Re: [Freeswitch-users] SIP Re-invite

2009-12-18 Thread DJB 
Thank you Mike for your suggestion on IRC.  We did what you recommend and found 
out it's the iptables issue that we thought it was not there at the beginning 
since we saw the first 2 invites from the far end fine, but somehow it has 
something to do with the 3rd invite.

I did close the Jira that I thought it was a bug.  Thank you again for the 
community and your support.

Dorn B.





From: Michael Jerris 
To: [email protected]
Sent: Fri, December 18, 2009 9:37:16 AM
Subject: Re: [Freeswitch-users] SIP Re-invite

I read through the trace, can you clarify where the missing invite is?  I think 
I see everything in the sofia trace.

Mike


On Dec 18, 2009, at 3:10 AM, DJB wrote:

Mike,
>
>
>My latest traces that I captured were done within the FS box:  
>http://pastebin.freeswitch.org/11541 
>
>
>Thank you,
>Dorn B.
>
>

From: Michael Jerris 
>To: [email protected]
>Sent: Thu, December 17, 2009 8:03:46 AM
>Subject: Re: [Freeswitch-users] SIP Re-invite
>
>are you doing this trace from the freeswitch box itself?
>
>
>Mike
>
>
>On Dec 17, 2009, at 10:48 AM, DJB wrote:
>
>Anthony,
>> 
>>I have pasted the invite sip trace here:  http://pastebin.freeswitch.org/11536
>>Please advise if you need further info.
>> 
>>Thank you.
>>
>
>___
>FreeSWITCH-users mailing list
>[email protected]
>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
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-18 Thread DJB 
Mike,

There are 2 traces in there.  One is from freeswitch/sofia siptrace debug and 
the other one from ngrep for your comparison.

The missing re-invite in FS is at 2009/12/17 17:25:55.207747 in ngrep portion, 
but it did not show in FS siptrace debug.

Thank you,
Dorn B.




From: Michael Jerris 
To: [email protected]
Sent: Fri, December 18, 2009 9:37:16 AM
Subject: Re: [Freeswitch-users] SIP Re-invite

I read through the trace, can you clarify where the missing invite is?  I think 
I see everything in the sofia trace.

Mike


On Dec 18, 2009, at 3:10 AM, DJB wrote:

Mike,
>
>
>My latest traces that I captured were done within the FS box:  
>http://pastebin.freeswitch.org/11541 
>
>
>Thank you,
>Dorn B.
>
>

From: Michael Jerris 
>To: [email protected]
>Sent: Thu, December 17, 2009 8:03:46 AM
>Subject: Re: [Freeswitch-users] SIP Re-invite
>
>are you doing this trace from the freeswitch box itself?
>
>
>Mike
>
>
>On Dec 17, 2009, at 10:48 AM, DJB wrote:
>
>Anthony,
>> 
>>I have pasted the invite sip trace here:  http://pastebin.freeswitch.org/11536
>>Please advise if you need further info.
>> 
>>Thank you.
>>
>
>___
>FreeSWITCH-users mailing list
>[email protected]
>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
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-18 Thread DJB 
Mike,

There are 2 traces in there.  One is from freeswitch/sofia siptrace debug and 
the other one from ngrep for your comparison.

The missing re-invite in FS is at 2009/12/17 17:25:55.207747 in ngrep portion, 
but it did not show in FS siptrace debug.

Thank you,
Dorn B.




From: Michael Jerris 
To: [email protected]
Sent: Fri, December 18, 2009 9:37:16 AM
Subject: Re: [Freeswitch-users] SIP Re-invite

I read through the trace, can you clarify where the missing invite is?  I think 
I see everything in the sofia trace.

Mike


On Dec 18, 2009, at 3:10 AM, DJB wrote:

Mike,
>
>
>My latest traces that I captured were done within the FS box:  
>http://pastebin.freeswitch.org/11541 
>
>
>Thank you,
>Dorn B.
>
>

From: Michael Jerris 
>To: [email protected]
>Sent: Thu, December 17, 2009 8:03:46 AM
>Subject: Re: [Freeswitch-users] SIP Re-invite
>
>are you doing this trace from the freeswitch box itself?
>
>
>Mike
>
>
>On Dec 17, 2009, at 10:48 AM, DJB wrote:
>
>Anthony,
>> 
>>I have pasted the invite sip trace here:  http://pastebin.freeswitch.org/11536
>>Please advise if you need further info.
>> 
>>Thank you.
>>
>
>___
>FreeSWITCH-users mailing list
>[email protected]
>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
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-18 Thread Michael Jerris 
I read through the trace, can you clarify where the missing invite is?  I think 
I see everything in the sofia trace.

Mike

On Dec 18, 2009, at 3:10 AM, DJB wrote:

> Mike,
> 
> My latest traces that I captured were done within the FS box:  
> http://pastebin.freeswitch.org/11541 
> 
> Thank you,
> Dorn B.
> From: Michael Jerris 
> To: [email protected]
> Sent: Thu, December 17, 2009 8:03:46 AM
> Subject: Re: [Freeswitch-users] SIP Re-invite
> 
> are you doing this trace from the freeswitch box itself?
> 
> Mike
> 
> On Dec 17, 2009, at 10:48 AM, DJB wrote:
> 
>> Anthony,
>>  
>> I have pasted the invite sip trace here:  
>> http://pastebin.freeswitch.org/11536
>> Please advise if you need further info.
>>  
>> Thank you.
> 
> 
> ___
> FreeSWITCH-users mailing list
> [email protected]
> 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
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-18 Thread DJB 
Mike,

My latest traces that I captured were done within the FS box:  
http://pastebin.freeswitch.org/11541 

Thank you,
Dorn B.



From: Michael Jerris 
To: [email protected]
Sent: Thu, December 17, 2009 8:03:46 AM
Subject: Re: [Freeswitch-users] SIP Re-invite

are you doing this trace from the freeswitch box itself?

Mike


On Dec 17, 2009, at 10:48 AM, DJB wrote:

Anthony,
> 
>I have pasted the invite sip trace here:  http://pastebin.freeswitch.org/11536
>Please advise if you need further info.
> 
>Thank you.
>



  ___
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-17 Thread DJB 
Please advise whether I should put a request in JIRA. 
http://pastebin.freeswitch.org/11541

Thank you.




From: DJB 
To: [email protected]
Sent: Thu, December 17, 2009 9:35:27 AM
Subject: Re: [Freeswitch-users] SIP Re-invite


Anthony/Michael,

I finally have a complete traces of a call at 
http://pastebin.freeswitch.org/11539

There are 2 traces in there from within the same box.  One is from 
freeswitch/sofia siptrace debug and the other one from ngrep for your 
comparison.

The missing re-invite in FS is at 2009/12/17 17:25:55.207747 


Thank you.




From: Anthony Minessale 
To: [email protected]
Sent: Thu, December 17, 2009 7:57:42 AM
Subject: Re: [Freeswitch-users] SIP Re-invite

The question was:

Are you doing the packet capture on the actual FS box using tshark or tcpdump?



