[Freeswitch-users] Issues with SIP+TCP?
Kristian Kielhofner
kristian.kielhofner at gmail.com
Fri Oct 23 11:14:07 PDT 2009
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)$">
<action application="set" data="call_timeout=60"/>
<action application="set" data="effective_caller_id_name=Voalte Test"/>
<action application="set"
data="effective_caller_id_number=19412848354"/>
<action application="bridge" data="sofia/avaya/7887 at 10.70.0.62"/>
</condition>
</extension>
dialplan (TCP):
<extension name="smhpbx">
<condition field="destination_number" expression="^(7887)$">
<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 at 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
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