[cisco-voip] One Way Voice on Cisco-Asterisk
Tony Kasule
timotsmith at gmail.com
Thu Jul 5 07:42:34 EDT 2018
Dear Sreekanth.
Thanks for your responses.
Please find my traces attached. The call from cisco to asterisk is fine,
asterisk to cisco has no audio, i have attached both traces.
regards,
On Thu, Jul 5, 2018 at 9:44 AM, Sreekanth <sknth.n at gmail.com> wrote:
> Dear Wilson,
>
> On 5 July 2018 at 11:56, Tony Kasule <timotsmith at gmail.com> wrote:
>
>> Dear Sreekanth,
>>
>> Thanks for your response.
>>
>> When I enabled MTP on the cisco call manager, I could no longer get audio
>> even o the cisco to asterisk calls (that were working before). Audio was
>> restored when I disabled MTP option on the call manager. I later came to
>> learn that the MTP option is not required when using he same codec both
>> sides.
>>
>
> MTPs are only required for functions such as dtmf mismatch and
> packetization mismatches between the 2 legs, or if you'd like to force
> Early Offer. The CUCM will invoke an MTP on its own if the call requires it.
>
>
>> I also checked on the cisco 7945 phone and during the call from asterisk
>> to cisco (which has no audio) and I noticed that Sender Packets is counting
>> and incrementing but Receiver Packets is 0. Does this mean that the cisco
>> phone is not receiving any packets, and if so, why?
>>
>
> What is the remote IP address and port? Yes this means that packets are
> not making it from remote end to the phone.
>
>
>> In the asterisk SIP debug, i saw Content-Length: 0 in the exchanges but I
>> wonder what would cause that.
>>
>
> Which message had the Content length 0? Can you paste a snippet here?
>
>
>>
>> Lastly, the SDP IP addresses appear to be ok. I only see the 2 IPs of the
>> devices communicating. I also went to asterisk's rtp.conf and disabled
>> strictrtp but no joy yet. Asterisk to cisco calls are silent yet cisco to
>> asterisk calls are ok.
>>
>>
> If you could paste the entire SIP dialog debug here, we can take a look to
> see what exactly is going on in the exchange.
>
>
>> Thanks for your help in advance.
>>
>> Regards,
>> wilson
>>
>>
>>
> Thanks
> Sreekanth
>
>
>> On Thu, Jul 5, 2018 at 8:30 AM, Sreekanth <sknth.n at gmail.com> wrote:
>>
>>> Tony,
>>> Are you seeing a complete SIP dialog between Asterisk and CUCM when
>>> making the calls from Asterisk towards the phone? Are the IPs and ports
>>> advertised in the SDP correct?
>>>
>>> I would start by taking a packet capture at the gateway or asterisk to
>>> see if 2 way RTP is flowing between them. If you enable MTP then you can
>>> also enable a pcap on the CUCM where the MTP is located.
>>> This would help isolate where the packets are being lost.
>>>
>>> Regards
>>> Sreekanth
>>>
>>> On 5 July 2018 at 10:52, Tony Kasule <timotsmith at gmail.com> wrote:
>>>
>>>> Dear Friends,
>>>>
>>>> I have CUCM 11 and Cisco gateway at my organisation where I am trying
>>>> to add a small asterisk call center.
>>>>
>>>> I created a SIP trunk between CUCM and Asterisk 15.4 on Centos 7 and
>>>> also did the same at the gateway. When I call from the PSTN to a dial-peer
>>>> that is mapped to asterisk, the call goes through well and we each each
>>>> other. However, when I call from asterisk to the PSTN, The call goes
>>>> through but there is total silence.
>>>>
>>>> Same issue with asterisk-CUCM. When I call from the Call manager to
>>>> asterisk, its fine but asterisk to cisco extension, there is no audio on
>>>> answering the call.
>>>>
>>>> I have been perplexed by this scenario. I extensively read online,
>>>> turned MTP on and off (at the sip trunk in CUCM), tried nat=yes and nat=no
>>>> at asterisk side etc but no joy yet.
>>>>
>>>> Has anyone else experiences this and any pointers on how to have it
>>>> resolved?
>>>>
>>>> Thank you so much.