On Thu, Dec 17, 2009 at 9:48 AM, DJB  wrote:

Anthony,
> 
>I have pasted the invite sip trace here:  http://pastebin.freeswitch.org/11536
>Please advise if you need further info.
> 
>Thank you.
>
>
>
>

 From: Anthony Minessale 
>To: [email protected]
>Sent: Wed, December 16, 2009 3:42:48 PM
>Subject: Re: [Freeswitch-users] SIP Re-invite
>
>
>>that means the invite is not matching the call dialog
>compare the via tags and call-id etc
>
>
>
>On Wed, Dec 16, 2009 at 5:29 PM, DJB  wrote:
>
>We have a customer that we are sending calls to off the FS and here is the 
>issue: 
>> 
>>Call is initially setup fine and they send a first re-invite with media 
>>0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first 
>>re-invite fine 
>> 
>>They then send a second re-invite with their media IP to cut through media 
>>and the FS sends a 200 OK to this fine. At this point the call is fine 
>> 
>>30 minutes later they send a third re-invite because according to them it is 
>>strictly for the purpose of “keep alive” per RFC 4028. This third re-invite 
>>has the exact same media IP and UDP pot information as the second re-invite 
>>does. The problem is FS does not respond to this third re-invite AT ALL. It 
>>doesn’t send a 100 trying a 200 OK nothing so this causes the call to be 
>>dropped as the other end does not recieve a response from FS.  
>>
>>
>>One more thing, we did not see the third re-invite in sofia siptrace, but we 
>>do see it in ethereal, which is kind of odds.
>>
>>
>>We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.
>>
>>
>>Thank you very much.
>>
>>___
>>FreeSWITCH-users mailing list
>>[email protected]
>>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/
>Twitter: http://twitter.com/FreeSWITCH_wire
>
>AIM: anthm
>MSN:[email protected]
>GTALK/JABBER/PAYPAL:[email protected]
>>IRC: irc.freenode.net #freeswitch
>
>FreeSWITCH Developer Conference
>sip:[email protected]
>iax:[email protected]/888
>googletalk:[email protected]
>>pstn:+19193869900
>
>
>___
>>FreeSWITCH-users mailing list
>[email protected]
>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/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:[email protected]
GTALK/JABBER/PAYPAL:[email protected]
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[email protected]
iax:[email protected]/888
googletalk:[email protected]
pstn:+19193869900


  ___
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-17 Thread DJB 
Yes, I have a complete trace here:  http://pastebin.freeswitch.org/11541




From: David Knell 
To: [email protected]
Sent: Thu, December 17, 2009 9:50:22 AM
Subject: Re: [Freeswitch-users] SIP Re-invite

Can you post the full packets with Ethernet, IP, UDP headers as well, or
upload a pcap file?

I'll add the change in 'Max-Forwards' from 70 to 69 between the two
packets to my things to be suspicious of list.

--Dave

> The trace that I pasted on the pastebin was from our
> analyzer,Tektronix spectra2 that was sitting between FS and customer.
> I also had the FS sip trace on and compare with the trace from Spectra
> when I found out about the 3rd re-invite was missing from FS.
>  
> Thank you.
> 
> 
> 
> __
> From: Anthony Minessale 
> To: [email protected]
> Sent: Thu, December 17, 2009 7:57:42 AM
> Subject: Re: [Freeswitch-users] SIP Re-invite
> 
> The question was:
> 
> Are you doing the packet capture on the actual FS box using tshark or
> tcpdump?
> 
> 
> On Thu, Dec 17, 2009 at 9:48 AM, DJB  wrote:
> Anthony,
>  
> I have pasted the invite sip trace here:
>http://pastebin.freeswitch.org/11536
> Please advise if you need further info.
>  
> Thank you.
>
>
>
> __
> From: Anthony Minessale 
>     To: [email protected]
> Sent: Wed, December 16, 2009 3:42:48 PM
> Subject: Re: [Freeswitch-users] SIP Re-invite
>
>
>
> that means the invite is not matching the call dialog
> compare the via tags and call-id etc
>
>
> On Wed, Dec 16, 2009 at 5:29 PM, DJB 
> wrote:
> We have a customer that we are sending calls to off
> the FS and here is the issue: 
>
>  
>
> Call is initially setup fine and they send a first
> re-invite with media 0.0.0.0 to place the caller on
> hold. FS sends a 200 ok to this first re-invite fine 
>
>  
>
> They then send a second re-invite with their media IP
> to cut through media and the FS sends a 200 OK to this
> fine. At this point the call is fine 
>
>  
>
> 30 minutes later they send a third re-invite because
> according to them it is strictly for the purpose of
> “keep alive” per RFC 4028. This third re-invite has
> the exact same media IP and UDP pot information as the
> second re-invite does. The problem is FS does not
> respond to this third re-invite AT ALL. It doesn’t
> send a 100 trying a 200 OK nothing so this causes the
> call to be dropped as the other end does not recieve a
> response from FS.  
>
>
> One more thing, we did not see the third re-invite in
> sofia siptrace, but we do see it in ethereal, which is
> kind of odds.
>
>
> We are running FreeSWITCH Version 1.0.trunk (15979) in
> bypass media mode.
>
>
> Thank you very much.
>
>
>
>
>
>
> ___
> FreeSWITCH-users mailing list
>[email protected]
>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/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:[email protected]
> GTALK/JABBER/PAYPAL:[email protected]
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:[email protected]

Re: [Freeswitch-users] SIP Re-invite

2009-12-17 Thread David Knell 
Can you post the full packets with Ethernet, IP, UDP headers as well, or
upload a pcap file?

I'll add the change in 'Max-Forwards' from 70 to 69 between the two
packets to my things to be suspicious of list.

--Dave

> The trace that I pasted on the pastebin was from our
> analyzer,Tektronix spectra2 that was sitting between FS and customer.
> I also had the FS sip trace on and compare with the trace from Spectra
> when I found out about the 3rd re-invite was missing from FS.
>  
> Thank you.
> 
> 
> 
> __
> From: Anthony Minessale 
> To: [email protected]
> Sent: Thu, December 17, 2009 7:57:42 AM
> Subject: Re: [Freeswitch-users] SIP Re-invite
> 
> The question was:
> 
> Are you doing the packet capture on the actual FS box using tshark or
> tcpdump?
> 
> 
> On Thu, Dec 17, 2009 at 9:48 AM, DJB  wrote:
> Anthony,
>  
> I have pasted the invite sip trace here:
> http://pastebin.freeswitch.org/11536
> Please advise if you need further info.
>  
> Thank you.
> 
> 
> 
> __
> From: Anthony Minessale 
>     To: [email protected]
> Sent: Wed, December 16, 2009 3:42:48 PM
> Subject: Re: [Freeswitch-users] SIP Re-invite
> 
> 
> 
> that means the invite is not matching the call dialog
> compare the via tags and call-id etc
> 
> 
> On Wed, Dec 16, 2009 at 5:29 PM, DJB 
> wrote:
> We have a customer that we are sending calls to off
> the FS and here is the issue: 
> 
>  
> 
> Call is initially setup fine and they send a first
> re-invite with media 0.0.0.0 to place the caller on
> hold. FS sends a 200 ok to this first re-invite fine 
> 
>  
> 
> They then send a second re-invite with their media IP
> to cut through media and the FS sends a 200 OK to this
> fine. At this point the call is fine 
> 
>  
> 
> 30 minutes later they send a third re-invite because
> according to them it is strictly for the purpose of
> “keep alive” per RFC 4028. This third re-invite has
> the exact same media IP and UDP pot information as the
> second re-invite does. The problem is FS does not
> respond to this third re-invite AT ALL. It doesn’t
> send a 100 trying a 200 OK nothing so this causes the
> call to be dropped as the other end does not recieve a
> response from FS.  
> 
> 
> One more thing, we did not see the third re-invite in
> sofia siptrace, but we do see it in ethereal, which is
> kind of odds.
> 
> 
> We are running FreeSWITCH Version 1.0.trunk (15979) in
> bypass media mode.
> 
> 
> Thank you very much.
> 
> 
> 
> 
> 
> 
> ___
> FreeSWITCH-users mailing list
> [email protected]
> 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/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:[email protected]
> GTALK/JABBER/PAYPAL:[email protected]
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:[email protected]
> iax:[email protected]/888
> googletalk:[email protected]
> pstn: +19193869900  +19193869900 
> 
> 
> 
>

Re: [Freeswitch-users] SIP Re-invite

2009-12-17 Thread DJB 
I am sorry; here is the complete one:  http://pastebin.freeswitch.org/11540

Thank you.




