Calls disconnect from provider using Cisco

ja133 · February 22, 2012, 11:53pm

Hi,
I am having a problem with a certain customer interconnecting with me - they are using a Cisco gateway. Some calls will drop after 8-18 minutes. I captured the last few moments of the call below. On my end I am using asterisk 1.6.2.7 with directmedia=yes and directrtpsetup=yes. When I turn off directmedia and my server handles the media, this issue will NEVER occur. In other words, it only occurs when I pass the media from the carrier to their switch. If anyone can assist, I would greatly appreciate it.

Thanks so much

<--- SIP read from UDP:54.35.24.88:5060 --->
INVITE sip:17182109675@184.55.33.74 SIP/2.0
Record-Route: <sip:54.35.24.88;lr=on;ftag=sansay90655509rdb1044;did=882.5b7b3a56>
Via: SIP/2.0/UDP 54.35.24.88;branch=z9hG4bKc59b.cabe8e86.0
Via: SIP/2.0/UDP 208.85.248.41:5060;branch=z9hG4bK1sansay90655509rdb1044
Record-Route: <sip:sansay90655509rdb1044@208.85.248.41:5060;lr;transport=udp>
To: <sip:17182109675@54.35.24.88>
From: <sip:7174537240@208.85.248.41>;tag=sansay90655509rdb1044
Call-ID: 421af690585eadc713c41980eb639f97a8649b5cb1c2f0eab08-0119-4162
CSeq: 1 INVITE
Contact: <sip:7174537240@208.85.248.41:5060>
Supported: replaces
P-Asserted-Identity: <sip:7174537240@208.85.248.41>
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
Max-Forwards: 13
Content-Type: application/sdp
Content-Length: 287

v=0
o=Sansay-VSXi 188 1 IN IP4 208.85.248.41
s=Session Controller
c=IN IP4 199.173.76.106
t=0 0
m=audio 53056 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: --- (16 headers 13 lines) ---
[2012-02-22 18:03:41] VERBOSE[1353] netsock.c:   == Using SIP RTP CoS mark 5
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Sending to 54.35.24.88 : 5060 (no NAT)
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Using INVITE request as basis request - 421af690585eadc713c41980eb639f97a8649b5cb1c2f0eab08-0119-4162
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Found peer 'ezcall-did' for '7174537240' from 54.35.24.88:5060
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Found RTP audio format 18
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Found RTP audio format 0
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Found RTP audio format 8
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Found RTP audio format 101
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Found audio description format G729 for ID 18
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Found audio description format PCMU for ID 0
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Found audio description format PCMA for ID 8
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Found audio description format telephone-event for ID 101
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Peer audio RTP is at port 199.173.76.106:53056
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Looking for 17182109675 in inbound-unlimited (domain 184.55.33.74)
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: list_route: hop: <sip:54.35.24.88;lr=on;ftag=sansay90655509rdb1044;did=882.5b7b3a56>
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: list_route: hop: <sip:sansay90655509rdb1044@208.85.248.41:5060;lr;transport=udp>
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: 
<--- Transmitting (no NAT) to 54.35.24.88:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 54.35.24.88;branch=z9hG4bKc59b.cabe8e86.0;received=54.35.24.88
Via: SIP/2.0/UDP 208.85.248.41:5060;branch=z9hG4bK1sansay90655509rdb1044
Record-Route: <sip:54.35.24.88;lr=on;ftag=sansay90655509rdb1044;did=882.5b7b3a56>
Record-Route: <sip:sansay90655509rdb1044@208.85.248.41:5060;lr;transport=udp>
From: <sip:7174537240@208.85.248.41>;tag=sansay90655509rdb1044
To: <sip:17182109675@54.35.24.88>
Call-ID: 421af690585eadc713c41980eb639f97a8649b5cb1c2f0eab08-0119-4162
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:17182109675@184.55.33.74>
Content-Length: 0


