[SR-Users] Kamalio & Asterisk integration erroneous behavior, setup issue
Aristeidis Tsitras
tsitras at gmail.com
Tue Sep 3 14:17:45 CEST 2019
HI. Attached you can find the following:
- tcpdump, named eth0.pcap
- asterisk traces
- kamailio's current config
- sip.conf
- extensions.conf
The attempt was to call from 2200 to 2201 and vice versa.
IPs:
1. System: 192.168.1.220
2. Windows PC 192.168.1.7
3. Android phone 192.168.1.124
Another thing that i noticed is that when i have the adsl attached on the
network then on the pcap file i see the public IP of the router something
like 2200 at 62.38.100.150 instead of 2200 at 192.168.1.7. When i unplug the adsl
from the router, then i get 2200 at 192.168.1.7. This happens only for the
windows PC.
Also there was an attempt from a second windows PC in the network to
register, but i could not see anything coming through. It is the extension
2202 and the IP is 192.168.1.8
I would appreciate any help available, please.
Στις Παρ, 30 Αυγ 2019 στις 4:09 μ.μ., ο/η David Villasmil <
david.villasmil.work at gmail.com> έγραψε:
> Log into asterisk’s cli and see what it has to say.
>
> On Fri, 30 Aug 2019 at 14:07, Aristeidis Tsitras <tsitras at gmail.com>
> wrote:
>
>> attached you can find the pcap.
>> IPs
>>
>> - 192.168.1.220. Kamailio at port 5060 and Asterisk at port 5080
>> - 192.168.1.7. Zoiper softphone in Windows PC. extension 2200
>> - 192.168.1.124. Softphone in Android phone. extension 2201
>>
>> 2200 calls 2201. 2200's softphone says call established, probably
>> voicemail, but 2201 never receives the call.
>>
>> Thanks in advance for your help
>>
>>
>> Στις Παρ, 30 Αυγ 2019 στις 2:19 μ.μ., ο/η David Villasmil <
>> david.villasmil.work at gmail.com> έγραψε:
>>
>>> The users are registered on kamailio, not asterisk, that’s why you don’t
>>> see them in asterisk.
>>>
>>> The voicemail is happening because asterisk doesn’t know where the user
>>> being called is. So I assume kamailio is not forwarding the registration
>>> location to asterisk.
>>>
>>> Make a trace with I.e.: sngrep while registering, you should the
>>> register forward happening.
>>>
>>> On Fri, 30 Aug 2019 at 09:55, Aristeidis Tsitras <tsitras at gmail.com>
>>> wrote:
>>>
>>>> new to the area and trying to setup Kamailio with Asterisk in a single
>>>> machine. All users will register to Kamailio's port and in case of need for
>>>> media, it will be forwarded to Asterisk, that is my intention. All of my
>>>> work is based on the following link
>>>> https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb.
>>>> Here is what i have done:
>>>>
>>>> - Debian 8, 64 bit machine with mysql and odbc
>>>> -
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> *root at kamast: ~ $ lsb_release -a No LSB modules are available.
>>>> Distributor ID: Debian Description: Debian GNU/Linux 8.11 (jessie)
>>>> Release: 8.11 Codename: jessie root at kamast: ~ $ uname -a Linux
>>>> kamast 3.16.0-10-amd64 #1 SMP Debian 3.16.72-1 (2019-08-13) x86_64
>>>> GNU/Linux root at kamast: ~ $ *
>>>> - Kamailio 5.2 installed from Kamailio's deb repository
>>>> - Asterisk 13LTS installed from source
>>>> - Used the same passwords such as kamailiorw and asterisk_password,
>>>> since this is a test system, for proof of concept.
>>>>
>>>> I did import to the mysql>asterisk database 3 users 2200, 2201 and
>>>> 2202. Then created in sip.conf the same 3 users with the same credentials.
>>>> Then on 3 PCs i used softphones (Jitsi, Zoiper) and registered each account
>>>> to a softphone. Problems:
>>>>
>>>> - Cannot see the users in the Asterisk's cli, sip show peers
>>>> - I can see users only in Kamailio with kamctl ul show
>>>> - A call between the extensions goes to voicemail. It never reaches
>>>> the other destination eg 2200 calls 2201 and in Asterisk's console i am
>>>> getting a message that 2201 is absent and it goes to voicemail. The same
>>>> with any other extension.
