Freeswitch not responding to sip messages from call out leg
Hello, We are using FS to bridge incoming call to an outbound telephone through a gateway provider. The bridge is created OK, but we found that after 3 minutes FS is no longer responding to sip messages from the call out provider. Thus, in case the provider sends sip-Invite message, it gets no response from FS and disconnect the call out leg. Note that FS is also not aware of the Bye message so leg A is still on the line and when it hangs up FS send Bye message to leg B and get as response "Unknown Dialog". I set the siptrace on and see that no packets arriving sofia (when the calling leg hanged up I do see the packets). Also I set all sofia components log level to 7 - this is what was printed out after the call answered: 2013-07-08 10:28:49.661014 [NOTICE] sofia.c:6288 Channel [sofia/external/97249126793] has been answered 2013-07-08 10:28:49.661014 [DEBUG] switch_channel.c:3291 sofia/external/97249126793 process sched_hangup +7199 DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45 alloted_timeout: sched_hangup(+7199 DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45 alloted_timeout) 2013-07-08 10:28:49.661014 [DEBUG] switch_scheduler.c:214 Added task 6 switch_ivr_schedule_hangup (DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45) to run at 1373275728 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/external/97249126793 [BREAK] 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/internal/972522977131@voxbone.com [BREAK] nta.c:8950 outgoing_timer_dk() nta: timer D fired, terminate INVITE (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/1 tout, 1/1 term, 1/2 free nta.c:8831 outgoing_timer_bf() nta: timer F fired, terminating ACK (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 tout, 0/0 term, 1/1 free We are running FreeSWITCH Version 1.2.7+git~20130307T054051Z~0a2e713593 (git 0a2e713 2013-03-07 05:40:51Z) On ubuntu release 12.04 Please help. Thanks Zvi Agmon Best regards Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109
Hello, Can any one help with this issue? We have no idea what is wrong. The problem reproduced with 2 different gateway providers and with the basic configuration that FS comes with (gateways and the bridge dial plan were added). Is there anything in the configuration that can cause such a problem? Here are the configurations *In public.xml:* <extension name="public_did"> <condition field="destination_number" "expression="^.*$"> <!--action application="info"/--> <action application="log" data="INFO in public_did" /> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="972544505109 XML bridge_call"/> </condition> </extension> *bridge_call dial plan:* <include> <context name="bridge_call"> <extension name="bridge_call"> <condition field="destination_number" expression="(.*)"> <action application="log" data="INFO bridge_call destination_number=${destination_number}" /> <action application="bridge" data="sofia/gateway/Jajah/${destination_number}"/> </condition> </extension> </context> </include> *The gateway:* <gateway name="Jajah"> <param name="realm" value="91.194.5.180"/> <param name="username" value=""/> <param name="password" value=""/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="sip_cid_type" value="none"/> </gateway> Please help...! Thanks! Best regards Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109 On Mon, Jul 8, 2013 at 10:55 AM, Zvi Agmon <zvi@lexifone.com> wrote:
Hello,
We are using FS to bridge incoming call to an outbound telephone through a gateway provider. The bridge is created OK, but we found that after 3 minutes FS is no longer responding to sip messages from the call out provider. Thus, in case the provider sends sip-Invite message, it gets no response from FS and disconnect the call out leg. Note that FS is also not aware of the Bye message so leg A is still on the line and when it hangs up FS send Bye message to leg B and get as response "Unknown Dialog".
I set the siptrace on and see that no packets arriving sofia (when the calling leg hanged up I do see the packets). Also I set all sofia components log level to 7 - this is what was printed out after the call answered:
2013-07-08 10:28:49.661014 [NOTICE] sofia.c:6288 Channel [sofia/external/97249126793] has been answered 2013-07-08 10:28:49.661014 [DEBUG] switch_channel.c:3291 sofia/external/97249126793 process sched_hangup +7199 DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45 alloted_timeout: sched_hangup(+7199 DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45alloted_timeout)
2013-07-08 10:28:49.661014 [DEBUG] switch_scheduler.c:214 Added task 6 switch_ivr_schedule_hangup (DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45) to run at 1373275728 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/external/97249126793 [BREAK] 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/internal/972522977131@voxbone.com [BREAK] nta.c:8950 outgoing_timer_dk() nta: timer D fired, terminate INVITE (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/1 tout, 1/1 term, 1/2 free nta.c:8831 outgoing_timer_bf() nta: timer F fired, terminating ACK (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 tout, 0/0 term, 1/1 free
We are running FreeSWITCH Version 1.2.7+git~20130307T054051Z~0a2e713593 (git 0a2e713 2013-03-07 05:40:51Z)
On ubuntu release 12.04
Please help.