<------------>
[2012-02-22 18:03:41] VERBOSE[6215] pbx.c:     -- Executing [17182109675@inbound-unlimited:1] Set("SIP/ezcall-did-00000819", "GROUP()=channel-a2-2948905451") in new stack
[2012-02-22 18:03:41] VERBOSE[6215] pbx.c:     -- Executing [17182109675@inbound-unlimited:2] GotoIf("SIP/ezcall-did-00000819", "0?20") in new stack
[2012-02-22 18:03:41] VERBOSE[6215] pbx.c:     -- Executing [17182109675@inbound-unlimited:3] Dial("SIP/ezcall-did-00000819", "SIP/2948905451/7182109675") in new stack
[2012-02-22 18:03:41] VERBOSE[6215] netsock.c:   == Using SIP RTP CoS mark 5
[2012-02-22 18:03:41] VERBOSE[6215] chan_sip.c: Audio is at 184.55.33.74 port 15688
[2012-02-22 18:03:41] VERBOSE[6215] chan_sip.c: Adding codec 0x100 (g729) to SDP
[2012-02-22 18:03:41] VERBOSE[6215] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-02-22 18:03:41] VERBOSE[6215] chan_sip.c: Reliably Transmitting (no NAT) to 65.35.85.122:5060:
INVITE sip:7182109675@65.35.85.122 SIP/2.0
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK59275998;rport
Max-Forwards: 70
From: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
To: <sip:7182109675@65.35.85.122>
Contact: <sip:7174537240@184.55.33.74>
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7
Date: Wed, 22 Feb 2012 23:03:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1356920958 1356920958 IN IP4 199.173.76.106
s=Asterisk PBX 1.6.2.7
c=IN IP4 199.173.76.106
t=0 0
m=audio 53056 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[2012-02-22 18:03:41] VERBOSE[6215] app_dial.c:     -- Called 2948905451/7182109675
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: 
<--- SIP read from UDP:65.35.85.122:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK59275998;rport
From: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
To: <sip:7182109675@65.35.85.122>
Date: Wed, 22 Feb 2012 23:03:39 GMT
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


<------------->
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: --- (10 headers 0 lines) ---
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: 
<--- SIP read from UDP:65.35.85.122:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK59275998;rport
From: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
To: <sip:7182109675@65.35.85.122>;tag=721A50-294
Date: Wed, 22 Feb 2012 23:03:39 GMT
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:7182109675@65.35.85.122:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 270

v=0
o=CiscoSystemsSIP-GW-UserAgent 5841 4320 IN IP4 65.35.85.122
s=SIP Call
c=IN IP4 65.35.85.122
t=0 0
m=audio 17588 RTP/AVP 18 101
c=IN IP4 65.35.85.122
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: --- (16 headers 12 lines) ---
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Found RTP audio format 18
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Found RTP audio format 101
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Found audio description format G729 for ID 18
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Found audio description format telephone-event for ID 101
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Peer audio RTP is at port 65.35.85.122:17588
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: list_route: hop: <sip:7182109675@65.35.85.122:5060>
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: set_destination: Parsing <sip:7182109675@65.35.85.122:5060> for address/port to send to
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: set_destination: set destination to 65.35.85.122, port 5060
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Transmitting (no NAT) to 65.35.85.122:5060:
ACK sip:7182109675@65.35.85.122:5060 SIP/2.0
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK30e18c78;rport
Max-Forwards: 70
From: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
To: <sip:7182109675@65.35.85.122>;tag=721A50-294
Contact: <sip:7174537240@184.55.33.74>
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.7
Content-Length: 0


