Bridging Lync and VoIPVoIP w/ Asterisk 1.8.6.0


#1

I am running Asterisk v1.8.6.0 and using it to bridge Microsoft Lync Server to VoIP VoIP (voipvoip.com). I am able to call from one to the other but no audio is transmitted or received.

To debug this, I have installed X-Lite to connect directly to Asterisk. I am able to successfully call from Lync->X-Lite and X-Lite->Lynx through Asterisk. Audio is working in both direction. I am also able to successfully call from VoIPVoIP->X-Lite and X-Lite to VoIPVoIP. Audio is also working in both directions. Based on this, I would think that going between Lync and VoIPVoIP would work but it doesn’t. Able to call and answer but no audio is transmitted in either direction. This would lead me to think it’s a codec problem and I have configured both Lync and VoIPVoIP in sip.conf with the following:

disallow=all
allow=ulaw
allow=alaw

I have also enabled full logging in Asterisk but didn’t see anything that stand out. Here is the full chatter between Lync and VoIPVoIP:

[Sep 21 15:58:03] VERBOSE[2672] config.c:   == Parsing '/etc/asterisk/logger.conf': [Sep 21 15:58:03] VERBOSE[2672] config.c:   == Found
[Sep 21 15:58:03] VERBOSE[2672] logger.c:  Asterisk Queue Logger restarted
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: 
<--- SIP read from TCP:10.1.1.24:61434 --->
INVITE sip:+18081234567@10.1.1.26;user=phone SIP/2.0
FROM: "John Doe"<sip:296@lync.domain.local;user=phone>;epid=34C8A8506A;tag=b81021fd62
TO: <sip:+18081234567@10.1.1.26;user=phone>
CSEQ: 3475 INVITE
CALL-ID: d57c1d34-e622-4fe9-b4a8-45121ab08d63
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 10.1.1.24:61434;branch=z9hG4bKde44b458
CONTACT: <sip:lync.domain.local:5068;transport=Tcp;maddr=10.1.1.24;ms-opaque=bac9e93852c1514c>
CONTENT-LENGTH: 330
SUPPORTED: 100rel
USER-AGENT: RTCC/4.0.0.0 MediationServer
CONTENT-TYPE: application/sdp
ALLOW: ACK
Allow: CANCEL,BYE,INVITE,PRACK,UPDATE

v=0
o=- 45 1 IN IP4 10.1.1.24
s=session
c=IN IP4 10.1.1.24
b=CT:1000
t=0 0
m=audio 57182 RTP/AVP 97 101 13 0 8
c=IN IP4 10.1.1.24
a=rtcp:57183
a=label:Audio
a=sendrecv
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: --- (14 headers 18 lines) ---
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Sending to 10.1.1.24:61434 (no NAT)
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Using INVITE request as basis request - d57c1d34-e622-4fe9-b4a8-45121ab08d63
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Found peer 'to-Lync_Server' for '296' from 10.1.1.24:61434
[Sep 21 15:58:09] VERBOSE[2720] netsock2.c:   == Using SIP RTP CoS mark 5
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Found RTP audio format 97
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Found RTP audio format 101
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Found RTP audio format 13
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Found RTP audio format 0
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Found RTP audio format 8
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Found unknown media description format RED for ID 97
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Found audio description format telephone-event for ID 101
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Found audio description format CN for ID 13
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Found audio description format PCMU for ID 0
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Found audio description format PCMA for ID 8
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Peer audio RTP is at port 10.1.1.24:57182
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: Looking for +18081234567 in from-Lync_Server (domain 10.1.1.26)
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: list_route: hop: <sip:lync.domain.local:5068;transport=Tcp;maddr=10.1.1.24;ms-opaque=bac9e93852c1514c>
[Sep 21 15:58:09] VERBOSE[2720] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.24:61434 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.1.1.24:61434;branch=z9hG4bKde44b458;received=10.1.1.24
From: "John Doe"<sip:296@lync.domain.local;user=phone>;epid=34C8A8506A;tag=b81021fd62
To: <sip:+18081234567@10.1.1.26;user=phone>
Call-ID: d57c1d34-e622-4fe9-b4a8-45121ab08d63
CSeq: 3475 INVITE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:+18081234567@10.1.1.26:5060;transport=TCP>
Content-Length: 0


<------------>
[Sep 21 15:58:09] VERBOSE[2721] pbx.c:     -- Executing [+18081234567@from-Lync_Server:1] Dial("SIP/to-Lync_Server-0000000a", "SIP/to-VoIP_VoIP/+18081234567") in new stack
[Sep 21 15:58:09] VERBOSE[2721] netsock2.c:   == Using SIP RTP CoS mark 5
[Sep 21 15:58:09] VERBOSE[2721] chan_sip.c: Audio is at 5060
[Sep 21 15:58:09] VERBOSE[2721] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 21 15:58:09] VERBOSE[2721] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Sep 21 15:58:09] VERBOSE[2721] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 21 15:58:09] VERBOSE[2721] chan_sip.c: Reliably Transmitting (no NAT) to 69.90.209.57:5060:
INVITE sip:+18081234567@sip3.voipvoip.com SIP/2.0
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK245fb217
Max-Forwards: 70
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>
Contact: <sip:5551112222@209.191.122.70:5060>
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Date: Wed, 21 Sep 2011 22:58:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 178561688 178561688 IN IP4 209.191.122.70
s=Asterisk PBX 1.8.6.0
c=IN IP4 209.191.122.70
t=0 0
m=audio 10616 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Sep 21 15:58:09] VERBOSE[2721] app_dial.c:     -- Called SIP/to-VoIP_VoIP/+18081234567
[Sep 21 15:58:09] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK245fb217
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 102 INVITE
Server: OpenSer (1.0.1 (i386/linux))
Content-Length: 0
Warning: 392 69.90.209.57:5060 "Noisy feedback tells: pid=7916 req_src_ip=209.191.122.70 req_src_port=5060 in_uri=sip:+18081234567@sip3.voipvoip.com out_uri=sip:+18081234567@sip3.voipvoip.com via_cnt==1"

