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New Member

SIP Trunk progress in-band issue

Hi, Dear Community Members

Our customer faces with an strange issue and I need some advice on how could it be resolved. There are two IPT networks (A and B) and the calls from A to B are transferred sometimes from one to another networks. Network A is managed by CUCM 7 and B is managed by Asterisk. When the call comes to B the caller hears normal calling beep. But, as it turned out, B sends some music which the caller from A should hear. The IT guy from B says that a progress in-band should be switched on on the SIP Trunk. I can not find the way to do it and I kindly ask you for the help.

10 REPLIES

SIP Trunk progress in-band issue

Do you have the MTP checkbox, checked?

HTH

Regards,

Yosh

HTH Regards, Yosh
VIP Super Bronze

SIP Trunk progress in-band issue

What is the actual call path here? Is this accurate?

IP Phone --(SCCP--> CUCM --(SIP Trunk)--> Asterisk --(SIP Line)--> IP Phone

When the call comes to B the caller hears normal calling beep. But, as it turned out, B sends some music which the caller from A should hear. 

So, to be clear, this is purely user A calling user B and hearing ringback instead of music on hold? No transfers or something else?

Please paste the SIP traces from CUCM for this call into the thread. There is no "progress in-band" for SIP, that's an ISDN term. We need to see what Asterisk tells CUCM (e.g. a 180 vs. a 183 with SDP) to answer your question.

Please remember to rate helpful responses and identify helpful or correct answers.

Please remember to rate helpful responses and identify helpful or
New Member

SIP Trunk progress in-band issue

Thank you, Jonathan

As soon as I will get the debugs I will send them back to you,

SIP Trunk progress in-band issue

To echo what Jonathan said, we'd definitely need the SIP Traces to see exactly what's going in. 

Since this is UCM 7.x (pre 8.5 Early Offer without MTP support) if we have MTP Required unchecked we'll be sending delayed offer INVITE, which means we'd need to either to enable MTP required for early offer or enable PRACK (Rel1xx) on the SIP Profile so the far end has our media info for inband ringback prior to connect.

New Member

SIP Trunk progress in-band issue

Where can I enable PRACK in CUCM 7?

New Member

SIP Trunk progress in-band issue

I've enabled Rel1xx in service parameters, but it did not help. The Asterisc guy saus that he just enabled progressinband=yes in SIP.CONF file in asterisk. What could be the corresponding setting in the CUCM?

SIP Trunk progress in-band issue

We'll need to look at the SIP Signaling to diagnose further. 

Without that, we have no way to determine if CUCM is already doing everything it can to provide media information prior to call connect. 

New Member

SIP Trunk progress in-band issue

Here is a traces of the call from CUCM. 1510 made call to the asterisk:

02/04/2014 16:46:40.004 CCM|//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 192.168.19.201:[5060]:

INVITE sip:6530@192.168.19.201:5060 SIP/2.0

Date: Tue, 04 Feb 2014 12:46:40 GMT

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

From: "Yunis F. Ibadullayev" <1510>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-18200612

Allow-Events: presence

P-Asserted-Identity: "Yunis F. Ibadullayev" <1510>

Supported: timer,resource-priority,replaces

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Min-SE:  1800

Remote-Party-ID: "Yunis F. Ibadullayev" <1510>;party=calling;screen=yes;privacy=off

Content-Length: 0

User-Agent: Cisco-CUCM7.1

To: <6530>

Contact: <1510>

Expires: 180

Call-ID: 632bfd80-2f01e130-1351e-26418ac@172.24.100.2

Via: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK39caf4e4a813b

CSeq: 101 INVITE

Session-Expires:  1800

Max-Forwards: 70

02/04/2014 16:46:40.009 CCM|//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 453 from 192.168.19.201:[5060]:

SIP/2.0 100 Trying

v: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK39caf4e4a813b;received=172.24.100.2

f: "Yunis F. Ibadullayev" <1510>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-18200612

t: <6530>

i: 632bfd80-2f01e130-1351e-26418ac@172.24.100.2

CSeq: 101 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

k: replaces

m: <6530>

l: 0

02/04/2014 16:46:40.014 CCM|//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 742 from 192.168.19.201:[5060]:

SIP/2.0 183 Session Progress

v: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK39caf4e4a813b;received=172.24.100.2

f: "Yunis F. Ibadullayev" <1510>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-18200612

t: <6530>;tag=as695c0554

i: 632bfd80-2f01e130-1351e-26418ac@172.24.100.2

CSeq: 101 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

k: replaces

m: <6530>

c: application/sdp

l: 242

v=0

o=root 2073 2073 IN IP4 192.168.19.201

s=session

c=IN IP4 192.168.19.201

t=0 0

m=audio 13096 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

02/04/2014 16:46:40.781 CCM|//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 192.168.19.201:[5060]:

CANCEL sip:6530@192.168.19.201:5060 SIP/2.0

Date: Tue, 04 Feb 2014 12:46:40 GMT

From: "Yunis F. Ibadullayev" <1510>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-18200612

Content-Length: 0

To: <6530>

Call-ID: 632bfd80-2f01e130-1351e-26418ac@172.24.100.2

Via: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK39caf4e4a813b

CSeq: 101 CANCEL

Max-Forwards: 70

02/04/2014 16:46:40.785 CCM|//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 450 from 192.168.19.201:[5060]:

SIP/2.0 487 Request Terminated

v: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK39caf4e4a813b;received=172.24.100.2

f: "Yunis F. Ibadullayev" <1510>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-18200612

t: <6530>;tag=as695c0554

i: 632bfd80-2f01e130-1351e-26418ac@172.24.100.2

CSeq: 101 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

k: replaces

l: 0

02/04/2014 16:46:40.785 CCM|//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 434 from 192.168.19.201:[5060]:

SIP/2.0 200 OK

v: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK39caf4e4a813b;received=172.24.100.2

f: "Yunis F. Ibadullayev" <1510>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-18200612

t: <6530>;tag=as695c0554

i: 632bfd80-2f01e130-1351e-26418ac@172.24.100.2

CSeq: 101 CANCEL

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

k: replaces

l: 0

02/04/2014 16:46:40.785 CCM|//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 192.168.19.201:[5060]:

ACK sip:6530@192.168.19.201:5060 SIP/2.0

Date: Tue, 04 Feb 2014 12:46:40 GMT

From: "Yunis F. Ibadullayev" <1510>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-18200612

Allow-Events: presence

Content-Length: 0

To: <6530>;tag=as695c0554

Call-ID: 632bfd80-2f01e130-1351e-26418ac@172.24.100.2

Via: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK39caf4e4a813b

CSeq: 101 ACK

Max-Forwards: 70

Re: SIP Trunk progress in-band issue

Looks like we're providing delayed offer with no PRACK.  In this case, there is no way for the far end to stream media prior to connect.  You have 4 options here:

1.   Enable PRACK.  This would require two changes:

     A.  On the CUCM side enable Rel1xx in the service param (7.x), or SIP Profile (8.x+)

     B.  COnfigure far end to send Supported: 100rel in their SIP Headers (far end currently not doing this)

2.  Early Offer using MTP Required

     A.  Check the "MTP Required" checkbox on the SIP Trunk

     B.  Provide an MTP with the appropriate codec and region configuration in the MRGL of the SIP Trunk

3.  Early Offer without MTP Required

     A.   Upgrade to CUCM 8.5 or later

     B.  Check the checkbox "Early Offer Support (insert MTP if needed)" checkbox on the SIP Profile of the trunk

4.  Configure far end to do out of band Ringing (180 Ringing instead o 183 with SDP)

Option 1:  no MTP needed, but configuration needed on both sides of the trunk

Option 2:  MTP inserted for every call.  Trunk limited to a single codec, no video support.

Option 3: Will do early offer without MTP if calling device supports SCCPv20+ (most SCCP phones do with recent firmware) or is SIP and calling endpoint sends early offer

Option 4: Requires configuration on far end only, but will not allow any tones or messages to be played prior to the call being answered. 