---
[2012-02-22 18:03:41] VERBOSE[6215] app_dial.c:     -- SIP/2948905451-0000081a answered SIP/ezcall-did-00000819
[2012-02-22 18:03:41] VERBOSE[6215] chan_sip.c: Audio is at 184.55.33.74 port 18260
[2012-02-22 18:03:41] VERBOSE[6215] chan_sip.c: Adding codec 0x100 (g729) to SDP
[2012-02-22 18:03:41] VERBOSE[6215] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-02-22 18:03:41] VERBOSE[6215] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to 54.35.24.88:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.35.24.88;branch=z9hG4bKc59b.cabe8e86.0;received=54.35.24.88
Via: SIP/2.0/UDP 208.85.248.41:5060;branch=z9hG4bK1sansay90655509rdb1044
Record-Route: <sip:54.35.24.88;lr=on;ftag=sansay90655509rdb1044;did=882.5b7b3a56>
Record-Route: <sip:sansay90655509rdb1044@208.85.248.41:5060;lr;transport=udp>
From: <sip:7174537240@208.85.248.41>;tag=sansay90655509rdb1044
To: <sip:17182109675@54.35.24.88>;tag=as05430591
Call-ID: 421af690585eadc713c41980eb639f97a8649b5cb1c2f0eab08-0119-4162
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:17182109675@184.55.33.74>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 308590160 308590160 IN IP4 65.35.85.122
s=Asterisk PBX 1.6.2.7
c=IN IP4 65.35.85.122
t=0 0
m=audio 17588 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[2012-02-22 18:03:41] VERBOSE[6215] rtp.c:     -- Found common audio codec for native bridge g729
[2012-02-22 18:03:41] VERBOSE[6215] rtp.c:     -- Native bridging SIP/ezcall-did-00000819 and SIP/2948905451-0000081a
[2012-02-22 18:03:41] VERBOSE[6215] chan_sip.c: set_destination: Parsing <sip:7182109675@65.35.85.122:5060> for address/port to send to
[2012-02-22 18:03:41] VERBOSE[6215] chan_sip.c: set_destination: set destination to 65.35.85.122, port 5060
[2012-02-22 18:03:41] VERBOSE[6215] chan_sip.c: Audio is at 184.55.33.74 port 15688
[2012-02-22 18:03:41] VERBOSE[6215] chan_sip.c: Adding codec 0x100 (g729) to SDP
[2012-02-22 18:03:41] VERBOSE[6215] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-02-22 18:03:41] VERBOSE[6215] chan_sip.c: Reliably Transmitting (no NAT) to 65.35.85.122:5060:
INVITE sip:7182109675@65.35.85.122:5060 SIP/2.0
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK4d70c319;rport
Max-Forwards: 70
From: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
To: <sip:7182109675@65.35.85.122>;tag=721A50-294
Contact: <sip:7174537240@184.55.33.74>
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1356920958 1356920959 IN IP4 199.173.76.106
s=Asterisk PBX 1.6.2.7
c=IN IP4 199.173.76.106
t=0 0
m=audio 53056 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: 
<--- SIP read from UDP:65.35.85.122:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK4d70c319;rport
From: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
To: <sip:7182109675@65.35.85.122>;tag=721A50-294
Date: Wed, 22 Feb 2012 23:03:39 GMT
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


<------------->
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: --- (10 headers 0 lines) ---
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: 
<--- SIP read from UDP:65.35.85.122:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK4d70c319;rport
From: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
To: <sip:7182109675@65.35.85.122>;tag=721A50-294
Date: Wed, 22 Feb 2012 23:03:39 GMT
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:7182109675@65.35.85.122:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Length: 270

v=0
o=CiscoSystemsSIP-GW-UserAgent 5841 4320 IN IP4 65.35.85.122
s=SIP Call
c=IN IP4 65.35.85.122
t=0 0
m=audio 17588 RTP/AVP 18 101
c=IN IP4 65.35.85.122
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: --- (15 headers 12 lines) ---
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: set_destination: Parsing <sip:7182109675@65.35.85.122:5060> for address/port to send to
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: set_destination: set destination to 65.35.85.122, port 5060
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Transmitting (no NAT) to 65.35.85.122:5060:
ACK sip:7182109675@65.35.85.122:5060 SIP/2.0
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK04100c60;rport
Max-Forwards: 70
From: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
To: <sip:7182109675@65.35.85.122>;tag=721A50-294
Contact: <sip:7174537240@184.55.33.74>
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.7
Content-Length: 0