From: DJB 
To: [email protected]
Sent: Thu, December 17, 2009 9:35:27 AM
Subject: Re: [Freeswitch-users] SIP Re-invite


Anthony/Michael,

I finally have a complete traces of a call at 
http://pastebin.freeswitch.org/11539

There are 2 traces in there from within the same box.  One is from 
freeswitch/sofia siptrace debug and the other one from ngrep for your 
comparison.

The missing re-invite in FS is at 2009/12/17 17:25:55.207747 


Thank you.




From: Anthony Minessale 
To: [email protected]
Sent: Thu, December 17, 2009 7:57:42 AM
Subject: Re: [Freeswitch-users] SIP Re-invite

The question was:

Are you doing the packet capture on the actual FS box using tshark or tcpdump?



On Thu, Dec 17, 2009 at 9:48 AM, DJB  wrote:

Anthony,
> 
>I have pasted the invite sip trace here:  http://pastebin.freeswitch.org/11536
>Please advise if you need further info.
> 
>Thank you.
>
>
>
>

 From: Anthony Minessale 
>To: [email protected]
>Sent: Wed, December 16, 2009 3:42:48 PM
>Subject: Re: [Freeswitch-users] SIP Re-invite
>
>
>>that means the invite is not matching the call dialog
>compare the via tags and call-id etc
>
>
>
>On Wed, Dec 16, 2009 at 5:29 PM, DJB  wrote:
>
>We have a customer that we are sending calls to off the FS and here is the 
>issue: 
>> 
>>Call is initially setup fine and they send a first re-invite with media 
>>0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first 
>>re-invite fine 
>> 
>>They then send a second re-invite with their media IP to cut through media 
>>and the FS sends a 200 OK to this fine. At this point the call is fine 
>> 
>>30 minutes later they send a third re-invite because according to them it is 
>>strictly for the purpose of “keep alive” per RFC 4028. This third re-invite 
>>has the exact same media IP and UDP pot information as the second re-invite 
>>does. The problem is FS does not respond to this third re-invite AT ALL. It 
>>doesn’t send a 100 trying a 200 OK nothing so this causes the call to be 
>>dropped as the other end does not recieve a response from FS.  
>>
>>
>>One more thing, we did not see the third re-invite in sofia siptrace, but we 
>>do see it in ethereal, which is kind of odds.
>>
>>
>>We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.
>>
>>
>>Thank you very much.
>>
>>___
>>FreeSWITCH-users mailing list
>>[email protected]
>>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/
>Twitter: http://twitter.com/FreeSWITCH_wire
>
>AIM: anthm
>MSN:[email protected]
>GTALK/JABBER/PAYPAL:[email protected]
>>IRC: irc.freenode.net #freeswitch
>
>FreeSWITCH Developer Conference
>sip:[email protected]
>iax:[email protected]/888
>googletalk:[email protected]
>>pstn:+19193869900
>
>
>___
>>FreeSWITCH-users mailing list
>[email protected]
>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/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:[email protected]
GTALK/JABBER/PAYPAL:[email protected]
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[email protected]
iax:[email protected]/888
googletalk:[email protected]
pstn:+19193869900


  ___
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-17 Thread DJB 
Anthony/Michael,

I finally have a complete traces of a call at 
http://pastebin.freeswitch.org/11539

There are 2 traces in there from within the same box.  One is from 
freeswitch/sofia siptrace debug and the other one from ngrep for your 
comparison.

The missing re-invite in FS is at 2009/12/17 17:25:55.207747 


Thank you.




From: Anthony Minessale 
To: [email protected]
Sent: Thu, December 17, 2009 7:57:42 AM
Subject: Re: [Freeswitch-users] SIP Re-invite

The question was:

Are you doing the packet capture on the actual FS box using tshark or tcpdump?



On Thu, Dec 17, 2009 at 9:48 AM, DJB  wrote:

Anthony,
> 
>I have pasted the invite sip trace here:  http://pastebin.freeswitch.org/11536
>Please advise if you need further info.
> 
>Thank you.
>
>
>
>

 From: Anthony Minessale 
>To: [email protected]
>Sent: Wed, December 16, 2009 3:42:48 PM
>Subject: Re: [Freeswitch-users] SIP Re-invite
>
>
>>that means the invite is not matching the call dialog
>compare the via tags and call-id etc
>
>
>
>On Wed, Dec 16, 2009 at 5:29 PM, DJB  wrote:
>
>We have a customer that we are sending calls to off the FS and here is the 
>issue: 
>> 
>>Call is initially setup fine and they send a first re-invite with media 
>>0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first 
>>re-invite fine 
>> 
>>They then send a second re-invite with their media IP to cut through media 
>>and the FS sends a 200 OK to this fine. At this point the call is fine 
>> 
>>30 minutes later they send a third re-invite because according to them it is 
>>strictly for the purpose of “keep alive” per RFC 4028. This third re-invite 
>>has the exact same media IP and UDP pot information as the second re-invite 
>>does. The problem is FS does not respond to this third re-invite AT ALL. It 
>>doesn’t send a 100 trying a 200 OK nothing so this causes the call to be 
>>dropped as the other end does not recieve a response from FS.  
>>
>>
>>One more thing, we did not see the third re-invite in sofia siptrace, but we 
>>do see it in ethereal, which is kind of odds.
>>
>>
>>We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.
>>
>>
>>Thank you very much.
>>
>>___
>>FreeSWITCH-users mailing list
>>[email protected]
>>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/
>Twitter: http://twitter.com/FreeSWITCH_wire
>
>AIM: anthm
>MSN:[email protected]
>GTALK/JABBER/PAYPAL:[email protected]
>>IRC: irc.freenode.net #freeswitch
>
>FreeSWITCH Developer Conference
>sip:[email protected]
>iax:[email protected]/888
>googletalk:[email protected]
>>pstn:+19193869900
>
>
>___
>>FreeSWITCH-users mailing list
>[email protected]
>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/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:[email protected]
GTALK/JABBER/PAYPAL:[email protected]
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[email protected]
iax:[email protected]/888
googletalk:[email protected]
pstn:+19193869900



  ___
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-17 Thread DJB 
It only happened to the calls from this customer that keeps sending re-invite 
every 30 minutes, since their switch is expecting a reply back from those 
re-invite and FS did not respond back to those re-invite.

Thank you. 





From: Michael Jerris 
To: [email protected]
Sent: Thu, December 17, 2009 7:36:44 AM
Subject: Re: [Freeswitch-users] SIP Re-invite

if you don't see it in sofia siptrace but do see it in tcpdump capture then 
something very ugly is going on.  Either sofia has hung up completely and is 
not listening on that port anymore (can other calls go through?) or the packet 
you see in tcpdump is not really going to the right port.  Can you confirm 
which one? 

Mike


On Dec 16, 2009, at 6:29 PM, DJB wrote:

We have a customer that we are sending calls to off the FS and here is the 
issue:
> 
>Call is initially setup fine and they send a first re-invite with media 
>0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite 
>fine
> 
>They then send a second re-invite with their media IP to cut through media and 
>the FS sends a 200 OK to this fine. At this point the call is fine
> 
>30 minutes later they send a third re-invite because according to them it is 
>strictly for the purpose of “keep alive” per RFC 4028. This third re-invite 
>has the exact same media IP and UDP pot information as the second re-invite 
>does. The problem is FS does not respond to this third re-invite AT ALL. It 
>doesn’t send a 100 trying a 200 OK nothing so this causes the call to be 
>dropped as the other end does not recieve a response from FS.  
>
>
>One more thing, we did not see the third re-invite in sofia siptrace, but we 
>do see it in ethereal, which is kind of odds.
>
>
>We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.



  ___
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-17 Thread DJB 
The trace that I pasted on the pastebin was from our analyzer,Tektronix 
spectra2 that was sitting between FS and customer.  I also had the FS sip trace 
on and compare with the trace from Spectra when I found out about the 3rd 
re-invite was missing from FS.

Thank you.





