Integration with Lync Server 2010

Asterisk Asterisk Support
1

Hi,
We have configured Lync Server 2010 and integrated this with the Asterisk PBX to forward external calls to our outside SIP Trunk in order to get connectivity.

This works good in the following way.
Place call from a Cellphone to a Lync extention
Place call from a Lync client to a Cellphone or another phone outside our network.
Conferencing Dial-in.

When we try to place a call using the Lync Mobile Client to an outside connection we just get a drop of the phone call.

What happends?
In Lync client we enter the desired number to call.
After about 5 seconds Lync calls us on the number given in the client using our SIP Trunk.
When we answer the call it should be forwarded to the external phone we want to place the call to.
This is not happening, the Lync client tells us that the other part is in a call or unreachable.

If we look at the logs in Asterisk we get the following.

== Using SIP RTP CoS mark 5
– Executing [+46xxxxxxx@from-lync:1] Dial(“SIP/Lync_Trunk-0000023e”, “SIP/+46xxxxx@external-trunk-name”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/+46xxxxxx@external-trunk-name
– SIP/external-trunk-name-0000023f is making progress passing it to SIP/Lync_Trunk-0000023e
– SIP/external-trunk-name-0000023f is ringing
– SIP/external-trunk-name-0000023f answered SIP/Lync_Trunk-0000023e

Remotely bridging SIP/Lync_Trunk-0000023e and SIP/external-trunk-name-0000023f
[Oct 16 18:04:07] WARNING[27276]: chan_sip.c:20497 handle_response_invite: just did sched_add waitid(277661) for sip_reinvite_retry for dialog 5307d74a-14b7-4112-90b8-af05da9ea8f9 in handle_response_invite
– Executing [h@from-lync:1] Dial(“SIP/Lync_Trunk-0000023e”, “SIP/h@external-trunk-name”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/h@external-trunk-name
== Spawn extension (from-lync, h, 1) exited non-zero on ‘SIP/Lync_Trunk-0000023e’
== Spawn extension (from-lync, +46xxxxxx, 1) exited non-zero on ‘SIP/Lync_Trunk-0000023e’
[Oct 16 18:04:07] WARNING[3179]: chan_sip.c:20406 handle_response_invite: Received response: “Forbidden” from ‘“User” sip:+46xxxxxxx@ip-of-Asteriskbox;tag=as0fc944b3’

Since Asterisk is trying to send h@external-trunk-name i suppose that this is why the external SIP trunk cant connect the call to the correct number.

What i know (from googling around) the h stands for hangup, so why is Asterisk trying to dial the h instead of sending the termination command?

Is there anyway to translate the variable h to the number that Lync is trying to bridge to?

Thanks in advance :smiley:

2

h is run when the calling party hangs up, or is hung up because it has run out of dialplan, or a called party cleared without any option to continue the dialplan being set. It is not allowed to do Dial, as it is running on a channel that is no longer up, so you have an invalid dialplan.

You will need to provide sip debug output to seem more precisely what is going wrong, but it looks like something doesn’t like re-invites.

3

Hi,

I think i have manage to get the sip debug output.
Is this telling you something about what can be wrong?

if you find something, where did you find it?

[Oct 16 23:50:55] VERBOSE[28565] config.c: == Parsing ‘/etc/asterisk/logger.conf’: [Oct 16 23:50:55] VERBOSE[28565] config.c: == Found
[Oct 16 23:50:55] VERBOSE[28565] logger.c: Asterisk Queue Logger restarted
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c:
<— SIP read from TCP:lync-ipaddress:56086 —>
INVITE sip:+46xxxxxxx@asterisk-ip;user=phone SIP/2.0
FROM: "David Lundqvist"sip:+46lyncphonenumber@lyncpool.fqdn;user=phone;epid=696CEF0FC3;tag=99e3cc3498
TO: sip:+46xxxxxxx@asterisk-ip;user=phone
CSEQ: 82508 INVITE
CALL-ID: ab8f6b05-9adf-4b7e-9639-f339670e4217
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP lync-ipaddress:56086;branch=z9hG4bKef2f4ac8
CONTACT: sip:lyncpool.fqdn:5060;transport=Tcp;maddr=lync-ipaddress;ms-opaque=99ea06d8ffc69dd3
CONTENT-LENGTH: 340
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=- 200 1 IN IP4 lync-ipaddress
s=session
c=IN IP4 lync-ipaddress
b=CT:1000
t=0 0
m=audio 56660 RTP/AVP 97 101 13 0 8
c=IN IP4 lync-ipaddress
a=rtcp:56661
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
<------------->
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: — (14 headers 18 lines) —
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Sending to lync-ipaddress:56086 (NAT)
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Using INVITE request as basis request - ab8f6b05-9adf-4b7e-9639-f339670e4217
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Found peer ‘Lync_Trunk’ for ‘+46lyncphonenumber’ from lync-ipaddress:56086
[Oct 16 23:51:15] VERBOSE[28567] netsock2.c: == Using SIP RTP CoS mark 5
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Found RTP audio format 97
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Found RTP audio format 101
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Found RTP audio format 13
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Found RTP audio format 0
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Found RTP audio format 8
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Found unknown media description format RED for ID 97
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Found audio description format telephone-event for ID 101
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Found audio description format CN for ID 13
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Found audio description format PCMU for ID 0
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Found audio description format PCMA for ID 8
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Peer audio RTP is at port lync-ipaddress:56660
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: Looking for +46xxxxxxx in from-lync (domain asterisk-ip)
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c: list_route: hop: sip:lyncpool.fqdn:5060;transport=Tcp;maddr=lync-ipaddress;ms-opaque=99ea06d8ffc69dd3
[Oct 16 23:51:15] VERBOSE[28567] chan_sip.c:
<— Transmitting (NAT) to lync-ipaddress:56086 —>
SIP/2.0 100 Trying
Via: SIP/2.0/TCP lync-ipaddress:56086;branch=z9hG4bKef2f4ac8;received=lync-ipaddress;rport=56086
From: "David Lundqvist"sip:+46lyncphonenumber@lyncpool.fqdn;user=phone;epid=696CEF0FC3;tag=99e3cc3498
To: sip:+46xxxxxxx@asterisk-ip;user=phone
Call-ID: ab8f6b05-9adf-4b7e-9639-f339670e4217
CSeq: 82508 INVITE
Server: Asterisk PBX 1.8.14.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:+46xxxxxxx@asterisk-ip:5060;transport=TCP
Content-Length: 0