HTH,

Adam

New Member

SIP Trunk progress in-band issue

Here is a trace from Asterisk Side:

<--- SIP read from 172.24.100.2:5060 --->
SUBSCRIBE sip:6536@192.168.19.201:5060 SIP/2.0
Date: Fri, 28 Feb 2014 13:25:52 GMT
From: <sip:087f4601-28c7-4a67-ba02-41c4991b3d2b@172.24.100.2>;tag=481045501
Event: presence
Content-Length: 0
User-Agent: Cisco-CUCM7.1
To: <sip:6536@192.168.19.201>
Contact: <sip:087f4601-28c7-4a67-ba02-41c4991b3d2b@172.24.100.2:5060>
Expires: 10800
Call-ID: d6fcb580-31018e60-1a1cf-26418ac@172.24.100.2
Accept: application/pidf+xml
Via: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e1966715d546
CSeq: 101 SUBSCRIBE
Max-Forwards: 69

<------------->
[Feb 28 17:25:52] --- (14 headers 0 lines) ---
[Feb 28 17:25:52]
<--- Transmitting (no NAT) to 172.24.100.2:5060 --->
SIP/2.0 403 Forbidden (policy)
v: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e1966715d546;received=172.24.100.2
f: <sip:087f4601-28c7-4a67-ba02-41c4991b3d2b@172.24.100.2>;tag=481045501
t: <sip:6536@192.168.19.201>;tag=as5c39c416
i: d6fcb580-31018e60-1a1cf-26418ac@172.24.100.2
CSeq: 101 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces
l: 0


[Feb 28 17:25:53]
<--- SIP read from 172.24.100.2:5060 --->
INVITE sip:6536@192.168.19.201:5060 SIP/2.0
Date: Fri, 28 Feb 2014 13:25:53 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
From: "Fuad A. Malikov" <sip:1506@172.24.100.2>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-19868037
Allow-Events: presence
P-Asserted-Identity: "Fuad A. Malikov" <sip:1506@172.24.100.2>
Supported: 100rel,timer,resource-priority,replaces
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Min-SE: 1800
Remote-Party-ID: "Fuad A. Malikov" <sip:1506@172.24.100.2>;party=calling;screen=yes;privacy=off
Content-Length: 0
User-Agent: Cisco-CUCM7.1
To: <sip:6536@192.168.19.201>
Contact: <sip:1506@172.24.100.2:5060>
Expires: 180
Call-ID: d7954c00-31018e61-1a1d0-26418ac@172.24.100.2
Via: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e19751bcc14b
CSeq: 101 INVITE
Session-Expires: 1800
Max-Forwards: 70

<------------->
[Feb 28 17:25:53] --- (21 headers 0 lines) ---
[Feb 28 17:25:53] Sending to 172.24.100.2 : 5060 (no NAT)
[Feb 28 17:25:53] Using INVITE request as basis request - d7954c00-31018e61-1a1d0-26418ac@172.24.100.2
[Feb 28 17:25:53] Found peer 'bank'
[Feb 28 17:25:53] Looking for 6536 in from_bank (domain 192.168.19.201)
[Feb 28 17:25:53] list_route: hop: <sip:1506@172.24.100.2:5060>
[Feb 28 17:25:53]
<--- Transmitting (NAT) to 172.24.100.2:5060 --->
SIP/2.0 100 Trying
v: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e19751bcc14b;received=172.24.100.2
f: "Fuad A. Malikov" <sip:1506@172.24.100.2>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-19868037
t: <sip:6536@192.168.19.201>
i: d7954c00-31018e61-1a1d0-26418ac@172.24.100.2
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces
m: <sip:6536@192.168.19.201>
l: 0


<------------>
[Feb 28 17:25:53] Audio is at 192.168.19.201 port 18244
[Feb 28 17:25:53] Adding codec 0x8 (alaw) to SDP
[Feb 28 17:25:53] Adding non-codec 0x1 (telephone-event) to SDP
[Feb 28 17:25:53]
<--- Transmitting (NAT) to 172.24.100.2:5060 --->
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e19751bcc14b;received=172.24.100.2
f: "Fuad A. Malikov" <sip:1506@172.24.100.2>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-19868037
t: <sip:6536@192.168.19.201>;tag=as464114bb
i: d7954c00-31018e61-1a1d0-26418ac@172.24.100.2
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces
m: <sip:6536@192.168.19.201>
c: application/sdp
l: 242

v=0
o=root 2092 2092 IN IP4 192.168.19.201
s=session
c=IN IP4 192.168.19.201
t=0 0
m=audio 18244 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

