Sebastian Arcus
2011-Oct-02 15:20 UTC
[asterisk-users] Sipgate trunk doesn't bridge with other trunk, but works with local extensions
Hello list, My setup is as follows: Trunks: 2 sip trunks, one with voipcheap.co.uk, one with sipgate.co.uk Extensions: 1 hardware sip phone Asterisk: 1.8.7.0 Everything is working fine, except bridging between the sipgate and voipcheap trunks. I'll explain: 1. If I call from an external phone my sipgate landline number, it connects to my internal hardware sip phone/extension and works fine. 2. If I use my hardware sip phone to make outgoing calls through the voipcheap.co.uk trunk - it all works fine. 3. However, I want the call coming in through the sipgate trunk to call my mobile phone through the voipcheap trunk - this is not working. It will ring the mobile number, but when I answer there is no sound at either end. I assume it is not: 1. A NAT problem, otherwise it would cause problems when making calls through voipcheap, or receiving through sipgate (but I could be wrong). 2. A codec problem - as I've forced everything on alaw I can't see any errors in the console either. Please find below my sip.conf, extensions.conf: #/etc/asterisk/sip.conf [general] canreinvite=no disallow=all allow=alaw allowguest=no externip=111.222.333.444 localnet=192.168.16.0/255.255.255.0 tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. registerattempts=0 register => 1234567:my_password at sipgate.co.uk/1234567 [sipgate] type = friend host=sipgate.co.uk fromdomain=sipgate.co.uk disallow=all allow=alaw qualify=yes nat=yes canreinvite=no [voipcheap] type=peer username=my_username fromdomain=sip.voipcheap.co.uk realm=sip.voipcheap.co.uk secret=my_password host=sip.voipcheap.co.uk disallow=all allow=alaw canreinvite=no [20] type=friend username=20 secret=my_password host=dynamic context=from_internal_sip qualify=yes #/etc/asterisk/extensions.conf [general] static=yes writeprotect=yes autofallthrough=yes priorityjumping=no [from_internal_sip] exten => _9.,1,Dial(SIP/${EXTEN:1}@voipcheap) exten => _9.,n,HangUp() [from_sipgate] exten => 6012878,1,Dial(SIP/0794012345 at voipcheap) exten => 6012878,n,HangUp() Any suggestions would be appreciated Sebastian
Bryant Zimmerman
2011-Oct-02 19:15 UTC
[asterisk-users] Sipgate trunk doesn't bridge with other trunk, but works with local extensions
verify your codec are the same on both trunks. make sure the both trunks are using the same codec. make sure you have the correct ports open. make sure you force all udp traffic to flow through your astrisk switch as well. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 ---------------------------------------- From: "Sebastian Arcus" <shop at open-t.co.uk> Sent: Sunday, October 02, 2011 11:20 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Subject: [asterisk-users] Sipgate trunk doesn't bridge with other trunk, but works with local extensions Hello list, My setup is as follows: Trunks: 2 sip trunks, one with voipcheap.co.uk, one with sipgate.co.uk Extensions: 1 hardware sip phone Asterisk: 1.8.7.0 Everything is working fine, except bridging between the sipgate and voipcheap trunks. I'll explain: 1. If I call from an external phone my sipgate landline number, it connects to my internal hardware sip phone/extension and works fine. 2. If I use my hardware sip phone to make outgoing calls through the voipcheap.co.uk trunk - it all works fine. 3. However, I want the call coming in through the sipgate trunk to call my mobile phone through the voipcheap trunk - this is not working. It will ring the mobile number, but when I answer there is no sound at either end. I assume it is not: 1. A NAT problem, otherwise it would cause problems when making calls through voipcheap, or receiving through sipgate (but I could be wrong). 2. A codec problem - as I've forced everything on alaw I can't see any errors in the console either. Please find below my sip.conf, extensions.conf: #/etc/asterisk/sip.conf [general] canreinvite=no disallow=all allow=alaw allowguest=no externip=111.222.333.444 localnet=192.168.16.0/255.255.255.0 tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. registerattempts=0 register => 1234567:my_password at sipgate.co.uk/1234567 [sipgate] type = friend host=sipgate.co.uk fromdomain=sipgate.co.uk disallow=all allow=alaw qualify=yes nat=yes canreinvite=no [voipcheap] type=peer username=my_username fromdomain=sip.voipcheap.co.uk realm=sip.voipcheap.co.uk secret=my_password host=sip.voipcheap.co.uk disallow=all allow=alaw canreinvite=no [20] type=friend username=20 secret=my_password host=dynamic context=from_internal_sip qualify=yes #/etc/asterisk/extensions.conf [general] static=yes writeprotect=yes autofallthrough=yes priorityjumping=no [from_internal_sip] exten => _9.,1,Dial(SIP/${EXTEN:1}@voipcheap) exten => _9.,n,HangUp() [from_sipgate] exten => 6012878,1,Dial(SIP/0794012345 at voipcheap) exten => 6012878,n,HangUp() Any suggestions would be appreciated Sebastian -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111002/cd2d8a60/attachment.htm>
dotnetdub
2011-Oct-02 19:42 UTC
[asterisk-users] Sipgate trunk doesn't bridge with other trunk, but works with local extensions
On 2 October 2011 16:20, Sebastian Arcus <shop at open-t.co.uk> wrote:> Hello list, > > My setup is as follows: > > Trunks: 2 sip trunks, one with voipcheap.co.uk, one with sipgate.co.uk > Extensions: 1 hardware sip phone > Asterisk: 1.8.7.0 > > Everything is working fine, except bridging between the sipgate and > voipcheap trunks. I'll explain: > >SNIP What kind of firewall are you using? If I was you I would take the FW out of the equation and test this. If possible put your server on a public IP and check. Regards Brian -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111002/04bd25b9/attachment.htm>