<------------>
[Oct 16 23:51:15] VERBOSE[28568] pbx.c: – Executing [+46xxxxxxx@from-lync:1] Dial(“SIP/Lync_Trunk-0000024a”, “SIP/+46xxxxxxx@outside.sip.trunk”) in new stack
[Oct 16 23:51:15] VERBOSE[28568] netsock2.c: == Using SIP RTP CoS mark 5
[Oct 16 23:51:15] VERBOSE[28568] chan_sip.c: Audio is at 12370
[Oct 16 23:51:15] VERBOSE[28568] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Oct 16 23:51:15] VERBOSE[28568] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[Oct 16 23:51:15] VERBOSE[28568] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Oct 16 23:51:15] VERBOSE[28568] chan_sip.c: Adding codec 0x800000000000 (testlaw) to SDP
[Oct 16 23:51:15] VERBOSE[28568] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 16 23:51:15] VERBOSE[28568] chan_sip.c: Reliably Transmitting (NAT) to external.ip:5060:
INVITE sip:+46xxxxxxx@outside.sip.trunk SIP/2.0
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK13da1ecc;rport
Max-Forwards: 70
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk
Contact: sip:+46lyncphonenumber@asterisk-ip:5060
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.14.1
Date: Tue, 16 Oct 2012 21:51:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 571805329 571805329 IN IP4 asterisk-ip
s=Asterisk PBX 1.8.14.1
c=IN IP4 asterisk-ip
t=0 0
m=audio 12370 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[Oct 16 23:51:15] VERBOSE[28568] app_dial.c: – Called SIP/+46xxxxxxx@outside.sip.trunk
[Oct 16 23:51:16] VERBOSE[28569] chan_sip.c:
<— SIP read from TCP:lync-ipaddress:56088 —>
OPTIONS sip:asterisk-ip SIP/2.0
FROM: sip:lyncpool.fqdn:5060;transport=Tcp;ms-opaque=99ea06d8ffc69dd3;epid=696CEF0FC3;tag=973fb086
TO: sip:asterisk-ip
CSEQ: 82509 OPTIONS
CALL-ID: f2fbdd9d7dd1415fa978a96ef9c36669
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP lync-ipaddress:56088;branch=z9hG4bK9735436f
CONTACT: sip:lyncpool.fqdn:5060;transport=Tcp;maddr=lync-ipaddress
CONTENT-LENGTH: 0
USER-AGENT: RTCC/4.0.0.0 MediationServer

<------------->
[Oct 16 23:51:16] VERBOSE[28569] chan_sip.c: — (10 headers 0 lines) —
[Oct 16 23:51:16] VERBOSE[28569] chan_sip.c: Looking for s in default (domain asterisk-ip)
[Oct 16 23:51:16] VERBOSE[28569] chan_sip.c:
<— Transmitting (NAT) to lync-ipaddress:56088 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP lync-ipaddress:56088;branch=z9hG4bK9735436f;received=lync-ipaddress;rport=56088
From: sip:lyncpool.fqdn:5060;transport=Tcp;ms-opaque=99ea06d8ffc69dd3;epid=696CEF0FC3;tag=973fb086
To: sip:asterisk-ip;tag=as3e10e4da
Call-ID: f2fbdd9d7dd1415fa978a96ef9c36669
CSeq: 82509 OPTIONS
Server: Asterisk PBX 1.8.14.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
[Oct 16 23:51:16] VERBOSE[28569] chan_sip.c: Scheduling destruction of SIP dialog ‘f2fbdd9d7dd1415fa978a96ef9c36669’ in 32000 ms (Method: OPTIONS)
[Oct 16 23:51:16] VERBOSE[3179] chan_sip.c:
<— SIP read from UDP:external.ip:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK13da1ecc
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 102 INVITE
Server: Kamailio at IP-Only
Content-Length: 0

