Resend: Issues with SIP + TCP?
I originally sent this last Friday but I've been unable to confirm it ever made it to the list. Hello everyone, I'm having some issues with SIP and TCP. I've used it before with success but I'm seeing some strange behavior... Level 7 debugs with siptrace on both profiles. UDP invite from softphone comes in on port 5062, it's supposed to bridge to 10.70.0.62. When configured to use UDP FS sends an INVITE (nothing currently answers) while TCP doesn't send anything (confirmed with siptrace and packet sniffer). I confirmed this behavior with a gateway configured for TCP and appending ;transport=tcp to a bridge line. This is trunk rev 15211 on an Intel Mac running Snow Leopard. I've also confirmed this behavior on an Intel Linux machine running Ubuntu (not sure of version ATM). TCP: http://pastebin.freeswitch.org/10825 UDP: http://pastebin.freeswitch.org/10826 dialplan (UDP): <extension name="smhpbx"> <condition field="destination_number" expression="^(7887)$"> <action application="set" data="call_timeout=60"/> <action application="set" data="effective_caller_id_name=Voalte Test"/> <action application="set" data="effective_caller_id_number=19412848354"/> <action application="bridge" data="sofia/avaya/7887@10.70.0.62"/> </condition> </extension> dialplan (TCP): <extension name="smhpbx"> <condition field="destination_number" expression="^(7887)$"> <action application="set" data="call_timeout=60"/> <action application="set" data="effective_caller_id_name=Voalte Test"/> <action application="set" data="effective_caller_id_number=19412848354"/> <action application="bridge" data="sofia/avaya/7887@10.70.0.62;transport=tcp"/> </condition> </extension> Any thoughts? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
i cant seem to reproduce it. originate sofia/internal/1235@conference.freeswitch.org;transport=tcp 9998 I get a working call and trace. Could you possibly have a dns error? I know it's an ip but it may still fail if it has no dns. try sofia loglevel all 9 and look for other errors. On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner < kristian.kielhofner@gmail.com> wrote:
I originally sent this last Friday but I've been unable to confirm it ever made it to the list.
Hello everyone,
I'm having some issues with SIP and TCP. I've used it before with success but I'm seeing some strange behavior...
Level 7 debugs with siptrace on both profiles. UDP invite from softphone comes in on port 5062, it's supposed to bridge to 10.70.0.62. When configured to use UDP FS sends an INVITE (nothing currently answers) while TCP doesn't send anything (confirmed with siptrace and packet sniffer). I confirmed this behavior with a gateway configured for TCP and appending ;transport=tcp to a bridge line.
This is trunk rev 15211 on an Intel Mac running Snow Leopard. I've also confirmed this behavior on an Intel Linux machine running Ubuntu (not sure of version ATM).
TCP:
http://pastebin.freeswitch.org/10825
UDP:
http://pastebin.freeswitch.org/10826
dialplan (UDP):
<extension name="smhpbx"> <condition field="destination_number" expression="^(7887)$"> <action application="set" data="call_timeout=60"/> <action application="set" data="effective_caller_id_name=Voalte Test"/> <action application="set" data="effective_caller_id_number=19412848354"/> <action application="bridge" data="sofia/avaya/7887@10.70.0.62"/> </condition> </extension>
dialplan (TCP):
<extension name="smhpbx"> <condition field="destination_number" expression="^(7887)$"> <action application="set" data="call_timeout=60"/> <action application="set" data="effective_caller_id_name=Voalte Test"/> <action application="set" data="effective_caller_id_number=19412848354"/> <action application="bridge" data="sofia/avaya/7887@10.70.0.62;transport=tcp"/> </condition> </extension>
Any thoughts?
Thanks!
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Tony, It seemed strange to me too (I'm using TCP in other places). I'll take another look at this with your suggestions for debugging. Thanks! On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
i cant seem to reproduce it.
originate sofia/internal/1235@conference.freeswitch.org;transport=tcp 9998
I get a working call and trace.
Could you possibly have a dns error? I know it's an ip but it may still fail if it has no dns.
try
sofia loglevel all 9
and look for other errors.