>>>>
>>>> Attached you can find:
>>>>
>>>> 1. Kamailio.cfg
>>>> 2. Asterisk's sip.conf
>>>> 3. Asterisk's extension.conf
>>>> 4. The import that i have done to mysql for the user creation.
>>>>
>>>>
>>>> I would appreciate if someone could point me to the error and help me
>>>> fix it please?
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Kamailio (SER) - Users Mailing List
>>>> sr-users at lists.kamailio.org
>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>> --
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> phone: +34669448337
>>> _______________________________________________
>>> Kamailio (SER) - Users Mailing List
>>> sr-users at lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users at lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
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root at kamast: ~ $ asterisk -rvvv
Asterisk 13.28.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 13.28.0 currently running on kamast (pid = 1057)
kamast*CLI> sip set debug on
SIP Debugging re-enabled
<--- SIP read from UDP:192.168.1.220:5060 --->
REGISTER sip:192.168.1.220:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bKf50d.f0e297f3000000000000000000000000.0
To: <sip:2201 at 192.168.1.220>
From: <sip:2201 at 192.168.1.220>;tag=b8aedf903b5aacaad04d669d0415f139-cb0e
CSeq: 10 REGISTER
Call-ID: 226cdfda2acb89c1-1362 at 192.168.1.220
Max-Forwards: 70
Content-Length: 0
User-Agent: kamailio (5.2.4 (x86_64/linux))
Contact: <sip:2201 at 192.168.1.220:5060>
Expires: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.220:5060 (NAT)
Sending to 192.168.1.220:5060 (NAT)
<--- Transmitting (NAT) to 192.168.1.220:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bKf50d.f0e297f3000000000000000000000000.0;received=192.168.1.220;rport=5060
From: <sip:2201 at 192.168.1.220>;tag=b8aedf903b5aacaad04d669d0415f139-cb0e
To: <sip:2201 at 192.168.1.220>;tag=as43390e06
Call-ID: 226cdfda2acb89c1-1362 at 192.168.1.220
CSeq: 10 REGISTER
Server: Asterisk PBX 13.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c21b584"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '226cdfda2acb89c1-1362 at 192.168.1.220' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.1.220:5060 --->
REGISTER sip:192.168.1.220:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK16a2.0de6b7d6000000000000000000000000.0
To: <sip:2201 at 192.168.1.220>
From: <sip:2201 at 192.168.1.220>;tag=b8aedf903b5aacaad04d669d0415f139-556c
CSeq: 10 REGISTER
Call-ID: 226cdfda2acb89c3-1360 at 192.168.1.220
Max-Forwards: 70
Content-Length: 0
User-Agent: kamailio (5.2.4 (x86_64/linux))
Contact: <sip:2201 at 192.168.1.220:5060>
Expires: 3600
<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.220:5060 (NAT)
Sending to 192.168.1.220:5060 (NAT)
<--- Transmitting (NAT) to 192.168.1.220:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK16a2.0de6b7d6000000000000000000000000.0;received=192.168.1.220;rport=5060
From: <sip:2201 at 192.168.1.220>;tag=b8aedf903b5aacaad04d669d0415f139-556c
To: <sip:2201 at 192.168.1.220>;tag=as21ba153d
Call-ID: 226cdfda2acb89c3-1360 at 192.168.1.220
CSeq: 10 REGISTER
Server: Asterisk PBX 13.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="639e90d5"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '226cdfda2acb89c3-1360 at 192.168.1.220' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.1.220:5060 --->
REGISTER sip:192.168.1.220:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK5544.e96aade7000000000000000000000000.0
To: <sip:2200 at 192.168.1.220>
From: <sip:2200 at 192.168.1.220>;tag=b8aedf903b5aacaad04d669d0415f139-ee1a
CSeq: 10 REGISTER
Call-ID: 226cdfda2acb89c2-1366 at 192.168.1.220