<------------->
[Oct 16 23:51:16] VERBOSE[3179] chan_sip.c: — (8 headers 0 lines) —
[Oct 16 23:51:19] VERBOSE[3179] chan_sip.c:
<— SIP read from UDP:external.ip:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK13da1ecc
Record-Route: sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY-
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk;tag=227084569
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 102 INVITE
Contact: sip:0709555078@sip-gw.ip-only.net:5060
Content-Type: application/sdp
Content-Length: 352

v=0
o=- 46639805 0 IN IP4 213.132.106.6
s=Cisco SDP 0
c=IN IP4 213.132.106.6
t=0 0
m=audio 18988 RTP/AVP 8 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
<------------->
[Oct 16 23:51:19] VERBOSE[3179] chan_sip.c: — (10 headers 15 lines) —
[Oct 16 23:51:19] VERBOSE[3179] chan_sip.c: list_route: hop: sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY-
[Oct 16 23:51:19] VERBOSE[3179] chan_sip.c: Found RTP audio format 8
[Oct 16 23:51:19] VERBOSE[3179] chan_sip.c: Found RTP audio format 101
[Oct 16 23:51:19] VERBOSE[3179] chan_sip.c: Found RTP audio format 100
[Oct 16 23:51:19] VERBOSE[3179] chan_sip.c: Found audio description format telephone-event for ID 101
[Oct 16 23:51:19] VERBOSE[3179] chan_sip.c: Found unknown media description format X-NSE for ID 100
[Oct 16 23:51:19] VERBOSE[3179] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Oct 16 23:51:19] VERBOSE[3179] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Oct 16 23:51:19] VERBOSE[3179] chan_sip.c: Peer audio RTP is at port 213.132.106.6:18988
[Oct 16 23:51:19] VERBOSE[28568] app_dial.c: – SIP/outside.sip.trunk-0000024b is making progress passing it to SIP/Lync_Trunk-0000024a
[Oct 16 23:51:19] VERBOSE[28568] chan_sip.c: Audio is at 19394
[Oct 16 23:51:19] VERBOSE[28568] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Oct 16 23:51:19] VERBOSE[28568] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Oct 16 23:51:19] VERBOSE[28568] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 16 23:51:19] VERBOSE[28568] chan_sip.c:
<— Transmitting (NAT) to lync-ipaddress:56086 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP lync-ipaddress:56086;branch=z9hG4bKef2f4ac8;received=lync-ipaddress;rport=56086
From: "David Lundqvist"sip:+46lyncphonenumber@lyncpool.fqdn;user=phone;epid=696CEF0FC3;tag=99e3cc3498
To: sip:+46xxxxxxx@asterisk-ip;user=phone;tag=as1b695f3f
Call-ID: ab8f6b05-9adf-4b7e-9639-f339670e4217
CSeq: 82508 INVITE
Server: Asterisk PBX 1.8.14.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:+46xxxxxxx@asterisk-ip:5060;transport=TCP
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 33029600 33029600 IN IP4 asterisk-ip
s=Asterisk PBX 1.8.14.1
c=IN IP4 asterisk-ip
t=0 0
m=audio 19394 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
[Oct 16 23:51:19] VERBOSE[3179] chan_sip.c:
<— SIP read from UDP:external.ip:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK13da1ecc
Record-Route: sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY-
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk;tag=227084569
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 102 INVITE
Contact: sip:0709555078@sip-gw.ip-only.net:5060
Content-Length: 0

<------------->
[Oct 16 23:51:19] VERBOSE[3179] chan_sip.c: — (9 headers 0 lines) —
[Oct 16 23:51:19] VERBOSE[3179] chan_sip.c: list_route: hop: sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY-
[Oct 16 23:51:19] VERBOSE[28568] app_dial.c: – SIP/outside.sip.trunk-0000024b is ringing
[Oct 16 23:51:19] VERBOSE[28568] chan_sip.c:
<— Transmitting (NAT) to lync-ipaddress:56086 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP lync-ipaddress:56086;branch=z9hG4bKef2f4ac8;received=lync-ipaddress;rport=56086
From: "David Lundqvist"sip:+46lyncphonenumber@lyncpool.fqdn;user=phone;epid=696CEF0FC3;tag=99e3cc3498
To: sip:+46xxxxxxx@asterisk-ip;user=phone;tag=as1b695f3f
Call-ID: ab8f6b05-9adf-4b7e-9639-f339670e4217
CSeq: 82508 INVITE
Server: Asterisk PBX 1.8.14.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:+46xxxxxxx@asterisk-ip:5060;transport=TCP
Content-Length: 0

<------------>
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c:
<— SIP read from UDP:external.ip:5060 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK13da1ecc
Record-Route: sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY-
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk;tag=227084569
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 102 INVITE
Contact: sip:0709555078@sip-gw.ip-only.net:5060
Supported: timer
Session-expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, BYE, CANCEL
Content-Type: application/sdp
Content-Length: 352

v=0
o=- 46639805 0 IN IP4 213.132.106.6
s=Cisco SDP 0
c=IN IP4 213.132.106.6
t=0 0
m=audio 18988 RTP/AVP 8 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
<------------->
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: — (14 headers 15 lines) —
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: list_route: hop: sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY-
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: set_destination: Parsing sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY- for address/port to send to
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: set_destination: set destination to external.ip:5060
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: Transmitting (NAT) to external.ip:5060:
ACK sip:0709555078@sip-gw.ip-only.net:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK09da02d2;rport
Route: sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY-
Max-Forwards: 70
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk;tag=227084569
Contact: sip:+46lyncphonenumber@asterisk-ip:5060
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.14.1
Content-Length: 0