Thanks Zvi Agmon Best regards
Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109
You'll need to get a full console debug log with sip trace and put in pastebin.freeswitch.org. The folks here will try to help and it's easier for them to do so if they have more information. See also the "reporting bugs" wiki page for more details on gathering data for troubleshooting. -MC On Jul 21, 2013 5:49 AM, "Zvi Agmon" <zvi@lexifone.com> wrote:
Hello,
Can any one help with this issue? We have no idea what is wrong. The problem reproduced with 2 different gateway providers and with the basic configuration that FS comes with (gateways and the bridge dial plan were added). Is there anything in the configuration that can cause such a problem?
Here are the configurations
*In public.xml:*
<extension name="public_did"> <condition field="destination_number" "expression="^.*$"> <!--action application="info"/--> <action application="log" data="INFO in public_did" /> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="972544505109 XML bridge_call"/> </condition> </extension>
*bridge_call dial plan:*
<include> <context name="bridge_call"> <extension name="bridge_call"> <condition field="destination_number" expression="(.*)"> <action application="log" data="INFO bridge_call destination_number=${destination_number}" /> <action application="bridge" data="sofia/gateway/Jajah/${destination_number}"/> </condition> </extension> </context> </include>
*The gateway:*
<gateway name="Jajah"> <param name="realm" value="91.194.5.180"/> <param name="username" value=""/> <param name="password" value=""/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="sip_cid_type" value="none"/> </gateway>
Please help...! Thanks!
Best regards
Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109
On Mon, Jul 8, 2013 at 10:55 AM, Zvi Agmon <zvi@lexifone.com> wrote:
Hello,
We are using FS to bridge incoming call to an outbound telephone through a gateway provider. The bridge is created OK, but we found that after 3 minutes FS is no longer responding to sip messages from the call out provider. Thus, in case the provider sends sip-Invite message, it gets no response from FS and disconnect the call out leg. Note that FS is also not aware of the Bye message so leg A is still on the line and when it hangs up FS send Bye message to leg B and get as response "Unknown Dialog".
I set the siptrace on and see that no packets arriving sofia (when the calling leg hanged up I do see the packets). Also I set all sofia components log level to 7 - this is what was printed out after the call answered:
2013-07-08 10:28:49.661014 [NOTICE] sofia.c:6288 Channel [sofia/external/97249126793] has been answered 2013-07-08 10:28:49.661014 [DEBUG] switch_channel.c:3291 sofia/external/97249126793 process sched_hangup +7199 DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45 alloted_timeout: sched_hangup(+7199 DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45alloted_timeout)
2013-07-08 10:28:49.661014 [DEBUG] switch_scheduler.c:214 Added task 6 switch_ivr_schedule_hangup (DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45) to run at 1373275728 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/external/97249126793 [BREAK] 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/internal/972522977131@voxbone.com [BREAK] nta.c:8950 outgoing_timer_dk() nta: timer D fired, terminate INVITE (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/1 tout, 1/1 term, 1/2 free nta.c:8831 outgoing_timer_bf() nta: timer F fired, terminating ACK (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 tout, 0/0 term, 1/1 free
We are running FreeSWITCH Version 1.2.7+git~20130307T054051Z~0a2e713593 (git 0a2e713 2013-03-07 05:40:51Z)
On ubuntu release 12.04
Please help.
Thanks Zvi Agmon Best regards
Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109
_________________________________________________________________________ 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
Hi, Michael - thanks for your response. I tried to follow the ""reporting bugs" instructions and open a Jira issue about 10 days ago - http://jira.freeswitch.org/browse/FS-5592 I added to this issue the relevant logs (FS and WireShark) and tried to clearly describe what is happening. Followed Anthony comment I reproduce the issue with a different gateway provider and added these logs also to Jira. Now, I also put the latest freeswitch log to pastebin - http://pastebin.freeswitch.org/21214 Please let me know if there's any information that's still missing or if the issue is not clear enough - I'll be more than happy to provide anything needed to track down the problem. Best regards Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109 On Sun, Jul 21, 2013 at 7:37 PM, Michael Collins <msc@freeswitch.org> wrote:
You'll need to get a full console debug log with sip trace and put in pastebin.freeswitch.org. The folks here will try to help and it's easier for them to do so if they have more information. See also the "reporting bugs" wiki page for more details on gathering data for troubleshooting.
