calls appear to be dropping ... from landlines
We're running FreeSwitch as part of a teleconferencing service, inside a telcom (so no internet latency/NAT issues) and using g.729 We are receiving some complaints of dropped calls, including from landlines. This means they join the conference, and x minutes in they simply drop. I know that cellphones tend to drop calls frequently, but landlines are pretty reliable, and we're hearing it a lot. From the FreeSwitch side of things, it just looks like those callers hung up (but then dialed back in just a moment later). I'm attaching two different snippets of the FS log files where these issues are occurring. Does anyone have any recommendations about how to troubleshoot this? Any known issues/patches in FS that could be biting us? Is there some SIP logging we can do to debug? Are there any paid contractors avail who would have the expertise to look into this? Any help appreciated ... this is a major issue for us! Thanks much, -Dale daletrub@gmail.com
I have experienced the same a while ago, I originated calls from my freeswitch server to some landlines and calls would simply drop after X minutes. I tried to debug the thing but found nothing relevant, maybe I had the same issue as you. Let me know if you figure it out what it was. Diego On Thu, May 21, 2009 at 10:15 PM, Dale Trub <daletrub@gmail.com> wrote:
We're running FreeSwitch as part of a teleconferencing service, inside a telcom (so no internet latency/NAT issues) and using g.729 We are receiving some complaints of dropped calls, including from landlines. This means they join the conference, and x minutes in they simply drop. I know that cellphones tend to drop calls frequently, but landlines are pretty reliable, and we're hearing it a lot. From the FreeSwitch side of things, it just looks like those callers hung up (but then dialed back in just a moment later). I'm attaching two different snippets of the FS log files where these issues are occurring.
Does anyone have any recommendations about how to troubleshoot this? Any known issues/patches in FS that could be biting us? Is there some SIP logging we can do to debug? Are there any paid contractors avail who would have the expertise to look into this? Any help appreciated ... this is a major issue for us! Thanks much, -Dale daletrub@gmail.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
On May 21, 2009, at 9:15 PM, Dale Trub wrote:
We're running FreeSwitch as part of a teleconferencing service, inside a telcom (so no internet latency/NAT issues) and using g.729
So you're using g729 with conferences?
We are receiving some complaints of dropped calls, including from landlines. This means they join the conference, and x minutes in they simply drop.
I know that cellphones tend to drop calls frequently, but landlines are pretty reliable, and we're hearing it a lot. From the FreeSwitch side of things, it just looks like those callers hung up (but then dialed back in just a moment later).
I'm attaching two different snippets of the FS log files where these issues are occurring.
Next time please call them .txt because you cause extra work to have to open them otherwise.
Does anyone have any recommendations about how to troubleshoot this?
Any known issues/patches in FS that could be biting us?
Depends you failed to include some very valid info such as what version or svn rev you're running and what linux distro.
Is there some SIP logging we can do to debug?
Yes covered on the wiki. http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
Are there any paid contractors avail who would have the expertise to look into this?
email consulting@freeswitch.org
Any help appreciated ... this is a major issue for us!
Thanks much,
-Dale
Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
Thanks Brian! To answer your questions: Freeswitch svn revision: 12148 Centos rev: 2.6.18-92.el5 And apologies, actually I guess we're using g711 not 729. Jason: I agree it would seem to be on the switch/telco side. And, the telco says many other people are in the same set-up as us and don't have any issues, so they're insisting it's on our end. On Thu, May 21, 2009 at 7:28 PM, Brian West <brian@freeswitch.org> wrote:
On May 21, 2009, at 9:15 PM, Dale Trub wrote:
We're running FreeSwitch as part of a teleconferencing service, inside a telcom (so no internet latency/NAT issues) and using g.729
So you're using g729 with conferences?
We are receiving some complaints of dropped calls, including from landlines. This means they join the conference, and x minutes in they simply drop.
I know that cellphones tend to drop calls frequently, but landlines are pretty reliable, and we're hearing it a lot. From the FreeSwitch side of things, it just looks like those callers hung up (but then dialed back in just a moment later).
I'm attaching two different snippets of the FS log files where these issues are occurring.
Next time please call them .txt because you cause extra work to have to open them otherwise.
Does anyone have any recommendations about how to troubleshoot this?
Any known issues/patches in FS that could be biting us?
Depends you failed to include some very valid info such as what version or svn rev you're running and what linux distro.
Is there some SIP logging we can do to debug?
Yes covered on the wiki. http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
Are there any paid contractors avail who would have the expertise to look into this?
email consulting@freeswitch.org
Any help appreciated ... this is a major issue for us!