Max-Forwards: 70
Content-Length: 0
User-Agent: kamailio (5.2.4 (x86_64/linux))
Contact: <sip:2200 at 192.168.1.220:5060>
Expires: 3600
<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.220:5060 (NAT)
Sending to 192.168.1.220:5060 (NAT)
<--- Transmitting (NAT) to 192.168.1.220:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK5544.e96aade7000000000000000000000000.0;received=192.168.1.220;rport=5060
From: <sip:2200 at 192.168.1.220>;tag=b8aedf903b5aacaad04d669d0415f139-ee1a
To: <sip:2200 at 192.168.1.220>;tag=as6368208e
Call-ID: 226cdfda2acb89c2-1366 at 192.168.1.220
CSeq: 10 REGISTER
Server: Asterisk PBX 13.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="553196a9"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '226cdfda2acb89c2-1366 at 192.168.1.220' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.1.220:5060 --->
INVITE sip:2200 at 192.168.1.220 SIP/2.0
Record-Route: <sip:192.168.1.220;lr=on;ftag=921537115>
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bKea02.587987c6ebf7679b11e43b0381d9cf5f.0
Via: SIP/2.0/UDP 192.168.1.124:53134;received=192.168.1.124;branch=z9hG4bK881078656;rport=53134
From: <sip:2201 at 192.168.1.220>;tag=921537115
To: <sip:2200 at 192.168.1.220>
Call-ID: 1686350530-53134-2 at BJC.BGI.B.BCE
CSeq: 11 INVITE
Contact: <sip:2201 at 192.168.1.124:53134>
Max-Forwards: 69
User-Agent: Grandstream Wave 1.0.3.29
Privacy: none
P-Preferred-Identity: <sip:2201 at 192.168.1.220>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 273
v=0
o=2201 8000 8000 IN IP4 192.168.1.124
s=SIP Call
c=IN IP4 192.168.1.124
t=0 0
m=audio 34322 RTP/AVP 0 8 101
a=sendrecv
a=rtcp:34323 IN IP4 192.168.1.124
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (18 headers 13 lines) ---
Sending to 192.168.1.220:5060 (NAT)
Sending to 192.168.1.220:5060 (NAT)
Using INVITE request as basis request - 1686350530-53134-2 at BJC.BGI.B.BCE
Found peer '2201' for '2201' from 192.168.1.220:5060
<--- Reliably Transmitting (NAT) to 192.168.1.220:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bKea02.587987c6ebf7679b11e43b0381d9cf5f.0;received=192.168.1.220;rport=5060
Via: SIP/2.0/UDP 192.168.1.124:53134;received=192.168.1.124;branch=z9hG4bK881078656;rport=53134
From: <sip:2201 at 192.168.1.220>;tag=921537115
To: <sip:2200 at 192.168.1.220>;tag=as6d84c401
Call-ID: 1686350530-53134-2 at BJC.BGI.B.BCE
CSeq: 11 INVITE
Server: Asterisk PBX 13.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5751ce2d"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1686350530-53134-2 at BJC.BGI.B.BCE' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.1.220:5060 --->
ACK sip:2200 at 192.168.1.220 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bKea02.587987c6ebf7679b11e43b0381d9cf5f.0
From: <sip:2201 at 192.168.1.220>;tag=921537115
To: <sip:2200 at 192.168.1.220>;tag=as6d84c401
Call-ID: 1686350530-53134-2 at BJC.BGI.B.BCE
CSeq: 11 ACK
Max-Forwards: 69
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '226cdfda2acb89c2-1360 at 192.168.1.220' Method: REGISTER
Really destroying SIP dialog '226cdfda2acb89c1-1364 at 192.168.1.220' Method: REGISTER
<--- SIP read from UDP:192.168.1.220:5060 --->
INVITE sip:2201 at 192.168.1.220:5060;transport=UDP SIP/2.0
Record-Route: <sip:192.168.1.220;lr=on;ftag=dc6a6e43>
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK65f4.6d66a28122068dc7b1d0e78ebbb5b6f1.0
Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---bf7b312adf2d2639
Max-Forwards: 69
Contact: <sip:2200 at 192.168.1.7:45108;transport=UDP>
To: <sip:2201 at 192.168.1.220:5060;transport=UDP>
From: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=dc6a6e43
Call-ID: lS-TaaAPk4bdBOlm9Md8Sw..