---
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: set_destination: Parsing <sip:7182109675@65.35.85.122:5060> for address/port to send to
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: set_destination: set destination to 65.35.85.122, port 5060
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Audio is at 184.55.33.74 port 15688
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Adding codec 0x100 (g729) to SDP
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Reliably Transmitting (no NAT) to 65.35.85.122:5060:
INVITE sip:7182109675@65.35.85.122:5060 SIP/2.0
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK433ed38b;rport
Max-Forwards: 70
From: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
To: <sip:7182109675@65.35.85.122>;tag=721A50-294
Contact: <sip:7174537240@184.55.33.74>
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.6.2.7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1356920958 1356920960 IN IP4 199.173.76.106
s=Asterisk PBX 1.6.2.7
c=IN IP4 199.173.76.106
t=0 0
m=audio 53056 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: 
<--- SIP read from UDP:54.35.24.88:5060 --->
ACK sip:17182109675@184.55.33.74 SIP/2.0
Via: SIP/2.0/UDP 54.35.24.88;branch=z9hG4bKc59b.cabe8e86.2
Via: SIP/2.0/UDP 208.85.248.41:5060;branch=z9hG4bK1sansay90655509rdb1044-200
To: <sip:17182109675@54.35.24.88>;tag=as05430591
From: <sip:7174537240@208.85.248.41>;tag=sansay90655509rdb1044
Call-ID: 421af690585eadc713c41980eb639f97a8649b5cb1c2f0eab08-0119-4162
CSeq: 1 ACK
Max-Forwards: 69
Content-Length: 0
P-hint: rr-enforced


<------------->
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: --- (10 headers 0 lines) ---
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: 
<--- SIP read from UDP:65.35.85.122:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK433ed38b;rport
From: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
To: <sip:7182109675@65.35.85.122>;tag=721A50-294
Date: Wed, 22 Feb 2012 23:03:39 GMT
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
CSeq: 104 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


<------------->
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: --- (10 headers 0 lines) ---
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: 
<--- SIP read from UDP:65.35.85.122:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK433ed38b;rport
From: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
To: <sip:7182109675@65.35.85.122>;tag=721A50-294
Date: Wed, 22 Feb 2012 23:03:39 GMT
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
CSeq: 104 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:7182109675@65.35.85.122:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Length: 270

v=0
o=CiscoSystemsSIP-GW-UserAgent 5841 4320 IN IP4 65.35.85.122
s=SIP Call
c=IN IP4 65.35.85.122
t=0 0
m=audio 17588 RTP/AVP 18 101
c=IN IP4 65.35.85.122
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: --- (15 headers 12 lines) ---
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: set_destination: Parsing <sip:7182109675@65.35.85.122:5060> for address/port to send to
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: set_destination: set destination to 65.35.85.122, port 5060
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: Transmitting (no NAT) to 65.35.85.122:5060:
ACK sip:7182109675@65.35.85.122:5060 SIP/2.0
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK4d566fcb;rport
Max-Forwards: 70
From: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
To: <sip:7182109675@65.35.85.122>;tag=721A50-294
Contact: <sip:7174537240@184.55.33.74>
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.6.2.7
Content-Length: 0