On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner <kristian.kielhofner@gmail.com> wrote:
I originally sent this last Friday but I've been unable to confirm it ever made it to the list.
Hello everyone,
I'm having some issues with SIP and TCP. I've used it before with success but I'm seeing some strange behavior...
Level 7 debugs with siptrace on both profiles. UDP invite from softphone comes in on port 5062, it's supposed to bridge to 10.70.0.62. When configured to use UDP FS sends an INVITE (nothing currently answers) while TCP doesn't send anything (confirmed with siptrace and packet sniffer). I confirmed this behavior with a gateway configured for TCP and appending ;transport=tcp to a bridge line.
This is trunk rev 15211 on an Intel Mac running Snow Leopard. I've also confirmed this behavior on an Intel Linux machine running Ubuntu (not sure of version ATM).
TCP:
http://pastebin.freeswitch.org/10825
UDP:
http://pastebin.freeswitch.org/10826
dialplan (UDP):
<extension name="smhpbx"> <condition field="destination_number" expression="^(7887)$"> <action application="set" data="call_timeout=60"/> <action application="set" data="effective_caller_id_name=Voalte Test"/> <action application="set" data="effective_caller_id_number=19412848354"/> <action application="bridge" data="sofia/avaya/7887@10.70.0.62"/> </condition> </extension>
dialplan (TCP):
<extension name="smhpbx"> <condition field="destination_number" expression="^(7887)$"> <action application="set" data="call_timeout=60"/> <action application="set" data="effective_caller_id_name=Voalte Test"/> <action application="set" data="effective_caller_id_number=19412848354"/> <action application="bridge" data="sofia/avaya/7887@10.70.0.62;transport=tcp"/> </condition> </extension>
Any thoughts?
Thanks!
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
I'm also having problems with this. When running FS compiled about 10 days ago it works fine (don't remember exact revision), but when using latest SVN it doesn't work anymore. I seems like it's trying to use UDP when it should use TCP. My setup is this: Avaya Communication Manager PBX -> Talks TLS to Avaya SIP SES Server -> Talks TCP to FreeSwitch. The replies from FS seems to be sent using UDP instead of TCP, and when I keep the config and revert to the 10 day old version it starts working again, so there is definately something wrong. I'll try to do some more testing, and get back with some SIP-traces as well. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Kristian Kielhofner Skickat: den 26 oktober 2009 16:47 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP? Tony, It seemed strange to me too (I'm using TCP in other places). I'll take another look at this with your suggestions for debugging. Thanks! On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
i cant seem to reproduce it.
originate sofia/internal/1235@conference.freeswitch.org;transport=tcp 9998
I get a working call and trace.
Could you possibly have a dns error? I know it's an ip but it may still fail if it has no dns.
try
sofia loglevel all 9
and look for other errors.
On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner <kristian.kielhofner@gmail.com> wrote:
I originally sent this last Friday but I've been unable to confirm it ever made it to the list.
Hello everyone,
I'm having some issues with SIP and TCP. I've used it before with success but I'm seeing some strange behavior...
Level 7 debugs with siptrace on both profiles. UDP invite from softphone comes in on port 5062, it's supposed to bridge to 10.70.0.62. When configured to use UDP FS sends an INVITE (nothing currently answers) while TCP doesn't send anything (confirmed with siptrace and packet sniffer). I confirmed this behavior with a gateway configured for TCP and appending ;transport=tcp to a bridge line.
This is trunk rev 15211 on an Intel Mac running Snow Leopard. I've also confirmed this behavior on an Intel Linux machine running Ubuntu (not sure of version ATM).
TCP:
http://pastebin.freeswitch.org/10825
UDP:
http://pastebin.freeswitch.org/10826
dialplan (UDP):
<extension name="smhpbx"> <condition field="destination_number" expression="^(7887)$"> <action application="set" data="call_timeout=60"/> <action application="set" data="effective_caller_id_name=Voalte Test"/> <action application="set" data="effective_caller_id_number=19412848354"/> <action application="bridge" data="sofia/avaya/7887@10.70.0.62"/> </condition> </extension>
dialplan (TCP):
<extension name="smhpbx"> <condition field="destination_number" expression="^(7887)$"> <action application="set" data="call_timeout=60"/> <action application="set" data="effective_caller_id_name=Voalte Test"/> <action application="set" data="effective_caller_id_number=19412848354"/> <action application="bridge" data="sofia/avaya/7887@10.70.0.62;transport=tcp"/> </condition> </extension>
Any thoughts?