From: Anthony Minessale 
To: [email protected]
Sent: Thu, December 17, 2009 7:57:42 AM
Subject: Re: [Freeswitch-users] SIP Re-invite

The question was:

Are you doing the packet capture on the actual FS box using tshark or tcpdump?



On Thu, Dec 17, 2009 at 9:48 AM, DJB  wrote:

Anthony,
>
>I have pasted the invite sip trace here:  http://pastebin.freeswitch.org/11536
>Please advise if you need further info.
>
>Thank you.
>
>
>
>

From: Anthony Minessale 
>To: [email protected]
>Sent: Wed, December 16, 2009 3:42:48 PM
>Subject: Re: [Freeswitch-users] SIP Re-invite
>
>
>that means the invite is not matching the call dialog
>compare the via tags and call-id etc
>
>
>
>On Wed, Dec 16, 2009 at 5:29 PM, DJB  wrote:
>
>We have a customer that we are sending calls to off the FS and here is the 
>issue: 
>> 
>>Call is initially setup fine and they send a first re-invite with media 
>>0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first 
>>re-invite fine 
>> 
>>They then send a second re-invite with their media IP to cut through media 
>>and the FS sends a 200 OK to this fine. At this point the call is fine 
>> 
>>30 minutes later they send a third re-invite because according to them it is 
>>strictly for the purpose of “keep alive” per RFC 4028. This third re-invite 
>>has the exact same media IP and UDP pot information as the second re-invite 
>>does. The problem is FS does not respond to this third re-invite AT ALL. It 
>>doesn’t send a 100 trying a 200 OK nothing so this causes the call to be 
>>dropped as the other end does not recieve a response from FS.  
>>
>>
>>One more thing, we did not see the third re-invite in sofia siptrace, but we 
>>do see it in ethereal, which is kind of odds.
>>
>>
>>We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.
>>
>>
>>Thank you very much.
>>
>>___
>>FreeSWITCH-users mailing list
>>[email protected]
>>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/
>Twitter: http://twitter.com/FreeSWITCH_wire
>
>AIM: anthm
>MSN:[email protected]
>GTALK/JABBER/PAYPAL:[email protected]
>IRC: irc.freenode.net #freeswitch
>
>FreeSWITCH Developer Conference
>sip:[email protected]
>iax:[email protected]/888
>googletalk:[email protected]
>pstn: +19193869900  +19193869900 
>
>
>___
>FreeSWITCH-users mailing list
>[email protected]
>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/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:[email protected]
GTALK/JABBER/PAYPAL:[email protected]
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[email protected]
iax:[email protected]/888
googletalk:[email protected]
pstn:+19193869900



  ___
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-17 Thread David Knell 
I'd be suspicious of:
(a) the CSeq going backwards from 4 to 3 between re-invites 2 and 3;
(b) the branch on the Via tag changing
(c) (not sure about this one) the SDP session ID and version changing
for what's the same session.

--Dave


> Anthony,
>  
> I have pasted the invite sip trace here:
> http://pastebin.freeswitch.org/11536
> Please advise if you need further info.
>  
> Thank you.
> 
> 
> 
> __
> From: Anthony Minessale 
> To: [email protected]
> Sent: Wed, December 16, 2009 3:42:48 PM
> Subject: Re: [Freeswitch-users] SIP Re-invite
> 
> that means the invite is not matching the call dialog
> compare the via tags and call-id etc
> 
> 
> On Wed, Dec 16, 2009 at 5:29 PM, DJB  wrote:
> We have a customer that we are sending calls to off the FS and
> here is the issue: 
> 
>  
> 
> Call is initially setup fine and they send a first re-invite
> with media 0.0.0.0 to place the caller on hold. FS sends a 200
> ok to this first re-invite fine 
> 
>  
> 
> They then send a second re-invite with their media IP to cut
> through media and the FS sends a 200 OK to this fine. At this
> point the call is fine 
> 
>  
> 
> 30 minutes later they send a third re-invite because according
> to them it is strictly for the purpose of “keep alive” per RFC
> 4028. This third re-invite has the exact same media IP and UDP
> pot information as the second re-invite does. The problem is
> FS does not respond to this third re-invite AT ALL. It doesn’t
> send a 100 trying a 200 OK nothing so this causes the call to
> be dropped as the other end does not recieve a response from
> FS.  
> 
> 
> One more thing, we did not see the third re-invite in sofia
> siptrace, but we do see it in ethereal, which is kind of odds.
> 
> 
> We are running FreeSWITCH Version 1.0.trunk (15979) in bypass
> media mode.
> 
> 
> Thank you very much.
> 
> 
> 
> 
> 
> 
> ___
> FreeSWITCH-users mailing list
> [email protected]
> 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/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:[email protected]
> GTALK/JABBER/PAYPAL:[email protected]
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:[email protected]
> iax:[email protected]/888
> googletalk:[email protected]
> pstn:+19193869900
> 
> 
> ___
> FreeSWITCH-users mailing list
> [email protected]
> 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
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-17 Thread Michael Jerris 
are you doing this trace from the freeswitch box itself?

Mike

On Dec 17, 2009, at 10:48 AM, DJB wrote:

> Anthony,
>  
> I have pasted the invite sip trace here:  http://pastebin.freeswitch.org/11536
> Please advise if you need further info.
>  
> Thank you.

___
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-17 Thread Anthony Minessale 
The question was:

Are you doing the packet capture on the actual FS box using tshark or
tcpdump?


On Thu, Dec 17, 2009 at 9:48 AM, DJB  wrote:

> Anthony,
>
> I have pasted the invite sip trace here:
> http://pastebin.freeswitch.org/11536
> Please advise if you need further info.
>
> Thank you.
>
>  --
> *From:* Anthony Minessale 
> *To:* [email protected]
> *Sent:* Wed, December 16, 2009 3:42:48 PM
> *Subject:* Re: [Freeswitch-users] SIP Re-invite
>
> that means the invite is not matching the call dialog
> compare the via tags and call-id etc
>
>
> On Wed, Dec 16, 2009 at 5:29 PM, DJB  wrote:
>
>>   We have a customer that we are sending calls to off the FS and here is
>> the issue:
>>
>>
>>
>> Call is initially setup fine and they send a first re-invite with media
>> 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first
>> re-invite fine
>>
>>
>>
>> They then send a second re-invite with their media IP to cut through media
>> and the FS sends a 200 OK to this fine. At this point the call is fine
>>
>>
>>
>> 30 minutes later they send a third re-invite because according to them it
>> is strictly for the purpose of “keep alive” per RFC 4028. This third
>> re-invite has the exact same media IP and UDP pot information as the second
>> re-invite does. The problem is FS does not respond to this third re-invite
>> AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the
>> call to be dropped as the other end does not recieve a response from FS.
>>
>>
>> One more thing, we did not see the third re-invite in sofia siptrace, but
>> we do see it in ethereal, which is kind of odds.
>>
>>
>> We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.
>>
>>
>> Thank you very much.
>>
>>
>> ___
>> FreeSWITCH-users mailing list
>> [email protected]
>> 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/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:[email protected] 
> GTALK/JABBER/PAYPAL:[email protected]
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:[email protected] 
> iax:[email protected]/888
> googletalk:[email protected]
> pstn:+19193869900
>
>
> ___
> FreeSWITCH-users mailing list
> [email protected]
> 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/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:[email protected] 
GTALK/JABBER/PAYPAL:[email protected]
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[email protected] 
iax:[email protected]/888
googletalk:[email protected]
pstn:+19193869900
___
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-17 Thread Anthony Minessale 
Is the packet capture running on the FS box itself?