<------------->
[Sep 21 15:58:09] VERBOSE[2496] chan_sip.c: --- (9 headers 0 lines) ---
[Sep 21 15:58:09] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK245fb217
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=d043eebb64d4461cc6a1e7afbfbe3159.45a3
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="voipvoip.com", nonce="4e7a6ca53cc9e0a7e2f7357bcdcd4dadc9324999"
Server: OpenSer (1.0.1 (i386/linux))
Content-Length: 0
Warning: 392 69.90.209.57:5060 "Noisy feedback tells: pid=7916 req_src_ip=209.191.122.70 req_src_port=5060 in_uri=sip:+18081234567@sip3.voipvoip.com out_uri=sip:+18081234567@sip3.voipvoip.com via_cnt==1"

<------------->
[Sep 21 15:58:09] VERBOSE[2496] chan_sip.c: --- (10 headers 0 lines) ---
[Sep 21 15:58:09] VERBOSE[2496] chan_sip.c: Transmitting (no NAT) to 69.90.209.57:5060:
ACK sip:+18081234567@sip3.voipvoip.com SIP/2.0
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK245fb217
Max-Forwards: 70
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=d043eebb64d4461cc6a1e7afbfbe3159.45a3
Contact: <sip:5551112222@209.191.122.70:5060>
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.6.0
Content-Length: 0


---
[Sep 21 15:58:09] VERBOSE[2496] chan_sip.c: Audio is at 5060
[Sep 21 15:58:09] VERBOSE[2496] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 21 15:58:09] VERBOSE[2496] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Sep 21 15:58:09] VERBOSE[2496] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 21 15:58:09] VERBOSE[2496] chan_sip.c: Reliably Transmitting (no NAT) to 69.90.209.57:5060:
INVITE sip:+18081234567@sip3.voipvoip.com SIP/2.0
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK72c6903e
Max-Forwards: 70
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>
Contact: <sip:5551112222@209.191.122.70:5060>
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Proxy-Authorization: Digest username="5551112222", realm="voipvoip.com", algorithm=MD5, uri="sip:+18081234567@sip3.voipvoip.com", nonce="4e7a6ca53cc9e0a7e2f7357bcdcd4dadc9324999", response="89f763491e60fb7d149085b5fb7ef942"
Date: Wed, 21 Sep 2011 22:58:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 178561688 178561689 IN IP4 209.191.122.70
s=Asterisk PBX 1.8.6.0
c=IN IP4 209.191.122.70
t=0 0
m=audio 10616 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Sep 21 15:58:09] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK72c6903e
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 103 INVITE
Server: OpenSer (1.0.1 (i386/linux))
Content-Length: 0
Warning: 392 69.90.209.57:5060 "Noisy feedback tells: pid=7915 req_src_ip=209.191.122.70 req_src_port=5060 in_uri=sip:+18081234567@sip3.voipvoip.com out_uri=sip:+18081234567@sip3.voipvoip.com via_cnt==1"

<------------->
[Sep 21 15:58:09] VERBOSE[2496] chan_sip.c: --- (9 headers 0 lines) ---
[Sep 21 15:58:09] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK72c6903e
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 103 INVITE
Server: OpenSer (1.0.1 (i386/linux))
Content-Length: 0
Warning: 392 69.90.209.57:5060 "Noisy feedback tells: pid=7915 req_src_ip=209.191.122.70 req_src_port=5060 in_uri=sip:+18081234567@sip3.voipvoip.com out_uri=sip:VoIP$18081234567@sipgw.voipvoip.com:5960 via_cnt==1"

<------------->
[Sep 21 15:58:09] VERBOSE[2496] chan_sip.c: --- (9 headers 0 lines) ---
[Sep 21 15:58:11] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:182.74.240.100:46591 --->


<------------->
[Sep 21 15:58:12] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK72c6903e
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 103 INVITE
Record-Route: <sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
Record-Route: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>
Contact: <sip:+18081234567@4.55.22.99:5060>
Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE
Content-Length: 231
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 11819 9676 IN IP4 4.55.22.99
s=SIP Media Capabilities
c=IN IP4 4.55.22.66
t=0 0
m=audio 28692 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
[Sep 21 15:58:12] VERBOSE[2496] chan_sip.c: --- (13 headers 11 lines) ---
[Sep 21 15:58:12] VERBOSE[2496] chan_sip.c: Found RTP audio format 0
[Sep 21 15:58:12] VERBOSE[2496] chan_sip.c: Found RTP audio format 101
[Sep 21 15:58:12] VERBOSE[2496] chan_sip.c: Found audio description format PCMU for ID 0
[Sep 21 15:58:12] VERBOSE[2496] chan_sip.c: Found audio description format telephone-event for ID 101
[Sep 21 15:58:12] VERBOSE[2496] chan_sip.c: Capabilities: us - 0x40c (ulaw|alaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Sep 21 15:58:12] VERBOSE[2496] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Sep 21 15:58:12] VERBOSE[2496] chan_sip.c: Peer audio RTP is at port 4.55.22.66:28692
[Sep 21 15:58:12] VERBOSE[2721] app_dial.c:     -- SIP/to-VoIP_VoIP-0000000b is ringing
[Sep 21 15:58:12] VERBOSE[2721] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.24:61434 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.1.1.24:61434;branch=z9hG4bKde44b458;received=10.1.1.24
From: "John Doe"<sip:296@lync.domain.local;user=phone>;epid=34C8A8506A;tag=b81021fd62
To: <sip:+18081234567@10.1.1.26;user=phone>;tag=as0a8394a8
Call-ID: d57c1d34-e622-4fe9-b4a8-45121ab08d63
CSeq: 3475 INVITE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:+18081234567@10.1.1.26:5060;transport=TCP>
Content-Length: 0


