07-25-2018 06:37 AM
Hi,
I need to add in "pinger" in the From field of an outbound Options Ping message. The SIP profile I use is below and works fine in https://cway.cisco.com/tools/SipProfileTest/
voice class sip-profiles 20
request OPTIONS sip-header From modify "<sip:(.*)>" "<sip:pinger@\1>"
I suspect it might have something to do with the fact that you don't see outboudn SIP Options PIng messages using debug ccsip messages maybe?
Has anyone actually been able to manipulate a SIP Options PINg message successfully before?
Solved! Go to Solution.
07-25-2018 12:29 PM
I jsut found the answer!!
The SIP profile doesn't work when using options-keepalive profiles on the dialpeer
I changed the dial-peer from this
dial-peer voice 12 voip
translation-profile outgoing OUT
session protocol sipv2
session transport udp
session server-group 1
destination e164-pattern-map 200
voice-class sip profiles 20
voice-class sip options-keepalive profile 10
voice-class sip bind control source-interface Port-channel1.402
voice-class sip bind media source-interface Port-channel1.402
dtmf-relay rtp-nte
codec g711alaw
no vad
to this & now it works!!
dial-peer voice 12 voip
translation-profile outgoing OUT
session protocol sipv2
session transport udp
destination target ipv4:X.X.X.X
destination e164-pattern-map 200
voice-class sip profiles 20
voice-class sip options-keepalive
voice-class sip bind control source-interface Port-channel1.402
voice-class sip bind media source-interface Port-channel1.402
dtmf-relay rtp-nte
codec g711alaw
no vad
07-25-2018 11:02 AM
The profiles usually work pretty well, but sometimes the recipient has requirements that you don't know about. The recent changes in debug displays (not including all messages) are supposed to clean things up for most situations, but if you want to see everything, do the usual 'debug ccsip messages' and then add 'debug ccsip non-call' and you should see the rest of them.
Mary Beth
07-25-2018 12:06 PM
Thanks - didn't know that "non call" debug. Helpful
So now that I can see the outbound Options message - the profile definitely isn't working......even though it should as per Cisco profile test site.
Below is Options message, profile & dialpeer....anyone see any reason why this should not work?
Sent:
OPTIONS sip:172.16.87.73:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.119.101:5060;branch=z9hG4bK11676AF7
From: <sip:172.16.119.101>;tag=6277BAC2-26E9
To: <sip:172.16.87.73>
Date: Wed, 25 Jul 2018 18:52:07 GMT
Call-ID: A9A7561F-8F7211E8-959293FF-3948EF15@172.16.119.101
User-Agent: Cisco-SIPGateway/IOS-16.7.1
Max-Forwards: 70
CSeq: 101 OPTIONS
Contact: <sip:172.16.119.101:5060>
Content-Length: 0
voice class sip-profiles 20
request OPTIONS sip-header From modify "<sip:(.*)>" "<sip:pinger@\1>"
dial-peer voice 12 voip
translation-profile outgoing OUT
session protocol sipv2
session transport udp
session server-group 1
destination e164-pattern-map 200
voice-class sip profiles 20
voice-class sip options-keepalive profile 10
voice-class sip bind control source-interface Port-channel1.402
voice-class sip bind media source-interface Port-channel1.402
dtmf-relay rtp-nte
codec g711alaw
no vad
07-25-2018 12:29 PM
I jsut found the answer!!
The SIP profile doesn't work when using options-keepalive profiles on the dialpeer
I changed the dial-peer from this
dial-peer voice 12 voip
translation-profile outgoing OUT
session protocol sipv2
session transport udp
session server-group 1
destination e164-pattern-map 200
voice-class sip profiles 20
voice-class sip options-keepalive profile 10
voice-class sip bind control source-interface Port-channel1.402
voice-class sip bind media source-interface Port-channel1.402
dtmf-relay rtp-nte
codec g711alaw
no vad
to this & now it works!!
dial-peer voice 12 voip
translation-profile outgoing OUT
session protocol sipv2
session transport udp
destination target ipv4:X.X.X.X
destination e164-pattern-map 200
voice-class sip profiles 20
voice-class sip options-keepalive
voice-class sip bind control source-interface Port-channel1.402
voice-class sip bind media source-interface Port-channel1.402
dtmf-relay rtp-nte
codec g711alaw
no vad
07-25-2018 12:33 PM
07-25-2018 12:31 PM
I would suspect something to do with the fact that this is not exactly a call, following that dial peer out of the box. You could try applying that profile globally, under voice service voip, sip area...
Mary Beth
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