<------------>
[Feb 28 17:25:56] NOTICE[2333]: chan_sip.c:8368 sip_reregister:    -- Re-registration for  3100119@109.235.199.58
[Feb 28 17:25:56] NOTICE[2333]: chan_sip.c:13978 handle_response_register: Outbound Registration: Expiry for 109.235.199.58 is 120 sec (Scheduling reregistration in 105 s)
[Feb 28 17:25:58] !! Unknown IE 210 (cs0)
[Feb 28 17:26:05] Reliably Transmitting (NAT) to 172.24.100.2:5060:
OPTIONS sip:172.24.100.2 SIP/2.0
v: SIP/2.0/UDP 192.168.19.201:5060;branch=z9hG4bK26e771c2;rport
f: "asterisk" <sip:asterisk@192.168.19.201>;tag=as747e0b2e
t: <sip:172.24.100.2>
m: <sip:asterisk@192.168.19.201>
i: 06cc43db1bed89c948f3ee74003cadba@192.168.19.201
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 28 Feb 2014 13:26:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces
l: 0


---
[Feb 28 17:26:05]
<--- SIP read from 172.24.100.2:5060 --->
SIP/2.0 200 OK
Date: Fri, 28 Feb 2014 13:26:05 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
From: "asterisk" <sip:asterisk@192.168.19.201>;tag=as747e0b2e
Content-Length: 0
To: <sip:172.24.100.2>;tag=745936969
Call-ID: 06cc43db1bed89c948f3ee74003cadba@192.168.19.201
Via: SIP/2.0/UDP 192.168.19.201:5060;branch=z9hG4bK26e771c2;rport
CSeq: 102 OPTIONS

[Feb 28 17:26:53] Audio is at 192.168.19.201 port 18244
[Feb 28 17:26:53] Adding codec 0x8 (alaw) to SDP
[Feb 28 17:26:53] Adding non-codec 0x1 (telephone-event) to SDP
[Feb 28 17:26:53]
<--- Transmitting (NAT) to 172.24.100.2:5060 --->
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e19751bcc14b;received=172.24.100.2
f: "Fuad A. Malikov" <sip:1506@172.24.100.2>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-19868037
t: <sip:6536@192.168.19.201>;tag=as464114bb
i: d7954c00-31018e61-1a1d0-26418ac@172.24.100.2
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces
m: <sip:6536@192.168.19.201>
c: application/sdp
l: 242

v=0
o=root 2092 2093 IN IP4 192.168.19.201
s=session
c=IN IP4 192.168.19.201
t=0 0
m=audio 18244 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

[Feb 28 17:27:05] Reliably Transmitting (NAT) to 172.24.100.2:5060:
OPTIONS sip:172.24.100.2 SIP/2.0
v: SIP/2.0/UDP 192.168.19.201:5060;branch=z9hG4bK309b5b62;rport
f: "asterisk" <sip:asterisk@192.168.19.201>;tag=as4357d77b
t: <sip:172.24.100.2>
m: <sip:asterisk@192.168.19.201>
i: 6061b4200cb380b3203a59ec60ae9ed9@192.168.19.201
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 28 Feb 2014 13:27:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces
l: 0


---
[Feb 28 17:27:05]
<--- SIP read from 172.24.100.2:5060 --->
SIP/2.0 200 OK
Date: Fri, 28 Feb 2014 13:27:05 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
From: "asterisk" <sip:asterisk@192.168.19.201>;tag=as4357d77b
Content-Length: 0
To: <sip:172.24.100.2>;tag=305350118
Call-ID: 6061b4200cb380b3203a59ec60ae9ed9@192.168.19.201
Via: SIP/2.0/UDP 192.168.19.201:5060;branch=z9hG4bK309b5b62;rport
CSeq: 102 OPTIONS