CSeq: 2 INVITE
Content-Type: application/sdp
User-Agent: Z 3.15.40006 rv2.8.20
Allow-Events: presence, kpml, talk
Content-Length: 237
v=0
o=Z 0 0 IN IP4 192.168.1.7
s=Z
c=IN IP4 192.168.1.7
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.1.220:5060 (NAT)
Sending to 192.168.1.220:5060 (NAT)
Using INVITE request as basis request - lS-TaaAPk4bdBOlm9Md8Sw..
Found peer '2200' for '2200' from 192.168.1.220:5060
<--- Reliably Transmitting (NAT) to 192.168.1.220:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK65f4.6d66a28122068dc7b1d0e78ebbb5b6f1.0;received=192.168.1.220;rport=5060
Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---bf7b312adf2d2639
From: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=dc6a6e43
To: <sip:2201 at 192.168.1.220:5060;transport=UDP>;tag=as3abe9d45
Call-ID: lS-TaaAPk4bdBOlm9Md8Sw..
CSeq: 2 INVITE
Server: Asterisk PBX 13.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="53e88fda"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'lS-TaaAPk4bdBOlm9Md8Sw..' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.1.220:5060 --->
ACK sip:2201 at 192.168.1.220:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK65f4.6d66a28122068dc7b1d0e78ebbb5b6f1.0
Max-Forwards: 69
To: <sip:2201 at 192.168.1.220:5060;transport=UDP>;tag=as3abe9d45
From: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=dc6a6e43
Call-ID: lS-TaaAPk4bdBOlm9Md8Sw..
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.1.220:5060 --->
INVITE sip:2201 at 192.168.1.220:5060;transport=UDP SIP/2.0
Record-Route: <sip:192.168.1.220;lr=on;ftag=dc6a6e43>
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK75f4.0f17c1d7306f5916f3c57249b76d8875.0
Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---3807847c5e55fd16
Max-Forwards: 69
Contact: <sip:2200 at 192.168.1.7:45108;transport=UDP>
To: <sip:2201 at 192.168.1.220:5060;transport=UDP>
From: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=dc6a6e43
Call-ID: lS-TaaAPk4bdBOlm9Md8Sw..
CSeq: 3 INVITE
Content-Type: application/sdp
User-Agent: Z 3.15.40006 rv2.8.20
Authorization: Digest username="2200",realm="asterisk",nonce="53e88fda",uri="sip:2201 at 192.168.1.220:5060;transport=UDP",response="4ef78b02387758ae5180d8dcbd789550",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 237
v=0
o=Z 0 0 IN IP4 192.168.1.7
s=Z
c=IN IP4 192.168.1.7
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (15 headers 12 lines) ---
Sending to 192.168.1.220:5060 (NAT)
Using INVITE request as basis request - lS-TaaAPk4bdBOlm9Md8Sw..
Found peer '2200' for '2200' from 192.168.1.220:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|g729|gsm), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.7:8000
Looking for 2201 in DialIn (domain 192.168.1.220)
sip_route_dump: route/path hop: <sip:192.168.1.220;lr=on;ftag=dc6a6e43>
<--- Transmitting (NAT) to 192.168.1.220:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK75f4.0f17c1d7306f5916f3c57249b76d8875.0;received=192.168.1.220;rport=5060
Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---3807847c5e55fd16
Record-Route: <sip:192.168.1.220;lr=on;ftag=dc6a6e43>
From: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=dc6a6e43
To: <sip:2201 at 192.168.1.220:5060;transport=UDP>
Call-ID: lS-TaaAPk4bdBOlm9Md8Sw..
CSeq: 3 INVITE
Server: Asterisk PBX 13.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:2201 at 192.168.1.220:5080>
Content-Length: 0
<------------>
-- Executing [2201 at DialIn:1] Dial("SIP/2200-00000000", "SIP/2201") in new stack
[Sep 3 15:05:08] WARNING[1432][C-00000002]: app_dial.c:2591 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [2201 at DialIn:2] VoiceMail("SIP/2200-00000000", "2201,u") in new stack
Audio is at 16900
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 192.168.1.220:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK75f4.0f17c1d7306f5916f3c57249b76d8875.0;received=192.168.1.220;rport=5060
Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---3807847c5e55fd16
Record-Route: <sip:192.168.1.220;lr=on;ftag=dc6a6e43>
From: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=dc6a6e43
To: <sip:2201 at 192.168.1.220:5060;transport=UDP>;tag=as4c7b3f98
Call-ID: lS-TaaAPk4bdBOlm9Md8Sw..