Thanks!
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ae5c6c832935743011996!
Hmm... I remembered incorrectly about my setup :) The Avaya PBX talks TLS to the Avaya SES Server, and then UDP to FS, not TCP - sorry, my bad! However, something that has changed the last 10 days seems to affect my setup so it doesn't work anymore. I'll do some more SIP tracing, and get back when I know more about it. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Kristian Kielhofner Skickat: den 26 oktober 2009 16:47 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP? Tony, It seemed strange to me too (I'm using TCP in other places). I'll take another look at this with your suggestions for debugging. Thanks! On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
i cant seem to reproduce it.
originate sofia/internal/1235@conference.freeswitch.org;transport=tcp 9998
I get a working call and trace.
Could you possibly have a dns error? I know it's an ip but it may still fail if it has no dns.
try
sofia loglevel all 9
and look for other errors.
On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner <kristian.kielhofner@gmail.com> wrote:
I originally sent this last Friday but I've been unable to confirm it ever made it to the list.
Hello everyone,
I'm having some issues with SIP and TCP. I've used it before with success but I'm seeing some strange behavior...
Level 7 debugs with siptrace on both profiles. UDP invite from softphone comes in on port 5062, it's supposed to bridge to 10.70.0.62. When configured to use UDP FS sends an INVITE (nothing currently answers) while TCP doesn't send anything (confirmed with siptrace and packet sniffer). I confirmed this behavior with a gateway configured for TCP and appending ;transport=tcp to a bridge line.
This is trunk rev 15211 on an Intel Mac running Snow Leopard. I've also confirmed this behavior on an Intel Linux machine running Ubuntu (not sure of version ATM).
TCP:
http://pastebin.freeswitch.org/10825
UDP:
http://pastebin.freeswitch.org/10826
dialplan (UDP):
<extension name="smhpbx"> <condition field="destination_number" expression="^(7887)$"> <action application="set" data="call_timeout=60"/> <action application="set" data="effective_caller_id_name=Voalte Test"/> <action application="set" data="effective_caller_id_number=19412848354"/> <action application="bridge" data="sofia/avaya/7887@10.70.0.62"/> </condition> </extension>
dialplan (TCP):
<extension name="smhpbx"> <condition field="destination_number" expression="^(7887)$"> <action application="set" data="call_timeout=60"/> <action application="set" data="effective_caller_id_name=Voalte Test"/> <action application="set" data="effective_caller_id_number=19412848354"/> <action application="bridge" data="sofia/avaya/7887@10.70.0.62;transport=tcp"/> </condition> </extension>
Any thoughts?
Thanks!
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ae5c6c832935743011996!
Finding the exact rev that broke it would be helpful. /b On Oct 26, 2009, at 12:06 PM, Peter Olsson wrote:
Hmm... I remembered incorrectly about my setup :) The Avaya PBX talks TLS to the Avaya SES Server, and then UDP to FS, not TCP - sorry, my bad!
However, something that has changed the last 10 days seems to affect my setup so it doesn't work anymore. I'll do some more SIP tracing, and get back when I know more about it.
/Peter
Yes, I know :) However, now I think this is related to the new headers introduced, it's probably not a TCP issue. Everything seems to work just fine until the 200 OK is sent, the Avaya PBX doesn't seem to accept that reply anymore. The only differences I've found between a working revision, and a non-working is this; In the working 200-OK transaction method UPDATE is listed in the Allow-header, and there is a header called "X-Actually-Support: UPDATE". In the non-working one I don't have these, and instead I have these headers; X-FS-Display-Name: 9099 X-FS-Display-Number: 9099 X-FS-Support: update_display P-Asserted-Identity: "9099" <9099> So I guess this could be related to the other thread going on right now "Downloaded tar vs latest SVN - 200 OK has more headers", and not a TCP issue. It's probably the Avaya doing something wrong (not the first time), but still it seems these changes affect more systems than mine. I'm just using it as a lab setup for now, so if anyone want me to test something, I can do it immediately. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Brian West Skickat: den 26 oktober 2009 18:23 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP? Finding the exact rev that broke it would be helpful. /b On Oct 26, 2009, at 12:06 PM, Peter Olsson wrote:
Hmm... I remembered incorrectly about my setup :) The Avaya PBX talks TLS to the Avaya SES Server, and then UDP to FS, not TCP - sorry, my bad!