-MC On Jul 21, 2013 5:49 AM, "Zvi Agmon" <zvi@lexifone.com> wrote:
Hello,
Can any one help with this issue? We have no idea what is wrong. The problem reproduced with 2 different gateway providers and with the basic configuration that FS comes with (gateways and the bridge dial plan were added). Is there anything in the configuration that can cause such a problem?
Here are the configurations
*In public.xml:*
<extension name="public_did"> <condition field="destination_number" "expression="^.*$"> <!--action application="info"/--> <action application="log" data="INFO in public_did" /> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="972544505109 XML bridge_call"/> </condition> </extension>
*bridge_call dial plan:*
<include> <context name="bridge_call"> <extension name="bridge_call"> <condition field="destination_number" expression="(.*)"> <action application="log" data="INFO bridge_call destination_number=${destination_number}" /> <action application="bridge" data="sofia/gateway/Jajah/${destination_number}"/> </condition> </extension> </context> </include>
*The gateway:*
<gateway name="Jajah"> <param name="realm" value="91.194.5.180"/> <param name="username" value=""/> <param name="password" value=""/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="sip_cid_type" value="none"/> </gateway>
Please help...! Thanks!
Best regards
Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109
On Mon, Jul 8, 2013 at 10:55 AM, Zvi Agmon <zvi@lexifone.com> wrote:
Hello,
We are using FS to bridge incoming call to an outbound telephone through a gateway provider. The bridge is created OK, but we found that after 3 minutes FS is no longer responding to sip messages from the call out provider. Thus, in case the provider sends sip-Invite message, it gets no response from FS and disconnect the call out leg. Note that FS is also not aware of the Bye message so leg A is still on the line and when it hangs up FS send Bye message to leg B and get as response "Unknown Dialog".
I set the siptrace on and see that no packets arriving sofia (when the calling leg hanged up I do see the packets). Also I set all sofia components log level to 7 - this is what was printed out after the call answered:
2013-07-08 10:28:49.661014 [NOTICE] sofia.c:6288 Channel [sofia/external/97249126793] has been answered 2013-07-08 10:28:49.661014 [DEBUG] switch_channel.c:3291 sofia/external/97249126793 process sched_hangup +7199 DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45 alloted_timeout: sched_hangup(+7199 DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45alloted_timeout)
2013-07-08 10:28:49.661014 [DEBUG] switch_scheduler.c:214 Added task 6 switch_ivr_schedule_hangup (DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45) to run at 1373275728 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/external/97249126793 [BREAK] 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/internal/972522977131@voxbone.com [BREAK] nta.c:8950 outgoing_timer_dk() nta: timer D fired, terminate INVITE (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/1 tout, 1/1 term, 1/2 free nta.c:8831 outgoing_timer_bf() nta: timer F fired, terminating ACK (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 tout, 0/0 term, 1/1 free
We are running FreeSWITCH Version 1.2.7+git~20130307T054051Z~0a2e713593 (git 0a2e713 2013-03-07 05:40:51Z)
On ubuntu release 12.04
Please help.
Thanks Zvi Agmon Best regards
Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109
_________________________________________________________________________ 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
Hi, Don't want to nag... but did not get any response... Any idea how to proceed? Best regards Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109 On Mon, Jul 22, 2013 at 9:46 AM, Zvi Agmon <zvi@lexifone.com> wrote:
Hi,
Michael - thanks for your response. I tried to follow the ""reporting bugs" instructions and open a Jira issue about 10 days ago - http://jira.freeswitch.org/browse/FS-5592
I added to this issue the relevant logs (FS and WireShark) and tried to clearly describe what is happening. Followed Anthony comment I reproduce the issue with a different gateway provider and added these logs also to Jira. Now, I also put the latest freeswitch log to pastebin - http://pastebin.freeswitch.org/21214
Please let me know if there's any information that's still missing or if the issue is not clear enough - I'll be more than happy to provide anything needed to track down the problem.