<------------>
[Sep 21 15:58:12] VERBOSE[2721] app_dial.c:     -- SIP/to-VoIP_VoIP-0000000b is making progress passing it to SIP/to-Lync_Server-0000000a
[Sep 21 15:58:12] VERBOSE[2721] chan_sip.c: Audio is at 5060
[Sep 21 15:58:12] VERBOSE[2721] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 21 15:58:12] VERBOSE[2721] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Sep 21 15:58:12] VERBOSE[2721] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 21 15:58:12] VERBOSE[2721] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.24:61434 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 10.1.1.24:61434;branch=z9hG4bKde44b458;received=10.1.1.24
From: "John Doe"<sip:296@lync.domain.local;user=phone>;epid=34C8A8506A;tag=b81021fd62
To: <sip:+18081234567@10.1.1.26;user=phone>;tag=as0a8394a8
Call-ID: d57c1d34-e622-4fe9-b4a8-45121ab08d63
CSeq: 3475 INVITE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:+18081234567@10.1.1.26:5060;transport=TCP>
Content-Type: application/sdp
Content-Length: 254

v=0
o=root 2132581636 2132581636 IN IP4 10.1.1.26
s=Asterisk PBX 1.8.6.0
c=IN IP4 10.1.1.26
t=0 0
m=audio 12420 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Sep 21 15:58:16] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK72c6903e
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 103 INVITE
Record-Route: <sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
Record-Route: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>
Contact: <sip:+18081234567@4.55.22.99:5060>
Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE
Content-Length: 231
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 11819 9676 IN IP4 4.55.22.99
s=SIP Media Capabilities
c=IN IP4 4.55.22.66
t=0 0
m=audio 28692 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
[Sep 21 15:58:16] VERBOSE[2496] chan_sip.c: --- (13 headers 11 lines) ---
[Sep 21 15:58:16] VERBOSE[2721] app_dial.c:     -- SIP/to-VoIP_VoIP-0000000b is ringing
[Sep 21 15:58:16] VERBOSE[2721] app_dial.c:     -- SIP/to-VoIP_VoIP-0000000b is making progress passing it to SIP/to-Lync_Server-0000000a
[Sep 21 15:58:18] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK72c6903e
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 103 INVITE
Record-Route: <sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
Record-Route: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+18081234567@4.55.22.99:5060>
Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE
Content-Length: 231
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 11819 9676 IN IP4 4.55.22.99
s=SIP Media Capabilities
c=IN IP4 4.55.22.66
t=0 0
m=audio 28692 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
[Sep 21 15:58:18] VERBOSE[2496] chan_sip.c: --- (14 headers 11 lines) ---
[Sep 21 15:58:18] VERBOSE[2496] chan_sip.c: list_route: hop: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>
[Sep 21 15:58:18] VERBOSE[2496] chan_sip.c: list_route: hop: <sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
[Sep 21 15:58:18] VERBOSE[2496] chan_sip.c: set_destination: Parsing <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on> for address/port to send to
[Sep 21 15:58:18] VERBOSE[2496] chan_sip.c: set_destination: set destination to 69.90.209.57:5060
[Sep 21 15:58:18] VERBOSE[2496] chan_sip.c: Transmitting (no NAT) to 69.90.209.57:5060:
ACK sip:+18081234567@4.55.22.99:5060 SIP/2.0
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK47f1372c
Route: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>,<sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
Max-Forwards: 70
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Contact: <sip:5551112222@209.191.122.70:5060>
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.6.0
Content-Length: 0


---
[Sep 21 15:58:18] VERBOSE[2721] app_dial.c:     -- SIP/to-VoIP_VoIP-0000000b answered SIP/to-Lync_Server-0000000a
[Sep 21 15:58:18] VERBOSE[2721] chan_sip.c: Audio is at 5060
[Sep 21 15:58:18] VERBOSE[2721] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 21 15:58:18] VERBOSE[2721] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Sep 21 15:58:18] VERBOSE[2721] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 21 15:58:18] VERBOSE[2721] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to 10.1.1.24:61434 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.1.1.24:61434;branch=z9hG4bKde44b458;received=10.1.1.24
From: "John Doe"<sip:296@lync.domain.local;user=phone>;epid=34C8A8506A;tag=b81021fd62
To: <sip:+18081234567@10.1.1.26;user=phone>;tag=as0a8394a8
Call-ID: d57c1d34-e622-4fe9-b4a8-45121ab08d63
CSeq: 3475 INVITE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:+18081234567@10.1.1.26:5060;transport=TCP>
Content-Type: application/sdp
Content-Length: 254

v=0
o=root 2132581636 2132581637 IN IP4 10.1.1.26
s=Asterisk PBX 1.8.6.0
c=IN IP4 10.1.1.26
t=0 0
m=audio 12420 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Sep 21 15:58:18] VERBOSE[2721] rtp_engine.c:     -- Remotely bridging SIP/to-Lync_Server-0000000a and SIP/to-VoIP_VoIP-0000000b
[Sep 21 15:58:18] VERBOSE[2721] chan_sip.c: set_destination: Parsing <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on> for address/port to send to
[Sep 21 15:58:18] VERBOSE[2721] chan_sip.c: set_destination: set destination to 69.90.209.57:5060
[Sep 21 15:58:18] VERBOSE[2721] chan_sip.c: Audio is at 5060
[Sep 21 15:58:18] VERBOSE[2721] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 21 15:58:18] VERBOSE[2721] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 21 15:58:18] VERBOSE[2721] chan_sip.c: Reliably Transmitting (no NAT) to 69.90.209.57:5060:
INVITE sip:+18081234567@4.55.22.99:5060 SIP/2.0
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK128cba51
Route: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>,<sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
Max-Forwards: 70
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Contact: <sip:5551112222@209.191.122.70:5060>
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 228

v=0
o=root 178561688 178561690 IN IP4 10.1.1.24
s=Asterisk PBX 1.8.6.0
c=IN IP4 10.1.1.24
t=0 0
m=audio 57182 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Sep 21 15:58:18] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK128cba51
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 104 INVITE
Server: OpenSer (1.0.1 (i386/linux))
Content-Length: 0
Warning: 392 69.90.209.57:5060 "Noisy feedback tells: pid=7917 req_src_ip=209.191.122.70 req_src_port=5060 in_uri=sip:+18081234567@4.55.22.99:5060 out_uri=sip:+18081234567@4.55.22.99:5060 via_cnt==1"