CSeq: 3 INVITE
Server: Asterisk PBX 13.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:2201 at 192.168.1.220:5080>
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 1924428461 1924428461 IN IP4 192.168.1.220
s=Asterisk PBX 13.28.0
c=IN IP4 192.168.1.220
t=0 0
m=audio 16900 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
<------------>
Really destroying SIP dialog '7c582d3252da826f5aaa393a7e07751c at 192.168.1.220:5080' Method: INVITE
<--- SIP read from UDP:192.168.1.220:5060 --->
ACK sip:2201 at 192.168.1.220:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK75f4.210aaf84a7a9186a6db02526407a9020.0
Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---a05573ff8a9b9267
Max-Forwards: 69
Contact: <sip:2200 at 192.168.1.7:45108;transport=UDP>
To: <sip:2201 at 192.168.1.220:5060;transport=UDP>;tag=as4c7b3f98
From: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=dc6a6e43
Call-ID: lS-TaaAPk4bdBOlm9Md8Sw..
CSeq: 3 ACK
User-Agent: Z 3.15.40006 rv2.8.20
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
[Sep 3 15:05:09] WARNING[1432][C-00000002]: app_voicemail.c:6726 leave_voicemail: No entry in voicemail config file for '2201'
-- Executing [2201 at DialIn:3] Hangup("SIP/2200-00000000", "") in new stack
== Spawn extension (DialIn, 2201, 3) exited non-zero on 'SIP/2200-00000000'
Scheduling destruction of SIP dialog 'lS-TaaAPk4bdBOlm9Md8Sw..' in 32000 ms (Method: ACK)
Reliably Transmitting (NAT) to 192.168.1.220:5060:
BYE sip:2200 at 192.168.1.7:45108;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5080;branch=z9hG4bK51b88b44;rport
Route: <sip:192.168.1.220;lr=on;ftag=dc6a6e43>
Max-Forwards: 70
From: <sip:2201 at 192.168.1.220:5060;transport=UDP>;tag=as4c7b3f98
To: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=dc6a6e43
Call-ID: lS-TaaAPk4bdBOlm9Md8Sw..
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.28.0
Proxy-Authorization: Digest username="2200", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.220", nonce="53e88fda", response="79068713ebb60b832e1b11128e7b30b3"
X-Asterisk-HangupCause: Subscriber absent
X-Asterisk-HangupCauseCode: 20
Content-Length: 0
---
[Sep 3 15:05:09] WARNING[1225]: res_odbc.c:1075 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified
[Sep 3 15:05:09] ERROR[1225]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle. CDR failed.
<--- SIP read from UDP:192.168.1.220:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.220:5080;received=192.168.1.220;branch=z9hG4bK51b88b44;rport=5080
Contact: <sip:2200 at 192.168.1.7:45108;transport=UDP>
To: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=dc6a6e43
From: <sip:2201 at 192.168.1.220:5060;transport=UDP>;tag=as4c7b3f98
Call-ID: lS-TaaAPk4bdBOlm9Md8Sw..
CSeq: 102 BYE
User-Agent: Z 3.15.40006 rv2.8.20
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'lS-TaaAPk4bdBOlm9Md8Sw..' Method: ACK
Really destroying SIP dialog '226cdfda2acb89c1-1362 at 192.168.1.220' Method: REGISTER
Really destroying SIP dialog '226cdfda2acb89c3-1360 at 192.168.1.220' Method: REGISTER
Really destroying SIP dialog '226cdfda2acb89c2-1366 at 192.168.1.220' Method: REGISTER
<--- SIP read from UDP:192.168.1.220:5060 --->
INVITE sip:2201 at 192.168.1.220:5060;transport=UDP SIP/2.0
Record-Route: <sip:192.168.1.220;lr=on;ftag=ea2c8d78>
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK0dae.243a824c763a788b3062fcc9ee025dcc.0
Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---6b58a840ab70676f
Max-Forwards: 69
Contact: <sip:2200 at 192.168.1.7:45108;transport=UDP>
To: <sip:2201 at 192.168.1.220:5060;transport=UDP>
From: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=ea2c8d78
Call-ID: 1od4Xzsm69VX6V2k73jWEw..
