I have a service provider sending me SIP OPTIONS pings as a keep alive for our SIP Trunk, but the responses back from my Netvanta 3140 are 501 Service Not Implemented. Due to this, we cannot keep the inbound SIP Trunk up and running and the provider would prefer not to turn the OPTIONS messages off.
Rx: UDP src=64.140.194.204:5060 dst=74.220.224.66:5060
11:45:25.966 SIP.STACK MSG OPTIONS sip:McCrillis@74.220.224.66:5060;transport=udp SIP/2.0
11:45:25.966 SIP.STACK MSG Via: SIP/2.0/UDP 64.140.194.204:5060;branch=z9hG4bKnbaudc3030sifm6m94q0.1
11:45:25.966 SIP.STACK MSG Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name
11:45:25.966 SIP.STACK MSG Max-Forwards: 69
11:45:25.966 SIP.STACK MSG Call-ID: F3BDFE2E@192.168.220.25
11:45:25.966 SIP.STACK MSG From: <sip:McCrillis@64.140.194.204:5060;transport=udp>;tag=192.168.220.25+1+247fb3b+627c111
11:45:25.966 SIP.STACK MSG CSeq: 802406716 OPTIONS
11:45:25.966 SIP.STACK MSG Organization: BRMSWA
11:45:25.966 SIP.STACK MSG Supported: resource-priority, siprec, 100rel
11:45:25.967 SIP.STACK MSG Content-Length: 0
11:45:25.967 SIP.STACK MSG Contact: <sip:McCrillis@64.140.194.204:5060;transport=udp>
11:45:25.967 SIP.STACK MSG To: <sip:McCrillis@74.220.224.66>
11:45:25.967 SIP.STACK MSG
11:45:25.969 SIP.STACK MSG Tx: UDP src=74.220.224.66:5060 dst=64.140.194.204:5060
11:45:25.969 SIP.STACK MSG SIP/2.0 501 Not Implemented
11:45:25.969 SIP.STACK MSG From: <sip:McCrillis@64.140.194.204:5060;transport=udp>;tag=192.168.220.25+1+247fb3b+627c111
11:45:25.969 SIP.STACK MSG To: <sip:McCrillis@74.220.224.66>;tag=529df600-7f000001-13c4-18400-37512323-18400
11:45:25.969 SIP.STACK MSG Call-ID: F3BDFE2E@192.168.220.25
11:45:25.970 SIP.STACK MSG CSeq: 802406716 OPTIONS
11:45:25.970 SIP.STACK MSG Via: SIP/2.0/UDP 64.140.194.204:5060;branch=z9hG4bKnbaudc3030sifm6m94q0.1
11:45:25.970 SIP.STACK MSG Content-Length: 0
This will fix your problem:
Configuring a Virtual User in AOS to respond to SIP OPTIONS over a SIP Trunk
-Mark
This will fix your problem:
Configuring a Virtual User in AOS to respond to SIP OPTIONS over a SIP Trunk
-Mark
Thanks Mark, worked like a charm!
anytime!
Glad that fixed your problem. Have a great day and thanks for using the Support Forums!
-Mark
Hello Mark,
Thanks for your post. I read the article and was able to respond to the SIP options from one ITSP who sent and sip-identity (in this case metaswitch) along with the SIP options request.
-----------------------------------------------------------
Session Initiation Protocol (OPTIONS)
Request-Line: OPTIONS sip:metaswitch@72.139.85.190:5060 SIP/2.0
Message Header
Via: SIP/2.0/UDP 74.216.209.100:5060;branch=z9hG4bK+0ce08e995e679d32574e3f5e6137b5e11+sip+5+a9a243ff
From: <sip:metaswitch@74.216.209.100>;tag=74.216.209.100+5+f42f3f+4db1f56b
Content-Length: 0
Supported: resource-priority, siprec, 100rel
To: <sip:metaswitch@72.139.85.190>
Contact: <sip:964f89d29f8e6ec84bd45290700216d1@74.216.209.100>
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name
Max-Forwards: 69
Call-ID: 0gQAAC8WAAACBAAALxYAAESsqG5MHAc0lSlam9/NaNDB101hXMPXzfPUg4SoltE0@74.216.209.100
CSeq: 984616213 OPTIONS
Organization: Metaswitch Networks
Accept: application/sdp, application/dtmf-relay
-----------------------------------------------------------
However, with another provider, they don't send an identity (ie there is no identify between sip: and the @ sign) So the Netvanta only responds with 501 Not Implemented. I've tried the sip-identity ping too but with no luck. Is there a way to have the Netvanta (3430 in my case) just respond to all SIP Options requests on a trunk?
-----------------------------------------------------------
Session Initiation Protocol (OPTIONS)
Request-Line: OPTIONS sip:customer12.lab.internetvoice.ca:5060;maddr=72.139.85.190 SIP/2.0
Message Header
Via: SIP/2.0/UDP 69.158.195.73:5060;branch=z9hG4bKg70sls307oeo790bvdf0
Call-ID: a9a28f318b0aa41e4378f778ded4a89f000qh60@69.158.195.73
To: sip:ping@customer12.lab.internetvoice.ca
From: <sip:ping@69.158.195.73>;tag=a8dfbb0014bba800f257b7caaeddad18000qh60
Max-Forwards: 0
CSeq: 7437 OPTIONS
-----------------------------------------------------------
You assistance is greatly appreicated.
Thanks,
Pat
Pat,
I have not run into this. You could try creating a voice user and then testing some things like trying customer12 or ping in the sip-identity command below replacing XXXXX. Or try different variants of customer12.lab.internetvoice.ca
(config)#voice user 1000
(config-1000)#sip-identity XXXXX T01
Let me know if that works or not. I am also pinging some other people to see what they think.
-Mark