<------------->
[Sep 21 15:58:18] VERBOSE[2496] chan_sip.c: --- (9 headers 0 lines) ---
[Sep 21 15:58:18] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK128cba51
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 104 INVITE
Record-Route: <sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
Record-Route: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+18081234567@4.55.22.99:5060>
Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE
Content-Length: 231
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 11819 9676 IN IP4 4.55.22.99
s=SIP Media Capabilities
c=IN IP4 4.55.22.66
t=0 0
m=audio 28692 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
[Sep 21 15:58:18] VERBOSE[2496] chan_sip.c: --- (14 headers 11 lines) ---
[Sep 21 15:58:18] VERBOSE[2496] chan_sip.c: set_destination: Parsing <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on> for address/port to send to
[Sep 21 15:58:18] VERBOSE[2496] chan_sip.c: set_destination: set destination to 69.90.209.57:5060
[Sep 21 15:58:18] VERBOSE[2496] chan_sip.c: Transmitting (no NAT) to 69.90.209.57:5060:
ACK sip:+18081234567@4.55.22.99:5060 SIP/2.0
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK056fd07f
Route: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>,<sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
Max-Forwards: 70
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Contact: <sip:5551112222@209.191.122.70:5060>
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.8.6.0
Content-Length: 0


---
[Sep 21 15:58:18] VERBOSE[2496] chan_sip.c: Really destroying SIP dialog '01973a2213d07a047ba035cc02254e04@10.1.1.26' Method: REGISTER
[Sep 21 15:58:19] VERBOSE[2720] chan_sip.c: 
<--- SIP read from TCP:10.1.1.24:61434 --->
ACK sip:+18081234567@10.1.1.26:5060;transport=TCP SIP/2.0
FROM: <sip:296@lync.domain.local;user=phone>;epid=34C8A8506A;tag=b81021fd62
TO: <sip:+18081234567@10.1.1.26;user=phone>;tag=as0a8394a8
CSEQ: 3475 ACK
CALL-ID: d57c1d34-e622-4fe9-b4a8-45121ab08d63
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 10.1.1.24:61434;branch=z9hG4bKdeed744e
CONTENT-LENGTH: 0
USER-AGENT: RTCC/4.0.0.0 MediationServer

<------------->
[Sep 21 15:58:19] VERBOSE[2720] chan_sip.c: --- (9 headers 0 lines) ---
[Sep 21 15:58:19] VERBOSE[2720] chan_sip.c: set_destination: Parsing <sip:lync.domain.local:5068;transport=Tcp;maddr=10.1.1.24;ms-opaque=bac9e93852c1514c> for address/port to send to
[Sep 21 15:58:19] VERBOSE[2720] chan_sip.c: set_destination: set destination to 10.1.1.24:5068
[Sep 21 15:58:19] VERBOSE[2720] chan_sip.c: Audio is at 5060
[Sep 21 15:58:19] VERBOSE[2720] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 21 15:58:19] VERBOSE[2720] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 21 15:58:19] VERBOSE[2720] chan_sip.c: Reliably Transmitting (no NAT) to 10.1.1.24:5068:
INVITE sip:lync.domain.local:5068;transport=Tcp;maddr=10.1.1.24;ms-opaque=bac9e93852c1514c SIP/2.0
Via: SIP/2.0/TCP 10.1.1.26:5060;branch=z9hG4bK0aef2b2c
Max-Forwards: 70
From: <sip:+18081234567@10.1.1.26;user=phone>;tag=as0a8394a8
To: "John Doe"<sip:296@lync.domain.local;user=phone>;epid=34C8A8506A;tag=b81021fd62
Contact: <sip:+18081234567@10.1.1.26:5060;transport=TCP>
Call-ID: d57c1d34-e622-4fe9-b4a8-45121ab08d63
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 2132581636 2132581638 IN IP4 4.55.22.66
s=Asterisk PBX 1.8.6.0
c=IN IP4 4.55.22.66
t=0 0
m=audio 28692 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Sep 21 15:58:19] VERBOSE[2720] chan_sip.c: 
<--- SIP read from TCP:10.1.1.24:61434 --->
SIP/2.0 100 Trying
FROM: <sip:+18081234567@10.1.1.26;user=phone>;tag=as0a8394a8
TO: "John Doe"<sip:296@lync.domain.local;user=phone>;tag=b81021fd62;epid=34C8A8506A
CSEQ: 102 INVITE
CALL-ID: d57c1d34-e622-4fe9-b4a8-45121ab08d63
VIA: SIP/2.0/TCP 10.1.1.26:5060;branch=z9hG4bK0aef2b2c
CONTENT-LENGTH: 0

<------------->
[Sep 21 15:58:19] VERBOSE[2720] chan_sip.c: --- (7 headers 0 lines) ---
[Sep 21 15:58:20] VERBOSE[2720] chan_sip.c: 
<--- SIP read from TCP:10.1.1.24:61434 --->
SIP/2.0 200 OK
FROM: <sip:+18081234567@10.1.1.26;user=phone>;tag=as0a8394a8
TO: "John Doe"<sip:296@lync.domain.local;user=phone>;epid=34C8A8506A;tag=b81021fd62
CSEQ: 102 INVITE
CALL-ID: d57c1d34-e622-4fe9-b4a8-45121ab08d63
VIA: SIP/2.0/TCP 10.1.1.26:5060;branch=z9hG4bK0aef2b2c
CONTACT: <sip:lync.domain.local:5068;transport=Tcp;maddr=10.1.1.24;ms-opaque=bac9e93852c1514c>
CONTENT-LENGTH: 245
SUPPORTED: 100rel
CONTENT-TYPE: application/sdp
SERVER: RTCC/4.0.0.0 MediationServer

