1 Jul
2014
1 Jul
'14
8:12 p.m.
Hello List
Hopefully someone can help. This is the problem when the call is hug up 20-30 seconds
after it initiates. The call is only hung on when the remote extension initiates the call.
If the remote extension receives the call there is no problem the call is not hung on. I
changed the remote cisco phone for a yealink and it is the same behavior. It thought it
was the phone.
This is what I am using in kamailio.cfg
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_ASTERISK
#!define WITH_USRLOCDB
#!define WITH_ANTIFLOOD
Remote User Internet
Internal network
Yealink IP TG28P ----DSL router ---|------Internet --------|-----Cisco ASA 5500
FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx Production
Server --------|------ PSTN
Thanks
Carlos Rangel
De: Carlos Rangel [mailto:crangel@globaltelesourcing.com]
Enviado el: jueves, 26 de junio de 2014 01:27 p.m.
Para: miconda(a)gmail.com; 'Kamailio (SER) - Users Mailing List'
Asunto: RE: [SR-Users] Kamailio Freepbx Integration Dropping Calls
Hi Daniel
Thank you so much for your response. Here is the SIP trace of one of the calls, I am not
sure where the call initiates but you can see at the end of the file in bold
X-Asterisk-HangupCause: No user responding. I am not sure why is it sending this message
though.
The variables are
Extension/Username=XXXXX
Ext_IP= Public IP
Internal_IP= Asterisk/Kamailio internal IP
Sorry for the long file but again I am not sure where the call initiates
This is the part where that call is hung on.
U 2014/06/26 13:36:11.831965 Kamailio_IP:5080 -> Kamailio_IP:5060
BYE sip:XXXXX@65.190.71.203:5060;user=phone;transport=udp SIP/2.0.
Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47.
Route: <sip:Kamailio_IP;lr=on;ftag=000653dc394000970f227678-1fafb4e2>.
Max-Forwards: 70.
From: <sip:919707249077@Kamailio_IP:5060>;tag=as2e670ea4.
To: "User"
<sip:XXXXX@Kamailio_IP:5060>;tag=000653dc394000970f227678-1fafb4e2.
Call-ID: 000653dc-3940000b-33caf1b2-20ccd185(a)192.168.0.22.
CSeq: 102 BYE.
User-Agent: FPBX-2.11.0(11.10.2).
X-Asterisk-HangupCause: No user responding.
X-Asterisk-HangupCauseCode: 18.
Content-Length: 0.
.
U 2014/06/26 13:36:11.832260 Kamailio_IP:5060 -> 65.190.71.203:5060
BYE sip:XXXXX@65.190.71.203:5060;user=phone;transport=udp SIP/2.0.
Via: SIP/2.0/UDP Kamailio_IP;branch=z9hG4bKcf68.d6ef5aa9cc5bd0fb0ab13a563b7cf284.0.
Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47.
Max-Forwards: 69.
From: <sip:919707249077@Kamailio_IP:5060>;tag=as2e670ea4.
To: "User"
<sip:XXXXX@Kamailio_IP:5060>;tag=000653dc394000970f227678-1fafb4e2.
Call-ID: 000653dc-3940000b-33caf1b2-20ccd185(a)192.168.0.22.
CSeq: 102 BYE.
User-Agent: FPBX-2.11.0(11.10.2).
X-Asterisk-HangupCause: No user responding.
X-Asterisk-HangupCauseCode: 18.
Content-Length: 0.
Descripción: Description: Description: DLR-Logo-No-TextCARLOS RANGEL | INFORMATION
TECHNOLOGY DIRECTOR
Global Telesourcing México, S. de R.L. de C.V. | Aarón Sáenz #1891-1 | Monterrey, N.L.,
México
Direct 703 894 1667 | Mobile US 703 894 1667 | Mobile MX +52 1 812 000 7362 |
crangel(a)globaltelesourcing.com
________________________________________________________________________________________________________________
The information contained in this e-mail and any attached documents may contain
information that is confidential or otherwise protected from disclosure. If you are not
the intended recipient of this message, or if this message has been sent to you in error,
please immediately alert the sender by reply e-mail and then delete this message,
including any attachments. Any dissemination, distribution or other use of the contents of
this message by anyone other than the intended recipient is strictly prohibited.
De: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org]
En nombre de Daniel-Constantin Mierla
Enviado el: jueves, 26 de junio de 2014 03:12 a.m.