[Oct 16 23:51:23] VERBOSE[28568] app_dial.c: – SIP/outside.sip.trunk-0000024b answered SIP/Lync_Trunk-0000024a
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Audio is at 19394
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c:
<— Reliably Transmitting (NAT) to lync-ipaddress:56086 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP lync-ipaddress:56086;branch=z9hG4bKef2f4ac8;received=lync-ipaddress;rport=56086
From: "David Lundqvist"sip:+46lyncphonenumber@lyncpool.fqdn;user=phone;epid=696CEF0FC3;tag=99e3cc3498
To: sip:+46xxxxxxx@asterisk-ip;user=phone;tag=as1b695f3f
Call-ID: ab8f6b05-9adf-4b7e-9639-f339670e4217
CSeq: 82508 INVITE
Server: Asterisk PBX 1.8.14.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:+46xxxxxxx@asterisk-ip:5060;transport=TCP
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 33029600 33029601 IN IP4 asterisk-ip
s=Asterisk PBX 1.8.14.1
c=IN IP4 asterisk-ip
t=0 0
m=audio 19394 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
[Oct 16 23:51:23] VERBOSE[28568] rtp_engine.c: – Remotely bridging SIP/Lync_Trunk-0000024a and SIP/outside.sip.trunk-0000024b
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: set_destination: Parsing sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY- for address/port to send to
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: set_destination: set destination to external.ip:5060
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Audio is at 12370
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Reliably Transmitting (NAT) to external.ip:5060:
INVITE sip:0709555078@sip-gw.ip-only.net:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK4c2d178e;rport
Route: sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY-
Max-Forwards: 70
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk;tag=227084569
Contact: sip:+46lyncphonenumber@asterisk-ip:5060
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.14.1
Session-Expires: 1800;refresher=uac
Min-SE: 90
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: 235

v=0
o=root 571805329 571805330 IN IP4 lync-ipaddress
s=Asterisk PBX 1.8.14.1
c=IN IP4 lync-ipaddress
t=0 0
m=audio 56660 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c:
<— SIP read from TCP:lync-ipaddress:56086 —>
ACK sip:+46xxxxxxx@asterisk-ip:5060;transport=TCP SIP/2.0
FROM: sip:+46lyncphonenumber@lyncpool.fqdn;user=phone;epid=696CEF0FC3;tag=99e3cc3498
TO: sip:+46xxxxxxx@asterisk-ip;user=phone;tag=as1b695f3f
CSEQ: 82508 ACK
CALL-ID: ab8f6b05-9adf-4b7e-9639-f339670e4217
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP lync-ipaddress:56086;branch=z9hG4bKb78c716e
CONTENT-LENGTH: 0
USER-AGENT: RTCC/4.0.0.0 MediationServer

<------------->
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c: — (9 headers 0 lines) —
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c: set_destination: Parsing sip:lyncpool.fqdn:5060;transport=Tcp;maddr=lync-ipaddress;ms-opaque=99ea06d8ffc69dd3 for address/port to send to
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c: set_destination: set destination to lync-ipaddress:5060
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c: Audio is at 19394
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c: Reliably Transmitting (NAT) to lync-ipaddress:56086:
INVITE sip:lyncpool.fqdn:5060;transport=Tcp;maddr=lync-ipaddress;ms-opaque=99ea06d8ffc69dd3 SIP/2.0
Via: SIP/2.0/TCP asterisk-ip:5060;branch=z9hG4bK0c9e0215;rport
Max-Forwards: 70
From: sip:+46xxxxxxx@asterisk-ip;user=phone;tag=as1b695f3f
To: "David Lundqvist"sip:+46lyncphonenumber@lyncpool.fqdn;user=phone;epid=696CEF0FC3;tag=99e3cc3498
Contact: sip:+46xxxxxxx@asterisk-ip:5060;transport=TCP
Call-ID: ab8f6b05-9adf-4b7e-9639-f339670e4217
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.14.1
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: 235

v=0
o=root 33029600 33029602 IN IP4 213.132.106.6
s=Asterisk PBX 1.8.14.1
c=IN IP4 213.132.106.6
t=0 0
m=audio 18988 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c:
<— SIP read from TCP:lync-ipaddress:56086 —>
SIP/2.0 100 Trying
FROM: sip:+46xxxxxxx@asterisk-ip;user=phone;tag=as1b695f3f
TO: "David Lundqvist"sip:+46lyncphonenumber@lyncpool.fqdn;user=phone;tag=99e3cc3498;epid=696CEF0FC3
CSEQ: 102 INVITE
CALL-ID: ab8f6b05-9adf-4b7e-9639-f339670e4217
VIA: SIP/2.0/TCP asterisk-ip:5060;branch=z9hG4bK0c9e0215;rport
CONTENT-LENGTH: 0

<------------->
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c: — (7 headers 0 lines) —
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c:
<— SIP read from UDP:external.ip:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK4c2d178e
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk;tag=227084569
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 103 INVITE
Server: Kamailio at IP-Only
Content-Length: 0