v=0
o=- 45 2 IN IP4 10.1.1.24
s=session
c=IN IP4 10.1.1.24
b=CT:1000
t=0 0
m=audio 57182 RTP/AVP 0 101
c=IN IP4 10.1.1.24
a=rtcp:57183
a=label:Audio
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
[Sep 21 15:58:20] VERBOSE[2720] chan_sip.c: --- (11 headers 14 lines) ---
[Sep 21 15:58:20] VERBOSE[2720] chan_sip.c: Found RTP audio format 0
[Sep 21 15:58:20] VERBOSE[2720] chan_sip.c: Found RTP audio format 101
[Sep 21 15:58:20] VERBOSE[2720] chan_sip.c: Found audio description format PCMU for ID 0
[Sep 21 15:58:20] VERBOSE[2720] chan_sip.c: Found audio description format telephone-event for ID 101
[Sep 21 15:58:20] VERBOSE[2720] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Sep 21 15:58:20] VERBOSE[2720] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Sep 21 15:58:20] VERBOSE[2720] chan_sip.c: Peer audio RTP is at port 10.1.1.24:57182
[Sep 21 15:58:20] VERBOSE[2720] chan_sip.c: set_destination: Parsing <sip:lync.domain.local:5068;transport=Tcp;maddr=10.1.1.24;ms-opaque=bac9e93852c1514c> for address/port to send to
[Sep 21 15:58:20] VERBOSE[2720] chan_sip.c: set_destination: set destination to 10.1.1.24:5068
[Sep 21 15:58:20] VERBOSE[2720] chan_sip.c: Transmitting (no NAT) to 10.1.1.24:5068:
ACK sip:lync.domain.local:5068;transport=Tcp;maddr=10.1.1.24;ms-opaque=bac9e93852c1514c SIP/2.0
Via: SIP/2.0/TCP 10.1.1.26:5060;branch=z9hG4bK387469bf
Max-Forwards: 70
From: <sip:+18081234567@10.1.1.26;user=phone>;tag=as0a8394a8
To: "John Doe"<sip:296@lync.domain.local;user=phone>;epid=34C8A8506A;tag=b81021fd62
Contact: <sip:+18081234567@10.1.1.26:5060;transport=TCP>
Call-ID: d57c1d34-e622-4fe9-b4a8-45121ab08d63
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.6.0
Content-Length: 0


---
[Sep 21 15:58:20] VERBOSE[2721] chan_sip.c: set_destination: Parsing <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on> for address/port to send to
[Sep 21 15:58:20] VERBOSE[2721] chan_sip.c: set_destination: set destination to 69.90.209.57:5060
[Sep 21 15:58:20] VERBOSE[2721] chan_sip.c: Audio is at 5060
[Sep 21 15:58:20] VERBOSE[2721] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 21 15:58:20] VERBOSE[2721] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 21 15:58:20] VERBOSE[2721] chan_sip.c: Reliably Transmitting (no NAT) to 69.90.209.57:5060:
INVITE sip:+18081234567@4.55.22.99:5060 SIP/2.0
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK2340a125
Route: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>,<sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
Max-Forwards: 70
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Contact: <sip:5551112222@209.191.122.70:5060>
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 105 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 228

v=0
o=root 178561688 178561691 IN IP4 10.1.1.24
s=Asterisk PBX 1.8.6.0
c=IN IP4 10.1.1.24
t=0 0
m=audio 57182 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Sep 21 15:58:20] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK2340a125
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 105 INVITE
Server: OpenSer (1.0.1 (i386/linux))
Content-Length: 0
Warning: 392 69.90.209.57:5060 "Noisy feedback tells: pid=7919 req_src_ip=209.191.122.70 req_src_port=5060 in_uri=sip:+18081234567@4.55.22.99:5060 out_uri=sip:+18081234567@4.55.22.99:5060 via_cnt==1"

<------------->
[Sep 21 15:58:20] VERBOSE[2496] chan_sip.c: --- (9 headers 0 lines) ---
[Sep 21 15:58:20] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK2340a125
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 105 INVITE
Record-Route: <sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
Record-Route: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+18081234567@4.55.22.99:5060>
Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE
Content-Length: 231
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 11819 9676 IN IP4 4.55.22.99
s=SIP Media Capabilities
c=IN IP4 4.55.22.66
t=0 0
m=audio 28692 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
[Sep 21 15:58:20] VERBOSE[2496] chan_sip.c: --- (14 headers 11 lines) ---
[Sep 21 15:58:20] VERBOSE[2496] chan_sip.c: set_destination: Parsing <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on> for address/port to send to
[Sep 21 15:58:20] VERBOSE[2496] chan_sip.c: set_destination: set destination to 69.90.209.57:5060
[Sep 21 15:58:20] VERBOSE[2496] chan_sip.c: Transmitting (no NAT) to 69.90.209.57:5060:
ACK sip:+18081234567@4.55.22.99:5060 SIP/2.0
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK078b1362
Route: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>,<sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
Max-Forwards: 70
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Contact: <sip:5551112222@209.191.122.70:5060>
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 105 ACK
User-Agent: Asterisk PBX 1.8.6.0
Content-Length: 0


---
[Sep 21 15:58:22] VERBOSE[2723] chan_sip.c: 
<--- SIP read from TCP:10.1.1.24:61445 --->
OPTIONS sip:10.1.1.26 SIP/2.0
FROM: <sip:lync.domain.local:5068;transport=Tcp;ms-opaque=bac9e93852c1514c>;epid=34C8A8506A;tag=e97a93ecde
TO: <sip:10.1.1.26>
CSEQ: 3476 OPTIONS
CALL-ID: c12499abe8874297bfe8970688a88ea7
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 10.1.1.24:61445;branch=z9hG4bK69bb3b3a
CONTACT: <sip:lync.domain.local:5068;transport=Tcp;maddr=10.1.1.24>
CONTENT-LENGTH: 0
USER-AGENT: RTCC/4.0.0.0 MediationServer

