486 Busy turns to 480 Temporarily Unavailable
Hello, I have a strange phenomen: When a target UA is busy, it returns "486 Busy" to Freeswitch. But Freeswitch then returns "480 Temporarily Unavailable" to the called party. Where does this come from and how can I change this behaviour? See (anonymized) SIP trace with ngrep: UA to Freeswitch: ======================== U 2012/01/04 13:59:44.928775 <fritzbox-IP>:5060 -> <freeswitch-IP>:5080 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP <freeswitch-IP>:5080;rport=5080;branch=z9hG4bKNZZDv0Syp4eyr. From: "026xxxxxxxx" <sip:026xxxxxxxx@<freeswitch-IP>>;tag=py094Kv7vr03a. To: <sip:069xxxxxxxx@<fritzbox-IP>;uniq=B05FE4881A55AEEB69361EFA327DB>;tag=E1C3374B97DAB2DE. Call-ID: d0d0d057-b176-122f-1f8d-001ec9b9da3c. CSeq: 22504928 INVITE. User-Agent: AVM FRITZ!Box 6360 Cable 85.05.07 (Sep 14 2011). Content-Length: 0. Freeswitch to Caller: ======================== U 2012/01/04 13:59:44.930387 <freeswitch-IP>:5060 -> <Sip-provider-IP>:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP <Sip-provider-IP>:5060;branch=z9hG4bK-4896-2830DFA. From: <sip:4926xxxxxxxx@sip.provider.de;user=phone>;tag=13517-HB-08a98588-2622da197. To: <sip:4969xxxxxxxx@<freeswitch-IP>;user=phone>;tag=XQtc5US24QgDa. Call-ID: 13517-SG-08a98587-0a352e121@sip.provider.de. CSeq: 134781549 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-68627e8 2011-11-21 13-52-28 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0. P-Asserted-Identity: "069xxxxxxxx" <sip:069xxxxxxxx@<freeswitch-IP>>. Best regards Peter
Hi, Thats really magical !! I wonder how would the console logs look like for this,. 1- Please do share the logs in pastebin. 2- take sip trace of the same call as well. 3- identify and paste the relevant dialplan code. As a new follower of FS this is really alarming for me to know how a 486 coming from carrier is relayed back to my sip client!! Regards, Sammy On Thu, Jan 5, 2012 at 3:59 PM, Peter P GMX <Prometheus001@gmx.net> wrote:
Hello,
I have a strange phenomen:
When a target UA is busy, it returns "486 Busy" to Freeswitch. But Freeswitch then returns "480 Temporarily Unavailable" to the called party. Where does this come from and how can I change this behaviour?
See (anonymized) SIP trace with ngrep:
UA to Freeswitch: ======================== U 2012/01/04 13:59:44.928775 <fritzbox-IP>:5060 -> <freeswitch-IP>:5080 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP <freeswitch-IP>:5080;rport=5080;branch=z9hG4bKNZZDv0Syp4eyr. From: "026xxxxxxxx" <sip:026xxxxxxxx@<freeswitch-IP>>;tag=py094Kv7vr03a. To: <sip:069xxxxxxxx@ <fritzbox-IP>;uniq=B05FE4881A55AEEB69361EFA327DB>;tag=E1C3374B97DAB2DE. Call-ID: d0d0d057-b176-122f-1f8d-001ec9b9da3c. CSeq: 22504928 INVITE. User-Agent: AVM FRITZ!Box 6360 Cable 85.05.07 (Sep 14 2011). Content-Length: 0.
