SIP Poke NO ANSWER

Asterisk Asterisk Support
1

Im hoping someone can shed some light on a very frustrating problem i am having.

I have Asterisk 1.2.14 and am having issues with my SNOM Phones staying registered.

Basically the SNOM phones on the internal network work fine, however we have remote sites connected through a VPN that are the ones having problems.

No my understanding of the sip_poke_noanswer is that Asterisk is sending a SIP Options Message tothe phone but not getting any response. Connectivity between the networks are fine, an average reponse ping is around 30-45 ms. There are no firewalls in the way as it is Internal to the Network via our VPN Routers.

When i register on the SNOM Phone it shows as registered in the asterisk CLI, but after a few seconds, it gives me the SIP POKE error.

Here is a copy of my SIP trace from my snom phone :-

Received from udp:192.168.32.222:5060 at 19/3/2007 16:52:29:460 (531 bytes):

OPTIONS sip:DKHome@192.168.22.13:2060;line=jjbaws4h SIP/2.0
Via: SIP/2.0/UDP 203.206.167.144:5060;branch=z9hG4bK48eafda6;rport
From: “asterisk” sip:asterisk@192.168.32.222;tag=as3d94fe91
To: sip:DKHome@192.168.22.13:2060;line=jjbaws4h
Contact: sip:asterisk@203.206.167.144
Call-ID: 0fd5e8ce0239080c3e54d1f05c691aca@192.168.32.222
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 19 Mar 2007 05:49:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

Sent to udp:203.206.167.144:5060 at 19/3/2007 16:52:29:460 (609 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.206.167.144:5060;branch=z9hG4bK48eafda6;rport=5060
From: “asterisk” sip:asterisk@192.168.32.222;tag=as3d94fe91
To: sip:DKHome@192.168.22.13:2060;line=jjbaws4h
Call-ID: 0fd5e8ce0239080c3e54d1f05c691aca@192.168.32.222
CSeq: 102 OPTIONS
Contact: sip:DKHome@192.168.22.13:2060;line=jjbaws4h;flow-id=1
User-Agent: snom320/6.5.2
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Content-Length: 0

These events are continually repeated.

Anyone can shed any light on what may be causing this??

Thanks in advance.

Daniel.

2

Here is a SIP trace on my Snom Phone. I think i can see what the problem is and am hoping someone can confirm and point me in the right direction

Received from udp:192.168.32.222:5060 at 20/3/2007 17:08:12:230 (531 bytes):

OPTIONS sip:DKHome@192.168.22.13:2051;line=jxk860uv SIP/2.0
Via: SIP/2.0/UDP 203.206.167.144:5060;branch=z9hG4bK7376f581;rport
From: “asterisk” sip:asterisk@192.168.32.222;tag=as2e8672e4
To: sip:DKHome@192.168.22.13:2051;line=jxk860uv
Contact: sip:asterisk@XXX.XXX.XXX.XXX
Call-ID: 41f551087779977201f3575f67807434@192.168.32.222
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 20 Mar 2007 06:04:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

Sent to udp:XXX.XXX.XXX.XXX:5060 at 20/3/2007 17:08:12:230 (609 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.206.167.144:5060;branch=z9hG4bK7376f581;rport=5060
From: “asterisk” sip:asterisk@192.168.32.222;tag=as2e8672e4
To: sip:DKHome@192.168.22.13:2051;line=jxk860uv
Call-ID: 41f551087779977201f3575f67807434@192.168.32.222
CSeq: 102 OPTIONS
Contact: sip:DKHome@192.168.22.13:2051;line=jxk860uv;flow-id=1
User-Agent: snom320/6.5.2
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Content-Length: 0

Its look like it is receiving the invite from the internal IP address 192.168.32.222 but when it goes to send it back out, its sending it to WAN IP address XXX.XXX.XXX.XXX

I assume the problem is with the line

Contact: sip:asterisk@XXX.XXX.XXX.XXX and the IP address of the WAN interface.

Can anyone shed some light on how to fix this …

Thanks.

3

[quote=“danielkelly”]I assume the problem is with the line

Contact: sip:asterisk@XXX.XXX.XXX.XXX and the IP address of the WAN interface.[/quote]

This must be right. Question is why Asterisk thinks it’s necessary to use that contact. Have you enabled NAT traversal? If all your end points are private - either LAN or VPN, you should not need to.