<------------->
[Sep 21 15:58:22] VERBOSE[2723] chan_sip.c: --- (10 headers 0 lines) ---
[Sep 21 15:58:22] VERBOSE[2723] chan_sip.c: Looking for s in default (domain 10.1.1.26)
[Sep 21 15:58:22] VERBOSE[2723] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.24:61445 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.1.1.24:61445;branch=z9hG4bK69bb3b3a;received=10.1.1.24
From: <sip:lync.domain.local:5068;transport=Tcp;ms-opaque=bac9e93852c1514c>;epid=34C8A8506A;tag=e97a93ecde
To: <sip:10.1.1.26>;tag=as6617dc36
Call-ID: c12499abe8874297bfe8970688a88ea7
CSeq: 3476 OPTIONS
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:10.1.1.26:5060;transport=TCP>
Accept: application/sdp
Content-Length: 0


<------------>
[Sep 21 15:58:22] VERBOSE[2723] chan_sip.c: Scheduling destruction of SIP dialog 'c12499abe8874297bfe8970688a88ea7' in 32000 ms (Method: OPTIONS)
[Sep 21 15:58:25] VERBOSE[2720] chan_sip.c: 
<--- SIP read from TCP:10.1.1.24:61434 --->
BYE sip:+18081234567@10.1.1.26:5060;transport=TCP SIP/2.0
FROM: <sip:296@lync.domain.local;user=phone>;epid=34C8A8506A;tag=b81021fd62
TO: <sip:+18081234567@10.1.1.26;user=phone>;tag=as0a8394a8
CSEQ: 3476 BYE
CALL-ID: d57c1d34-e622-4fe9-b4a8-45121ab08d63
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 10.1.1.24:61434;branch=z9hG4bKbc1190cc
CONTACT: <sip:lync.domain.local:5068;transport=Tcp;maddr=10.1.1.24>
CONTENT-LENGTH: 0
USER-AGENT: RTCC/4.0.0.0 MediationServer

<------------->
[Sep 21 15:58:25] VERBOSE[2720] chan_sip.c: --- (10 headers 0 lines) ---
[Sep 21 15:58:25] VERBOSE[2720] chan_sip.c: Sending to 10.1.1.24:61434 (no NAT)
[Sep 21 15:58:25] VERBOSE[2720] chan_sip.c: Scheduling destruction of SIP dialog 'd57c1d34-e622-4fe9-b4a8-45121ab08d63' in 6400 ms (Method: BYE)
[Sep 21 15:58:25] VERBOSE[2720] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.24:61434 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.1.1.24:61434;branch=z9hG4bKbc1190cc;received=10.1.1.24
From: <sip:296@lync.domain.local;user=phone>;epid=34C8A8506A;tag=b81021fd62
To: <sip:+18081234567@10.1.1.26;user=phone>;tag=as0a8394a8
Call-ID: d57c1d34-e622-4fe9-b4a8-45121ab08d63
CSeq: 3476 BYE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Sep 21 15:58:25] VERBOSE[2721] chan_sip.c: set_destination: Parsing <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on> for address/port to send to
[Sep 21 15:58:25] VERBOSE[2721] chan_sip.c: set_destination: set destination to 69.90.209.57:5060
[Sep 21 15:58:25] VERBOSE[2721] chan_sip.c: Audio is at 5060
[Sep 21 15:58:25] VERBOSE[2721] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 21 15:58:25] VERBOSE[2721] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 21 15:58:25] VERBOSE[2721] chan_sip.c: Reliably Transmitting (no NAT) to 69.90.209.57:5060:
INVITE sip:+18081234567@4.55.22.99:5060 SIP/2.0
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK482331f1
Route: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>,<sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
Max-Forwards: 70
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Contact: <sip:5551112222@209.191.122.70:5060>
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 106 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 178561688 178561692 IN IP4 209.191.122.70
s=Asterisk PBX 1.8.6.0
c=IN IP4 209.191.122.70
t=0 0
m=audio 10616 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Sep 21 15:58:25] VERBOSE[2721] pbx.c:     -- Executing [h@from-Lync_Server:1] Dial("SIP/to-Lync_Server-0000000a", "SIP/to-VoIP_VoIP/h") in new stack
[Sep 21 15:58:25] VERBOSE[2721] netsock2.c:   == Using SIP RTP CoS mark 5
[Sep 21 15:58:25] VERBOSE[2721] chan_sip.c: Audio is at 5060
[Sep 21 15:58:25] VERBOSE[2721] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 21 15:58:25] VERBOSE[2721] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Sep 21 15:58:25] VERBOSE[2721] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 21 15:58:25] VERBOSE[2721] chan_sip.c: Reliably Transmitting (no NAT) to 69.90.209.57:5060:
INVITE sip:h@sip3.voipvoip.com SIP/2.0
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK400d26c0
Max-Forwards: 70
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as4875aec1
To: <sip:h@sip3.voipvoip.com>
Contact: <sip:5551112222@209.191.122.70:5060>
Call-ID: 73aaae5c719a2b302a6bf5580e3cd99d@sip3.voipvoip.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Date: Wed, 21 Sep 2011 22:58:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 1808997917 1808997917 IN IP4 209.191.122.70
s=Asterisk PBX 1.8.6.0
c=IN IP4 209.191.122.70
t=0 0
m=audio 14928 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Sep 21 15:58:25] VERBOSE[2721] app_dial.c:     -- Called SIP/to-VoIP_VoIP/h
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK482331f1
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 106 INVITE
Server: OpenSer (1.0.1 (i386/linux))
Content-Length: 0
Warning: 392 69.90.209.57:5060 "Noisy feedback tells: pid=7916 req_src_ip=209.191.122.70 req_src_port=5060 in_uri=sip:+18081234567@4.55.22.99:5060 out_uri=sip:+18081234567@4.55.22.99:5060 via_cnt==1"

