Discussion:
[Freeswitch-users] Freeswitch hangup cause code issue
Ankit Gandhi
2009-02-06 08:57:07 UTC
Permalink
Hello,
I want my caller to detect hangup cause as 34 so that he can try next
provider according to lcr.
Here is my setup.
Caller -> switch (fs) -> Terminator.
Now when terminator sends "503 Service Unavailable", I want to override this
cause, so that the caller gets hangup case 34. (According to the terminator
he is sending hangup cause 34 from his side, but in freeswitch we are
getting hangup cause 41 for that call and the same hangup cause on caller
side).
When I tried asterisk as caller, I get hangup cause 34 in that case. But
when I tried freeswitch as caller, then we are getting hangup cause 41, the
same as we are getting in switch (fs).
From the switch, I tried one of this condition through javascript to
override the hangup cause before sending to caller:
-> session.execute("respond","503");
-> session.execute("hangup","NORMAL_CIRCUIT_CONGESTION");
-> session.execute("hangup","34");
-> session.hangup(34);
In all the above cases, asterisk properly detects the hangup cause 34, but
freeswitch does not detect that. It detects the same hangup cause 41 for the
call. Other callers also get the same hangup cause 41 for such calls.
How can I override this cause, so that the caller gets hangup cause 34 in
such cases?

Here is the sip trace, on the caller side returned through switch (fs).
ss = switch
cc = caller
--------------------------------------------------------------------
U ss.ss.ss.ss:5060 -> cc.cc.cc.cc:5080
SIP/2.0 503 Service Unavailable.
Via: SIP/2.0/UDP
cc.cc.cc.cc:5080;rport=5080;branch=z9hG4bK8HvDU4Z88KU5m;received=122.169.29.122.
From: "654321" <sip:654321 at cc.cc.cc.cc>;tag=8p7aF18aN0mSj.
To: <sip:123456 at ss.ss.ss.ss>;tag=Nj8ypjDyvKUXe.
Call-ID: fd1ab774-6ec5-122c-9fac-001cc086141d.
CSeq: 110835048 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION".
Content-Length: 0.
-----------------------------------------------------------------------
--
View this message in context: http://www.nabble.com/Freeswitch-hangup-cause-code-issue-tp21868781p21868781.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
Anthony Minessale
2009-02-06 15:09:08 UTC
Permalink
you would need to provide the console output of FreeSWITCH of the entire
call.
with TPORT_LOG=1 env var set and console loglevel debug (press f8)
Post by Ankit Gandhi
Hello,
I want my caller to detect hangup cause as 34 so that he can try next
provider according to lcr.
Here is my setup.
Caller -> switch (fs) -> Terminator.
Now when terminator sends "503 Service Unavailable", I want to override this
cause, so that the caller gets hangup case 34. (According to the terminator
he is sending hangup cause 34 from his side, but in freeswitch we are
getting hangup cause 41 for that call and the same hangup cause on caller
side).
When I tried asterisk as caller, I get hangup cause 34 in that case. But
when I tried freeswitch as caller, then we are getting hangup cause 41, the
same as we are getting in switch (fs).
From the switch, I tried one of this condition through javascript to
-> session.execute("respond","503");
-> session.execute("hangup","NORMAL_CIRCUIT_CONGESTION");
-> session.execute("hangup","34");
-> session.hangup(34);
In all the above cases, asterisk properly detects the hangup cause 34, but
freeswitch does not detect that. It detects the same hangup cause 41 for the
call. Other callers also get the same hangup cause 41 for such calls.
How can I override this cause, so that the caller gets hangup cause 34 in
such cases?
Here is the sip trace, on the caller side returned through switch (fs).
ss = switch
cc = caller
--------------------------------------------------------------------
U ss.ss.ss.ss:5060 -> cc.cc.cc.cc:5080
SIP/2.0 503 Service Unavailable.
Via: SIP/2.0/UDP
cc.cc.cc.cc:5080
;rport=5080;branch=z9hG4bK8HvDU4Z88KU5m;received=122.169.29.122.
From: "654321" <sip:654321 at cc.cc.cc.cc <sip%3A654321 at cc.cc.cc.cc>
;tag=8p7aF18aN0mSj.
To: <sip:123456 at ss.ss.ss.ss>;tag=Nj8ypjDyvKUXe.
Call-ID: fd1ab774-6ec5-122c-9fac-001cc086141d.
CSeq: 110835048 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION".
Content-Length: 0.