Thanks much,
-Dale
Brian West brian@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
If I had a few bucks for every time the telco has said this to me I could just about retire! You using 100% SIP? /b On May 21, 2009, at 10:09 PM, Dale Trub wrote:
And, the telco says many other people are in the same set-up as us and don't have any issues, so they're insisting it's on our end.
Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
1) update to lastest trunk (you are at least 1000 revisions behind) 2) disable the presence debug in sofia.conf 3) enable sip trace instead "sofia profile internal siptrace on" 4) reproduce your problem. Make sure you include more of the log from before the hangup happened. The one you posted here is missing some of the info from the few seconds prior but with the incomplete info it looks like the other side sent a BYE ending the call. On Thu, May 21, 2009 at 10:09 PM, Dale Trub <daletrub@gmail.com> wrote:
Thanks Brian! To answer your questions: Freeswitch svn revision: 12148 Centos rev: 2.6.18-92.el5
And apologies, actually I guess we're using g711 not 729.
Jason: I agree it would seem to be on the switch/telco side. And, the telco says many other people are in the same set-up as us and don't have any issues, so they're insisting it's on our end.
On Thu, May 21, 2009 at 7:28 PM, Brian West <brian@freeswitch.org> wrote:
On May 21, 2009, at 9:15 PM, Dale Trub wrote:
We're running FreeSwitch as part of a teleconferencing service, inside a telcom (so no internet latency/NAT issues) and using g.729
So you're using g729 with conferences?
We are receiving some complaints of dropped calls, including from landlines. This means they join the conference, and x minutes in they simply drop.
I know that cellphones tend to drop calls frequently, but landlines are pretty reliable, and we're hearing it a lot. From the FreeSwitch side of things, it just looks like those callers hung up (but then dialed back in just a moment later).
I'm attaching two different snippets of the FS log files where these issues are occurring.
Next time please call them .txt because you cause extra work to have to open them otherwise.
Does anyone have any recommendations about how to troubleshoot this?
Any known issues/patches in FS that could be biting us?
Depends you failed to include some very valid info such as what version or svn rev you're running and what linux distro.
Is there some SIP logging we can do to debug?
Yes covered on the wiki. http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
Are there any paid contractors avail who would have the expertise to look into this?
email consulting@freeswitch.org
Any help appreciated ... this is a major issue for us!
Thanks much,
-Dale
Brian West brian@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony, Thank you for your suggestions! We are working on 1), but need to re-integrate code we've changed, and do regression testing. That's in progress, and we expect to be able to upgrade by the end of next week. We did manage to do 3) and 4), and we now have SIP logs (attached). Are you able to see anything that's out of the ordinary that we should be paying attention to? Best, Dale On Fri, May 22, 2009 at 5:47 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
1) update to lastest trunk (you are at least 1000 revisions behind) 2) disable the presence debug in sofia.conf 3) enable sip trace instead "sofia profile internal siptrace on" 4) reproduce your problem.
Make sure you include more of the log from before the hangup happened. The one you posted here is missing some of the info from the few seconds prior but with the incomplete info it looks like the other side sent a BYE ending the call.
On Thu, May 21, 2009 at 10:09 PM, Dale Trub <daletrub@gmail.com> wrote:
Thanks Brian! To answer your questions: Freeswitch svn revision: 12148 Centos rev: 2.6.18-92.el5
And apologies, actually I guess we're using g711 not 729.
Jason: I agree it would seem to be on the switch/telco side. And, the telco says many other people are in the same set-up as us and don't have any issues, so they're insisting it's on our end.
On Thu, May 21, 2009 at 7:28 PM, Brian West <brian@freeswitch.org> wrote:
On May 21, 2009, at 9:15 PM, Dale Trub wrote:
We're running FreeSwitch as part of a teleconferencing service, inside a telcom (so no internet latency/NAT issues) and using g.729
So you're using g729 with conferences?
We are receiving some complaints of dropped calls, including from landlines. This means they join the conference, and x minutes in they simply drop.
I know that cellphones tend to drop calls frequently, but landlines are pretty reliable, and we're hearing it a lot. From the FreeSwitch side of things, it just looks like those callers hung up (but then dialed back in just a moment later).
I'm attaching two different snippets of the FS log files where these issues are occurring.
Next time please call them .txt because you cause extra work to have to open them otherwise.
Does anyone have any recommendations about how to troubleshoot this?
Any known issues/patches in FS that could be biting us?
Depends you failed to include some very valid info such as what version or svn rev you're running and what linux distro.
Is there some SIP logging we can do to debug?
Yes covered on the wiki. http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
Are there any paid contractors avail who would have the expertise to look into this?
email consulting@freeswitch.org
Any help appreciated ... this is a major issue for us!