<------------->
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: --- (9 headers 0 lines) ---
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK400d26c0
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as4875aec1
To: <sip:h@sip3.voipvoip.com>
Call-ID: 73aaae5c719a2b302a6bf5580e3cd99d@sip3.voipvoip.com
CSeq: 102 INVITE
Server: OpenSer (1.0.1 (i386/linux))
Content-Length: 0
Warning: 392 69.90.209.57:5060 "Noisy feedback tells: pid=7914 req_src_ip=209.191.122.70 req_src_port=5060 in_uri=sip:h@sip3.voipvoip.com out_uri=sip:h@sip3.voipvoip.com via_cnt==1"

<------------->
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: --- (9 headers 0 lines) ---
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK400d26c0
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as4875aec1
To: <sip:h@sip3.voipvoip.com>;tag=d043eebb64d4461cc6a1e7afbfbe3159.3186
Call-ID: 73aaae5c719a2b302a6bf5580e3cd99d@sip3.voipvoip.com
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="voipvoip.com", nonce="4e7a6cb3bba2043b68c00941f8acdb4bb6bbce03"
Server: OpenSer (1.0.1 (i386/linux))
Content-Length: 0
Warning: 392 69.90.209.57:5060 "Noisy feedback tells: pid=7914 req_src_ip=209.191.122.70 req_src_port=5060 in_uri=sip:h@sip3.voipvoip.com out_uri=sip:h@sip3.voipvoip.com via_cnt==1"

<------------->
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: --- (10 headers 0 lines) ---
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: Transmitting (no NAT) to 69.90.209.57:5060:
ACK sip:h@sip3.voipvoip.com SIP/2.0
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK400d26c0
Max-Forwards: 70
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as4875aec1
To: <sip:h@sip3.voipvoip.com>;tag=d043eebb64d4461cc6a1e7afbfbe3159.3186
Contact: <sip:5551112222@209.191.122.70:5060>
Call-ID: 73aaae5c719a2b302a6bf5580e3cd99d@sip3.voipvoip.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.6.0
Content-Length: 0


---
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: Audio is at 5060
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: Reliably Transmitting (no NAT) to 69.90.209.57:5060:
INVITE sip:h@sip3.voipvoip.com SIP/2.0
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK1338be4b
Max-Forwards: 70
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as4875aec1
To: <sip:h@sip3.voipvoip.com>
Contact: <sip:5551112222@209.191.122.70:5060>
Call-ID: 73aaae5c719a2b302a6bf5580e3cd99d@sip3.voipvoip.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Proxy-Authorization: Digest username="5551112222", realm="voipvoip.com", algorithm=MD5, uri="sip:h@sip3.voipvoip.com", nonce="4e7a6cb3bba2043b68c00941f8acdb4bb6bbce03", response="70e2aebca94541d88b4893d1d9e785e8"
Date: Wed, 21 Sep 2011 22:58:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 1808997917 1808997918 IN IP4 209.191.122.70
s=Asterisk PBX 1.8.6.0
c=IN IP4 209.191.122.70
t=0 0
m=audio 14928 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK1338be4b
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as4875aec1
To: <sip:h@sip3.voipvoip.com>
Call-ID: 73aaae5c719a2b302a6bf5580e3cd99d@sip3.voipvoip.com
CSeq: 103 INVITE
Server: OpenSer (1.0.1 (i386/linux))
Content-Length: 0
Warning: 392 69.90.209.57:5060 "Noisy feedback tells: pid=7915 req_src_ip=209.191.122.70 req_src_port=5060 in_uri=sip:h@sip3.voipvoip.com out_uri=sip:h@sip3.voipvoip.com via_cnt==1"

<------------->
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: --- (9 headers 0 lines) ---
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK1338be4b
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as4875aec1
To: <sip:h@sip3.voipvoip.com>;tag=d043eebb64d4461cc6a1e7afbfbe3159.acf0
Call-ID: 73aaae5c719a2b302a6bf5580e3cd99d@sip3.voipvoip.com
CSeq: 103 INVITE
Server: OpenSer (1.0.1 (i386/linux))
Content-Length: 0
Warning: 392 69.90.209.57:5060 "Noisy feedback tells: pid=7915 req_src_ip=209.191.122.70 req_src_port=5060 in_uri=sip:h@sip3.voipvoip.com out_uri=sip:h@sip3.voipvoip.com via_cnt==1"

<------------->
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: --- (9 headers 0 lines) ---
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c:     -- Got SIP response 484 "Address Incomplete" back from 69.90.209.57:5060
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: Transmitting (no NAT) to 69.90.209.57:5060:
ACK sip:h@sip3.voipvoip.com SIP/2.0
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK1338be4b
Max-Forwards: 70
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as4875aec1
To: <sip:h@sip3.voipvoip.com>;tag=d043eebb64d4461cc6a1e7afbfbe3159.acf0
Contact: <sip:5551112222@209.191.122.70:5060>
Call-ID: 73aaae5c719a2b302a6bf5580e3cd99d@sip3.voipvoip.com
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.6.0
Content-Length: 0


---
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK1338be4b
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as4875aec1
To: <sip:h@sip3.voipvoip.com>
Call-ID: 73aaae5c719a2b302a6bf5580e3cd99d@sip3.voipvoip.com
CSeq: 103 INVITE
Server: OpenSer (1.0.1 (i386/linux))
Content-Length: 0
Warning: 392 69.90.209.57:5060 "Noisy feedback tells: pid=7915 req_src_ip=209.191.122.70 req_src_port=5060 in_uri=sip:h@sip3.voipvoip.com out_uri=sip:h@sip3.voipvoip.com via_cnt==1"

