Hi all I have been trying to figure out where Im going wrong with following this guide: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
The only thing that is different is I am using Asterisk 1.8.22.0
Kamailio works fine and I am able to register a phone to it but the forward register fails to register with Asterisk.
Its like Asterisk doesn't see the request even running sip debug on the Asterisk Console it doesn't show the register attempts. I have to use ngrep to see any thing.
Realtime works fine when I register the phone on Asterisk.
Although I can get round the register using AGI I get the same issue when passing a call to asterisk, Asterisk doesn't see the invite.
Am I missing something so obvious :-/
This is what I have in sip.conf "IP change to protect the innocent!"
[kamailio-2] type=peer host=1.1.1.1 fromdomain=1.1.1.1 context=outgoing insecure=invite,port directmedia=nonat qualify=yes disallow=all allow=all
and this is the sip trace for the register
U 2013/09/13 16:54:50.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000@127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: sip:102@172.16.0.112:5060. Expires: 120. .
U 2013/09/13 16:54:54.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000@127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: sip:102@172.16.0.112:5060. Expires: 120.
and I have this for my Asterisk binds in kamamilio.cfg
#!ifdef WITH_ASTERISK asterisk.bindip = "172.16.0.110" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "172.16.0.112" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif
Hi,
Could be something network related, or firewall? I know you stated phones could register directly to Asterisk but if nothing is showing in Asterisk SIP log, then the requests are never getting to it.
Was your trace performed on Kamailio server or Asterisk?
Cheers,
Charles
On 13 September 2013 17:12, imperium broadcast <imperium.broadcast@gmail.com
wrote:
Hi all I have been trying to figure out where Im going wrong with following this guide: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
The only thing that is different is I am using Asterisk 1.8.22.0
Kamailio works fine and I am able to register a phone to it but the forward register fails to register with Asterisk.
Its like Asterisk doesn't see the request even running sip debug on the Asterisk Console it doesn't show the register attempts. I have to use ngrep to see any thing.
Realtime works fine when I register the phone on Asterisk.
Although I can get round the register using AGI I get the same issue when passing a call to asterisk, Asterisk doesn't see the invite.
Am I missing something so obvious :-/
This is what I have in sip.conf "IP change to protect the innocent!"
[kamailio-2] type=peer host=1.1.1.1 fromdomain=1.1.1.1 context=outgoing insecure=invite,port directmedia=nonat qualify=yes disallow=all allow=all
and this is the sip trace for the register
U 2013/09/13 16:54:50.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000@127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: sip:102@172.16.0.112:5060. Expires: 120. .
U 2013/09/13 16:54:54.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000@127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: sip:102@172.16.0.112:5060. Expires: 120.
and I have this for my Asterisk binds in kamamilio.cfg
#!ifdef WITH_ASTERISK asterisk.bindip = "172.16.0.110" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "172.16.0.112" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif
-- Regards Mick
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Charles, the sip trace was carried out on the asterisk server. I have also disable the fw on both servers. I suspect it might be network issue but I have no idea what. I have also set up another server running asterisk 11 same problem so I know its not asterisk 1.8. I get no errors on either sever and logs don't show any issues. Its just an odd issue thats got me baffled.
Regards Mick On 14 Sep 2013 09:08, "Charles Chance" charles.chance@sipcentric.com wrote:
Hi,
Could be something network related, or firewall? I know you stated phones could register directly to Asterisk but if nothing is showing in Asterisk SIP log, then the requests are never getting to it.
Was your trace performed on Kamailio server or Asterisk?
Cheers,
Charles
On 13 September 2013 17:12, imperium broadcast < imperium.broadcast@gmail.com> wrote:
Hi all I have been trying to figure out where Im going wrong with following this guide:
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
The only thing that is different is I am using Asterisk 1.8.22.0
Kamailio works fine and I am able to register a phone to it but the forward register fails to register with Asterisk.
Its like Asterisk doesn't see the request even running sip debug on the Asterisk Console it doesn't show the register attempts. I have to use ngrep to see any thing.
Realtime works fine when I register the phone on Asterisk.
Although I can get round the register using AGI I get the same issue when passing a call to asterisk, Asterisk doesn't see the invite.
Am I missing something so obvious :-/
This is what I have in sip.conf "IP change to protect the innocent!"
[kamailio-2] type=peer host=1.1.1.1 fromdomain=1.1.1.1 context=outgoing insecure=invite,port directmedia=nonat qualify=yes disallow=all allow=all
and this is the sip trace for the register
U 2013/09/13 16:54:50.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000@127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: sip:102@172.16.0.112:5060. Expires: 120. .