Best regards
Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109
On Sun, Jul 21, 2013 at 7:37 PM, Michael Collins <msc@freeswitch.org>wrote:
You'll need to get a full console debug log with sip trace and put in pastebin.freeswitch.org. The folks here will try to help and it's easier for them to do so if they have more information. See also the "reporting bugs" wiki page for more details on gathering data for troubleshooting.
-MC On Jul 21, 2013 5:49 AM, "Zvi Agmon" <zvi@lexifone.com> wrote:
Hello,
Can any one help with this issue? We have no idea what is wrong. The problem reproduced with 2 different gateway providers and with the basic configuration that FS comes with (gateways and the bridge dial plan were added). Is there anything in the configuration that can cause such a problem?
Here are the configurations
*In public.xml:*
<extension name="public_did"> <condition field="destination_number" "expression="^.*$"> <!--action application="info"/--> <action application="log" data="INFO in public_did" /> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="972544505109 XML bridge_call"/> </condition> </extension>
*bridge_call dial plan:*
<include> <context name="bridge_call"> <extension name="bridge_call"> <condition field="destination_number" expression="(.*)"> <action application="log" data="INFO bridge_call destination_number=${destination_number}" /> <action application="bridge" data="sofia/gateway/Jajah/${destination_number}"/> </condition> </extension> </context> </include>
*The gateway:*
<gateway name="Jajah"> <param name="realm" value="91.194.5.180"/> <param name="username" value=""/> <param name="password" value=""/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="sip_cid_type" value="none"/> </gateway>
Please help...! Thanks!
Best regards
Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109
On Mon, Jul 8, 2013 at 10:55 AM, Zvi Agmon <zvi@lexifone.com> wrote:
Hello,
We are using FS to bridge incoming call to an outbound telephone through a gateway provider. The bridge is created OK, but we found that after 3 minutes FS is no longer responding to sip messages from the call out provider. Thus, in case the provider sends sip-Invite message, it gets no response from FS and disconnect the call out leg. Note that FS is also not aware of the Bye message so leg A is still on the line and when it hangs up FS send Bye message to leg B and get as response "Unknown Dialog".
I set the siptrace on and see that no packets arriving sofia (when the calling leg hanged up I do see the packets). Also I set all sofia components log level to 7 - this is what was printed out after the call answered:
2013-07-08 10:28:49.661014 [NOTICE] sofia.c:6288 Channel [sofia/external/97249126793] has been answered 2013-07-08 10:28:49.661014 [DEBUG] switch_channel.c:3291 sofia/external/97249126793 process sched_hangup +7199 DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45 alloted_timeout: sched_hangup(+7199 DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45alloted_timeout)
2013-07-08 10:28:49.661014 [DEBUG] switch_scheduler.c:214 Added task 6 switch_ivr_schedule_hangup (DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45) to run at 1373275728 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/external/97249126793 [BREAK] 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/internal/972522977131@voxbone.com [BREAK] nta.c:8950 outgoing_timer_dk() nta: timer D fired, terminate INVITE (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/1 tout, 1/1 term, 1/2 free nta.c:8831 outgoing_timer_bf() nta: timer F fired, terminating ACK (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 tout, 0/0 term, 1/1 free
We are running FreeSWITCH Version 1.2.7+git~20130307T054051Z~0a2e713593 (git 0a2e713 2013-03-07 05:40:51Z)
On ubuntu release 12.04
Please help.
Thanks Zvi Agmon Best regards
Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109
_________________________________________________________________________ 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
Following this issue, We tried to reproduce this issue with another SIP server (Voxeo Prophecy) and we saw that the SIP server replies back an answer on the outgoing port to INVITE requests. Is there any configuration property which might be causing FreeSwitch to stop listen the port of the outgoing leg? This issue is very important and we would appreciate your response. Best regards, Doron. On Thu, Jul 25, 2013 at 2:59 PM, Zvi Agmon <zvi@lexifone.com> wrote:
Hi,
Don't want to nag... but did not get any response... Any idea how to proceed?