<------------->
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: — (8 headers 0 lines) —
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c:
<— SIP read from TCP:lync-ipaddress:56086 —>
SIP/2.0 491 Proxy side reinvite failed, pass result to GW.
FROM: sip:+46xxxxxxx@asterisk-ip;user=phone;tag=as1b695f3f
TO: "David Lundqvist"sip:+46lyncphonenumber@lyncpool.fqdn;user=phone;epid=696CEF0FC3;tag=99e3cc3498
CSEQ: 102 INVITE
CALL-ID: ab8f6b05-9adf-4b7e-9639-f339670e4217
VIA: SIP/2.0/TCP asterisk-ip:5060;branch=z9hG4bK0c9e0215;rport
CONTENT-LENGTH: 0
SUPPORTED: 100rel
SERVER: RTCC/4.0.0.0 MediationServer

<------------->
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c: — (9 headers 0 lines) —
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c: set_destination: Parsing sip:lyncpool.fqdn:5060;transport=Tcp;maddr=lync-ipaddress;ms-opaque=99ea06d8ffc69dd3 for address/port to send to
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c: set_destination: set destination to lync-ipaddress:5060
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c: Transmitting (NAT) to lync-ipaddress:56086:
ACK sip:lyncpool.fqdn:5060;transport=Tcp;maddr=lync-ipaddress;ms-opaque=99ea06d8ffc69dd3 SIP/2.0
Via: SIP/2.0/TCP asterisk-ip:5060;branch=z9hG4bK0c9e0215;rport
Max-Forwards: 70
From: sip:+46xxxxxxx@asterisk-ip;user=phone;tag=as1b695f3f
To: "David Lundqvist"sip:+46lyncphonenumber@lyncpool.fqdn;user=phone;epid=696CEF0FC3;tag=99e3cc3498
Contact: sip:+46xxxxxxx@asterisk-ip:5060;transport=TCP
Call-ID: ab8f6b05-9adf-4b7e-9639-f339670e4217
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.14.1
Content-Length: 0

[Oct 16 23:51:23] WARNING[28567] chan_sip.c: just did sched_add waitid(278953) for sip_reinvite_retry for dialog ab8f6b05-9adf-4b7e-9639-f339670e4217 in handle_response_invite
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c:
<— SIP read from UDP:external.ip:5060 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK4c2d178e
Record-Route: sip:external.ip;lr=on;ftag=as52c79849
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk;tag=227084569
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 103 INVITE
Contact: sip:0709555078@sip-gw.ip-only.net:5060
Supported: timer
Session-expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, BYE, CANCEL
Content-Type: application/sdp
Content-Length: 352

v=0
o=- 46639805 1 IN IP4 213.132.106.6
s=Cisco SDP 0
c=IN IP4 213.132.106.6
t=0 0
m=audio 18988 RTP/AVP 8 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
<------------->
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: — (14 headers 15 lines) —
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: Found RTP audio format 8
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: Found RTP audio format 101
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: Found RTP audio format 100
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: Found audio description format telephone-event for ID 101
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: Found unknown media description format X-NSE for ID 100
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: Peer audio RTP is at port 213.132.106.6:18988
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: set_destination: Parsing sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY- for address/port to send to
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: set_destination: set destination to external.ip:5060
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: Transmitting (NAT) to external.ip:5060:
ACK sip:0709555078@sip-gw.ip-only.net:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK693994d9;rport
Route: sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY-
Max-Forwards: 70
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk;tag=227084569
Contact: sip:+46lyncphonenumber@asterisk-ip:5060
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.14.1
Content-Length: 0

[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c:
<— SIP read from TCP:lync-ipaddress:56086 —>
BYE sip:+46xxxxxxx@asterisk-ip:5060;transport=TCP SIP/2.0
FROM: sip:+46lyncphonenumber@lyncpool.fqdn;user=phone;epid=696CEF0FC3;tag=99e3cc3498
TO: sip:+46xxxxxxx@asterisk-ip;user=phone;tag=as1b695f3f
CSEQ: 82509 BYE
CALL-ID: ab8f6b05-9adf-4b7e-9639-f339670e4217
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP lync-ipaddress:56086;branch=z9hG4bKec593290
CONTACT: sip:lyncpool.fqdn:5060;transport=Tcp;maddr=lync-ipaddress
CONTENT-LENGTH: 0
USER-AGENT: RTCC/4.0.0.0 MediationServer

<------------->
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c: — (10 headers 0 lines) —
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c: Sending to lync-ipaddress:56086 (NAT)
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c: Scheduling destruction of SIP dialog ‘ab8f6b05-9adf-4b7e-9639-f339670e4217’ in 6400 ms (Method: BYE)
[Oct 16 23:51:23] VERBOSE[28567] chan_sip.c:
<— Transmitting (NAT) to lync-ipaddress:56086 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP lync-ipaddress:56086;branch=z9hG4bKec593290;received=lync-ipaddress;rport=56086
From: sip:+46lyncphonenumber@lyncpool.fqdn;user=phone;epid=696CEF0FC3;tag=99e3cc3498
To: sip:+46xxxxxxx@asterisk-ip;user=phone;tag=as1b695f3f
Call-ID: ab8f6b05-9adf-4b7e-9639-f339670e4217
CSeq: 82509 BYE
Server: Asterisk PBX 1.8.14.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: set_destination: Parsing sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY- for address/port to send to
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: set_destination: set destination to external.ip:5060
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Audio is at 12370
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Reliably Transmitting (NAT) to external.ip:5060:
INVITE sip:0709555078@sip-gw.ip-only.net:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK765017d6;rport
Route: sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY-
Max-Forwards: 70
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk;tag=227084569
Contact: sip:+46lyncphonenumber@asterisk-ip:5060
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.8.14.1
Session-Expires: 1800;refresher=uac
Min-SE: 90
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: 241

