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kctrey at gmail.com
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 Posted: Wed Feb 10, 2016 5:47 pm Post subject: [asterisk-users] Unexpected termination of the call when pic |
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How are you initiating the call out to that server? Are you dialing from an internal phone or doing it from the CLI? It looks like it is from an internal extension, if I were guessing, but that side of the call isn't in your log.
If it is from an internal extension, I think a SIP trace on that side would help.
On Wed, Feb 10, 2016, 3:20 PM Dmitriy Serov <serov.d.p@gmail.com (serov.d.p@gmail.com)> wrote:
| Quote: | Please help find the cause of strange behavior res_pjsip.
Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up!
Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4d
No diagnostic messages why asterisk suddenly decided to hangup i don't found 
There are suggestions or strong belief about the reasons of such behavior?
Thanks.
Dmitriy.
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serov.d.p at gmail.com
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| Posted: Thu Feb 11, 2016 2:33 am Post subject: [asterisk-users] Unexpected termination of the call when pic |
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The call initiated from internal extension.
I have made two test call:
Successful call from device on res_pjsip via endpoint on chan_sip: http://pastebin.com/LWeDYstj
Unsuccessful call from device on res_pjsip via endpoint on res_pjsip: http://pastebin.com/hepVb6Nu
And ones again i don't see anything that would make asterisk send BYE.
I would be grateful for any ideas.
11.02.2016 1:47, Trey Hilyard пишет:
| Quote: |
How are you initiating the call out to that server? Are you dialing from an internal phone or doing it from the CLI? It looks like it is from an internal extension, if I were guessing, but that side of the call isn't in your log.
If it is from an internal extension, I think a SIP trace on that side would help.
On Wed, Feb 10, 2016, 3:20 PM Dmitriy Serov <[url=mailto:serov.d.p@gmail.com]serov.d.p@gmail.com (serov.d.p@gmail.com)[/url]> wrote:
| Quote: | Please help find the cause of strange behavior res_pjsip.
Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up!
Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4d
No diagnostic messages why asterisk suddenly decided to hangup i don't found
There are suggestions or strong belief about the reasons of such behavior?
Thanks.
Dmitriy.
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kctrey at gmail.com
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| Posted: Thu Feb 11, 2016 10:26 am Post subject: [asterisk-users] Unexpected termination of the call when pic |
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I am stumped so far. What is most interesting to me is that Asterisk is actually sending two BYE transactions for the same dialog, at basically the same time. I am still going through your traces again, but maybe someone else has suggestions on how to add more debug to the res_pjsip logging that would prove useful.
On Thu, Feb 11, 2016 at 1:33 AM Dmitriy Serov <serov.d.p@gmail.com (serov.d.p@gmail.com)> wrote:
| Quote: | The call initiated from internal extension.
I have made two test call:
Successful call from device on res_pjsip via endpoint on chan_sip: http://pastebin.com/LWeDYstj
Unsuccessful call from device on res_pjsip via endpoint on res_pjsip: http://pastebin.com/hepVb6Nu
And ones again i don't see anything that would make asterisk send BYE.
I would be grateful for any ideas.
11.02.2016 1:47, Trey Hilyard пишет:
| Quote: |
How are you initiating the call out to that server? Are you dialing from an internal phone or doing it from the CLI? It looks like it is from an internal extension, if I were guessing, but that side of the call isn't in your log.
If it is from an internal extension, I think a SIP trace on that side would help.
On Wed, Feb 10, 2016, 3:20 PM Dmitriy Serov <[url=mailto:serov.d.p@gmail.com]serov.d.p@gmail.com (serov.d.p@gmail.com)[/url]> wrote:
| Quote: | Please help find the cause of strange behavior res_pjsip.
Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up!
Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4d
No diagnostic messages why asterisk suddenly decided to hangup i don't found
There are suggestions or strong belief about the reasons of such behavior?
Thanks.
Dmitriy.
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_____________________________________________________________________
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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_____________________________________________________________________
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
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jcolp at digium.com
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| Posted: Thu Feb 11, 2016 10:51 am Post subject: [asterisk-users] Unexpected termination of the call when pic |
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Dmitriy Serov wrote:
| Quote: | The call initiated from internal extension.
I have made two test call:
Successful call from device on res_pjsip via endpoint on chan_sip:
http://pastebin.com/LWeDYstj
Unsuccessful call from device on res_pjsip via endpoint on res_pjsip:
http://pastebin.com/hepVb6Nu
And ones again i don't see anything that would make asterisk send BYE.
I would be grateful for any ideas.
|
Kia ora,
I have a feeling it may be an off-nominal SDP negotiation issue, causing
two paths to get triggered which both send a BYE. I'd suggest filing an
issue[1] with the traces you've provided. We can potentially use them to
construct a sipp scenario that reproduces the issue. The configuration
would also be needed.
You can also try to narrow it down slightly by disabling the video
codecs and seeing if that changes things. If it does then it's with
video involved.
[1] https://issues.asterisk.org/jira
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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serov.d.p at gmail.com
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| Posted: Thu Feb 11, 2016 1:51 pm Post subject: [asterisk-users] Unexpected termination of the call when pic |
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11.02.2016 18:50, Joshua Colp пишет:
| Quote: | Dmitriy Serov wrote:
| Quote: | The call initiated from internal extension.
I have made two test call:
Successful call from device on res_pjsip via endpoint on chan_sip:
http://pastebin.com/LWeDYstj
Unsuccessful call from device on res_pjsip via endpoint on res_pjsip:
http://pastebin.com/hepVb6Nu
And ones again i don't see anything that would make asterisk send BYE.
I would be grateful for any ideas.
|
Kia ora,
I have a feeling it may be an off-nominal SDP negotiation issue,
causing two paths to get triggered which both send a BYE. I'd suggest
filing an issue[1] with the traces you've provided. We can potentially
use them to construct a sipp scenario that reproduces the issue. The
configuration would also be needed.
You can also try to narrow it down slightly by disabling the video
codecs and seeing if that changes things. If it does then it's with
video involved.
[1] https://issues.asterisk.org/jira
|
Joshua, Thanks! Disabling all codecs except alaw (I guess video codecs)
makes call successful.
https://issues.asterisk.org/jira/browse/ASTERISK-25772
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_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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