---
[2012-02-22 18:03:42] VERBOSE[1353] chan_sip.c: Really destroying SIP dialog '5b2c201b2b944a2f60586630121603b7@184.106.129.94' Method: REGISTER
[2012-02-22 18:03:42] VERBOSE[1353] chan_sip.c: Really destroying SIP dialog '0140f53a67bd1a1767375ee971c27c99@50.56.88.215' Method: REGISTER
[2012-02-22 18:03:43] VERBOSE[1353] chan_sip.c: Really destroying SIP dialog '1d772c1e0a718b3225d034641053b8ac@184.106.228.165' Method: OPTIONS
[2012-02-22 18:03:44] VERBOSE[1353] chan_sip.c: Really destroying SIP dialog '5d2943df0c727f7771696c6d15847450@50.57.140.121' Method: OPTIONS
[2012-02-22 18:03:44] VERBOSE[1353] chan_sip.c: Really destroying SIP dialog '59f5aa4e4283ccbd1ce117b45310ac30@184.106.233.13' Method: OPTIONS
[2012-02-22 18:03:45] VERBOSE[1353] chan_sip.c: Really destroying SIP dialog '0a50f5f2069cb4a82488fd4b5a592afc@192.168.1.8' Method: REGISTER
[2012-02-22 18:03:46] VERBOSE[1353] chan_sip.c: Really destroying SIP dialog '298600f21488a134126293db703248c4@184.106.229.160' Method: OPTIONS
[2012-02-22 18:03:47] VERBOSE[1353] chan_sip.c: Really destroying SIP dialog '5fbe6a627b89a41c36f0063f2e8757e2@184.106.69.109' Method: OPTIONS
[2012-02-22 18:03:48] VERBOSE[1353] chan_sip.c: REGISTER 12 headers, 0 lines
[2012-02-22 18:03:48] VERBOSE[1353] chan_sip.c: Reliably Transmitting (no NAT) to 208.138.47.173:5060:
REGISTER sip:bizvoip.cwjamaica.com SIP/2.0
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK0986b3b3;rport
Max-Forwards: 70
From: <sip:8766569072@bizvoip.cwjamaica.com>;tag=as1854417d
To: <sip:8766569072@bizvoip.cwjamaica.com>
Call-ID: 5a6a9e236b07c29216be0a8614e57927@184.55.33.74
CSeq: 2001 REGISTER
User-Agent: Asterisk PBX 1.6.2.7
Authorization: Digest username="8766569072", realm="bizvoip.cwjamaica.com", algorithm=MD5, uri="sip:bizvoip.cwjamaica.com", nonce="BroadWorksXgyxzeniaTwzdtawBW", response="32317842c2702881e584f06b163a6879", qop=auth, cnonce="151ef709", nc=00000740
Expires: 120
Contact: <sip:8766569072@184.55.33.74>
Content-Length: 0


---
[2012-02-22 18:03:48] VERBOSE[1353] chan_sip.c: 
<--- SIP read from UDP:208.138.47.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK0986b3b3;rport=5060
From: <sip:8766569072@bizvoip.cwjamaica.com>;tag=as1854417d
To: <sip:8766569072@bizvoip.cwjamaica.com>;tag=gK049fb30f
Call-ID: 5a6a9e236b07c29216be0a8614e57927@184.55.33.74
CSeq: 2001 REGISTER
Contact: <sip:8766569072@184.55.33.74>;expires=45
Content-Length: 0


<------------->
[2012-02-22 18:03:48] VERBOSE[1353] chan_sip.c: --- (8 headers 0 lines) ---
[2012-02-22 18:03:48] VERBOSE[1353] chan_sip.c: Reliably Transmitting (no NAT) to 50.57.90.69:5060:
OPTIONS sip:50.57.90.69 SIP/2.0
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK4190eb64;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@184.55.33.74>;tag=as2010f0d6
To: <sip:50.57.90.69>
Contact: <sip:asterisk@184.55.33.74>
Call-ID: 0964831e5a7033290f2ef9ac456651fe@184.55.33.74
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.7
Date: Wed, 22 Feb 2012 23:03:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[2012-02-22 18:03:48] VERBOSE[1353] chan_sip.c: Reliably Transmitting (no NAT) to 67.15.128.14:5060:
OPTIONS sip:sip1.didx.net SIP/2.0
Via: SIP/2.0/UDP 184.55.33.74:5060;branch=z9hG4bK0db778ec;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@184.55.33.74>;tag=as10431238
To: <sip:sip1.didx.net>
Contact: <sip:asterisk@184.55.33.74>
Call-ID: 3f24f88c3f6aa5ae78447ca66a93c7e6@184.55.33.74
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.7
Date: Wed, 22 Feb 2012 23:03:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