Para: Kamailio (SER) - Users Mailing List
Asunto: Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls
Hello,
can you gran the SIP trace on kamailio server for such case?
You can use ngrep, like:
ngrep -d any -qt -W byline port 5060
and send the output to the mailing list. You can replace any sensitive information (e.g.,
ip address) before sending to mailing list.
The typical call drop after 30-40 secs is when ACK is not routed properly, but we have to
see that in the sip trace.
Cheers,
Daniel
On 25/06/14 18:50, Carlos Rangel wrote:
Hello
I have successfully (I believe) implemented Kamailio 4.1.4 integration with Freepbx 5.2.11
taking as a guide Daniel’s tutorial
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb.
I just did not create the voicemail tables because voice mail is handled by Freepbx. I
installed the system in a separate box for testing and connected to the Freepbx Production
server via IAX trunk.
The system is behind a Cisco Firewall and looks like this
Remote User Internet
Internal network
Cisco 7960 ----DSL router ---|------Internet --------|-----Cisco ASA 5500
FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx Production
Server --------|------ PSTN
I have configured the FW to allow UDP and TCP traffic from the corresponding IP as well as
tfpt that is needed for the Ciscos to pick up the configuration from the server. I have a
few remotes Cisco 7960 phones that can register remotely in Kamailio as long as the user
is added with kamctl add user password and as long as the extension is created in Freepbx.
The problem that I have is when try to make a call from the remote Ciscos the call is
dropped after 30 or 40 seconds. I can see from the logs that the problem appears to be
that the server is not receiving responses from the phone
06-25 10:57:30] WARNING[1814] chan_sip.c: Retransmission timeout reached on transmission
000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 for seqno 102 (Critical Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
[2014-06-25 10:57:30] WARNING[1814] chan_sip.c: Hanging up call
000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 - no reply to our critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Is this something that we can adjust in kamailio or could it be related to the FW
configuration?? Sorry but I am very new to kamailio and sip.
Thanks
Carlos
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
1 Jul
1 Jul
8:26 p.m.
New subject: Kamailio Freepbx Integration Dropping Calls
do you have an ngrep of the sip traffic? This can happen if the sip/rtp
cannot connect (perhaps blocked by the dsl router)
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)
On 07/01/2014 01:12 PM, Carlos Rangel wrote:
Hello List
Hopefully someone can help. This is the problem when the call is hug up 20-30 seconds
after it initiates. The call is only hung on when the remote extension initiates the call.
If the remote extension receives the call there is no problem the call is not hung on. I
changed the remote cisco phone for a yealink and it is the same behavior. It thought it
was the phone.
This is what I am using in kamailio.cfg
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_ASTERISK
#!define WITH_USRLOCDB
#!define WITH_ANTIFLOOD
Remote User Internet
Internal network
Yealink IP TG28P ----DSL router ---|------Internet --------|-----Cisco ASA 5500
FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx Production
Server --------|------ PSTN
Thanks
Carlos Rangel
De: Carlos Rangel [mailto:crangel@globaltelesourcing.com]
Enviado el: jueves, 26 de junio de 2014 01:27 p.m.
Para: miconda(a)gmail.com; 'Kamailio (SER) - Users Mailing List'
Asunto: RE: [SR-Users] Kamailio Freepbx Integration Dropping Calls
Hi Daniel
Thank you so much for your response. Here is the SIP trace of one of the calls, I am not
sure where the call initiates but you can see at the end of the file in bold
X-Asterisk-HangupCause: No user responding. I am not sure why is it sending this message
though.
The variables are
Extension/Username=XXXXX
Ext_IP= Public IP
Internal_IP= Asterisk/Kamailio internal IP
Sorry for the long file but again I am not sure where the call initiates
This is the part where that call is hung on.
U 2014/06/26 13:36:11.831965 Kamailio_IP:5080 -> Kamailio_IP:5060
BYE sip:XXXXX@65.190.71.203:5060;user=phone;transport=udp SIP/2.0.
Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47.
Route: <sip:Kamailio_IP;lr=on;ftag=000653dc394000970f227678-1fafb4e2>.
Max-Forwards: 70.
From: <sip:919707249077@Kamailio_IP:5060>;tag=as2e670ea4.
To: "User"
<sip:XXXXX@Kamailio_IP:5060>;tag=000653dc394000970f227678-1fafb4e2.
