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[Freeswitch-users] calls appear to be dropping ... from landlines


 
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diego.viola at gmail.com
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PostPosted: Thu May 21, 2009 9:24 pm    Post subject: [Freeswitch-users] calls appear to be dropping ... from land Reply with quote

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:
Quote:
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
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http://www.freeswitch.org



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brian at freeswitch.org
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PostPosted: Thu May 21, 2009 9:28 pm    Post subject: [Freeswitch-users] calls appear to be dropping ... from land Reply with quote

On May 21, 2009, at 9:15 PM, Dale Trub wrote:
Quote:
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?

Quote:
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.

Quote:
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.

Quote:
Is there some SIP logging we can do to debug?


Yes covered on the wiki. http://wiki.freeswitch.org/wiki/Debugging_Freeswitch

Quote:
Are there any paid contractors avail who would have the expertise to look into this?


email consulting@freeswitch.org (consulting@freeswitch.org)

Quote:
Any help appreciated ... this is a major issue for us!


Thanks much,


-Dale


Brian West
brian@freeswitch.org (brian@freeswitch.org)



-- Meet us at ClueCon! http://www.cluecon.com
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daletrub at gmail.com
Guest





PostPosted: Thu May 21, 2009 10:10 pm    Post subject: [Freeswitch-users] calls appear to be dropping ... from land Reply with quote

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 (brian@freeswitch.org)> wrote:
Quote:

On May 21, 2009, at 9:15 PM, Dale Trub wrote:

Quote:
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?

Quote:
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.

Quote:
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.

Quote:
Is there some SIP logging we can do to debug?



Yes covered on the wiki.  http://wiki.freeswitch.org/wiki/Debugging_Freeswitch

Quote:
Are there any paid contractors avail who would have the expertise to look into this?



email consulting@freeswitch.org (consulting@freeswitch.org)

Quote:
Any help appreciated ... this is a major issue for us!


Thanks much,


-Dale




Brian West
brian@freeswitch.org (brian@freeswitch.org)



-- Meet us at ClueCon!  http://www.cluecon.com










_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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brian at freeswitch.org
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PostPosted: Thu May 21, 2009 11:48 pm    Post subject: [Freeswitch-users] calls appear to be dropping ... from land Reply with quote

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:
Quote:
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 (brian@freeswitch.org)



-- Meet us at ClueCon! http://www.cluecon.com
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anthony.minessale at g...
Guest





PostPosted: Fri May 22, 2009 8:02 am    Post subject: [Freeswitch-users] calls appear to be dropping ... from land Reply with quote

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 (daletrub@gmail.com)> wrote:
Quote:
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 (brian@freeswitch.org)> wrote:


Quote:


On May 21, 2009, at 9:15 PM, Dale Trub wrote:

Quote:
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?

Quote:
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.

Quote:
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.

Quote:
Is there some SIP logging we can do to debug?



Yes covered on the wiki.  http://wiki.freeswitch.org/wiki/Debugging_Freeswitch

Quote:
Are there any paid contractors avail who would have the expertise to look into this?



email consulting@freeswitch.org (consulting@freeswitch.org)

Quote:
Any help appreciated ... this is a major issue for us!


Thanks much,


-Dale




Brian West
brian@freeswitch.org (brian@freeswitch.org)



-- Meet us at ClueCon!  http://www.cluecon.com












_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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
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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 ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
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daletrub at gmail.com
Guest





PostPosted: Wed May 27, 2009 7:00 pm    Post subject: [Freeswitch-users] calls appear to be dropping ... from land Reply with quote

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 (anthony.minessale@gmail.com)> wrote:
Quote:
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 (daletrub@gmail.com)> wrote:
Quote:
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 (brian@freeswitch.org)> wrote:


Quote:


On May 21, 2009, at 9:15 PM, Dale Trub wrote:

Quote:
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?

Quote:
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.

Quote:
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.

Quote:
Is there some SIP logging we can do to debug?



Yes covered on the wiki.  http://wiki.freeswitch.org/wiki/Debugging_Freeswitch

Quote:
Are there any paid contractors avail who would have the expertise to look into this?



email consulting@freeswitch.org (consulting@freeswitch.org)

Quote:
Any help appreciated ... this is a major issue for us!


Thanks much,


-Dale




Brian West
brian@freeswitch.org (brian@freeswitch.org)



-- Meet us at ClueCon!  http://www.cluecon.com












_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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 (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 ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

_______________________________________________
Freeswitch-users mailing list
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anthony.minessale at g...
Guest





PostPosted: Thu May 28, 2009 8:11 am    Post subject: [Freeswitch-users] calls appear to be dropping ... from land Reply with quote

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 (daletrub@gmail.com)> wrote:
Quote:
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 (anthony.minessale@gmail.com)> wrote:
Quote:
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 (daletrub@gmail.com)> wrote:
Quote:
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 (brian@freeswitch.org)> wrote:


Quote:


On May 21, 2009, at 9:15 PM, Dale Trub wrote:

Quote:
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?

Quote:
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.

Quote:
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.

Quote:
Is there some SIP logging we can do to debug?



Yes covered on the wiki.  http://wiki.freeswitch.org/wiki/Debugging_Freeswitch

Quote:
Are there any paid contractors avail who would have the expertise to look into this?



email consulting@freeswitch.org (consulting@freeswitch.org)

Quote:
Any help appreciated ... this is a major issue for us!


Thanks much,


-Dale




Brian West
brian@freeswitch.org (brian@freeswitch.org)



-- Meet us at ClueCon!  http://www.cluecon.com












_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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 (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 ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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 (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 ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
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anthony.minessale at g...
Guest





PostPosted: Thu May 28, 2009 8:23 am    Post subject: [Freeswitch-users] calls appear to be dropping ... from land Reply with quote

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 (anthony.minessale@gmail.com)> wrote:
Quote:
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 (daletrub@gmail.com)> wrote:
Quote:
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 (anthony.minessale@gmail.com)> wrote:
Quote:
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 (daletrub@gmail.com)> wrote:
Quote:
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 (brian@freeswitch.org)> wrote:


Quote:


On May 21, 2009, at 9:15 PM, Dale Trub wrote:

Quote:
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?

Quote:
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.

Quote:
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.

Quote:
Is there some SIP logging we can do to debug?



Yes covered on the wiki.  http://wiki.freeswitch.org/wiki/Debugging_Freeswitch

Quote:
Are there any paid contractors avail who would have the expertise to look into this?



email consulting@freeswitch.org (consulting@freeswitch.org)

Quote:
Any help appreciated ... this is a major issue for us!


Thanks much,


-Dale




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jim at evolutiontel.net
Guest





PostPosted: Tue Jun 02, 2009 7:27 am    Post subject: [Freeswitch-users] calls appear to be dropping ... from land Reply with quote

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:
Quote:
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:
Quote:

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:
Quote:

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:
Quote:

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:
Quote:

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:
Quote:

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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ClueCon http://www.cluecon.com/

AIM: anthm
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GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
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iax:guest@conference.freeswitch.org/888
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--
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
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pstn:213-799-1400

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