[2012-02-22 18:03:59] VERBOSE[1353] chan_sip.c: --- (7 headers 0 lines) ---
[2012-02-22 18:03:59] VERBOSE[1353] chan_sip.c: Really destroying SIP dialog '61565baa18e6d5f60062a427250101a9@184.55.33.74' Method: OPTIONS
[2012-02-22 18:03:59] VERBOSE[1353] chan_sip.c: 
<--- SIP read from UDP:65.35.85.122:55036 --->
INVITE sip:7174537240@184.55.33.74:5060 SIP/2.0
Via: SIP/2.0/UDP 65.35.85.122:5060;branch=z9hG4bK451974
From: <sip:7182109675@65.35.85.122>;tag=721A50-294
To: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
Date: Wed, 22 Feb 2012 23:03:57 GMT
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1255718427-1558188513-2159264725-2090755461
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1329951837
Contact: <sip:7182109675@65.35.85.122:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 270

v=0
o=CiscoSystemsSIP-GW-UserAgent 5841 4320 IN IP4 65.35.85.122
s=SIP Call
c=IN IP4 65.35.85.122
t=0 0
m=audio 17588 RTP/AVP 18 101
c=IN IP4 65.35.85.122
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
[2012-02-22 18:03:59] VERBOSE[1353] chan_sip.c: --- (19 headers 12 lines) ---
[2012-02-22 18:03:59] VERBOSE[1353] chan_sip.c: Sending to 65.35.85.122 : 5060 (no NAT)
[2012-02-22 18:03:59] VERBOSE[1353] chan_sip.c: 
<--- Transmitting (no NAT) to 65.35.85.122:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 65.35.85.122:5060;branch=z9hG4bK451974;received=65.35.85.122
From: <sip:7182109675@65.35.85.122>;tag=721A50-294
To: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
CSeq: 101 INVITE
Server: Asterisk PBX 1.6.2.7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:7174537240@184.55.33.74>
Content-Length: 0


<------------>
[2012-02-22 18:03:59] VERBOSE[1353] chan_sip.c: Audio is at 184.55.33.74 port 15688
[2012-02-22 18:03:59] VERBOSE[1353] chan_sip.c: Adding codec 0x100 (g729) to SDP
[2012-02-22 18:03:59] VERBOSE[1353] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-02-22 18:03:59] VERBOSE[1353] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to 65.35.85.122:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 65.35.85.122:5060;branch=z9hG4bK451974;received=65.35.85.122
From: <sip:7182109675@65.35.85.122>;tag=721A50-294
To: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
CSeq: 101 INVITE
Server: Asterisk PBX 1.6.2.7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:7174537240@184.55.33.74>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1356920958 1356920960 IN IP4 199.173.76.106
s=Asterisk PBX 1.6.2.7
c=IN IP4 199.173.76.106
t=0 0
m=audio 53056 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[2012-02-22 18:03:59] VERBOSE[1353] chan_sip.c: 
<--- SIP read from UDP:65.35.85.122:55036 --->
ACK sip:7174537240@184.55.33.74:5060 SIP/2.0
Via: SIP/2.0/UDP 65.35.85.122:5060;branch=z9hG4bK461E54
From: <sip:7182109675@65.35.85.122>;tag=721A50-294
To: "7174537240" <sip:7174537240@184.55.33.74>;tag=as73093239
Date: Wed, 22 Feb 2012 23:03:57 GMT
Call-ID: 2aaa76c752cca5004a5dd01958af6a85@184.55.33.74
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

david55 · February 23, 2012, 8:40am

The call is still up at the end of that trace (no BYE transaction).

ja133 · February 23, 2012, 10:18pm

Maybe I posted the wrong set of the log. It should be this:

[2012-02-22 17:57:50] VERBOSE[1353] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to 70.42.22.112:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 70.42.22.112;branch=z9hG4bK4dc3.159e61f2.0;received=70.42.22.112
Via: SIP/2.0/UDP 208.85.248.41:5060;branch=z9hG4bK2sansay90646569rdb2320
From: <sip:7174557240@208.85.248.41>;tag=sansay90646569rdb2320
To: <sip:17182109675@70.42.22.112>;tag=as55a65300
Call-ID: 40f95b30505eadc713c419816573a174498f614f7d2cb9ded8-0117-6435
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:17182109675@184.55.42.13>
Content-Type: application/sdp
Content-Length: 107

v=0
o=root 182627518 182627519 IN IP4 64.82.142.84
s=Asterisk PBX 1.6.2.7
c=IN IP4 64.82.142.84
t=0 0
m=audio 0 RTP/AVP 0 8

