to accommodate 480->320

On Thu, Apr 16, 2015 at 1:38 AM, Max Clark <max....@gmail.com> wrote:
I should have said that that log was PSTN -> 236 -> 218.

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Max Clark

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Apr 15, 2015, 8:05:51 PM4/15/15
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 Should I be concerned that this setting doesn't exist in my configuration file?

George Niculae

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Apr 15, 2015, 8:10:04 PM4/15/15
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Yes, what version  of sipxcom?


On Thursday, April 16, 2015, Max Clark <max....@gmail.com> wrote:
 Should I be concerned that this setting doesn't exist in my configuration file?
 
<param name="disable-rtp-auto-adjust" value="true"/>

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Max Clark

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Apr 15, 2015, 8:12:44 PM4/15/15
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14.10.20150204095804

George Niculae

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Apr 15, 2015, 8:14:05 PM4/15/15
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Actually it was just added in 15.06, so not an issue
Put it after

<param name="rtp-ip" value="$${local_ip_v4}"/>

Max Clark

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Apr 15, 2015, 8:22:27 PM4/15/15
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Thanks George. I'm looking for another person onsite that can make a test call for me.

Max Clark

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Apr 16, 2015, 12:00:13 PM4/16/15
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George,

This did not fix the problem. The greeting is still not being played.


Extension 0 transferring to 236.

Max

George Niculae

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Apr 16, 2015, 1:11:39 PM4/16/15
to Max Clark, sipxco...@googlegroups.com
I must admit I am running out of ideas, the PTIME issue doesn't appear in logs so looks like the param was correctly applied. The other thing that I see (though I am very sure it's benign) is the one below
2015-04-16 08:54:26.772198 [DEBUG] switch_core_file.c:211 File /var/sipxdata/mediaserver/data/mailstore/236/standard.wav sample rate 8000 doesn't match requested rate 16000
db60cbaa-e450-11e4-9623-5d701def911c 2015-04-16 08:54:26.772198 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@16000hz 1 channels 20ms
Next step would be to take a pcap and send it across, we have to check where / if RTP flows ...

George

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Max Clark

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Apr 16, 2015, 1:13:54 PM4/16/15
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Should I flag any specific tcpdump options?
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Max Clark

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Apr 16, 2015, 1:15:44 PM4/16/15
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What's crazy about this is if the user transferring uses the 8+ext option to deposit directly to voicemail the greeting is played.
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George Niculae

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Apr 16, 2015, 1:21:13 PM4/16/15
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Yeah, that's really weird...

Todd Hodgen

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Apr 16, 2015, 1:46:32 PM4/16/15
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Have you tried a capture on the server with tcpdump.  You can then play that call on wireshark, and hear the audio in both direction.  It might point out something.

 

Todd R. Hodgen

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Max Clark

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Apr 16, 2015, 3:17:40 PM4/16/15
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Practically speaking - where do I go from here?

Voicemail greetings are on the system, I can download them and listen to them on my computer (no corruption). They play just fine when the extension is called either via a DID or transferred from the Auto Attendant. They even work when a transfer to the voicemail deposit (8+ext) is used. This problem is present when using a blind transfer from extension to extension. And even with that, after the voicemail greeting should play, and then caller voicemail recording the system plays the voicemail confirmation prompt back to the caller. So even at this stage media is working properly.

This log is from a blind transfer from ext 0 to ext 234 (the extension actually doesn't have a greeting recorded so sipx uses the default number prompts):


Every step after executing a playback there is a message "sample rate 8000 doesn't match requested rate 16000" which on the surface I would be suspicious of, except audio does work (although selectively).

Key events are:

2. local_stream://moh (audio) https://dpaste.de/UabC#L6702
3. default_greeting_prefix.wav (no audio) https://dpaste.de/UabC#L6735,6736
4. 02.wav (no audio) https://dpaste.de/UabC#L6741,6742
5. 03.wav (no audio) https://dpaste.de/UabC#L6747,6748
6. 04.wav (no audio) https://dpaste.de/UabC#L6753,6754
7. is_not_available.wav (no audio) https://dpaste.de/UabC#L6759,6760
8. please_leave_a_msg.wav (no audio) https://dpaste.de/UabC#L6765,6766
9. deposit_greeting_extn.wav (audio) https://dpaste.de/UabC#L6771,6772

I should note that #1 (silence), #2 (moh), and #9 (deposit_greeting_extn.wav) are what the caller hears as the transfer happens. I have downloaded all of the wav files and verified that they do in fact have normal audio and play.

Could this be an issue with moh? Can I disable moh on transfer and give the caller something else?

Thanks,
Max

Max Clark

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Apr 16, 2015, 4:59:54 PM4/16/15
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Is it possible to disable music on hold for transfers?

George Niculae

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Apr 16, 2015, 5:10:03 PM4/16/15
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You can disable it from phones, e.g. for Polycoms go to Lines > Registration, Show Advanced Settings and remove value for setting below

(Default: )

On Thu, Apr 16, 2015 at 10:59 PM, Max Clark <max....@gmail.com> wrote:
Is it possible to disable music on hold for transfers?

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Max Clark

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Apr 16, 2015, 5:19:20 PM4/16/15
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Thanks I'm looking for that on the spa504g configuration now.

Michael Picher

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Apr 16, 2015, 7:46:17 PM4/16/15
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Oh, you didn't say toy were using those. Try with polycom


On Thu, Apr 16, 2015, 5:19 PM Max Clark <max....@gmail.com> wrote:
Thanks I'm looking for that on the spa504g configuration now.

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Max Clark

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Apr 16, 2015, 7:48:09 PM4/16/15
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I would love to get rid of the SPA... just assuming this is an issue that is SPA caused, is there a fix (disable MOH, disable other functionality) that I should look at?

tgra...@commonwealthal.com

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Apr 20, 2015, 1:35:24 PM4/20/15
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I think it is the REFER. Blind transfers to extensions work, but when that fails to pickup there wouldbe a REFER message. If the call comes in on a ITSP the ITSP needs to support refer, if by PSTN (pots/pri gateway), the gateway needs to support REFER. To test this theory point the inbound call to an auto attendant and see if it will transfer.


On Thursday, April 16, 2015 at 7:48:09 PM UTC-4, Max Clark wrote:
I would love to get rid of the SPA... just assuming this is an issue that is SPA caused, is there a fix (disable MOH, disable other functionality) that I should look at?

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Max Clark

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Apr 21, 2015, 6:52:03 PM4/21/15
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I got approval for a maintenance window tonight. We're going to go through all the scenarios we can think of and try to isolate this.

Max Clark

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Apr 22, 2015, 11:48:57 AM4/22/15
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Followup on this issue....

I received a recommendation (https://groups.google.com/forum/#!topic/sipxcom-users/MBZP6YLe7Fo) to add this configuration to FreeSWITCH (/etc/sipxpbx/freeswitch/sofia.conf.xml.vm):


<param name="rtp-autofix-timing" value="false"/>

Do not do this. Applying this configuration and sending profiles to the server adversely impacts the voicemail quality. All recordings and messages are garbled afterwards.

We found an option available under the phone group in "Call Feature Settings" called "Blind Attn Xfer Enable" (attached). Enabling this and sending profiles to the SPA phones did resolve the problem. We're waiting for a full day of normal use before closing this matter - but based on all of our testing last night confidence is high.

Thanks for all of the help.

Max
Call_Feature_Settings.png
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