v=0
o=root 571805329 571805331 IN IP4 asterisk-ip
s=Asterisk PBX 1.8.14.1
c=IN IP4 asterisk-ip
t=0 0
m=audio 12370 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[Oct 16 23:51:23] VERBOSE[28568] pbx.c: – Executing [h@from-lync:1] Dial(“SIP/Lync_Trunk-0000024a”, “SIP/h@outside.sip.trunk”) in new stack
[Oct 16 23:51:23] VERBOSE[28568] netsock2.c: == Using SIP RTP CoS mark 5
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Audio is at 10776
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Adding codec 0x800000000000 (testlaw) to SDP
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Reliably Transmitting (NAT) to 82.99.32.90:5060:
INVITE sip:h@outside.sip.trunk SIP/2.0
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK4923f8d3;rport
Max-Forwards: 70
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as56a9325a
To: sip:h@outside.sip.trunk
Contact: sip:+46lyncphonenumber@asterisk-ip:5060
Call-ID: 73a4d6e1344f6fdd4ec5d419361d04f3@asterisk-ip:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.14.1
Date: Tue, 16 Oct 2012 21:51:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1404944550 1404944550 IN IP4 asterisk-ip
s=Asterisk PBX 1.8.14.1
c=IN IP4 asterisk-ip
t=0 0
m=audio 10776 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[Oct 16 23:51:23] VERBOSE[28568] app_dial.c: – Called SIP/h@outside.sip.trunk
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Scheduling destruction of SIP dialog ‘73a4d6e1344f6fdd4ec5d419361d04f3@asterisk-ip:5060’ in 32000 ms (Method: INVITE)
[Oct 16 23:51:23] VERBOSE[28568] features.c: == Spawn extension (from-lync, h, 1) exited non-zero on ‘SIP/Lync_Trunk-0000024a’
[Oct 16 23:51:23] VERBOSE[28568] chan_sip.c: Scheduling destruction of SIP dialog ‘49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060’ in 32000 ms (Method: INVITE)
[Oct 16 23:51:23] VERBOSE[28568] pbx.c: == Spawn extension (from-lync, +46xxxxxxx, 1) exited non-zero on ‘SIP/Lync_Trunk-0000024a’
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c:
<— SIP read from UDP:external.ip:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK765017d6
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk;tag=227084569
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 104 INVITE
Server: Kamailio at IP-Only
Content-Length: 0

<------------->
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: — (8 headers 0 lines) —
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c:
<— SIP read from UDP:82.99.32.90:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK4923f8d3
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as56a9325a
To: sip:h@outside.sip.trunk;tag=5c7119ba1804add736967138edd37679.c0f5
Call-ID: 73a4d6e1344f6fdd4ec5d419361d04f3@asterisk-ip:5060
CSeq: 102 INVITE
Server: Kamailio at IP-Only
Content-Length: 0

<------------->
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: — (8 headers 0 lines) —
[Oct 16 23:51:23] VERBOSE[3179] chan_sip.c: Transmitting (NAT) to 82.99.32.90:5060:
ACK sip:h@outside.sip.trunk SIP/2.0
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK4923f8d3;rport
Max-Forwards: 70
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as56a9325a
To: sip:h@outside.sip.trunk;tag=5c7119ba1804add736967138edd37679.c0f5
Contact: sip:+46lyncphonenumber@asterisk-ip:5060
Call-ID: 73a4d6e1344f6fdd4ec5d419361d04f3@asterisk-ip:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.14.1
Content-Length: 0

[Oct 16 23:51:23] WARNING[3179] chan_sip.c: Received response: “Forbidden” from ‘“David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as56a9325a’
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c:
<— SIP read from UDP:external.ip:5060 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK765017d6
Record-Route: sip:external.ip;lr=on;ftag=as52c79849
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk;tag=227084569
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 104 INVITE
Contact: sip:0709555078@sip-gw.ip-only.net:5060
Supported: timer
Session-expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, BYE, CANCEL
Content-Type: application/sdp
Content-Length: 352

v=0
o=- 46639805 2 IN IP4 213.132.106.6
s=Cisco SDP 0
c=IN IP4 213.132.106.6
t=0 0
m=audio 18988 RTP/AVP 8 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
<------------->
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: — (14 headers 15 lines) —
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Found RTP audio format 8
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Found RTP audio format 101
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Found RTP audio format 100
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Found audio description format telephone-event for ID 101
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Found unknown media description format X-NSE for ID 100
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Peer audio RTP is at port 213.132.106.6:18988
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: set_destination: Parsing sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY- for address/port to send to
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: set_destination: set destination to external.ip:5060
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Transmitting (NAT) to external.ip:5060:
ACK sip:0709555078@sip-gw.ip-only.net:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK0a39c446;rport
Route: sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY-
Max-Forwards: 70
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk;tag=227084569
Contact: sip:+46lyncphonenumber@asterisk-ip:5060
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.8.14.1
Content-Length: 0