<------------>
[2012-02-22 17:57:50] VERBOSE[1353] chan_sip.c: 
<--- SIP read from UDP:70.42.22.112:5060 --->
ACK sip:17182109675@184.55.42.13 SIP/2.0
Via: SIP/2.0/UDP 70.42.22.112;branch=z9hG4bK4dc3.159e61f2.2
Via: SIP/2.0/UDP 208.85.248.41:5060;branch=z9hG4bK2sansay90646569rdb2320-200
To: <sip:17182109675@70.42.22.112>;tag=as55a65300
From: <sip:7174557240@208.85.248.41>;tag=sansay90646569rdb2320
Call-ID: 40f95b30505eadc713c419816573a174498f614f7d2cb9ded8-0117-6435
CSeq: 2 ACK
Max-Forwards: 69
Content-Length: 0
P-hint: rr-enforced


<------------->
[2012-02-22 17:57:50] VERBOSE[1353] chan_sip.c: --- (10 headers 0 lines) ---
[2012-02-22 17:57:50] VERBOSE[1353] chan_sip.c: 
<--- SIP read from UDP:70.42.22.112:5060 --->
BYE sip:17182109675@184.55.42.13 SIP/2.0
Via: SIP/2.0/UDP 70.42.22.112;branch=z9hG4bK5dc3.4b0ab447.0
Via: SIP/2.0/UDP 208.85.248.41:5060;branch=z9hG4bK3sansay90646569rdb2320
To: <sip:17182109675@70.42.22.112>;tag=as55a65300
From: <sip:7174557240@208.85.248.41>;tag=sansay90646569rdb2320
Call-ID: 40f95b30505eadc713c419816573a174498f614f7d2cb9ded8-0117-6435
CSeq: 3 BYE
User-agent: CS2000_NGSS/9.0
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
Max-Forwards: 66
Content-Length: 0
P-hint: rr-enforced


<------------->
[2012-02-22 17:57:50] VERBOSE[1353] chan_sip.c: --- (12 headers 0 lines) ---
[2012-02-22 17:57:50] VERBOSE[1353] chan_sip.c: Sending to 70.42.22.112 : 5060 (no NAT)
[2012-02-22 17:57:50] VERBOSE[1353] chan_sip.c: 
<--- Transmitting (no NAT) to 70.42.22.112:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 70.42.22.112;branch=z9hG4bK5dc3.4b0ab447.0;received=70.42.22.112
Via: SIP/2.0/UDP 208.85.248.41:5060;branch=z9hG4bK3sansay90646569rdb2320
From: <sip:7174557240@208.85.248.41>;tag=sansay90646569rdb2320
To: <sip:17182109675@70.42.22.112>;tag=as55a65300
Call-ID: 40f95b30505eadc713c419816573a174498f614f7d2cb9ded8-0117-6435
CSeq: 3 BYE
Server: Asterisk PBX 1.6.2.7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[2012-02-22 17:57:50] VERBOSE[6029] chan_sip.c: Scheduling destruction of SIP dialog '29d5f51e27aa07886a6fcdcf122164e3@184.55.42.13' in 32000 ms (Method: ACK)
[2012-02-22 17:57:50] VERBOSE[6029] chan_sip.c: set_destination: Parsing <sip:7182109675@64.82.142.84:5060> for address/port to send to
[2012-02-22 17:57:50] VERBOSE[6029] chan_sip.c: set_destination: set destination to 64.82.142.84, port 5060
[2012-02-22 17:57:50] VERBOSE[6029] chan_sip.c: Reliably Transmitting (no NAT) to 64.82.142.84:5060:
BYE sip:7182109675@64.82.142.84:5060 SIP/2.0
Via: SIP/2.0/UDP 184.55.42.13:5060;branch=z9hG4bK3f1927d4;rport
Max-Forwards: 70
From: "7174557240" <sip:7174557240@184.55.42.13>;tag=as6a795a07
To: <sip:7182109675@64.82.142.84>;tag=5B1FC4-2497
Call-ID: 29d5f51e27aa07886a6fcdcf122164e3@184.55.42.13
CSeq: 105 BYE
User-Agent: Asterisk PBX 1.6.2.7
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[2012-02-22 17:57:50] VERBOSE[6029] pbx.c:   == Spawn extension (inbound-unlimited, 17182109675, 3) exited non-zero on 'SIP/ezcall-did-000007f9'
[2012-02-22 17:57:50] VERBOSE[1353] chan_sip.c: 
<--- SIP read from UDP:64.82.142.84:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.55.42.13:5060;branch=z9hG4bK3f1927d4;rport
From: "7174557240" <sip:7174557240@184.55.42.13>;tag=as6a795a07
To: <sip:7182109675@64.82.142.84>;tag=5B1FC4-2497
Date: Wed, 22 Feb 2012 22:57:48 GMT
Call-ID: 29d5f51e27aa07886a6fcdcf122164e3@184.55.42.13
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 105 BYE
Reason: Q.850;cause=16
Content-Length: 0