Call-ID: 000653dc-3940000b-33caf1b2-20ccd185(a)192.168.0.22.
CSeq: 102 BYE.
User-Agent: FPBX-2.11.0(11.10.2).
X-Asterisk-HangupCause: No user responding.
X-Asterisk-HangupCauseCode: 18.
Content-Length: 0.
.
U 2014/06/26 13:36:11.832260 Kamailio_IP:5060 -> 65.190.71.203:5060
BYE sip:XXXXX@65.190.71.203:5060;user=phone;transport=udp SIP/2.0.
Via: SIP/2.0/UDP Kamailio_IP;branch=z9hG4bKcf68.d6ef5aa9cc5bd0fb0ab13a563b7cf284.0.
Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47.
Max-Forwards: 69.
From: <sip:919707249077@Kamailio_IP:5060>;tag=as2e670ea4.
To: "User"
<sip:XXXXX@Kamailio_IP:5060>;tag=000653dc394000970f227678-1fafb4e2.
Call-ID: 000653dc-3940000b-33caf1b2-20ccd185(a)192.168.0.22.
CSeq: 102 BYE.
User-Agent: FPBX-2.11.0(11.10.2).
X-Asterisk-HangupCause: No user responding.
X-Asterisk-HangupCauseCode: 18.
Content-Length: 0.
Descripción: Description: Description: DLR-Logo-No-TextCARLOS RANGEL | INFORMATION
TECHNOLOGY DIRECTOR
Global Telesourcing México, S. de R.L. de C.V. | Aarón Sáenz #1891-1 | Monterrey, N.L.,
México
Direct 703 894 1667 | Mobile US 703 894 1667 | Mobile MX +52 1 812 000 7362 |
crangel(a)globaltelesourcing.com
________________________________________________________________________________________________________________
The information contained in this e-mail and any attached documents may contain
information that is confidential or otherwise protected from disclosure. If you are not
the intended recipient of this message, or if this message has been sent to you in error,
please immediately alert the sender by reply e-mail and then delete this message,
including any attachments. Any dissemination, distribution or other use of the contents of
this message by anyone other than the intended recipient is strictly prohibited.
De: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org]
En nombre de Daniel-Constantin Mierla
Enviado el: jueves, 26 de junio de 2014 03:12 a.m.
Para: Kamailio (SER) - Users Mailing List
Asunto: Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls
Hello,
can you gran the SIP trace on kamailio server for such case?
You can use ngrep, like:
ngrep -d any -qt -W byline port 5060
and send the output to the mailing list. You can replace any sensitive information (e.g.,
ip address) before sending to mailing list.
The typical call drop after 30-40 secs is when ACK is not routed properly, but we have to
see that in the sip trace.
Cheers,
Daniel
On 25/06/14 18:50, Carlos Rangel wrote:
Hello
I have successfully (I believe) implemented Kamailio 4.1.4 integration with Freepbx
5.2.11 taking as a guide Daniel’s tutorial
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb.
I just did not create the voicemail tables because voice mail is handled by Freepbx. I
installed the system in a separate box for testing and connected to the Freepbx Production
server via IAX trunk.
The system is behind a Cisco Firewall and looks like this
Remote User Internet
Internal network
Cisco 7960 ----DSL router ---|------Internet --------|-----Cisco ASA 5500
FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx Production
Server --------|------ PSTN
I have configured the FW to allow UDP and TCP traffic from the corresponding IP as well
as tfpt that is needed for the Ciscos to pick up the configuration from the server. I have
a few remotes Cisco 7960 phones that can register remotely in Kamailio as long as the
user is added with kamctl add user password and as long as the extension is created in
Freepbx.
The problem that I have is when try to make a call from the remote Ciscos the call is
dropped after 30 or 40 seconds. I can see from the logs that the problem appears to be
that the server is not receiving responses from the phone
06-25 10:57:30] WARNING[1814] chan_sip.c: Retransmission timeout reached on transmission
000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 for seqno 102 (Critical Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
[2014-06-25 10:57:30] WARNING[1814] chan_sip.c: Hanging up call
000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 - no reply to our critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Is this something that we can adjust in kamailio or could it be related to the FW
configuration?? Sorry but I am very new to kamailio and sip.
Thanks
Carlos
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
3797
days inactive
3797
days old
1 comments
2 participants
participants (2)
-
Carlos Rangel -
Fred Posner