On Thu, Dec 17, 2009 at 9:36 AM, Michael Jerris  wrote:

> if you don't see it in sofia siptrace but do see it in tcpdump capture then
> something very ugly is going on.  Either sofia has hung up completely and is
> not listening on that port anymore (can other calls go through?) or the
> packet you see in tcpdump is not really going to the right port.  Can you
> confirm which one?
>
> Mike
>
> On Dec 16, 2009, at 6:29 PM, DJB wrote:
>
> We have a customer that we are sending calls to off the FS and here is the
> issue:
>
>
>
> Call is initially setup fine and they send a first re-invite with media
> 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first
> re-invite fine
>
>
>
> They then send a second re-invite with their media IP to cut through media
> and the FS sends a 200 OK to this fine. At this point the call is fine
>
>
>
> 30 minutes later they send a third re-invite because according to them it
> is strictly for the purpose of “keep alive” per RFC 4028. This third
> re-invite has the exact same media IP and UDP pot information as the second
> re-invite does. The problem is FS does not respond to this third re-invite
> AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the
> call to be dropped as the other end does not recieve a response from FS.
>
>
> One more thing, we did not see the third re-invite in sofia siptrace, but
> we do see it in ethereal, which is kind of odds.
>
>
> We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.
>
>
>
> ___
> FreeSWITCH-users mailing list
> [email protected]
> 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/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:[email protected] 
GTALK/JABBER/PAYPAL:[email protected]
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[email protected] 
iax:[email protected]/888
googletalk:[email protected]
pstn:+19193869900
___
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-17 Thread DJB 
Anthony,

I have pasted the invite sip trace here:  http://pastebin.freeswitch.org/11536
Please advise if you need further info.

Thank you.





From: Anthony Minessale 
To: [email protected]
Sent: Wed, December 16, 2009 3:42:48 PM
Subject: Re: [Freeswitch-users] SIP Re-invite

that means the invite is not matching the call dialog
compare the via tags and call-id etc



On Wed, Dec 16, 2009 at 5:29 PM, DJB  wrote:

We have a customer that we are sending calls to off the FS and here is the 
issue: 
> 
>Call is initially setup fine and they send a first re-invite with media 
>0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite 
>fine 
> 
>They then send a second re-invite with their media IP to cut through media and 
>the FS sends a 200 OK to this fine. At this point the call is fine 
> 
>30 minutes later they send a third re-invite because according to them it is 
>strictly for the purpose of “keep alive” per RFC 4028. This third re-invite 
>has the exact same media IP and UDP pot information as the second re-invite 
>does. The problem is FS does not respond to this third re-invite AT ALL. It 
>doesn’t send a 100 trying a 200 OK nothing so this causes the call to be 
>dropped as the other end does not recieve a response from FS.  
>
>
>One more thing, we did not see the third re-invite in sofia siptrace, but we 
>do see it in ethereal, which is kind of odds.
>
>
>We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.
>
>
>Thank you very much.
>
>___
>FreeSWITCH-users mailing list
>[email protected]
>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/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:[email protected]
GTALK/JABBER/PAYPAL:[email protected]
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[email protected]
iax:[email protected]/888
googletalk:[email protected]
pstn:+19193869900



  ___
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-17 Thread Michael Jerris 
if you don't see it in sofia siptrace but do see it in tcpdump capture then 
something very ugly is going on.  Either sofia has hung up completely and is 
not listening on that port anymore (can other calls go through?) or the packet 
you see in tcpdump is not really going to the right port.  Can you confirm 
which one?

Mike

On Dec 16, 2009, at 6:29 PM, DJB wrote:

> We have a customer that we are sending calls to off the FS and here is the 
> issue:
> 
>  
> 
> Call is initially setup fine and they send a first re-invite with media 
> 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first 
> re-invite fine
> 
>  
> 
> They then send a second re-invite with their media IP to cut through media 
> and the FS sends a 200 OK to this fine. At this point the call is fine
> 
>  
> 
> 30 minutes later they send a third re-invite because according to them it is 
> strictly for the purpose of “keep alive” per RFC 4028. This third re-invite 
> has the exact same media IP and UDP pot information as the second re-invite 
> does. The problem is FS does not respond to this third re-invite AT ALL. It 
> doesn’t send a 100 trying a 200 OK nothing so this causes the call to be 
> dropped as the other end does not recieve a response from FS.  
> 
> 
> 
> One more thing, we did not see the third re-invite in sofia siptrace, but we 
> do see it in ethereal, which is kind of odds.
> 
> 
> 
> We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.
> 

___
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-16 Thread Michael Giagnocavo 
FWIW, we’ve seen the same thing intermittently, haven’t had time/been able to 
get a solid repro to capture debug information.

Call ID and tags are all matching. After the re-invite fails and the remote end 
sends a BYE, FS does indeed respond to the re-invite.

-Michael

From: [email protected] 
[mailto:[email protected]] On Behalf Of DJB
Sent: Wednesday, December 16, 2009 6:00 PM
To: [email protected]
Subject: Re: [Freeswitch-users] SIP Re-invite

Call-ID are the same for 1st, 2nd, and 3rd INVITE.  The only thing I saw 
difference was the Via Branch value.  Would that be a problem, since 1st and 
2nd INVITE was also different and was okay.

Is there any other values that I should look at?

Thank you.


From: Anthony Minessale 
To: [email protected]
Sent: Wed, December 16, 2009 3:42:48 PM
Subject: Re: [Freeswitch-users] SIP Re-invite

that means the invite is not matching the call dialog
compare the via tags and call-id etc

On Wed, Dec 16, 2009 at 5:29 PM, DJB 
mailto:[email protected]>> wrote:
We have a customer that we are sending calls to off the FS and here is the 
issue:

Call is initially setup fine and they send a first re-invite with media 0.0.0.0 
to place the caller on hold. FS sends a 200 ok to this first re-invite fine

They then send a second re-invite with their media IP to cut through media and 
the FS sends a 200 OK to this fine. At this point the call is fine

30 minutes later they send a third re-invite because according to them it is 
strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has 
the exact same media IP and UDP pot information as the second re-invite does. 
The problem is FS does not respond to this third re-invite AT ALL. It doesn’t 
send a 100 trying a 200 OK nothing so this causes the call to be dropped as the 
other end does not recieve a response from FS.

One more thing, we did not see the third re-invite in sofia siptrace, but we do 
see it in ethereal, which is kind of odds.

We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.

Thank you very much.


___
FreeSWITCH-users mailing list
[email protected]<mailto:[email protected]>
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/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:[email protected]<mailto:msn%[email protected]>
GTALK/JABBER/PAYPAL:[email protected]<mailto:paypal%[email protected]>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:[email protected]<mailto:sip%[email protected]>
iax:[email protected]/888<http://iax:[email protected]/888>
googletalk:[email protected]<mailto:googletalk%3aconf%[email protected]>
pstn:+19193869900

___
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-16 Thread DJB 
Call-ID are the same for 1st, 2nd, and 3rd INVITE.  The only thing I saw 
difference was the Via Branch value.  Would that be a problem, since 1st and 
2nd INVITE was also different and was okay.

Is there any other values that I should look at?  

Thank you.