>>>>
>>>> Timothy
>>>>
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>>
>>>
>>
>
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fsm*CLI> sip set debug peer 0997
SIP Debugging Enabled for IP: 172.16.0.143
<--- SIP read from UDP:172.16.0.143:51519 --->
INVITE sip:0045 at 172.16.1.55 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.143:51519;branch=z9hG4bK-524287-1---d531a64ae3db0a29;rport
Max-Forwards: 70
Contact: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>
To: <sip:0045 at 172.16.1.55>
From: "0997"<sip:0997 at 172.16.1.55>;tag=728d3e0b
Call-ID: 90533ZWEwODAwZmIxMWUxMzJiMjk5ZTc2NjhkNDRiZDU3Yzk
CSeq: 1 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.2.0 stamp 90533
Content-Length: 333
v=0
o=- 1530790576080613 1 IN IP4 172.16.0.143
s=X-Lite release 5.2.0 stamp 90533
c=IN IP4 172.16.0.143
t=0 0
m=audio 61866 RTP/AVP 9 8 120 0 84 101
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:84 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (13 headers 12 lines) ---
Sending to 172.16.0.143:51519 (no NAT)
Sending to 172.16.0.143:51519 (no NAT)
Using INVITE request as basis request - 90533ZWEwODAwZmIxMWUxMzJiMjk5ZTc2NjhkNDRiZDU3Yzk
Found peer '0997' for '0997' from 172.16.0.143:51519
<--- Reliably Transmitting (no NAT) to 172.16.0.143:51519 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.0.143:51519;branch=z9hG4bK-524287-1---d531a64ae3db0a29;received=172.16.0.143;rport=51519
From: "0997"<sip:0997 at 172.16.1.55>;tag=728d3e0b
To: <sip:0045 at 172.16.1.55>;tag=as3187a097
Call-ID: 90533ZWEwODAwZmIxMWUxMzJiMjk5ZTc2NjhkNDRiZDU3Yzk
CSeq: 1 INVITE
Server: Asterisk PBX 15.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3afe135e"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '90533ZWEwODAwZmIxMWUxMzJiMjk5ZTc2NjhkNDRiZDU3Yzk' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:172.16.0.143:51519 --->
ACK sip:0045 at 172.16.1.55 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.143:51519;branch=z9hG4bK-524287-1---d531a64ae3db0a29;rport
Max-Forwards: 70
To: <sip:0045 at 172.16.1.55>;tag=as3187a097
From: "0997"<sip:0997 at 172.16.1.55>;tag=728d3e0b
Call-ID: 90533ZWEwODAwZmIxMWUxMzJiMjk5ZTc2NjhkNDRiZDU3Yzk
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:172.16.0.143:51519 --->
INVITE sip:0045 at 172.16.1.55 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.143:51519;branch=z9hG4bK-524287-1---d08e0b75d92c4e2e;rport
Max-Forwards: 70
Contact: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>
To: <sip:0045 at 172.16.1.55>
From: "0997"<sip:0997 at 172.16.1.55>;tag=728d3e0b
Call-ID: 90533ZWEwODAwZmIxMWUxMzJiMjk5ZTc2NjhkNDRiZDU3Yzk
CSeq: 2 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.2.0 stamp 90533
Authorization: Digest username="0997",realm="asterisk",nonce="3afe135e",uri="sip:0045 at 172.16.1.55",response="0d8ebe1425631344125a1fb4eebbfb97",algorithm=MD5
Content-Length: 333
v=0
o=- 1530790576080613 1 IN IP4 172.16.0.143
s=X-Lite release 5.2.0 stamp 90533
c=IN IP4 172.16.0.143
t=0 0
m=audio 61866 RTP/AVP 9 8 120 0 84 101
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:84 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 172.16.0.143:51519 (no NAT)
Using INVITE request as basis request - 90533ZWEwODAwZmIxMWUxMzJiMjk5ZTc2NjhkNDRiZDU3Yzk
Found peer '0997' for '0997' from 172.16.0.143:51519