However, something that has changed the last 10 days seems to affect my setup so it doesn't work anymore. I'll do some more SIP tracing, and get back when I know more about it.
/Peter
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Bet your hardware just barfs on those like others have... I mean really I HATE SIP. This is stupid. /b On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote:
In the non-working one I don't have these, and instead I have these headers; X-FS-Display-Name: 9099 X-FS-Display-Number: 9099 X-FS-Support: update_display P-Asserted-Identity: "9099" <9099>
try r15230 add the profile param <param name="pass-callee-id" value="false"/> On Mon, Oct 26, 2009 at 12:46 PM, Brian West <brian@freeswitch.org> wrote:
Bet your hardware just barfs on those like others have... I mean really I HATE SIP. This is stupid.
/b
On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote:
In the non-working one I don't have these, and instead I have these headers; X-FS-Display-Name: 9099 X-FS-Display-Number: 9099 X-FS-Support: update_display P-Asserted-Identity: "9099" <9099>
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
I understand your frustration :) We deal with SIP integration with about 10 different PBX vendors today, And it's always something that doesn't work as it should. Right now I don't have anything more connected to FS though. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Brian West Skickat: den 26 oktober 2009 18:46 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP? Bet your hardware just barfs on those like others have... I mean really I HATE SIP. This is stupid. /b On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote:
In the non-working one I don't have these, and instead I have these headers; X-FS-Display-Name: 9099 X-FS-Display-Number: 9099 X-FS-Support: update_display P-Asserted-Identity: "9099" <9099>
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ae5e23832938073513968!
At some point we'll have to NO NO NO fix your broken crap. :P The reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with me... "NO!" /b On Oct 26, 2009, at 1:27 PM, Peter Olsson wrote:
I understand your frustration :) We deal with SIP integration with about 10 different PBX vendors today, And it's always something that doesn't work as it should. Right now I don't have anything more connected to FS though.
/Peter
On Mon, Oct 26, 2009 at 11:40 AM, Brian West <brian@freeswitch.org> wrote:
At some point we'll have to NO NO NO fix your broken crap. :P The reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with me... "NO!"
/b
I was wondering... does anyone make a "SIP certification" program kinda like a pen-tester except to find all the ways your SIP setup is broken? Just curious. -MC
You have SIPit, which was the SIP Backoff till Pillsbury got their panties in a wad. /b On Oct 26, 2009, at 3:03 PM, Michael Collins wrote:
I was wondering... does anyone make a "SIP certification" program kinda like a pen-tester except to find all the ways your SIP setup is broken? Just curious. -MC
Woof! On Mon, 26 Oct 2009 16:03:59 -0400, Michael Collins <msc@freeswitch.org> wrote:
I was wondering... does anyone make a "SIP certification" program kinda like a pen-tester except to find all the ways your SIP setup is broken? Just curious.
Here is a start: http://interop.sipxecs.org/ It's best for testing phone implementations, but it can handle other UA's as well. --Woof!
Thanks Brian, it works now. I'll try to learn to say NO next time :) /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Brian West Skickat: den 26 oktober 2009 19:40 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP? At some point we'll have to NO NO NO fix your broken crap. :P The reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with me... "NO!" /b On Oct 26, 2009, at 1:27 PM, Peter Olsson wrote:
I understand your frustration :) We deal with SIP integration with about 10 different PBX vendors today, And it's always something that doesn't work as it should. Right now I don't have anything more connected to FS though.
/Peter
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participants (6)
-
Andy Spitzer
-
Anthony Minessale
-
Brian West
-
Kristian Kielhofner
-
Michael Collins
-
Peter Olsson