Hi all,
We have a customer with SIP trunks from a small carrier. They've worked fine for months until a week ago. The carrier made a change for better security and now outbound calls will remain up for 18 seconds then drop. VERY consistent. We get a Bye message from carrier and we drop. Inbound calls work fine.
One segment of the trace looks like this (with XX to mask site specifics) - note the last line that says "no element found". Any idea what that means? We get a lot of "OK" messages - and I wonder if we're not getting a message and carrier wants an ACK. THANKS
164526mS SIP Call Rx: 19
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.236.XXX.68:5060;rport=5060;branch=z9hG4bK3b48ab5bee464c62a3e8d70a1130e035
From: "XXXXXX6758" <sip:XXXXXX6758@XX.XX.47.63>;tag=3f6cd5a49cc28d8a
To: <sip:9XXXXXX1583@XX.XX.47.63>;tag=12D2C4-F76
Date: Sat, 01 Jan 2000 00:20:54 GMT
Call-ID: b0f6240c9c895a6093166a0851db00c2
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1880277584 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:57296143#9XXXXXX1583@XX.XX.47.62:5060>
Record-Route: <sip:XX.XX.47.63;lr=on;ftag=3f6cd5a49cc28d8a;did=268.d68a14b5>
Supported: replaces
Session-Expires: 2400;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 255
v=0
o=CiscoSystemsSIP-GW-UserAgent 4850 4192 IN IP4 XX.XX.47.62
s=SIP Call
c=IN IP4 XX.XX.47.62
t=0 0
m=audio 18062 RTP/AVP 18 101
c=IN IP4 XX.XX.47.62
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
164527mS Sip: 19.1020.0 2 SIPTrunk Endpoint(f521dc60) ProcessInboundSIPResponse calling CheckSessionExpiresField, session expires is 2400 refresher_is_ipo 1
164528mS PRN: MZ stubs sip_cbk_fetchTxn no element found
We have a customer with SIP trunks from a small carrier. They've worked fine for months until a week ago. The carrier made a change for better security and now outbound calls will remain up for 18 seconds then drop. VERY consistent. We get a Bye message from carrier and we drop. Inbound calls work fine.
One segment of the trace looks like this (with XX to mask site specifics) - note the last line that says "no element found". Any idea what that means? We get a lot of "OK" messages - and I wonder if we're not getting a message and carrier wants an ACK. THANKS
164526mS SIP Call Rx: 19
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.236.XXX.68:5060;rport=5060;branch=z9hG4bK3b48ab5bee464c62a3e8d70a1130e035
From: "XXXXXX6758" <sip:XXXXXX6758@XX.XX.47.63>;tag=3f6cd5a49cc28d8a
To: <sip:9XXXXXX1583@XX.XX.47.63>;tag=12D2C4-F76
Date: Sat, 01 Jan 2000 00:20:54 GMT
Call-ID: b0f6240c9c895a6093166a0851db00c2
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1880277584 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:57296143#9XXXXXX1583@XX.XX.47.62:5060>
Record-Route: <sip:XX.XX.47.63;lr=on;ftag=3f6cd5a49cc28d8a;did=268.d68a14b5>
Supported: replaces
Session-Expires: 2400;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 255
v=0
o=CiscoSystemsSIP-GW-UserAgent 4850 4192 IN IP4 XX.XX.47.62
s=SIP Call
c=IN IP4 XX.XX.47.62
t=0 0
m=audio 18062 RTP/AVP 18 101
c=IN IP4 XX.XX.47.62
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
164527mS Sip: 19.1020.0 2 SIPTrunk Endpoint(f521dc60) ProcessInboundSIPResponse calling CheckSessionExpiresField, session expires is 2400 refresher_is_ipo 1
164528mS PRN: MZ stubs sip_cbk_fetchTxn no element found