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 120
Found RTP audio format 0
Found RTP audio format 84
Found RTP audio format 101
Found audio description format opus for ID 120
Found audio description format speex for ID 84
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g722|speex16|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.16.0.143:61866
Looking for 0045 in public (domain 172.16.1.55)
sip_route_dump: route/path hop: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>
<--- Transmitting (no NAT) to 172.16.0.143:51519 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.0.143:51519;branch=z9hG4bK-524287-1---d08e0b75d92c4e2e;received=172.16.0.143;rport=51519
From: "0997"<sip:0997 at 172.16.1.55>;tag=728d3e0b
To: <sip:0045 at 172.16.1.55>
Call-ID: 90533ZWEwODAwZmIxMWUxMzJiMjk5ZTc2NjhkNDRiZDU3Yzk
CSeq: 2 INVITE
Server: Asterisk PBX 15.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0045 at 172.16.1.55:5060>
Content-Length: 0
<------------>
-- Executing [0045 at public:1] Dial("SIP/0997-00000014", "SIP/0045 at 172.16.1.253") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/0045 at 172.16.1.253
-- SIP/172.16.1.253-00000015 is ringing
<--- Transmitting (no NAT) to 172.16.0.143:51519 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.0.143:51519;branch=z9hG4bK-524287-1---d08e0b75d92c4e2e;received=172.16.0.143;rport=51519
From: "0997"<sip:0997 at 172.16.1.55>;tag=728d3e0b
To: <sip:0045 at 172.16.1.55>;tag=as1a283668
Call-ID: 90533ZWEwODAwZmIxMWUxMzJiMjk5ZTc2NjhkNDRiZDU3Yzk
CSeq: 2 INVITE
Server: Asterisk PBX 15.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0045 at 172.16.1.55:5060>
Content-Length: 0
<------------>
-- SIP/172.16.1.253-00000015 answered SIP/0997-00000014
Audio is at 16628
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 172.16.0.143:51519 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.143:51519;branch=z9hG4bK-524287-1---d08e0b75d92c4e2e;received=172.16.0.143;rport=51519
From: "0997"<sip:0997 at 172.16.1.55>;tag=728d3e0b
To: <sip:0045 at 172.16.1.55>;tag=as1a283668
Call-ID: 90533ZWEwODAwZmIxMWUxMzJiMjk5ZTc2NjhkNDRiZDU3Yzk
CSeq: 2 INVITE
Server: Asterisk PBX 15.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0045 at 172.16.1.55:5060>
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 585514165 585514165 IN IP4 172.16.1.55
s=Asterisk PBX 15.4.1
c=IN IP4 172.16.1.55
t=0 0
m=audio 16628 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=maxptime:150
a=sendrecv
<------------>
-- Channel SIP/172.16.1.253-00000015 joined 'simple_bridge' basic-bridge <bbee9883-fe74-419d-b3d8-f0a73f344564>
-- Channel SIP/0997-00000014 joined 'simple_bridge' basic-bridge <bbee9883-fe74-419d-b3d8-f0a73f344564>
> Bridge bbee9883-fe74-419d-b3d8-f0a73f344564: switching from simple_bridge technology to native_rtp
> Locally RTP bridged 'SIP/0997-00000014' and 'SIP/172.16.1.253-00000015' in stack
<--- SIP read from UDP:172.16.0.143:51519 --->
ACK sip:0045 at 172.16.1.55:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.143:51519;branch=z9hG4bK-524287-1---d133af119fb09872;rport
Max-Forwards: 70
Contact: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>
To: <sip:0045 at 172.16.1.55>;tag=as1a283668
From: "0997"<sip:0997 at 172.16.1.55>;tag=728d3e0b
Call-ID: 90533ZWEwODAwZmIxMWUxMzJiMjk5ZTc2NjhkNDRiZDU3Yzk
CSeq: 2 ACK