From: Anthony Minessale 
To: [email protected]
Sent: Wed, December 16, 2009 3:42:48 PM
Subject: Re: [Freeswitch-users] SIP Re-invite

that means the invite is not matching the call dialog
compare the via tags and call-id etc



On Wed, Dec 16, 2009 at 5:29 PM, DJB  wrote:

We
>have a customer that we are sending calls to off the FS and here is the issue: 
> 
>Call
>is initially setup fine and they send a first re-invite with media 0.0.0.0 to
>place the caller on hold. FS sends a 200 ok to this first re-invite fine 
> 
>They
>then send a second re-invite with their media IP to cut through media and the
>FS sends a 200 OK to this fine. At this point the call is fine 
> 
>30
>minutes later they send a third re-invite because according to them it is
>strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has
>the exact same media IP and UDP pot information as the second re-invite does.
>The problem is FS does not respond to this third re-invite AT ALL. It doesn’t
>send a 100 trying a 200 OK nothing so this causes the call to be dropped as the
>other end does not recieve a response from FS.  
>
>
>One more thing, we did not see the third re-invite in sofia siptrace, but we 
>do see it in ethereal, which is kind of odds.
>
>
>We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.
>
>
>Thank you very much.
>
>___
>>FreeSWITCH-users mailing list
>[email protected]
>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/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:[email protected]
GTALK/JABBER/PAYPAL:[email protected]
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[email protected]
iax:[email protected]/888
googletalk:[email protected]
pstn:+19193869900



  ___
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP Re-invite

2009-12-16 Thread Anthony Minessale 
that means the invite is not matching the call dialog
compare the via tags and call-id etc


On Wed, Dec 16, 2009 at 5:29 PM, DJB  wrote:

> We have a customer that we are sending calls to off the FS and here is the
> issue:
>
>
>
> Call is initially setup fine and they send a first re-invite with media
> 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first
> re-invite fine
>
>
>
> They then send a second re-invite with their media IP to cut through media
> and the FS sends a 200 OK to this fine. At this point the call is fine
>
>
>
> 30 minutes later they send a third re-invite because according to them it
> is strictly for the purpose of “keep alive” per RFC 4028. This third
> re-invite has the exact same media IP and UDP pot information as the second
> re-invite does. The problem is FS does not respond to this third re-invite
> AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the
> call to be dropped as the other end does not recieve a response from FS.
>
>
> One more thing, we did not see the third re-invite in sofia siptrace, but
> we do see it in ethereal, which is kind of odds.
>
>
> We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.
>
>
> Thank you very much.
>
>
> ___
> FreeSWITCH-users mailing list
> [email protected]
> 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/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:[email protected] 
GTALK/JABBER/PAYPAL:[email protected]
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[email protected] 
iax:[email protected]/888
googletalk:[email protected]
pstn:+19193869900
___
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP re-invite / bypass_media

2009-07-02 Thread Anthony Minessale 
no problem


On Thu, Jul 2, 2009 at 12:27 PM, Phillip Jones  wrote:

> Used:
>
> session.execute("set","bypass_media_after_bridge=true");
> in the confirm.js script and that works perfectly!
>
> Thank you for you help!
> On Thu, Jul 2, 2009 at 1:05 PM, Anthony Minessale <
> [email protected]> wrote:
>
>> try setting bypass_media_after_bridge=true on the session in your confirm
>> script
>>
>>
>>
>> On Thu, Jul 2, 2009 at 11:53 AM, Phillip Jones wrote:
>>
>>> Thanks for responding and for your help.
>>>
>>> The xml and confirm.js are attached below. Basically trying to
>>> bypass_media after the leg B presses 1 to accept the call. I tried,
>>> using bypass_media_after_bridge=true, but the re-invite appears to be done
>>> before the confirm.js, So the media is successfully rerouted, but BEFORE the
>>> leg b never gets hear a prompt or gets the opportunity to press 1.
>>>
>>> To get round this I am trying to manually bypass_media in the confirm.js
>>> script with apiExecute("uuid_media", "off " + session.uuid);. However only
>>> the B leg is reinvited (and media is routed correctly). I don't see the A
>>> leg reinvite, and then a BYE is issueed on both legs.
>>>
>>> 
>>> 
>>>
>>> 
>>> 
>>> 
>>> <
>>> 
>>>
>>> 
>>> 
>>> 
>>>
>>> This is the confirm.js:
>>>
>>> // confirm.js - FreeSwitch call confirmation script
>>> // (c) 2009 - St‚phane Alnet
>>> // License: GPL2 or above
>>> console_log("info", "Destination: "+ session.destination + "\n");
>>> if(!session.getVariable('leg_confirm'))
>>> {
>>> console_log("info", "No need to confirm, connect the call!\n");
>>> exit();
>>> }
>>> var confirmed = false;
>>> var confirmation_digit = "1";
>>> var try_count = 6;
>>> var prompt_file = "prompts/ToAcceptThisCallPress1.wav";
>>> function onInput( session, type, data, arg ) {
>>> if ( type == "dtmf" ) {
>>> console_log( "info", "Got digit " + data.digit + "\n" );
>>> if ( data.digit == confirmation_digit ) {
>>> confirmed = true;
>>> console_log( "info", "Confirming session..\n" );
>>> return(false);
>>> }
>>> }
>>> return(true);
>>> }
>>> if ( session.ready() )
>>> {
>>> session.answer();
>>> session.flushDigits();
>>> console_log("info", "Starting confirmation\n");
>>> var count = try_count;
>>> while( session.ready() && ! confirmed && count-- > 0 )
>>> {
>>> session.execute("sleep","200");
>>> session.streamFile( prompt_file, onInput );
>>> }
>>>
>>> if( ! confirmed )
>>> {
>>> console_log("info", "Not confirmed\n");
>>> session.hangup();
>>> }
>>> else
>>> {
>>>  *apiExecute("uuid_media", "off " + session.uuid);*
>>> console_log("info", "Confirmed\n");
>>> }
>>> }
>>> else
>>> {
>>> console_log("info", "Session is not ready.\n");
>>> }
>>>
>>>
>>>
>>>
>>> On Thu, Jul 2, 2009 at 12:19 PM, Anthony Minessale <
>>> [email protected]> wrote:
>>>
 I would need to know more details about what you are doing.

 you could set the variable bypass_media_after_bridge=true on the a leg
 before you call the b leg and use the group_confirm feature to get the
 caller
 to press the key.



 On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones wrote:

> Thanks for that.
>
> That seems to successfully re-invite and re-route the the B leg - but
> does not reinvite the A leg and then immediately issues a "bye" on both
> legs.
>
> Do I have to do something to reinvite that A leg?
>
>   On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale <
> [email protected]> wrote:
>
>> try
>> apiExecute("uuid_media", "off " + session.uuid);
>>
>>
>>
>>   On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones > > wrote:
>>
>>>   Hi there,
>>>
>>> I was wondering whether it is possible to have FreeSwitch go into
>>> bypass_media mode on demand?
>>>
>>> For instance, leg a bridges to leg b - leg b is invited to accept the
>>> call by pressing 1. I want to go to bypass_media (do a SIP reinvite to
>>> reroute the media) after the one is pressed.
>>>
>>> Currently I am issuing the following from my js script that prompts
>>> for the 1:
>>>
>>> session.apiExecute("uuid_media",session.uuid);
>>>
>>> Not working however.
>>>
>>> Any help to get me going would be appreciated.
>>>
>>> Thanks
>>>
>>> Phillip Jones.
>>>
>>> ___
>>> Freeswitch-users mailing list
>>> [email protected]
>>> 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:[email protected]
>> GTALK/JABBER/PAYPAL:[email protected]
>> IRC: irc.freenode.ne

Re: [Freeswitch-users] SIP re-invite / bypass_media

2009-07-02 Thread Phillip Jones 
Used:

session.execute("set","bypass_media_after_bridge=true");
in the confirm.js script and that works perfectly!