[Feb 28 17:27:39]
<--- SIP read from 172.24.100.2:5060 --->
CANCEL sip:6536@192.168.19.201:5060 SIP/2.0
Date: Fri, 28 Feb 2014 13:25:53 GMT
From: "Fuad A. Malikov" <sip:1506@172.24.100.2>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-19868037
Content-Length: 0
To: <sip:6536@192.168.19.201>
Call-ID: d7954c00-31018e61-1a1d0-26418ac@172.24.100.2
Via: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e19751bcc14b
CSeq: 101 CANCEL
Max-Forwards: 70

<------------->
[Feb 28 17:27:39] --- (9 headers 0 lines) ---
[Feb 28 17:27:39] Sending to 172.24.100.2 : 5060 (NAT)
[Feb 28 17:27:39]
<--- Reliably Transmitting (NAT) to 172.24.100.2:5060 --->
SIP/2.0 487 Request Terminated
v: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e19751bcc14b;received=172.24.100.2
f: "Fuad A. Malikov" <sip:1506@172.24.100.2>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-19868037
t: <sip:6536@192.168.19.201>;tag=as464114bb
i: d7954c00-31018e61-1a1d0-26418ac@172.24.100.2
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces
l: 0


<------------>
[Feb 28 17:27:39]
<--- Transmitting (NAT) to 172.24.100.2:5060 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e19751bcc14b;received=172.24.100.2
f: "Fuad A. Malikov" <sip:1506@172.24.100.2>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-19868037
t: <sip:6536@192.168.19.201>;tag=as464114bb
i: d7954c00-31018e61-1a1d0-26418ac@172.24.100.2
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces
l: 0


<------------>
[Feb 28 17:27:39]
<--- SIP read from 172.24.100.2:5060 --->
ACK sip:6536@192.168.19.201:5060 SIP/2.0
Date: Fri, 28 Feb 2014 13:25:53 GMT
From: "Fuad A. Malikov" <sip:1506@172.24.100.2>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-19868037
Allow-Events: presence
Content-Length: 0
To: <sip:6536@192.168.19.201>;tag=as464114bb
Call-ID: d7954c00-31018e61-1a1d0-26418ac@172.24.100.2
Via: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e19751bcc14b
CSeq: 101 ACK
Max-Forwards: 70

[Feb 28 17:27:40]
<--- SIP read from 172.24.100.2:5060 --->
SUBSCRIBE sip:6536@192.168.19.201:5060 SIP/2.0
Date: Fri, 28 Feb 2014 13:27:40 GMT
From: <sip:087f4601-28c7-4a67-ba02-41c4991b3d2b@172.24.100.2>;tag=1931971158
Event: presence
Content-Length: 0
User-Agent: Cisco-CUCM7.1
To: <sip:6536@192.168.19.201>
Contact: <sip:087f4601-28c7-4a67-ba02-41c4991b3d2b@172.24.100.2:5060>
Expires: 10800
Call-ID: 175c3380-31018ecc-1a1db-26418ac@172.24.100.2
Accept: application/pidf+xml
Via: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e1b4138fc2f2
CSeq: 101 SUBSCRIBE
Max-Forwards: 69

<------------->
[Feb 28 17:27:40] --- (14 headers 0 lines) ---
[Feb 28 17:27:40]
<--- Transmitting (no NAT) to 172.24.100.2:5060 --->
SIP/2.0 403 Forbidden (policy)
v: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e1b4138fc2f2;received=172.24.100.2
f: <sip:087f4601-28c7-4a67-ba02-41c4991b3d2b@172.24.100.2>;tag=1931971158
t: <sip:6536@192.168.19.201>;tag=as5b0d7c88
i: 175c3380-31018ecc-1a1db-26418ac@172.24.100.2
CSeq: 101 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces
l: 0