User-Agent: X-Lite release 5.2.0 stamp 90533
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:172.16.0.143:51519 --->
BYE sip:0045 at 172.16.1.55:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.143:51519;branch=z9hG4bK-524287-1---ebdca36bd824cb2f;rport
Max-Forwards: 70
Contact: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>
To: <sip:0045 at 172.16.1.55>;tag=as1a283668
From: "0997"<sip:0997 at 172.16.1.55>;tag=728d3e0b
Call-ID: 90533ZWEwODAwZmIxMWUxMzJiMjk5ZTc2NjhkNDRiZDU3Yzk
CSeq: 3 BYE
User-Agent: X-Lite release 5.2.0 stamp 90533
Authorization: Digest username="0997",realm="asterisk",nonce="3afe135e",uri="sip:0045 at 172.16.1.55:5060",response="c44a7195f3f3dc91405d33266a1d09a8",algorithm=MD5
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 172.16.0.143:51519 (no NAT)
Scheduling destruction of SIP dialog '90533ZWEwODAwZmIxMWUxMzJiMjk5ZTc2NjhkNDRiZDU3Yzk' in 6400 ms (Method: BYE)
<--- Transmitting (no NAT) to 172.16.0.143:51519 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.143:51519;branch=z9hG4bK-524287-1---ebdca36bd824cb2f;received=172.16.0.143;rport=51519
From: "0997"<sip:0997 at 172.16.1.55>;tag=728d3e0b
To: <sip:0045 at 172.16.1.55>;tag=as1a283668
Call-ID: 90533ZWEwODAwZmIxMWUxMzJiMjk5ZTc2NjhkNDRiZDU3Yzk
CSeq: 3 BYE
Server: Asterisk PBX 15.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
-- Channel SIP/0997-00000014 left 'native_rtp' basic-bridge <bbee9883-fe74-419d-b3d8-f0a73f344564>
-- Channel SIP/172.16.1.253-00000015 left 'native_rtp' basic-bridge <bbee9883-fe74-419d-b3d8-f0a73f344564>
== Spawn extension (public, 0045, 1) exited non-zero on 'SIP/0997-00000014'
Reliably Transmitting (no NAT) to 172.16.0.143:51519:
OPTIONS sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf SIP/2.0
Via: SIP/2.0/UDP 172.16.1.55:5060;branch=z9hG4bK65ef6bda
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 172.16.1.55>;tag=as01cb05f1
To: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>
Contact: <sip:asterisk at 172.16.1.55:5060>
Call-ID: 4375352e75e900687cddd01252a60aed at 172.16.1.55:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.4.1
Date: Thu, 05 Jul 2018 11:40:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:172.16.0.143:51519 --->
<------------->
<--- SIP read from UDP:172.16.0.143:51519 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.55:5060;branch=z9hG4bK65ef6bda
Contact: <sip:172.16.0.143:51519>
To: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>;tag=c3e7d209
From: "asterisk" <sip:asterisk at 172.16.1.55>;tag=as01cb05f1
Call-ID: 4375352e75e900687cddd01252a60aed at 172.16.1.55:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Supported: replaces
User-Agent: X-Lite release 5.2.0 stamp 90533
Allow-Events: talk, hold
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '4375352e75e900687cddd01252a60aed at 172.16.1.55:5060' Method: OPTIONS
Really destroying SIP dialog '90533ZWEwODAwZmIxMWUxMzJiMjk5ZTc2NjhkNDRiZDU3Yzk' Method: BYE
<--- SIP read from UDP:172.16.0.143:51519 --->
<------------->
fsm*CLI>
-------------- next part --------------
fsm*CLI>
== Using SIP RTP CoS mark 5
-- Executing [997 at public:1] Dial("SIP/1004-00000016", "SIP/997&SIP/0997") in new stack
[Jul 5 14:42:13] WARNING[21583][C-0000000f]: chan_sip.c:6283 create_addr: Purely numeric hostname (997), and not a peer--rejecting!