Thanks much,
-Dale
Brian West brian@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Also you should be putting these details in a jira report. http://jira.freeswitch.org open an issue report and attach all relevant logs, do not attach tarballs or gzipped files and make sure text files have a .txt extension. On Wed, May 27, 2009 at 6:58 PM, Dale Trub <daletrub@gmail.com> wrote:
Anthony,
Thank you for your suggestions! We are working on 1), but need to re-integrate code we've changed, and do regression testing. That's in progress, and we expect to be able to upgrade by the end of next week.
We did manage to do 3) and 4), and we now have SIP logs (attached). Are you able to see anything that's out of the ordinary that we should be paying attention to?
Best, Dale
On Fri, May 22, 2009 at 5:47 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
1) update to lastest trunk (you are at least 1000 revisions behind) 2) disable the presence debug in sofia.conf 3) enable sip trace instead "sofia profile internal siptrace on" 4) reproduce your problem.
Make sure you include more of the log from before the hangup happened. The one you posted here is missing some of the info from the few seconds prior but with the incomplete info it looks like the other side sent a BYE ending the call.
On Thu, May 21, 2009 at 10:09 PM, Dale Trub <daletrub@gmail.com> wrote:
Thanks Brian! To answer your questions: Freeswitch svn revision: 12148 Centos rev: 2.6.18-92.el5
And apologies, actually I guess we're using g711 not 729.
Jason: I agree it would seem to be on the switch/telco side. And, the telco says many other people are in the same set-up as us and don't have any issues, so they're insisting it's on our end.
On Thu, May 21, 2009 at 7:28 PM, Brian West <brian@freeswitch.org>wrote:
On May 21, 2009, at 9:15 PM, Dale Trub wrote:
We're running FreeSwitch as part of a teleconferencing service, inside a telcom (so no internet latency/NAT issues) and using g.729
So you're using g729 with conferences?
We are receiving some complaints of dropped calls, including from landlines. This means they join the conference, and x minutes in they simply drop.
I know that cellphones tend to drop calls frequently, but landlines are pretty reliable, and we're hearing it a lot. From the FreeSwitch side of things, it just looks like those callers hung up (but then dialed back in just a moment later).
I'm attaching two different snippets of the FS log files where these issues are occurring.
Next time please call them .txt because you cause extra work to have to open them otherwise.
Does anyone have any recommendations about how to troubleshoot this?
Any known issues/patches in FS that could be biting us?
Depends you failed to include some very valid info such as what version or svn rev you're running and what linux distro.
Is there some SIP logging we can do to debug?
Yes covered on the wiki. http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
Are there any paid contractors avail who would have the expertise to look into this?
email consulting@freeswitch.org
Any help appreciated ... this is a major issue for us!
Thanks much,
-Dale
Brian West brian@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
btw, 3 and 4 are not useful without 1 we only debug issues with svn trunk On Thu, May 28, 2009 at 8:07 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Also you should be putting these details in a jira report. http://jira.freeswitch.org
open an issue report and attach all relevant logs, do not attach tarballs or gzipped files and make sure text files have a .txt extension.
On Wed, May 27, 2009 at 6:58 PM, Dale Trub <daletrub@gmail.com> wrote:
Anthony,
Thank you for your suggestions! We are working on 1), but need to re-integrate code we've changed, and do regression testing. That's in progress, and we expect to be able to upgrade by the end of next week.
We did manage to do 3) and 4), and we now have SIP logs (attached). Are you able to see anything that's out of the ordinary that we should be paying attention to?
Best, Dale
On Fri, May 22, 2009 at 5:47 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
1) update to lastest trunk (you are at least 1000 revisions behind) 2) disable the presence debug in sofia.conf 3) enable sip trace instead "sofia profile internal siptrace on" 4) reproduce your problem.
Make sure you include more of the log from before the hangup happened. The one you posted here is missing some of the info from the few seconds prior but with the incomplete info it looks like the other side sent a BYE ending the call.
On Thu, May 21, 2009 at 10:09 PM, Dale Trub <daletrub@gmail.com> wrote:
Thanks Brian! To answer your questions: Freeswitch svn revision: 12148 Centos rev: 2.6.18-92.el5
And apologies, actually I guess we're using g711 not 729.
Jason: I agree it would seem to be on the switch/telco side. And, the telco says many other people are in the same set-up as us and don't have any issues, so they're insisting it's on our end.
On Thu, May 21, 2009 at 7:28 PM, Brian West <brian@freeswitch.org>wrote:
On May 21, 2009, at 9:15 PM, Dale Trub wrote:
We're running FreeSwitch as part of a teleconferencing service, inside a telcom (so no internet latency/NAT issues) and using g.729
So you're using g729 with conferences?
We are receiving some complaints of dropped calls, including from landlines. This means they join the conference, and x minutes in they simply drop.