<------------>
[Feb 28 17:27:41]
<--- SIP read from 172.24.100.2:5060 --->
INVITE sip:6530@192.168.19.201:5060 SIP/2.0
Date: Fri, 28 Feb 2014 13:27:41 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
From: "Fuad A. Malikov" <sip:1506@172.24.100.2>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-19868406
Allow-Events: presence
P-Asserted-Identity: "Fuad A. Malikov" <sip:1506@172.24.100.2>
Supported: 100rel,timer,resource-priority,replaces
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Min-SE: 1800
Remote-Party-ID: "Fuad A. Malikov" <sip:1506@172.24.100.2>;party=calling;screen=yes;privacy=off
Content-Length: 0
User-Agent: Cisco-CUCM7.1
To: <sip:6530@192.168.19.201>
Contact: <sip:1506@172.24.100.2:5060>
Expires: 180
Call-ID: 17f4ca00-31018ecd-1a1dc-26418ac@172.24.100.2
Via: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e1b566bb205a
CSeq: 101 INVITE
Session-Expires: 1800
Max-Forwards: 70

<------------->
[Feb 28 17:27:41] --- (21 headers 0 lines) ---
[Feb 28 17:27:41] Sending to 172.24.100.2 : 5060 (no NAT)
[Feb 28 17:27:41] Using INVITE request as basis request - 17f4ca00-31018ecd-1a1dc-26418ac@172.24.100.2
[Feb 28 17:27:41] Found peer 'bank'
[Feb 28 17:27:41] Looking for 6530 in from_bank (domain 192.168.19.201)
[Feb 28 17:27:41] list_route: hop: <sip:1506@172.24.100.2:5060>
[Feb 28 17:27:41]
<--- Transmitting (NAT) to 172.24.100.2:5060 --->
SIP/2.0 100 Trying
v: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e1b566bb205a;received=172.24.100.2
f: "Fuad A. Malikov" <sip:1506@172.24.100.2>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-19868406
t: <sip:6530@192.168.19.201>
i: 17f4ca00-31018ecd-1a1dc-26418ac@172.24.100.2
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces
m: <sip:6530@192.168.19.201>
l: 0


<------------>
[Feb 28 17:27:41] Audio is at 192.168.19.201 port 10472
[Feb 28 17:27:41] Adding codec 0x8 (alaw) to SDP
[Feb 28 17:27:41] Adding non-codec 0x1 (telephone-event) to SDP
[Feb 28 17:27:41]
<--- Transmitting (NAT) to 172.24.100.2:5060 --->
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e1b566bb205a;received=172.24.100.2
f: "Fuad A. Malikov" <sip:1506@172.24.100.2>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-19868406
t: <sip:6530@192.168.19.201>;tag=as0f4bf446
i: 17f4ca00-31018ecd-1a1dc-26418ac@172.24.100.2
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces
m: <sip:6530@192.168.19.201>
c: application/sdp
l: 242

v=0
o=root 2092 2092 IN IP4 192.168.19.201
s=session
c=IN IP4 192.168.19.201
t=0 0
m=audio 10472 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

<------------>
[Feb 28 17:27:41] NOTICE[2333]: chan_sip.c:13978 handle_response_register: Outbound Registration: Expiry for 109.235.199.58 is 120 sec (Scheduling reregistration in 105 s)
[Feb 28 17:27:44] Audio is at 192.168.19.201 port 10472
[Feb 28 17:27:44] Adding codec 0x8 (alaw) to SDP
[Feb 28 17:27:44] Adding non-codec 0x1 (telephone-event) to SDP
[Feb 28 17:27:44]
<--- Reliably Transmitting (NAT) to 172.24.100.2:5060 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e1b566bb205a;received=172.24.100.2
f: "Fuad A. Malikov" <sip:1506@172.24.100.2>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-19868406
t: <sip:6530@192.168.19.201>;tag=as0f4bf446
i: 17f4ca00-31018ecd-1a1dc-26418ac@172.24.100.2
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces
m: <sip:6530@192.168.19.201>
c: application/sdp
l: 242

v=0
o=root 2092 2093 IN IP4 192.168.19.201
s=session
c=IN IP4 192.168.19.201
t=0 0
m=audio 10472 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