CSeq: 2 INVITE
Content-Type: application/sdp
User-Agent: Z 3.15.40006 rv2.8.20
Allow-Events: presence, kpml, talk
Content-Length: 237
v=0
o=Z 0 0 IN IP4 192.168.1.7
s=Z
c=IN IP4 192.168.1.7
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.1.220:5060 (NAT)
Sending to 192.168.1.220:5060 (NAT)
Using INVITE request as basis request - 1od4Xzsm69VX6V2k73jWEw..
Found peer '2200' for '2200' from 192.168.1.220:5060
<--- Reliably Transmitting (NAT) to 192.168.1.220:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK0dae.243a824c763a788b3062fcc9ee025dcc.0;received=192.168.1.220;rport=5060
Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---6b58a840ab70676f
From: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=ea2c8d78
To: <sip:2201 at 192.168.1.220:5060;transport=UDP>;tag=as2dd9dd23
Call-ID: 1od4Xzsm69VX6V2k73jWEw..
CSeq: 2 INVITE
Server: Asterisk PBX 13.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5da84077"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1od4Xzsm69VX6V2k73jWEw..' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.1.220:5060 --->
ACK sip:2201 at 192.168.1.220:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK0dae.243a824c763a788b3062fcc9ee025dcc.0
Max-Forwards: 69
To: <sip:2201 at 192.168.1.220:5060;transport=UDP>;tag=as2dd9dd23
From: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=ea2c8d78
Call-ID: 1od4Xzsm69VX6V2k73jWEw..
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.1.220:5060 --->
INVITE sip:2201 at 192.168.1.220:5060;transport=UDP SIP/2.0
Record-Route: <sip:192.168.1.220;lr=on;ftag=ea2c8d78>
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK1dae.8048dfe2e2bb5b27777e466752e27231.0
Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---7efa4b7e878e8087
Max-Forwards: 69
Contact: <sip:2200 at 192.168.1.7:45108;transport=UDP>
To: <sip:2201 at 192.168.1.220:5060;transport=UDP>
From: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=ea2c8d78
Call-ID: 1od4Xzsm69VX6V2k73jWEw..
CSeq: 3 INVITE
Content-Type: application/sdp
User-Agent: Z 3.15.40006 rv2.8.20
Authorization: Digest username="2200",realm="asterisk",nonce="5da84077",uri="sip:2201 at 192.168.1.220:5060;transport=UDP",response="71781c02a2d439bc80981d7662b2d87a",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 237
v=0
o=Z 0 0 IN IP4 192.168.1.7
s=Z
c=IN IP4 192.168.1.7
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (15 headers 12 lines) ---
Sending to 192.168.1.220:5060 (NAT)
Using INVITE request as basis request - 1od4Xzsm69VX6V2k73jWEw..
Found peer '2200' for '2200' from 192.168.1.220:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|g729|gsm), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.7:8000
Looking for 2201 in DialIn (domain 192.168.1.220)
sip_route_dump: route/path hop: <sip:192.168.1.220;lr=on;ftag=ea2c8d78>
<--- Transmitting (NAT) to 192.168.1.220:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK1dae.8048dfe2e2bb5b27777e466752e27231.0;received=192.168.1.220;rport=5060
Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---7efa4b7e878e8087
Record-Route: <sip:192.168.1.220;lr=on;ftag=ea2c8d78>
From: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=ea2c8d78
To: <sip:2201 at 192.168.1.220:5060;transport=UDP>
Call-ID: 1od4Xzsm69VX6V2k73jWEw..