Thank you for you help!
On Thu, Jul 2, 2009 at 1:05 PM, Anthony Minessale <
[email protected]> wrote:

> try setting bypass_media_after_bridge=true on the session in your confirm
> script
>
>
>
> On Thu, Jul 2, 2009 at 11:53 AM, Phillip Jones wrote:
>
>> Thanks for responding and for your help.
>>
>> The xml and confirm.js are attached below. Basically trying to
>> bypass_media after the leg B presses 1 to accept the call. I tried,
>> using bypass_media_after_bridge=true, but the re-invite appears to be done
>> before the confirm.js, So the media is successfully rerouted, but BEFORE the
>> leg b never gets hear a prompt or gets the opportunity to press 1.
>>
>> To get round this I am trying to manually bypass_media in the confirm.js
>> script with apiExecute("uuid_media", "off " + session.uuid);. However only
>> the B leg is reinvited (and media is routed correctly). I don't see the A
>> leg reinvite, and then a BYE is issueed on both legs.
>>
>> 
>> 
>>
>> 
>> 
>> 
>> <
>> 
>>
>> 
>> 
>> 
>>
>> This is the confirm.js:
>>
>> // confirm.js - FreeSwitch call confirmation script
>> // (c) 2009 - St‚phane Alnet
>> // License: GPL2 or above
>> console_log("info", "Destination: "+ session.destination + "\n");
>> if(!session.getVariable('leg_confirm'))
>> {
>> console_log("info", "No need to confirm, connect the call!\n");
>> exit();
>> }
>> var confirmed = false;
>> var confirmation_digit = "1";
>> var try_count = 6;
>> var prompt_file = "prompts/ToAcceptThisCallPress1.wav";
>> function onInput( session, type, data, arg ) {
>> if ( type == "dtmf" ) {
>> console_log( "info", "Got digit " + data.digit + "\n" );
>> if ( data.digit == confirmation_digit ) {
>> confirmed = true;
>> console_log( "info", "Confirming session..\n" );
>> return(false);
>> }
>> }
>> return(true);
>> }
>> if ( session.ready() )
>> {
>> session.answer();
>> session.flushDigits();
>> console_log("info", "Starting confirmation\n");
>> var count = try_count;
>> while( session.ready() && ! confirmed && count-- > 0 )
>> {
>> session.execute("sleep","200");
>> session.streamFile( prompt_file, onInput );
>> }
>>
>> if( ! confirmed )
>> {
>> console_log("info", "Not confirmed\n");
>> session.hangup();
>> }
>> else
>> {
>>  *apiExecute("uuid_media", "off " + session.uuid);*
>> console_log("info", "Confirmed\n");
>> }
>> }
>> else
>> {
>> console_log("info", "Session is not ready.\n");
>> }
>>
>>
>>
>>
>> On Thu, Jul 2, 2009 at 12:19 PM, Anthony Minessale <
>> [email protected]> wrote:
>>
>>> I would need to know more details about what you are doing.
>>>
>>> you could set the variable bypass_media_after_bridge=true on the a leg
>>> before you call the b leg and use the group_confirm feature to get the
>>> caller
>>> to press the key.
>>>
>>>
>>>
>>> On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones wrote:
>>>
 Thanks for that.

 That seems to successfully re-invite and re-route the the B leg - but
 does not reinvite the A leg and then immediately issues a "bye" on both
 legs.

 Do I have to do something to reinvite that A leg?

   On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale <
 [email protected]> wrote:

> try
> apiExecute("uuid_media", "off " + session.uuid);
>
>
>
>   On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones 
> wrote:
>
>>   Hi there,
>>
>> I was wondering whether it is possible to have FreeSwitch go into
>> bypass_media mode on demand?
>>
>> For instance, leg a bridges to leg b - leg b is invited to accept the
>> call by pressing 1. I want to go to bypass_media (do a SIP reinvite to
>> reroute the media) after the one is pressed.
>>
>> Currently I am issuing the following from my js script that prompts
>> for the 1:
>>
>> session.apiExecute("uuid_media",session.uuid);
>>
>> Not working however.
>>
>> Any help to get me going would be appreciated.
>>
>> Thanks
>>
>> Phillip Jones.
>>
>> ___
>> Freeswitch-users mailing list
>> [email protected]
>> 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:[email protected]
> GTALK/JABBER/PAYPAL:[email protected]
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:[email protected]
> iax:[email protected]/888
> googletalk:[email protected]
> pstn:213-799-1400
>
> _

Re: [Freeswitch-users] SIP re-invite / bypass_media

2009-07-02 Thread Anthony Minessale 
try setting bypass_media_after_bridge=true on the session in your confirm
script


On Thu, Jul 2, 2009 at 11:53 AM, Phillip Jones  wrote:

> Thanks for responding and for your help.
>
> The xml and confirm.js are attached below. Basically trying to bypass_media
> after the leg B presses 1 to accept the call. I tried,
> using bypass_media_after_bridge=true, but the re-invite appears to be done
> before the confirm.js, So the media is successfully rerouted, but BEFORE the
> leg b never gets hear a prompt or gets the opportunity to press 1.
>
> To get round this I am trying to manually bypass_media in the confirm.js
> script with apiExecute("uuid_media", "off " + session.uuid);. However only
> the B leg is reinvited (and media is routed correctly). I don't see the A
> leg reinvite, and then a BYE is issueed on both legs.
>
> 
> 
>
> 
> 
> 
> <
> 
>
> 
> 
> 
>
> This is the confirm.js:
>
> // confirm.js - FreeSwitch call confirmation script
> // (c) 2009 - St‚phane Alnet
> // License: GPL2 or above
> console_log("info", "Destination: "+ session.destination + "\n");
> if(!session.getVariable('leg_confirm'))
> {
> console_log("info", "No need to confirm, connect the call!\n");
> exit();
> }
> var confirmed = false;
> var confirmation_digit = "1";
> var try_count = 6;
> var prompt_file = "prompts/ToAcceptThisCallPress1.wav";
> function onInput( session, type, data, arg ) {
> if ( type == "dtmf" ) {
> console_log( "info", "Got digit " + data.digit + "\n" );
> if ( data.digit == confirmation_digit ) {
> confirmed = true;
> console_log( "info", "Confirming session..\n" );
> return(false);
> }
> }
> return(true);
> }
> if ( session.ready() )
> {
> session.answer();
> session.flushDigits();
> console_log("info", "Starting confirmation\n");
> var count = try_count;
> while( session.ready() && ! confirmed && count-- > 0 )
> {
> session.execute("sleep","200");
> session.streamFile( prompt_file, onInput );
> }
>
> if( ! confirmed )
> {
> console_log("info", "Not confirmed\n");
> session.hangup();
> }
> else
> {
> *apiExecute("uuid_media", "off " + session.uuid);*
> console_log("info", "Confirmed\n");
> }
> }
> else
> {
> console_log("info", "Session is not ready.\n");
> }
>
>
>
>
> On Thu, Jul 2, 2009 at 12:19 PM, Anthony Minessale <
> [email protected]> wrote:
>
>> I would need to know more details about what you are doing.
>>
>> you could set the variable bypass_media_after_bridge=true on the a leg
>> before you call the b leg and use the group_confirm feature to get the
>> caller
>> to press the key.
>>
>>
>>
>> On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones wrote:
>>
>>> Thanks for that.
>>>
>>> That seems to successfully re-invite and re-route the the B leg - but
>>> does not reinvite the A leg and then immediately issues a "bye" on both
>>> legs.
>>>
>>> Do I have to do something to reinvite that A leg?
>>>
>>>   On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale <
>>> [email protected]> wrote:
>>>
 try
 apiExecute("uuid_media", "off " + session.uuid);



   On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones 
 wrote:

>   Hi there,
>
> I was wondering whether it is possible to have FreeSwitch go into
> bypass_media mode on demand?
>
> For instance, leg a bridges to leg b - leg b is invited to accept the
> call by pressing 1. I want to go to bypass_media (do a SIP reinvite to
> reroute the media) after the one is pressed.
>
> Currently I am issuing the following from my js script that prompts for
> the 1:
>
> session.apiExecute("uuid_media",session.uuid);
>
> Not working however.
>
> Any help to get me going would be appreciated.
>
> Thanks
>
> Phillip Jones.
>
> ___
> Freeswitch-users mailing list
> [email protected]
> 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:[email protected] 
 GTALK/JABBER/PAYPAL:[email protected]
 IRC: irc.freenode.net #freeswitch

 FreeSWITCH Developer Conference
 sip:[email protected] 
 iax:[email protected]/888
 googletalk:[email protected]
 pstn:213-799-1400

 ___
 Freeswitch-users mailing list
 [email protected]
 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

Re: [Freeswitch-users] SIP re-invite / bypass_media

2009-07-02 Thread Phillip Jones 
 Thanks for responding and for your help.

The xml and confirm.js are attached below. Basically trying to bypass_media
after the leg B presses 1 to accept the call. I tried,
using bypass_media_after_bridge=true, but the re-invite appears to be done
before the confirm.js, So the media is successfully rerouted, but BEFORE the
leg b never gets hear a prompt or gets the opportunity to press 1.