U 2013/09/13 16:54:54.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000@127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: sip:102@172.16.0.112:5060. Expires: 120.
and I have this for my Asterisk binds in kamamilio.cfg
#!ifdef WITH_ASTERISK asterisk.bindip = "172.16.0.110" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "172.16.0.112" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif
-- Regards Mick
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
www.sipcentric.com
Follow us on twitter @sipcentric http://twitter.com/sipcentric
Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Mick,
What is the output of the following commands on Asterisk machine?
netstat -lnp | grep asterisk iptables -L sestatus
Regards,
Charles
On 14 September 2013 12:43, imperium broadcast <imperium.broadcast@gmail.com
wrote:
Hi Charles, the sip trace was carried out on the asterisk server. I have also disable the fw on both servers. I suspect it might be network issue but I have no idea what. I have also set up another server running asterisk 11 same problem so I know its not asterisk 1.8. I get no errors on either sever and logs don't show any issues. Its just an odd issue thats got me baffled.
Regards Mick On 14 Sep 2013 09:08, "Charles Chance" charles.chance@sipcentric.com wrote:
Hi,
Could be something network related, or firewall? I know you stated phones could register directly to Asterisk but if nothing is showing in Asterisk SIP log, then the requests are never getting to it.
Was your trace performed on Kamailio server or Asterisk?
Cheers,
Charles
On 13 September 2013 17:12, imperium broadcast < imperium.broadcast@gmail.com> wrote:
Hi all I have been trying to figure out where Im going wrong with following this guide:
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
The only thing that is different is I am using Asterisk 1.8.22.0
Kamailio works fine and I am able to register a phone to it but the forward register fails to register with Asterisk.
Its like Asterisk doesn't see the request even running sip debug on the Asterisk Console it doesn't show the register attempts. I have to use ngrep to see any thing.
Realtime works fine when I register the phone on Asterisk.
Although I can get round the register using AGI I get the same issue when passing a call to asterisk, Asterisk doesn't see the invite.
Am I missing something so obvious :-/
This is what I have in sip.conf "IP change to protect the innocent!"
[kamailio-2] type=peer host=1.1.1.1 fromdomain=1.1.1.1 context=outgoing insecure=invite,port directmedia=nonat qualify=yes disallow=all allow=all
and this is the sip trace for the register
U 2013/09/13 16:54:50.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000@127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: sip:102@172.16.0.112:5060. Expires: 120. .
U 2013/09/13 16:54:54.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000@127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: sip:102@172.16.0.112:5060. Expires: 120.
and I have this for my Asterisk binds in kamamilio.cfg
#!ifdef WITH_ASTERISK asterisk.bindip = "172.16.0.110" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "172.16.0.112" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif
-- Regards Mick
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
www.sipcentric.com
Follow us on twitter @sipcentric http://twitter.com/sipcentric
Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Another thing, you do have bindport=5080 in your sip.conf, don't you?
Regards,
Charles
On 14 September 2013 13:42, Charles Chance charles.chance@sipcentric.comwrote:
Hi Mick,
What is the output of the following commands on Asterisk machine?
netstat -lnp | grep asterisk iptables -L sestatus
Regards,
Charles
On 14 September 2013 12:43, imperium broadcast < imperium.broadcast@gmail.com> wrote:
Hi Charles, the sip trace was carried out on the asterisk server. I have also disable the fw on both servers. I suspect it might be network issue but I have no idea what. I have also set up another server running asterisk 11 same problem so I know its not asterisk 1.8. I get no errors on either sever and logs don't show any issues. Its just an odd issue thats got me baffled.
Regards Mick On 14 Sep 2013 09:08, "Charles Chance" charles.chance@sipcentric.com wrote:
Hi,
Could be something network related, or firewall? I know you stated phones could register directly to Asterisk but if nothing is showing in Asterisk SIP log, then the requests are never getting to it.
Was your trace performed on Kamailio server or Asterisk?
Cheers,
Charles
On 13 September 2013 17:12, imperium broadcast < imperium.broadcast@gmail.com> wrote:
Hi all I have been trying to figure out where Im going wrong with following this guide:
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
The only thing that is different is I am using Asterisk 1.8.22.0
Kamailio works fine and I am able to register a phone to it but the forward register fails to register with Asterisk.
Its like Asterisk doesn't see the request even running sip debug on the Asterisk Console it doesn't show the register attempts. I have to use ngrep to see any thing.
Realtime works fine when I register the phone on Asterisk.