CSeq: 3 INVITE
Server: Asterisk PBX 13.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:2201 at 192.168.1.220:5080>
Content-Length: 0
<------------>
-- Executing [2201 at DialIn:1] Dial("SIP/2200-00000001", "SIP/2201") in new stack
[Sep 3 15:05:23] WARNING[1435][C-00000003]: app_dial.c:2591 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [2201 at DialIn:2] VoiceMail("SIP/2200-00000001", "2201,u") in new stack
Audio is at 19634
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 192.168.1.220:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK1dae.8048dfe2e2bb5b27777e466752e27231.0;received=192.168.1.220;rport=5060
Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---7efa4b7e878e8087
Record-Route: <sip:192.168.1.220;lr=on;ftag=ea2c8d78>
From: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=ea2c8d78
To: <sip:2201 at 192.168.1.220:5060;transport=UDP>;tag=as564764f2
Call-ID: 1od4Xzsm69VX6V2k73jWEw..
CSeq: 3 INVITE
Server: Asterisk PBX 13.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:2201 at 192.168.1.220:5080>
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 104362391 104362391 IN IP4 192.168.1.220
s=Asterisk PBX 13.28.0
c=IN IP4 192.168.1.220
t=0 0
m=audio 19634 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
<------------>
Really destroying SIP dialog '47b487740987fcc6581b748f605b07ad at 192.168.1.220:5080' Method: INVITE
<--- SIP read from UDP:192.168.1.220:5060 --->
ACK sip:2201 at 192.168.1.220:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220;branch=z9hG4bK1dae.210333542ab05a25de4f08108f0ba08a.0
Via: SIP/2.0/UDP 192.168.1.7:45108;branch=z9hG4bK-524287-1---1fdacb4bc80c606c
Max-Forwards: 69
Contact: <sip:2200 at 192.168.1.7:45108;transport=UDP>
To: <sip:2201 at 192.168.1.220:5060;transport=UDP>;tag=as564764f2
From: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=ea2c8d78
Call-ID: 1od4Xzsm69VX6V2k73jWEw..
CSeq: 3 ACK
User-Agent: Z 3.15.40006 rv2.8.20
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
[Sep 3 15:05:24] WARNING[1435][C-00000003]: app_voicemail.c:6726 leave_voicemail: No entry in voicemail config file for '2201'
-- Executing [2201 at DialIn:3] Hangup("SIP/2200-00000001", "") in new stack
== Spawn extension (DialIn, 2201, 3) exited non-zero on 'SIP/2200-00000001'
Scheduling destruction of SIP dialog '1od4Xzsm69VX6V2k73jWEw..' in 32000 ms (Method: ACK)
Reliably Transmitting (NAT) to 192.168.1.220:5060:
BYE sip:2200 at 192.168.1.7:45108;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5080;branch=z9hG4bK58ff235d;rport
Route: <sip:192.168.1.220;lr=on;ftag=ea2c8d78>
Max-Forwards: 70
From: <sip:2201 at 192.168.1.220:5060;transport=UDP>;tag=as564764f2
To: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=ea2c8d78
Call-ID: 1od4Xzsm69VX6V2k73jWEw..
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.28.0
Proxy-Authorization: Digest username="2200", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.220", nonce="5da84077", response="925eb75d16d876118d4e1b6b3680dc03"
X-Asterisk-HangupCause: Subscriber absent
X-Asterisk-HangupCauseCode: 20
Content-Length: 0
---
[Sep 3 15:05:24] WARNING[1225]: res_odbc.c:1075 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified
[Sep 3 15:05:24] ERROR[1225]: cdr_odbc.c:176 odbc_log: Unable to retrieve database handle. CDR failed.
<--- SIP read from UDP:192.168.1.220:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.220:5080;received=192.168.1.220;branch=z9hG4bK58ff235d;rport=5080
Contact: <sip:2200 at 192.168.1.7:45108;transport=UDP>
To: <sip:2200 at 192.168.1.220:5060;transport=UDP>;tag=ea2c8d78
From: <sip:2201 at 192.168.1.220:5060;transport=UDP>;tag=as564764f2
Call-ID: 1od4Xzsm69VX6V2k73jWEw..
CSeq: 102 BYE
User-Agent: Z 3.15.40006 rv2.8.20
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '1od4Xzsm69VX6V2k73jWEw..' Method: ACK
kamast*CLI> sip set debug off
SIP Debugging Disabled
kamast*CLI>
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