Best regards
Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109
On Mon, Jul 22, 2013 at 9:46 AM, Zvi Agmon <zvi@lexifone.com> wrote:
Hi,
Michael - thanks for your response. I tried to follow the ""reporting bugs" instructions and open a Jira issue about 10 days ago - http://jira.freeswitch.org/browse/FS-5592
I added to this issue the relevant logs (FS and WireShark) and tried to clearly describe what is happening. Followed Anthony comment I reproduce the issue with a different gateway provider and added these logs also to Jira. Now, I also put the latest freeswitch log to pastebin - http://pastebin.freeswitch.org/21214
Please let me know if there's any information that's still missing or if the issue is not clear enough - I'll be more than happy to provide anything needed to track down the problem.
Best regards
Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109
On Sun, Jul 21, 2013 at 7:37 PM, Michael Collins <msc@freeswitch.org>wrote:
You'll need to get a full console debug log with sip trace and put in pastebin.freeswitch.org. The folks here will try to help and it's easier for them to do so if they have more information. See also the "reporting bugs" wiki page for more details on gathering data for troubleshooting.
-MC On Jul 21, 2013 5:49 AM, "Zvi Agmon" <zvi@lexifone.com> wrote:
Hello,
Can any one help with this issue? We have no idea what is wrong. The problem reproduced with 2 different gateway providers and with the basic configuration that FS comes with (gateways and the bridge dial plan were added). Is there anything in the configuration that can cause such a problem?
Here are the configurations
*In public.xml:*
<extension name="public_did"> <condition field="destination_number" "expression="^.*$"> <!--action application="info"/--> <action application="log" data="INFO in public_did" /> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="972544505109 XML bridge_call"/> </condition> </extension>
*bridge_call dial plan:*
<include> <context name="bridge_call"> <extension name="bridge_call"> <condition field="destination_number" expression="(.*)"> <action application="log" data="INFO bridge_call destination_number=${destination_number}" /> <action application="bridge" data="sofia/gateway/Jajah/${destination_number}"/> </condition> </extension> </context> </include>
*The gateway:*
<gateway name="Jajah"> <param name="realm" value="91.194.5.180"/> <param name="username" value=""/> <param name="password" value=""/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="sip_cid_type" value="none"/> </gateway>
Please help...! Thanks!
Best regards
Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109
On Mon, Jul 8, 2013 at 10:55 AM, Zvi Agmon <zvi@lexifone.com> wrote:
Hello,
We are using FS to bridge incoming call to an outbound telephone through a gateway provider. The bridge is created OK, but we found that after 3 minutes FS is no longer responding to sip messages from the call out provider. Thus, in case the provider sends sip-Invite message, it gets no response from FS and disconnect the call out leg. Note that FS is also not aware of the Bye message so leg A is still on the line and when it hangs up FS send Bye message to leg B and get as response "Unknown Dialog".
I set the siptrace on and see that no packets arriving sofia (when the calling leg hanged up I do see the packets). Also I set all sofia components log level to 7 - this is what was printed out after the call answered:
2013-07-08 10:28:49.661014 [NOTICE] sofia.c:6288 Channel [sofia/external/97249126793] has been answered 2013-07-08 10:28:49.661014 [DEBUG] switch_channel.c:3291 sofia/external/97249126793 process sched_hangup +7199 DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45 alloted_timeout: sched_hangup(+7199 DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45alloted_timeout)
2013-07-08 10:28:49.661014 [DEBUG] switch_scheduler.c:214 Added task 6 switch_ivr_schedule_hangup (DJIWR4KKCZB3DGAS7LYLI76IPE@81.201.82.45) to run at 1373275728 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/external/97249126793 [BREAK] 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/internal/972522977131@voxbone.com [BREAK] nta.c:8950 outgoing_timer_dk() nta: timer D fired, terminate INVITE (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/1 tout, 1/1 term, 1/2 free nta.c:8831 outgoing_timer_bf() nta: timer F fired, terminating ACK (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 tout, 0/0 term, 1/1 free
We are running FreeSWITCH Version 1.2.7+git~20130307T054051Z~0a2e713593 (git 0a2e713 2013-03-07 05:40:51Z)
On ubuntu release 12.04
Please help.
Thanks Zvi Agmon Best regards
Zvi Agmon Lexifone email: zvi@lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109
_________________________________________________________________________ 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
_________________________________________________________________________ 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
-- Doron Kruh, Senior Developer Lexifone , Hapalyam 7 Haifa. Mobile: 972-52-3823361 Office: 972-4-9126792
participants (3)
-
Doron Kruh
-
Michael Collins
-
Zvi Agmon