Although I can get round the register using AGI I get the same issue when passing a call to asterisk, Asterisk doesn't see the invite.
Am I missing something so obvious :-/
This is what I have in sip.conf "IP change to protect the innocent!"
[kamailio-2] type=peer host=1.1.1.1 fromdomain=1.1.1.1 context=outgoing insecure=invite,port directmedia=nonat qualify=yes disallow=all allow=all
and this is the sip trace for the register
U 2013/09/13 16:54:50.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000@127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: sip:102@172.16.0.112:5060. Expires: 120. .
U 2013/09/13 16:54:54.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000@127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: sip:102@172.16.0.112:5060. Expires: 120.
and I have this for my Asterisk binds in kamamilio.cfg
#!ifdef WITH_ASTERISK asterisk.bindip = "172.16.0.110" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "172.16.0.112" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif
-- Regards Mick
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
www.sipcentric.com
Follow us on twitter @sipcentric http://twitter.com/sipcentric
Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Charles, yes I have asterisk bindport set to 5080. Here is the outputs for those commands. Asterisk Server: root@testsvr001:/etc/asterisk# netstat -lnp | grep asterisk tcp 0 0 0.0.0.0:2000 0.0.0.0:* LISTEN 1572/asterisk udp 0 0 0.0.0.0:5000 0.0.0.0:* 1572/asterisk udp 0 0 0.0.0.0:4520 0.0.0.0:* 1572/asterisk udp 0 0 0.0.0.0:5080 0.0.0.0:* 1572/asterisk udp 0 0 0.0.0.0:4569 0.0.0.0:* 1572/asterisk unix 2 [ ACC ] STREAM LISTENING 203576 1572/asterisk /var/run/asterisk/asterisk.ctl
root@testsvr001:/etc/asterisk#
root@testsvr001:/etc/asterisk# iptables -L Chain INPUT (policy ACCEPT) target prot opt source destination
Chain FORWARD (policy ACCEPT) target prot opt source destination
Chain OUTPUT (policy ACCEPT) target prot opt source destination
root@testsvr001:/etc/asterisk# sestatus SELinux status: disabled
Ran the commands on the Kamailio as well just to see:
root@VSwitch001:~# netstat -lnp | grep kamailio tcp 0 0 1.1.1.1:5060 0.0.0.0:* LISTEN 8944/kamailio tcp 0 0 172.16.0.112:5060 0.0.0.0:* LISTEN 8944/kamailio tcp 0 0 127.0.0.1:5060 0.0.0.0:* LISTEN 8944/kamailio udp 0 0 1.1.1.1:5060 0.0.0.0:* 8922/kamailio udp 0 0 172.16.0.112:5060 0.0.0.0:* 8922/kamailio udp 0 0 127.0.0.1:5060 0.0.0.0:* 8922/kamailio raw 0 0 0.0.0.0:255 0.0.0.0:* 7 8922/kamailio unix 2 [ ACC ] STREAM LISTENING 38061 8938/kamailio /tmp/kamailio_ctl
root@VSwitch001:/usr/local/etc/kamailio# iptables -L Chain INPUT (policy ACCEPT) target prot opt source destination
Chain FORWARD (policy ACCEPT) target prot opt source destination
Chain OUTPUT (policy ACCEPT) target prot opt source destination
root@VSwitch001:/usr/local/etc/kamailio# sestatus SELinux status: disabled
Thanks for your help.
Regards Mick
On 14 September 2013 13:52, Charles Chance charles.chance@sipcentric.comwrote:
Another thing, you do have bindport=5080 in your sip.conf, don't you?
Regards,
Charles
On 14 September 2013 13:42, Charles Chance charles.chance@sipcentric.comwrote:
Hi Mick,
What is the output of the following commands on Asterisk machine?
netstat -lnp | grep asterisk iptables -L sestatus
Regards,
Charles
On 14 September 2013 12:43, imperium broadcast < imperium.broadcast@gmail.com> wrote:
Hi Charles, the sip trace was carried out on the asterisk server. I have also disable the fw on both servers. I suspect it might be network issue but I have no idea what. I have also set up another server running asterisk 11 same problem so I know its not asterisk 1.8. I get no errors on either sever and logs don't show any issues. Its just an odd issue thats got me baffled.
Regards Mick On 14 Sep 2013 09:08, "Charles Chance" charles.chance@sipcentric.com wrote:
Hi,
Could be something network related, or firewall? I know you stated phones could register directly to Asterisk but if nothing is showing in Asterisk SIP log, then the requests are never getting to it.