Freeswitch to Caller: ======================== U 2012/01/04 13:59:44.930387 <freeswitch-IP>:5060 -> <Sip-provider-IP>:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP <Sip-provider-IP>:5060;branch=z9hG4bK-4896-2830DFA. From: <sip:4926xxxxxxxx@sip.provider.de ;user=phone>;tag=13517-HB-08a98588-2622da197. To: <sip:4969xxxxxxxx@<freeswitch-IP>;user=phone>;tag=XQtc5US24QgDa. Call-ID: 13517-SG-08a98587-0a352e121@sip.provider.de. CSeq: 134781549 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-68627e8 2011-11-21 13-52-28 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0. P-Asserted-Identity: "069xxxxxxxx" <sip:069xxxxxxxx@<freeswitch-IP>>.
Best regards Peter
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
I have the same problem. An external gateway is returning 486 to FS but FS returns 480 to the caller... I am testing following dialplan: <extension name="elastix"> <condition field="destination_number" expression="^77.+$"> <action application="bridge" data="sofia/gateway/elastix16/8888"/> </condition> </extension> 2012/1/5 Peter P GMX <Prometheus001@gmx.net>
Hello,
I have a strange phenomen:
When a target UA is busy, it returns "486 Busy" to Freeswitch. But Freeswitch then returns "480 Temporarily Unavailable" to the called party. Where does this come from and how can I change this behaviour?
See (anonymized) SIP trace with ngrep:
UA to Freeswitch: ======================== U 2012/01/04 13:59:44.928775 <fritzbox-IP>:5060 -> <freeswitch-IP>:5080 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP <freeswitch-IP>:5080;rport=5080;branch=z9hG4bKNZZDv0Syp4eyr. From: "026xxxxxxxx" <sip:026xxxxxxxx@<freeswitch-IP>>;tag=py094Kv7vr03a. To: <sip:069xxxxxxxx@ <fritzbox-IP>;uniq=B05FE4881A55AEEB69361EFA327DB>;tag=E1C3374B97DAB2DE. Call-ID: d0d0d057-b176-122f-1f8d-001ec9b9da3c. CSeq: 22504928 INVITE. User-Agent: AVM FRITZ!Box 6360 Cable 85.05.07 (Sep 14 2011). Content-Length: 0.
Freeswitch to Caller: ======================== U 2012/01/04 13:59:44.930387 <freeswitch-IP>:5060 -> <Sip-provider-IP>:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP <Sip-provider-IP>:5060;branch=z9hG4bK-4896-2830DFA. From: <sip:4926xxxxxxxx@sip.provider.de ;user=phone>;tag=13517-HB-08a98588-2622da197. To: <sip:4969xxxxxxxx@<freeswitch-IP>;user=phone>;tag=XQtc5US24QgDa. Call-ID: 13517-SG-08a98587-0a352e121@sip.provider.de. CSeq: 134781549 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-68627e8 2011-11-21 13-52-28 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0. P-Asserted-Identity: "069xxxxxxxx" <sip:069xxxxxxxx@<freeswitch-IP>>.
Best regards Peter
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
On 07/01/12 14:45, Juan Antonio Ibañez Santorum wrote:
I have the same problem. An external gateway is returning 486 to FS but FS returns 480 to the caller... I am testing following dialplan:
<extension name="elastix"> <condition field="destination_number" expression="^77.+$"> <action application="bridge" data="sofia/gateway/elastix16/8888"/> </condition> </extension>
Sounds like a bad interaction with some continue_on_foo configuration. Does it help if you do: <action application="hangup" data="${originate_disposition}"/> after the action-bridge? Regards,
No, same result, 480 instead 486 2012/1/7 Paul Cupis <paul@cupis.co.uk>
On 07/01/12 14:45, Juan Antonio Ibañez Santorum wrote:
I have the same problem. An external gateway is returning 486 to FS but FS returns 480 to the caller... I am testing following dialplan:
<extension name="elastix"> <condition field="destination_number" expression="^77.+$"> <action application="bridge" data="sofia/gateway/elastix16/8888"/> </condition> </extension>
Sounds like a bad interaction with some continue_on_foo configuration.
Does it help if you do:
<action application="hangup" data="${originate_disposition}"/>
after the action-bridge?