[2012-02-22 18:03:38] VERBOSE[1353] chan_sip.c: --- (13 headers 0 lines) ---
[2012-02-22 18:03:38] VERBOSE[1353] chan_sip.c: Really destroying SIP dialog '21e1d323380fc2595d3dc86162a61961@184.55.42.13' Method: OPTIONS
[2012-02-22 18:03:38] VERBOSE[1353] chan_sip.c: Really destroying SIP dialog '4e6be1954497f5bc505f3a992210106f@74.90.68.26' Method: OPTIONS
[2012-02-22 18:03:39] VERBOSE[1353] chan_sip.c: Really destroying SIP dialog '2dcfe416106f13562cb7874b7ec269f8@173.203.202.55' Method: OPTIONS
[2012-02-22 18:03:41] VERBOSE[1353] chan_sip.c: 
<--- SIP read from UDP:70.42.22.112:5060 --->
INVITE sip:17182109675@184.55.42.13 SIP/2.0
Record-Route: <sip:70.42.22.112;lr=on;ftag=sansay90655509rdb1044;did=882.5b7b3a56>
Via: SIP/2.0/UDP 70.42.22.112;branch=z9hG4bKc59b.cabe8e86.0
Via: SIP/2.0/UDP 208.85.248.41:5060;branch=z9hG4bK1sansay90655509rdb1044
Record-Route: <sip:sansay90655509rdb1044@208.85.248.41:5060;lr;transport=udp>
To: <sip:17182109675@70.42.22.112>
From: <sip:7174557240@208.85.248.41>;tag=sansay90655509rdb1044
Call-ID: 421af690585eadc713c41980eb639f97a8649b5cb1c2f0eab08-0119-4162
CSeq: 1 INVITE
Contact: <sip:7174557240@208.85.248.41:5060>
Supported: replaces
P-Asserted-Identity: <sip:7174557240@208.85.248.41>
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
Max-Forwards: 13
Content-Type: application/sdp
Content-Length: 287

v=0
o=Sansay-VSXi 188 1 IN IP4 208.85.248.41
s=Session Controller
c=IN IP4 199.173.76.106
t=0 0
m=audio 53056 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Topic		Replies	Views
Call end after 15 minutes Asterisk Support	1	568	February 12, 2014
Cisco -> Asterisk Asterisk Support	1	188	December 1, 2007
Problem after upgrading asterisk to 10.6-7 from 10.5 Asterisk Support	3	268	August 13, 2012
Cisco SIP phone - are there any timeout parameters? Asterisk Support	6	324	August 14, 2009
Calls dropped when transferred to outside Asterisk Support	11	429	November 29, 2012

Calls disconnect from provider using Cisco

Related topics