-----------------------------------------------------------------------
--
http://www.nabble.com/Freeswitch-hangup-cause-code-issue-tp21868781p21868781.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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Ankit Gandhi
2009-02-06 08:57:07 UTC
Permalink
Hello,
I want my caller to detect hangup cause as 34 so that he can try next
provider according to lcr.
Here is my setup.
Caller -> switch (fs) -> Terminator.
Now when terminator sends "503 Service Unavailable", I want to override this
cause, so that the caller gets hangup case 34. (According to the terminator
he is sending hangup cause 34 from his side, but in freeswitch we are
getting hangup cause 41 for that call and the same hangup cause on caller
side).
When I tried asterisk as caller, I get hangup cause 34 in that case. But
when I tried freeswitch as caller, then we are getting hangup cause 41, the
same as we are getting in switch (fs).
From the switch, I tried one of this condition through javascript to
override the hangup cause before sending to caller:
-> session.execute("respond","503");
-> session.execute("hangup","NORMAL_CIRCUIT_CONGESTION");
-> session.execute("hangup","34");
-> session.hangup(34);
In all the above cases, asterisk properly detects the hangup cause 34, but
freeswitch does not detect that. It detects the same hangup cause 41 for the
call. Other callers also get the same hangup cause 41 for such calls.
How can I override this cause, so that the caller gets hangup cause 34 in
such cases?

Here is the sip trace, on the caller side returned through switch (fs).
ss = switch
cc = caller
--------------------------------------------------------------------
U ss.ss.ss.ss:5060 -> cc.cc.cc.cc:5080
SIP/2.0 503 Service Unavailable.
Via: SIP/2.0/UDP
cc.cc.cc.cc:5080;rport=5080;branch=z9hG4bK8HvDU4Z88KU5m;received=122.169.29.122.
From: "654321" <sip:654321 at cc.cc.cc.cc>;tag=8p7aF18aN0mSj.
To: <sip:123456 at ss.ss.ss.ss>;tag=Nj8ypjDyvKUXe.
Call-ID: fd1ab774-6ec5-122c-9fac-001cc086141d.
CSeq: 110835048 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION".
Content-Length: 0.
-----------------------------------------------------------------------
--
View this message in context: http://www.nabble.com/Freeswitch-hangup-cause-code-issue-tp21868781p21868781.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
Anthony Minessale
2009-02-06 15:09:08 UTC
Permalink
you would need to provide the console output of FreeSWITCH of the entire
call.
with TPORT_LOG=1 env var set and console loglevel debug (press f8)
Post by Ankit Gandhi
Hello,
I want my caller to detect hangup cause as 34 so that he can try next
provider according to lcr.
Here is my setup.
Caller -> switch (fs) -> Terminator.
Now when terminator sends "503 Service Unavailable", I want to override this
cause, so that the caller gets hangup case 34. (According to the terminator
he is sending hangup cause 34 from his side, but in freeswitch we are
getting hangup cause 41 for that call and the same hangup cause on caller
side).
When I tried asterisk as caller, I get hangup cause 34 in that case. But
when I tried freeswitch as caller, then we are getting hangup cause 41, the
same as we are getting in switch (fs).
From the switch, I tried one of this condition through javascript to
-> session.execute("respond","503");
-> session.execute("hangup","NORMAL_CIRCUIT_CONGESTION");
-> session.execute("hangup","34");
-> session.hangup(34);
In all the above cases, asterisk properly detects the hangup cause 34, but
freeswitch does not detect that. It detects the same hangup cause 41 for the
call. Other callers also get the same hangup cause 41 for such calls.
How can I override this cause, so that the caller gets hangup cause 34 in
such cases?
Here is the sip trace, on the caller side returned through switch (fs).
ss = switch
cc = caller
--------------------------------------------------------------------
U ss.ss.ss.ss:5060 -> cc.cc.cc.cc:5080
SIP/2.0 503 Service Unavailable.
Via: SIP/2.0/UDP
cc.cc.cc.cc:5080
;rport=5080;branch=z9hG4bK8HvDU4Z88KU5m;received=122.169.29.122.
From: "654321" <sip:654321 at cc.cc.cc.cc <sip%3A654321 at cc.cc.cc.cc>
;tag=8p7aF18aN0mSj.
To: <sip:123456 at ss.ss.ss.ss>;tag=Nj8ypjDyvKUXe.
Call-ID: fd1ab774-6ec5-122c-9fac-001cc086141d.
CSeq: 110835048 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION".
Content-Length: 0.
-----------------------------------------------------------------------
--
http://www.nabble.com/Freeswitch-hangup-cause-code-issue-tp21868781p21868781.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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