[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: set_destination: Parsing sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY- for address/port to send to
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: set_destination: set destination to external.ip:5060
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Reliably Transmitting (NAT) to external.ip:5060:
BYE sip:0709555078@sip-gw.ip-only.net:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK43fd9540;rport
Route: sip:external.ip;lr=on;ftag=as52c79849;d=93b.3fac3fe3;vsf=AAAAABsAAAEABQYDAgtyCnEAHwMfAx8DFgAWABgHNjY-
Max-Forwards: 70
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk;tag=227084569
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 105 BYE
User-Agent: Asterisk PBX 1.8.14.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Scheduling destruction of SIP dialog ‘49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060’ in 32000 ms (Method: INVITE)
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c:
<— SIP read from UDP:external.ip:5060 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP asterisk-ip:5060;branch=z9hG4bK43fd9540
From: “David Lundqvist” sip:+46lyncphonenumber@asterisk-ip;tag=as52c79849
To: sip:+46xxxxxxx@outside.sip.trunk;tag=227084569
Call-ID: 49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060
CSeq: 105 BYE
Content-Length: 0

<------------->
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: — (7 headers 0 lines) —
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Really destroying SIP dialog ‘49dae6066041ed8f5a00e1c2769bcf90@asterisk-ip:5060’ Method: INVITE
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: set_destination: Parsing sip:lyncpool.fqdn:5060;transport=Tcp;maddr=lync-ipaddress;ms-opaque=99ea06d8ffc69dd3 for address/port to send to
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: set_destination: set destination to lync-ipaddress:5060
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Audio is at 19394
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 16 23:51:24] VERBOSE[3179] chan_sip.c: Reliably Transmitting (NAT) to lync-ipaddress:56086:
INVITE sip:lyncpool.fqdn:5060;transport=Tcp;maddr=lync-ipaddress;ms-opaque=99ea06d8ffc69dd3 SIP/2.0
Via: SIP/2.0/TCP asterisk-ip:5060;branch=z9hG4bK7e975efa;rport
Max-Forwards: 70
From: sip:+46lyncphonenumber@lyncpool.fqdn;user=phone;epid=696CEF0FC3;tag=99e3cc3498
To: sip:+46xxxxxxx@asterisk-ip;user=phone;tag=as1b695f3f
Contact: sip:+46xxxxxxx@asterisk-ip:5060;transport=TCP
Call-ID: ab8f6b05-9adf-4b7e-9639-f339670e4217
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.14.1
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: 262

v=0
o=root 33029600 33029603 IN IP4 213.132.106.6
s=Asterisk PBX 1.8.14.1
c=IN IP4 213.132.106.6
t=0 0
m=audio 18988 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

[Oct 16 23:51:24] VERBOSE[28567] chan_sip.c:
<— SIP read from TCP:lync-ipaddress:56086 —>
SIP/2.0 100 Trying
FROM: sip:+46lyncphonenumber@lyncpool.fqdn;user=phone;tag=99e3cc3498;epid=696CEF0FC3
TO: sip:+46xxxxxxx@asterisk-ip;user=phone;tag=as1b695f3f
CSEQ: 103 INVITE
CALL-ID: ab8f6b05-9adf-4b7e-9639-f339670e4217
VIA: SIP/2.0/TCP asterisk-ip:5060;branch=z9hG4bK7e975efa;rport
CONTENT-LENGTH: 0

<------------->
[Oct 16 23:51:24] VERBOSE[28567] chan_sip.c: — (7 headers 0 lines) —
[Oct 16 23:51:24] VERBOSE[28567] chan_sip.c:
<— SIP read from TCP:lync-ipaddress:56086 —>
SIP/2.0 481 Call Leg/Transaction Does Not Exist
FROM: sip:+46lyncphonenumber@lyncpool.fqdn;user=phone;tag=99e3cc3498;epid=696CEF0FC3
TO: sip:+46xxxxxxx@asterisk-ip;user=phone;tag=as1b695f3f
CSEQ: 103 INVITE
CALL-ID: ab8f6b05-9adf-4b7e-9639-f339670e4217
VIA: SIP/2.0/TCP asterisk-ip:5060;branch=z9hG4bK7e975efa;rport
CONTENT-LENGTH: 0
SERVER: RTCC/4.0.0.0

<------------->
[Oct 16 23:51:24] VERBOSE[28567] chan_sip.c: — (8 headers 0 lines) —
[Oct 16 23:51:30] VERBOSE[3179] chan_sip.c: Really destroying SIP dialog ‘ab8f6b05-9adf-4b7e-9639-f339670e4217’ Method: BYE
[Oct 16 23:51:40] VERBOSE[3179] chan_sip.c: Really destroying SIP dialog ‘770dd01a395345ecb57154e3c01d7fd7’ Method: OPTIONS
[Oct 16 23:51:48] VERBOSE[3179] chan_sip.c: Really destroying SIP dialog ‘f2fbdd9d7dd1415fa978a96ef9c36669’ Method: OPTIONS
[Oct 16 23:51:48] VERBOSE[28565] asterisk.c: – Remote UNIX connection disconnected
[Oct 16 23:51:48] VERBOSE[3179] chan_sip.c: Reliably Transmitting (NAT) to lync-ipaddress:5060:
OPTIONS sip:lync-ipaddress SIP/2.0
Via: SIP/2.0/TCP asterisk-ip:5060;branch=z9hG4bK58c3dc07;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@asterisk-ip;tag=as1636fcb5
To: sip:lync-ipaddress
Contact: sip:asterisk@asterisk-ip:5060;transport=TCP
Call-ID: 6ea5147727a13a6e3d24defb3491d6e1@asterisk-ip:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.14.1
Date: Tue, 16 Oct 2012 21:51:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