To get round this I am trying to manually bypass_media in the confirm.js
script with apiExecute("uuid_media", "off " + session.uuid);. However only
the B leg is reinvited (and media is routed correctly). I don't see the A
leg reinvite, and then a BYE is issueed on both legs.







<






This is the confirm.js:

// confirm.js - FreeSwitch call confirmation script
// (c) 2009 - St‚phane Alnet
// License: GPL2 or above
console_log("info", "Destination: "+ session.destination + "\n");
if(!session.getVariable('leg_confirm'))
{
console_log("info", "No need to confirm, connect the call!\n");
exit();
}
var confirmed = false;
var confirmation_digit = "1";
var try_count = 6;
var prompt_file = "prompts/ToAcceptThisCallPress1.wav";
function onInput( session, type, data, arg ) {
if ( type == "dtmf" ) {
console_log( "info", "Got digit " + data.digit + "\n" );
if ( data.digit == confirmation_digit ) {
confirmed = true;
console_log( "info", "Confirming session..\n" );
return(false);
}
}
return(true);
}
if ( session.ready() )
{
session.answer();
session.flushDigits();
console_log("info", "Starting confirmation\n");
var count = try_count;
while( session.ready() && ! confirmed && count-- > 0 )
{
session.execute("sleep","200");
session.streamFile( prompt_file, onInput );
}

if( ! confirmed )
{
console_log("info", "Not confirmed\n");
session.hangup();
}
else
{
*apiExecute("uuid_media", "off " + session.uuid);*
console_log("info", "Confirmed\n");
}
}
else
{
console_log("info", "Session is not ready.\n");
}




On Thu, Jul 2, 2009 at 12:19 PM, Anthony Minessale <
[email protected]> wrote:

> I would need to know more details about what you are doing.
>
> you could set the variable bypass_media_after_bridge=true on the a leg
> before you call the b leg and use the group_confirm feature to get the
> caller
> to press the key.
>
>
>
> On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones wrote:
>
>> Thanks for that.
>>
>> That seems to successfully re-invite and re-route the the B leg - but does
>> not reinvite the A leg and then immediately issues a "bye" on both legs.
>>
>> Do I have to do something to reinvite that A leg?
>>
>>   On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale <
>> [email protected]> wrote:
>>
>>> try
>>> apiExecute("uuid_media", "off " + session.uuid);
>>>
>>>
>>>
>>>   On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones wrote:
>>>
   Hi there,

 I was wondering whether it is possible to have FreeSwitch go into
 bypass_media mode on demand?

 For instance, leg a bridges to leg b - leg b is invited to accept the
 call by pressing 1. I want to go to bypass_media (do a SIP reinvite to
 reroute the media) after the one is pressed.

 Currently I am issuing the following from my js script that prompts for
 the 1:

 session.apiExecute("uuid_media",session.uuid);

 Not working however.

 Any help to get me going would be appreciated.

 Thanks

 Phillip Jones.

 ___
 Freeswitch-users mailing list
 [email protected]
 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:[email protected] 
>>> GTALK/JABBER/PAYPAL:[email protected]
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:[email protected] 
>>> iax:[email protected]/888
>>> googletalk:[email protected]
>>> pstn:213-799-1400
>>>
>>> ___
>>> Freeswitch-users mailing list
>>> [email protected]
>>> 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
>> [email protected]
>> 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:[email protected] 
> GTALK/JABBER/PAYPAL:anthony.miness...

Re: [Freeswitch-users] SIP re-invite / bypass_media

2009-07-02 Thread Anthony Minessale 
I would need to know more details about what you are doing.

you could set the variable bypass_media_after_bridge=true on the a leg
before you call the b leg and use the group_confirm feature to get the
caller
to press the key.


On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones  wrote:

> Thanks for that.
>
> That seems to successfully re-invite and re-route the the B leg - but does
> not reinvite the A leg and then immediately issues a "bye" on both legs.
>
> Do I have to do something to reinvite that A leg?
>
> On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale <
> [email protected]> wrote:
>
>> try
>> apiExecute("uuid_media", "off " + session.uuid);
>>
>>
>>
>>   On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones wrote:
>>
>>>   Hi there,
>>>
>>> I was wondering whether it is possible to have FreeSwitch go into
>>> bypass_media mode on demand?
>>>
>>> For instance, leg a bridges to leg b - leg b is invited to accept the
>>> call by pressing 1. I want to go to bypass_media (do a SIP reinvite to
>>> reroute the media) after the one is pressed.
>>>
>>> Currently I am issuing the following from my js script that prompts for
>>> the 1:
>>>
>>> session.apiExecute("uuid_media",session.uuid);
>>>
>>> Not working however.
>>>
>>> Any help to get me going would be appreciated.
>>>
>>> Thanks
>>>
>>> Phillip Jones.
>>>
>>> ___
>>> Freeswitch-users mailing list
>>> [email protected]
>>> 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:[email protected] 
>> GTALK/JABBER/PAYPAL:[email protected]
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:[email protected] 
>> iax:[email protected]/888
>> googletalk:[email protected]
>> pstn:213-799-1400
>>
>> ___
>> Freeswitch-users mailing list
>> [email protected]
>> 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
> [email protected]
> 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:[email protected] 
GTALK/JABBER/PAYPAL:[email protected]
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[email protected] 
iax:[email protected]/888
googletalk:[email protected]
pstn:213-799-1400
___
Freeswitch-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP re-invite / bypass_media

2009-07-02 Thread Phillip Jones 
Thanks for that.

That seems to successfully re-invite and re-route the the B leg - but does
not reinvite the A leg and then immediately issues a "bye" on both legs.

Do I have to do something to reinvite that A leg?

On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale <
[email protected]> wrote:

> try
> apiExecute("uuid_media", "off " + session.uuid);
>
>
>
>   On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones wrote:
>
>>   Hi there,
>>
>> I was wondering whether it is possible to have FreeSwitch go into
>> bypass_media mode on demand?
>>
>> For instance, leg a bridges to leg b - leg b is invited to accept the call
>> by pressing 1. I want to go to bypass_media (do a SIP reinvite to reroute
>> the media) after the one is pressed.
>>
>> Currently I am issuing the following from my js script that prompts for
>> the 1:
>>
>> session.apiExecute("uuid_media",session.uuid);
>>
>> Not working however.
>>
>> Any help to get me going would be appreciated.
>>
>> Thanks
>>
>> Phillip Jones.
>>
>> ___
>> Freeswitch-users mailing list
>> [email protected]
>> 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:[email protected] 
> GTALK/JABBER/PAYPAL:[email protected]
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:[email protected] 
> iax:[email protected]/888
> googletalk:[email protected]
> pstn:213-799-1400
>
> ___
> Freeswitch-users mailing list
> [email protected]
> 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
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] SIP re-invite / bypass_media

2009-07-01 Thread Anthony Minessale 
try
apiExecute("uuid_media", "off " + session.uuid);



On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones  wrote:

> Hi there,
>
> I was wondering whether it is possible to have FreeSwitch go into
> bypass_media mode on demand?
>
> For instance, leg a bridges to leg b - leg b is invited to accept the call
> by pressing 1. I want to go to bypass_media (do a SIP reinvite to reroute
> the media) after the one is pressed.
>
> Currently I am issuing the following from my js script that prompts for the
> 1:
>
> session.apiExecute("uuid_media",session.uuid);
>
> Not working however.
>
> Any help to get me going would be appreciated.
>
> Thanks
>
> Phillip Jones.
>
> ___
> Freeswitch-users mailing list
> [email protected]
> 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:[email protected] 
GTALK/JABBER/PAYPAL:[email protected]
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[email protected] 
iax:[email protected]/888
googletalk:[email protected]
pstn:213-799-1400
___
Freeswitch-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org