Regards,
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
On 07/01/12 16:32, Juan Antonio Ibañez Santorum wrote:
No, same result, 480 instead 486
Can you show us the full log of the call, please? http://pastebin.freeswitch.org/
Of course: Caller: 192.168.2.2 FS: 192.168.2.238 Gateway (Asterisk): 192.168.2.239 http://pastebin.freeswitch.org/18099 Regards 2012/1/7 Paul Cupis <paul@cupis.co.uk>
On 07/01/12 16:32, Juan Antonio Ibañez Santorum wrote:
No, same result, 480 instead 486
Can you show us the full log of the call, please?
http://pastebin.freeswitch.org/
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
Can you try adding: <action application="set" data="hangup_after_bridge=true"/> before your bridge action, please? Regards,
No news. FS still sends 480 instead 486 setting that variable. 2012/1/7 Paul Cupis <paul@cupis.co.uk>
Can you try adding:
<action application="set" data="hangup_after_bridge=true"/>
before your bridge action, please?
Regards,
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
Why are you executing hangup ${originate_disposition} after your bridge? You'll see that FreeSWITCH is passing originate_disposition=success, probably because the remote end is returning early media (which FreeSWITCH considers a successful bridge). Do one (or both) of these: 1) Add ignore_early_media=true to your bridge line. 2) Remove the hangup ${originate_disposition} after your bridge and let FreeSWITCH run of things to execute on its own. Depending on your specific goals there are better options but these actions should isolate your problem. 2012/1/7 Juan Antonio Ibañez Santorum <juanito1982@gmail.com>:
Of course:
Caller: 192.168.2.2 FS: 192.168.2.238 Gateway (Asterisk): 192.168.2.239
http://pastebin.freeswitch.org/18099
Regards
2012/1/7 Paul Cupis <paul@cupis.co.uk>
On 07/01/12 16:32, Juan Antonio Ibañez Santorum wrote:
No, same result, 480 instead 486
Can you show us the full log of the call, please?
http://pastebin.freeswitch.org/
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
-- Kristian Kielhofner
What about if not possible to ignore early media? 2012/1/8 Kristian Kielhofner <kris@kriskinc.com>
Why are you executing hangup ${originate_disposition} after your bridge?
You'll see that FreeSWITCH is passing originate_disposition=success, probably because the remote end is returning early media (which FreeSWITCH considers a successful bridge).
Do one (or both) of these: 1) Add ignore_early_media=true to your bridge line.
2) Remove the hangup ${originate_disposition} after your bridge and let FreeSWITCH run of things to execute on its own.
Depending on your specific goals there are better options but these actions should isolate your problem.
2012/1/7 Juan Antonio Ibañez Santorum <juanito1982@gmail.com>:
Of course:
Caller: 192.168.2.2 FS: 192.168.2.238 Gateway (Asterisk): 192.168.2.239
http://pastebin.freeswitch.org/18099
Regards
2012/1/7 Paul Cupis <paul@cupis.co.uk>
On 07/01/12 16:32, Juan Antonio Ibañez Santorum wrote:
No, same result, 480 instead 486
Can you show us the full log of the call, please?
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
-- Kristian Kielhofner
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
On Thu, Jan 5, 2012 at 7:59 PM, Peter P GMX <Prometheus001@gmx.net> wrote:
Hello,
I have a strange phenomen:
When a target UA is busy, it returns "486 Busy" to Freeswitch. But Freeswitch then returns "480 Temporarily Unavailable" to the called party. Where does this come from and how can I change this behaviour?