[Oct 16 23:51:48] VERBOSE[8109] chan_sip.c:
<— SIP read from TCP:lync-ipaddress:5060 —>
SIP/2.0 200 OK
FROM: "asterisk"sip:asterisk@asterisk-ip;tag=as1636fcb5
TO: sip:lync-ipaddress;tag=3eeebfebb
CSEQ: 102 OPTIONS
CALL-ID: 6ea5147727a13a6e3d24defb3491d6e1@asterisk-ip:5060
VIA: SIP/2.0/TCP asterisk-ip:5060;branch=z9hG4bK58c3dc07;rport
ACCEPT: application/sdp
CONTENT-LENGTH: 0
ACCEPT-ENCODING: gzip
ACCEPT-LANGUAGE: en
ALLOW: NOTIFY
ALLOW: BENOTIFY
SERVER: RTCC/4.0.0.0 MediationServer

<------------->
[Oct 16 23:51:48] VERBOSE[8109] chan_sip.c: — (13 headers 0 lines) —
[Oct 16 23:51:49] VERBOSE[3179] chan_sip.c: Really destroying SIP dialog ‘6ea5147727a13a6e3d24defb3491d6e1@asterisk-ip:5060’ Method: OPTIONS
[Oct 16 23:51:55] VERBOSE[3179] chan_sip.c: Really destroying SIP dialog ‘73a4d6e1344f6fdd4ec5d419361d04f3@asterisk-ip:5060’ Method: INVITE
[Oct 16 23:52:28] VERBOSE[3039] asterisk.c: – Remote UNIX connection
[Oct 16 23:52:28] VERBOSE[28577] chan_sip.c:
<— SIP read from TCP:lync.external.ip:53110 —>
OPTIONS sip:asterisk-ip SIP/2.0
FROM: sip:lync.fqdn.internal.name:5060;transport=Tcp;ms-opaque=22e2f4011e37d134;epid=8D60B4E98B;tag=4fb28b87c3
TO: sip:asterisk-ip
CSEQ: 5949 OPTIONS
CALL-ID: c25560843b1f4897ad13faf58a158ea9
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 195.198.26.34:62119;branch=z9hG4bKdc1814c9
CONTACT: sip:lync.fqdn.internal.name:5060;transport=Tcp;maddr=195.198.26.34
CONTENT-LENGTH: 0
USER-AGENT: RTCC/4.0.0.0 MediationServer

<------------->
[Oct 16 23:52:28] VERBOSE[28577] chan_sip.c: — (10 headers 0 lines) —
[Oct 16 23:52:28] VERBOSE[28577] chan_sip.c: Looking for s in default (domain asterisk-ip)
[Oct 16 23:52:28] VERBOSE[28577] chan_sip.c:
<— Transmitting (NAT) to lync.external.ip:53110 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 195.198.26.34:62119;branch=z9hG4bKdc1814c9;received=lync.external.ip;rport=53110
From: sip:lync.fqdn.internal.name:5060;transport=Tcp;ms-opaque=22e2f4011e37d134;epid=8D60B4E98B;tag=4fb28b87c3
To: sip:asterisk-ip;tag=as7517a653
Call-ID: c25560843b1f4897ad13faf58a158ea9
CSeq: 5949 OPTIONS
Server: Asterisk PBX 1.8.14.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
[Oct 16 23:52:28] VERBOSE[28577] chan_sip.c: Scheduling destruction of SIP dialog ‘c25560843b1f4897ad13faf58a158ea9’ in 32000 ms (Method: OPTIONS)

Thanks in advance again! :smiley:

4

[quote]<— SIP read from TCP:lync-ipaddress:56086 —>
SIP/2.0 491 Proxy side reinvite failed, pass result to GW.
[/quote]
Lync has rejected the attempt to have it send media direct to the other party, and followed up by dropping the call. I’m not familiar with the subtleties of 491, but it may be that it is reflecting a limitation of what is on the far side of it.

The work around is to disable this optimisation by setting directmedia to no, in the sip.conf device section for the Lync server.

5

[quote=“david55”][quote]<— SIP read from TCP:lync-ipaddress:56086 —>
SIP/2.0 491 Proxy side reinvite failed, pass result to GW.
[/quote]
Lync has rejected the attempt to have it send media direct to the other party, and followed up by dropping the call. I’m not familiar with the subtleties of 491, but it may be that it is reflecting a limitation of what is on the far side of it.

The work around is to disable this optimisation by setting directmedia to no, in the sip.conf device section for the Lync server.[/quote]

Hi David,
I managed to get the call working.
I searched around a bit on the error you provided "491 Proxy side reinvite failed, pass result to GW."
Apparantly Lync server has an option for “Enable Refer”.
I disabled this option.
Together with this i added the G.711 codec to the Lync trunk on the Asterisk side.

Now the call gets transfered to the phonenumber, but when you answer you can only hear the calling part. The calling part can not hear your voice, do you have any idea of whats goin on?

Thanks.

6

There were no REFERs with Asterisk. I still presume that something on the Lync side doesn’t understand how to do direct media, so you need to disable this.

7

Hi David,
I changed
canreinvite to no also…

Now the calls are connected and the audio is crystal clear…

Thanks for all your help!

8

canreinvite is the obsolete form of directmedia, but is still recognized by the configuration file parser.