I know that cellphones tend to drop calls frequently, but landlines are pretty reliable, and we're hearing it a lot. From the FreeSwitch side of things, it just looks like those callers hung up (but then dialed back in just a moment later).
I'm attaching two different snippets of the FS log files where these issues are occurring.
Next time please call them .txt because you cause extra work to have to open them otherwise.
Does anyone have any recommendations about how to troubleshoot this?
Any known issues/patches in FS that could be biting us?
Depends you failed to include some very valid info such as what version or svn rev you're running and what linux distro.
Is there some SIP logging we can do to debug?
Yes covered on the wiki. http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
Are there any paid contractors avail who would have the expertise to look into this?
email consulting@freeswitch.org
Any help appreciated ... this is a major issue for us!
Thanks much,
-Dale
Brian West brian@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Hey Gents, What is the Jira for this issue? Dale, Did you get any SIP traces. I am interested to have a look. You can use NGREP if your system is Linux. Regards, Jim On Thu, May 28, 2009 at 11:08 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
btw,
3 and 4 are not useful without 1 we only debug issues with svn trunk
On Thu, May 28, 2009 at 8:07 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Also you should be putting these details in a jira report. http://jira.freeswitch.org
open an issue report and attach all relevant logs, do not attach tarballs or gzipped files and make sure text files have a .txt extension.
On Wed, May 27, 2009 at 6:58 PM, Dale Trub <daletrub@gmail.com> wrote:
Anthony,
Thank you for your suggestions! We are working on 1), but need to re-integrate code we've changed, and do regression testing. That's in progress, and we expect to be able to upgrade by the end of next week.
We did manage to do 3) and 4), and we now have SIP logs (attached). Are you able to see anything that's out of the ordinary that we should be paying attention to?
Best, Dale
On Fri, May 22, 2009 at 5:47 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
1) update to lastest trunk (you are at least 1000 revisions behind) 2) disable the presence debug in sofia.conf 3) enable sip trace instead "sofia profile internal siptrace on" 4) reproduce your problem.
Make sure you include more of the log from before the hangup happened. The one you posted here is missing some of the info from the few seconds prior but with the incomplete info it looks like the other side sent a BYE ending the call.
On Thu, May 21, 2009 at 10:09 PM, Dale Trub <daletrub@gmail.com> wrote:
Thanks Brian! To answer your questions: Freeswitch svn revision: 12148 Centos rev: 2.6.18-92.el5 And apologies, actually I guess we're using g711 not 729. Jason: I agree it would seem to be on the switch/telco side. And, the telco says many other people are in the same set-up as us and don't have any issues, so they're insisting it's on our end. On Thu, May 21, 2009 at 7:28 PM, Brian West <brian@freeswitch.org> wrote:
On May 21, 2009, at 9:15 PM, Dale Trub wrote:
We're running FreeSwitch as part of a teleconferencing service, inside a telcom (so no internet latency/NAT issues) and using g.729
So you're using g729 with conferences?
We are receiving some complaints of dropped calls, including from landlines. This means they join the conference, and x minutes in they simply drop. I know that cellphones tend to drop calls frequently, but landlines are pretty reliable, and we're hearing it a lot. From the FreeSwitch side of things, it just looks like those callers hung up (but then dialed back in just a moment later). I'm attaching two different snippets of the FS log files where these issues are occurring.
Next time please call them .txt because you cause extra work to have to open them otherwise.
Does anyone have any recommendations about how to troubleshoot this? Any known issues/patches in FS that could be biting us?
Depends you failed to include some very valid info such as what version or svn rev you're running and what linux distro.
Is there some SIP logging we can do to debug?
Yes covered on the wiki. http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
Are there any paid contractors avail who would have the expertise to look into this?
email consulting@freeswitch.org
Any help appreciated ... this is a major issue for us! Thanks much, -Dale
Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
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AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
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Dale Trub <daletrub@gmail.com> wrote:
Does anyone have any recommendations about how to troubleshoot this? sofia profile external siptrace on and watch the SIP traces to see what happens.
Any known issues/patches in FS that could be biting us?
You didn't say which version you were running. Does the problem still occur with the latest version from svn trunk? I assume there is a SIP service provider somewhere between your FreeSWITCH system and the PSTN. It is quite possible that the problem is on their end, especially if their system is terminating the calls. From the logs, it just looks like a standard hangup, which leads me to wonder whether the other side is responsible. Collecting SIP traces would seem to be a good idea. Disclaimer: I'm not an expert, and I'm sure there are paid contractors in the community who will gladly help, for a fee.
participants (6)
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Anthony Minessale
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Brian West
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Dale Trub
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Diego Viola
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Jason White
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Jim Burke