<------------>
[Feb 28 17:27:44]
<--- SIP read from 172.24.100.2:5060 --->
ACK sip:6530@192.168.19.201:5060 SIP/2.0
Date: Fri, 28 Feb 2014 13:27:41 GMT
From: "Fuad A. Malikov" <sip:1506@172.24.100.2>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-19868406
Allow-Events: presence
Content-Length: 213
To: <sip:6530@192.168.19.201>;tag=as0f4bf446
Content-Type: application/sdp
Call-ID: 17f4ca00-31018ecd-1a1dc-26418ac@172.24.100.2
Via: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e1b74ba55334
CSeq: 101 ACK
Max-Forwards: 70

v=0
o=CiscoSystemsCCM-SIP 2000 2 IN IP4 172.24.100.2
s=SIP Call
c=IN IP4 172.24.100.31
t=0 0
m=audio 27442 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[Feb 28 17:27:44] --- (11 headers 10 lines) ---
[Feb 28 17:27:44] Found RTP audio format 8
[Feb 28 17:27:44] Found RTP audio format 101
[Feb 28 17:27:44] Found audio description format PCMA for ID 8
[Feb 28 17:27:44] Found audio description format telephone-event for ID 101
[Feb 28 17:27:44] Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
[Feb 28 17:27:44] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 28 17:27:44] Peer audio RTP is at port 172.24.100.31:27442
[Feb 28 17:28:00]
<--- SIP read from 172.24.100.2:5060 --->
BYE sip:6530@192.168.19.201:5060 SIP/2.0
Date: Fri, 28 Feb 2014 13:27:41 GMT
From: "Fuad A. Malikov" <sip:1506@172.24.100.2>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-19868406
P-Asserted-Identity: "Fuad A. Malikov" <sip:1506@172.24.100.2>
Content-Length: 0
User-Agent: Cisco-CUCM7.1
To: <sip:6530@192.168.19.201>;tag=as0f4bf446
Call-ID: 17f4ca00-31018ecd-1a1dc-26418ac@172.24.100.2
Via: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e1bb36ab473a
CSeq: 102 BYE
Max-Forwards: 70

<------------->
[Feb 28 17:28:00] --- (11 headers 0 lines) ---
[Feb 28 17:28:00] Sending to 172.24.100.2 : 5060 (NAT)
[Feb 28 17:28:00]
<--- Transmitting (NAT) to 172.24.100.2:5060 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 172.24.100.2:5060;branch=z9hG4bK4e1bb36ab473a;received=172.24.100.2
f: "Fuad A. Malikov" <sip:1506@172.24.100.2>;tag=bfaebb53-f38a-455e-8657-16e67a930eb7-19868406
t: <sip:6530@192.168.19.201>;tag=as0f4bf446
i: 17f4ca00-31018ecd-1a1dc-26418ac@172.24.100.2
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces
l: 0


[Feb 28 17:28:05] Reliably Transmitting (NAT) to 172.24.100.2:5060:
OPTIONS sip:172.24.100.2 SIP/2.0
v: SIP/2.0/UDP 192.168.19.201:5060;branch=z9hG4bK2c3c2fd6;rport
f: "asterisk" <sip:asterisk@192.168.19.201>;tag=as4a064ef0
t: <sip:172.24.100.2>
m: <sip:asterisk@192.168.19.201>
i: 343d0e8423ae712c0361112d33b70788@192.168.19.201
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 28 Feb 2014 13:28:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
k: replaces
l: 0


---
[Feb 28 17:28:05]
<--- SIP read from 172.24.100.2:5060 --->
SIP/2.0 200 OK
Date: Fri, 28 Feb 2014 13:28:05 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
From: "asterisk" <sip:asterisk@192.168.19.201>;tag=as4a064ef0
Content-Length: 0
To: <sip:172.24.100.2>;tag=1750061705
Call-ID: 343d0e8423ae712c0361112d33b70788@192.168.19.201
Via: SIP/2.0/UDP 192.168.19.201:5060;branch=z9hG4bK2c3c2fd6;rport
CSeq: 102 OPTIONS

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