See (anonymized) SIP trace with ngrep:
UA to Freeswitch: ======================== U 2012/01/04 13:59:44.928775 <fritzbox-IP>:5060 -> <freeswitch-IP>:5080 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP <freeswitch-IP>:5080;rport=5080;branch=z9hG4bKNZZDv0Syp4eyr. From: "026xxxxxxxx" <sip:026xxxxxxxx@<freeswitch-IP>>;tag=py094Kv7vr03a. To: <sip:069xxxxxxxx@ <fritzbox-IP>;uniq=B05FE4881A55AEEB69361EFA327DB>;tag=E1C3374B97DAB2DE. Call-ID: d0d0d057-b176-122f-1f8d-001ec9b9da3c. CSeq: 22504928 INVITE. User-Agent: AVM FRITZ!Box 6360 Cable 85.05.07 (Sep 14 2011). Content-Length: 0.
Freeswitch to Caller: ======================== U 2012/01/04 13:59:44.930387 <freeswitch-IP>:5060 -> <Sip-provider-IP>:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP <Sip-provider-IP>:5060;branch=z9hG4bK-4896-2830DFA. From: <sip:4926xxxxxxxx@sip.provider.de ;user=phone>;tag=13517-HB-08a98588-2622da197. To: <sip:4969xxxxxxxx@<freeswitch-IP>;user=phone>;tag=XQtc5US24QgDa. Call-ID: 13517-SG-08a98587-0a352e121@sip.provider.de. CSeq: 134781549 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-68627e8 2011-11-21 13-52-28 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0. P-Asserted-Identity: "069xxxxxxxx" <sip:069xxxxxxxx@<freeswitch-IP>>.
Best regards Peter
I believe this is a bug. I have opened a jira ticket: http://jira.freeswitch.org/browse/FS-3810 regards, Takeshi
On Fri, Jan 13, 2012 at 3:07 PM, mayamatakeshi <mayamatakeshi@gmail.com>wrote:
On Thu, Jan 5, 2012 at 7:59 PM, Peter P GMX <Prometheus001@gmx.net> wrote:
Hello,
I have a strange phenomen:
When a target UA is busy, it returns "486 Busy" to Freeswitch. But Freeswitch then returns "480 Temporarily Unavailable" to the called party. Where does this come from and how can I change this behaviour?
See (anonymized) SIP trace with ngrep:
UA to Freeswitch: ======================== U 2012/01/04 13:59:44.928775 <fritzbox-IP>:5060 -> <freeswitch-IP>:5080 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP <freeswitch-IP>:5080;rport=5080;branch=z9hG4bKNZZDv0Syp4eyr. From: "026xxxxxxxx" <sip:026xxxxxxxx@<freeswitch-IP>>;tag=py094Kv7vr03a. To: <sip:069xxxxxxxx@ <fritzbox-IP>;uniq=B05FE4881A55AEEB69361EFA327DB>;tag=E1C3374B97DAB2DE. Call-ID: d0d0d057-b176-122f-1f8d-001ec9b9da3c. CSeq: 22504928 INVITE. User-Agent: AVM FRITZ!Box 6360 Cable 85.05.07 (Sep 14 2011). Content-Length: 0.
Freeswitch to Caller: ======================== U 2012/01/04 13:59:44.930387 <freeswitch-IP>:5060 -> <Sip-provider-IP>:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP <Sip-provider-IP>:5060;branch=z9hG4bK-4896-2830DFA. From: <sip:4926xxxxxxxx@sip.provider.de ;user=phone>;tag=13517-HB-08a98588-2622da197. To: <sip:4969xxxxxxxx@<freeswitch-IP>;user=phone>;tag=XQtc5US24QgDa. Call-ID: 13517-SG-08a98587-0a352e121@sip.provider.de. CSeq: 134781549 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-68627e8 2011-11-21 13-52-28 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0. P-Asserted-Identity: "069xxxxxxxx" <sip:069xxxxxxxx@<freeswitch-IP>>.
Best regards Peter
I believe this is a bug. I have opened a jira ticket: http://jira.freeswitch.org/browse/FS-3810
Solved.
participants (6)
-
Juan Antonio Ibañez Santorum
-
Kristian Kielhofner
-
mayamatakeshi
-
Paul Cupis
-
Peter P GMX
-
Sammy Govind