Hi,
we have and Asterisk server connected to a Patton Smartnode 4638 with
4 BRI.
We configured 4 SIP account on Patton (1001, 1002, 1003, 1004).
The system is fully functional, but we have a problem to recognize
incoming calls from Asterisk: when a call come from SIP/1001 (BRI 1 on
Patton) or SIP/1002 (BRI 2) or SIP/1003 (BRI 3) Asterisk record a call
coming from SIP/1004.
I have contacted Patton support, I have send configuration and debug
and they told me that there is a problem of Asterisk configuration.
In the sip debug on Asterisk I have seen (SIP/1001 incoming call)
...
Sending to 192.168.2.122 : 5060 (no NAT)
Using INVITE request as basis request -
89c9689349c54649aae566e9192c529b at 192.168.2.122
Found peer '1004'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
...
(192.168.2.122 is the ip address of Smartnode.)
In the Patton configuration
...
gateway sip ASTERISK
bind interface LAN router
service default
defaultserver manual 192.168.2.121 5060 loose-router
registration manual 192.168.2.121
user 1001 authenticate password 36ocYTYpKxk= encrypted register
display-name 1001
gateway sip ASTERISK
no shutdown
...
(192.168.2.121 is the ip address of Asterisk server)
The call is coming from SIP/1001, but the INVITE request founds peer
1004.
The problem come when I try to use FOP: I am not able to correctly
connect button to trunk.
Someone can help me?
Thanks in advance.
Eco
Il giorno 30/apr/10, alle ore 10:01, A.Santoro ha scritto:> Hi, > we have and Asterisk server connected to a Patton Smartnode 4638 with > 4 BRI. > [...]Hi Eco, I think the problem is in your sip.conf. Have you tried setting "insecure=port,invite" in the sip.conf for each sip account? Bye, Carlo
2010/4/30 A.Santoro <ng at ecoricerche.it>> Hi, > we have and Asterisk server connected to a Patton Smartnode 4638 with > 4 BRI. [...] >Hi Eco, I think the problem is in your sip.conf. Have you tried setting "insecure=port,invite" in the sip.conf for each sip account? Bye, Carlo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100430/184b8b60/attachment.htm
Hi!> calls from Asterisk: when a call come from SIP/1001 (BRI 1 on Patton) > or SIP/1002 (BRI 2) or SIP/1003 (BRI 3) Asterisk record a call coming > from SIP/1004.Read up on how Asterisk does user/peer matching in sip.conf on inbound calls: With all users/peers having the same IP and hostname it is the entry that was defined last in sip.conf that wins. Here's a starter: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer Olle has often posted in more detail about this here. Either you simply do not differentiate between the different lines and treat them all as one single trunk (why exactly do you need to know which line is in use?), or you have to consider other ways like assigning different SIP ports on the Patton (a SIP gateway for each line), or maybe use different usernames when calling asterisk, or check if using different SIP domains (see [general] section in sip.conf) can help you. See also: http://www.mail-archive.com/asterisk-dev at lists.digium.com/msg39355.html https://issues.asterisk.org/view.php?id=14340 https://issues.asterisk.org/view.php?id=14250 Note: Your issue is Patton --> Asterisk, while the registration part of the Patton config that you posted matters for Asterisk --> Patton calls. Philipp
On Fri, 30 Apr 2010 10:39:12 +0200, Carlo Dimaggio <jaasmailing at gmail.com> wrote:>2010/4/30 A.Santoro <ng at ecoricerche.it> > >> Hi, >> we have and Asterisk server connected to a Patton Smartnode 4638 with >> 4 BRI. [...] >> >Have you tried setting "insecure=port,invite" in the sip.conf for each sip >account? >Hi Carlo, thanks for your answer. Now I tried... and nothing is changed. In the following lines one the sip account of the peers (in sip.conf) [1001] username=1001 type=friend secret=xxxx dtmfmode=auto insecure=very host=dynamic port=5060 context=inbound qualify=yes disallow=all allow=ulaw allow=alaw canreinvite=no Bye. Eco
On Fri, 30 Apr 2010 14:16:14 +0200, Philipp von Klitzing <klitzing at pool.informatik.rwth-aachen.de> wrote:>Hi! > >> calls from Asterisk: when a call come from SIP/1001 (BRI 1 on Patton) >> or SIP/1002 (BRI 2) or SIP/1003 (BRI 3) Asterisk record a call coming >> from SIP/1004. > >Read up on how Asterisk does user/peer matching in sip.conf on inbound >calls: With all users/peers having the same IP and hostname it is the >entry that was defined last in sip.conf that wins.Philipp thanks for your answer. This clears all my doubts, is not my configuration problem.>why exactly do you need to know which >line is in use?We have 4 trunk and 4 company in our office, I was testing FOP and I would want to show the occupied trunks for inbound and outbound calls for single company. Thanks again. Bye Eco