[Jul 5 14:42:13] WARNING[21583][C-0000000f]: app_dial.c:2512 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Using SIP RTP CoS mark 5
Audio is at 17714
Adding codec gsm to SDP
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.16.0.143:51519:
INVITE sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf SIP/2.0
Via: SIP/2.0/UDP 172.16.1.55:5060;branch=z9hG4bK5342abdf
Max-Forwards: 70
From: "ICT Hardware" <sip:0045 at 172.16.1.55>;tag=as41633653
To: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>
Contact: <sip:0045 at 172.16.1.55:5060>
Call-ID: 45bd10ca276e64e12c270151281f174e at 172.16.1.55:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 15.4.1
Date: Thu, 05 Jul 2018 11:42:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284
v=0
o=root 1891102906 1891102906 IN IP4 172.16.1.55
s=Asterisk PBX 15.4.1
c=IN IP4 172.16.1.55
t=0 0
m=audio 17714 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
---
-- Called SIP/0997
<--- SIP read from UDP:172.16.0.143:51519 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.1.55:5060;branch=z9hG4bK5342abdf
To: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>
From: "ICT Hardware" <sip:0045 at 172.16.1.55>;tag=as41633653
Call-ID: 45bd10ca276e64e12c270151281f174e at 172.16.1.55:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:172.16.0.143:51519 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.1.55:5060;branch=z9hG4bK5342abdf
Contact: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>
To: "0997"<sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>;tag=62e37b63
From: "ICT Hardware" <sip:0045 at 172.16.1.55>;tag=as41633653
Call-ID: 45bd10ca276e64e12c270151281f174e at 172.16.1.55:5060
CSeq: 102 INVITE
User-Agent: X-Lite release 5.2.0 stamp 90533
Allow-Events: talk, hold
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>
-- SIP/0997-00000017 is ringing
<--- SIP read from UDP:172.16.0.143:51519 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.55:5060;branch=z9hG4bK5342abdf
Contact: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>
To: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>;tag=62e37b63
From: "ICT Hardware" <sip:0045 at 172.16.1.55>;tag=as41633653
Call-ID: 45bd10ca276e64e12c270151281f174e at 172.16.1.55:5060
CSeq: 102 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.2.0 stamp 90533
Content-Length: 204
v=0
o=- 3929938815 3 IN IP4 172.16.0.143
s=X-Lite release 5.2.0 stamp 90533
c=IN IP4 172.16.0.143
t=0 0
m=audio 57646 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.16.0.143:57646
sip_route_dump: route/path hop: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>
set_destination: Parsing <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf> for address/port to send to
set_destination: set destination to 172.16.0.143:51519
Transmitting (no NAT) to 172.16.0.143:51519:
ACK sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf SIP/2.0
Via: SIP/2.0/UDP 172.16.1.55:5060;branch=z9hG4bK67e3b6ce
Max-Forwards: 70
From: "ICT Hardware" <sip:0045 at 172.16.1.55>;tag=as41633653
To: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>;tag=62e37b63
Contact: <sip:0045 at 172.16.1.55:5060>
Call-ID: 45bd10ca276e64e12c270151281f174e at 172.16.1.55:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.4.1
Content-Length: 0
---
-- SIP/0997-00000017 answered SIP/1004-00000016
-- Channel SIP/0997-00000017 joined 'simple_bridge' basic-bridge <c8574c51-9f3b-4d71-8c4f-6be465fd54dc>
-- Channel SIP/1004-00000016 joined 'simple_bridge' basic-bridge <c8574c51-9f3b-4d71-8c4f-6be465fd54dc>
<--- SIP read from UDP:172.16.0.143:51519 --->
<------------->
-- Channel SIP/1004-00000016 left 'simple_bridge' basic-bridge <c8574c51-9f3b-4d71-8c4f-6be465fd54dc>
-- Channel SIP/0997-00000017 left 'simple_bridge' basic-bridge <c8574c51-9f3b-4d71-8c4f-6be465fd54dc>
Scheduling destruction of SIP dialog '45bd10ca276e64e12c270151281f174e at 172.16.1.55:5060' in 6400 ms (Method: INVITE)
== Spawn extension (public, 997, 1) exited non-zero on 'SIP/1004-00000016'
set_destination: Parsing <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf> for address/port to send to
set_destination: set destination to 172.16.0.143:51519
Reliably Transmitting (no NAT) to 172.16.0.143:51519:
BYE sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf SIP/2.0
Via: SIP/2.0/UDP 172.16.1.55:5060;branch=z9hG4bK28ae7815
Max-Forwards: 70
From: "ICT Hardware" <sip:0045 at 172.16.1.55>;tag=as41633653
To: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>;tag=62e37b63
Call-ID: 45bd10ca276e64e12c270151281f174e at 172.16.1.55:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 15.4.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:172.16.0.143:51519 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.55:5060;branch=z9hG4bK28ae7815
Contact: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>
To: <sip:0997 at 172.16.0.143:51519;rinstance=1d47821b5d101daf>;tag=62e37b63
From: "ICT Hardware" <sip:0045 at 172.16.1.55>;tag=as41633653
Call-ID: 45bd10ca276e64e12c270151281f174e at 172.16.1.55:5060
CSeq: 103 BYE
User-Agent: X-Lite release 5.2.0 stamp 90533
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '45bd10ca276e64e12c270151281f174e at 172.16.1.55:5060' Method: INVITE
Reliably T
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