Was your trace performed on Kamailio server or Asterisk?
Cheers,
Charles
On 13 September 2013 17:12, imperium broadcast < imperium.broadcast@gmail.com> wrote:
Hi all I have been trying to figure out where Im going wrong with following this guide:
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
The only thing that is different is I am using Asterisk 1.8.22.0
Kamailio works fine and I am able to register a phone to it but the forward register fails to register with Asterisk.
Its like Asterisk doesn't see the request even running sip debug on the Asterisk Console it doesn't show the register attempts. I have to use ngrep to see any thing.
Realtime works fine when I register the phone on Asterisk.
Although I can get round the register using AGI I get the same issue when passing a call to asterisk, Asterisk doesn't see the invite.
Am I missing something so obvious :-/
This is what I have in sip.conf "IP change to protect the innocent!"
[kamailio-2] type=peer host=1.1.1.1 fromdomain=1.1.1.1 context=outgoing insecure=invite,port directmedia=nonat qualify=yes disallow=all allow=all
and this is the sip trace for the register
U 2013/09/13 16:54:50.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000@127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: sip:102@172.16.0.112:5060. Expires: 120. .
U 2013/09/13 16:54:54.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000@127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: sip:102@172.16.0.112:5060. Expires: 120.
and I have this for my Asterisk binds in kamamilio.cfg
#!ifdef WITH_ASTERISK asterisk.bindip = "172.16.0.110" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "172.16.0.112" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif
-- Regards Mick
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
www.sipcentric.com
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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Had a bit of breakthrough \o/
By setting the asterisk bindip to the external IP of the asterisk server it registers with the asterisk, I am unable to make a test call yet but its jump in the right direction.
Regards Mick
On 14 September 2013 13:52, Charles Chance charles.chance@sipcentric.comwrote:
Another thing, you do have bindport=5080 in your sip.conf, don't you?
Regards,
Charles
On 14 September 2013 13:42, Charles Chance charles.chance@sipcentric.comwrote:
Hi Mick,
What is the output of the following commands on Asterisk machine?
netstat -lnp | grep asterisk iptables -L sestatus
Regards,
Charles
On 14 September 2013 12:43, imperium broadcast < imperium.broadcast@gmail.com> wrote:
Hi Charles, the sip trace was carried out on the asterisk server. I have also disable the fw on both servers. I suspect it might be network issue but I have no idea what. I have also set up another server running asterisk 11 same problem so I know its not asterisk 1.8. I get no errors on either sever and logs don't show any issues. Its just an odd issue thats got me baffled.
Regards Mick On 14 Sep 2013 09:08, "Charles Chance" charles.chance@sipcentric.com wrote:
Hi,
Could be something network related, or firewall? I know you stated phones could register directly to Asterisk but if nothing is showing in Asterisk SIP log, then the requests are never getting to it.
Was your trace performed on Kamailio server or Asterisk?
Cheers,
Charles
On 13 September 2013 17:12, imperium broadcast < imperium.broadcast@gmail.com> wrote:
Hi all I have been trying to figure out where Im going wrong with following this guide:
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
The only thing that is different is I am using Asterisk 1.8.22.0
Kamailio works fine and I am able to register a phone to it but the forward register fails to register with Asterisk.
Its like Asterisk doesn't see the request even running sip debug on the Asterisk Console it doesn't show the register attempts. I have to use ngrep to see any thing.
Realtime works fine when I register the phone on Asterisk.
Although I can get round the register using AGI I get the same issue when passing a call to asterisk, Asterisk doesn't see the invite.
Am I missing something so obvious :-/
This is what I have in sip.conf "IP change to protect the innocent!"
[kamailio-2] type=peer host=1.1.1.1 fromdomain=1.1.1.1 context=outgoing insecure=invite,port directmedia=nonat qualify=yes disallow=all allow=all
and this is the sip trace for the register
U 2013/09/13 16:54:50.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000@127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: sip:102@172.16.0.112:5060. Expires: 120. .
U 2013/09/13 16:54:54.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000@127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: sip:102@172.16.0.112:5060. Expires: 120.
and I have this for my Asterisk binds in kamamilio.cfg
#!ifdef WITH_ASTERISK asterisk.bindip = "172.16.0.110" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "172.16.0.112" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif
-- Regards Mick
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
www.sipcentric.com
Follow us on twitter @sipcentric http://twitter.com/sipcentric
Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
www.sipcentric.com
Follow us on twitter @sipcentric http://twitter.com/sipcentric
Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Charles Chance -
imperium broadcast