<------------->
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: --- (9 headers 0 lines) ---
[Sep 21 15:58:25] VERBOSE[2721] app_dial.c:   == Everyone is busy/congested at this time (1:0/0/1)
[Sep 21 15:58:25] VERBOSE[2721] pbx.c:     -- Executing [h@from-Lync_Server:2] Hangup("SIP/to-Lync_Server-0000000a", "") in new stack
[Sep 21 15:58:25] VERBOSE[2721] features.c:   == Spawn extension (from-Lync_Server, h, 2) exited non-zero on 'SIP/to-Lync_Server-0000000a'
[Sep 21 15:58:25] VERBOSE[2721] chan_sip.c: Scheduling destruction of SIP dialog '66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com' in 6400 ms (Method: INVITE)
[Sep 21 15:58:25] VERBOSE[2721] pbx.c:   == Spawn extension (from-Lync_Server, +18081234567, 1) exited non-zero on 'SIP/to-Lync_Server-0000000a'
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK482331f1
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 106 INVITE
Record-Route: <sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
Record-Route: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+18081234567@4.55.22.99:5060>
Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE
Content-Length: 231
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 11819 9676 IN IP4 4.55.22.99
s=SIP Media Capabilities
c=IN IP4 4.55.22.66
t=0 0
m=audio 28692 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: --- (14 headers 11 lines) ---
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: set_destination: Parsing <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on> for address/port to send to
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: set_destination: set destination to 69.90.209.57:5060
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: Transmitting (no NAT) to 69.90.209.57:5060:
ACK sip:+18081234567@4.55.22.99:5060 SIP/2.0
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK0bc38dbc
Route: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>,<sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
Max-Forwards: 70
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Contact: <sip:5551112222@209.191.122.70:5060>
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 106 ACK
User-Agent: Asterisk PBX 1.8.6.0
Content-Length: 0


---
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: set_destination: Parsing <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on> for address/port to send to
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: set_destination: set destination to 69.90.209.57:5060
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: Reliably Transmitting (no NAT) to 69.90.209.57:5060:
BYE sip:+18081234567@4.55.22.99:5060 SIP/2.0
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK0d4d714c
Route: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>,<sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
Max-Forwards: 70
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 107 BYE
User-Agent: Asterisk PBX 1.8.6.0
Proxy-Authorization: Digest username="5551112222", realm="voipvoip.com", algorithm=MD5, uri="sip:+18081234567@4.55.22.99:5060", nonce="4e7a6ca53cc9e0a7e2f7357bcdcd4dadc9324999", response="09914ed4053ab33a8496f62cb132438a"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: Scheduling destruction of SIP dialog '66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com' in 6400 ms (Method: INVITE)
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: Really destroying SIP dialog '73aaae5c719a2b302a6bf5580e3cd99d@sip3.voipvoip.com' Method: INVITE
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.191.122.70:5060;branch=z9hG4bK0d4d714c
From: "John Doe" <sip:5551112222@sip3.voipvoip.com>;tag=as6a38aa96
To: <sip:+18081234567@sip3.voipvoip.com>;tag=gK0c866797
Call-ID: 66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com
CSeq: 107 BYE
Record-Route: <sip:69.90.209.55:5960;lr=on;ftag=as6a38aa96>
Record-Route: <sip:69.90.209.57:5060;ftag=as6a38aa96;lr=on>
Content-Length: 0

<------------->
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: --- (9 headers 0 lines) ---
[Sep 21 15:58:25] VERBOSE[2496] chan_sip.c: Really destroying SIP dialog '66f5c4a737c95f8842902ce82a93c423@sip3.voipvoip.com' Method: INVITE
[Sep 21 15:58:28] VERBOSE[2496] chan_sip.c: 
<--- SIP read from UDP:69.90.209.57:5060 --->

<------------->
[Sep 21 15:58:31] VERBOSE[2496] chan_sip.c: Really destroying SIP dialog 'd57c1d34-e622-4fe9-b4a8-45121ab08d63' Method: BYE

#2

That’s not a codec problem, as the joined code gets to ulaw:

[quote][Sep 21 15:58:12] VERBOSE[2496] chan_sip.c: Capabilities: us - 0x40c (ulaw|alaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[/quote]

The problem is the redirect, asterisk is executing, visible here:

[quote][Sep 21 15:58:18] VERBOSE[2721] rtp_engine.c: – Remotely bridging SIP/to-Lync_Server-0000000a and SIP/to-VoIP_VoIP-0000000b
[/quote]

As SIP/to-Lync_Server resides on an internal IP (10.1.1.26) this rtp-bridge will fail as this IP is not reachable for the provider. You should set up the correct NAT-settings in sip.conf (nat=yes and directmedia=no), than it should work as expected (with voice).


#3

I haven’t looked carefully at this case, but most uses of nat=yes are wrong.

The reason this probably seems to work with X-Lite is that X-Lite doesn’t support re-invites. In fact, for at least some versions, it doesn’t support it so badly that the re-invite times out and Asterisk kills the call after 30 seconds.


#4

I don’t think it’s a NAT issue as the Asterisk server is actually dual homed (2 NIC’s). One NIC has an public IP and one NIC has an internal IP. If you look at the traffic I posted, you will see the following addresses used:

10.1.1.24 - Lync Server
10.1.1.26 - Asterisk Server
209.191.122.70 - Asterisk Server
69.90.209.57 - VoIP VoIP

You will see that Asterisk communicates with Lync as 10.1.1.26 while communicating with VoIP VoIP as 209.191.122.70. There is no NAT involved with any communication between Asterisk<=>Lync and Asterisk<=>VoIPVoIP.


#5

Things that stand out in the trace in reversed order of relevance:

  1. ‘Audio is at 5060’ This is probably asterisk bug, hopefully it only affects logging not call establishment.

  2. There is a problem with dialplan:
    [Sep 21 15:58:25] INVITE sip:h@sip3.voipvoip.com SIP/2.0
    extension h should terminate the call, not attempt to dial. I suspect the dialplan is written as:
    exten => _., blah
    instead of
    exten => _X.,blah

  3. as already pointed by abw1oim, you should change media handling to prevent remote bridging.
    I am not sure what the right option in 1.8 is but I would try

directmediadeny=0.0.0.0/0

More options: asterisk.org/doxygen/trunk/Config_sip.html