No compatible codecs, not accepting this offer!

Asterisk Asterisk Support
1

Hi everyone

I’m trying to use g729 codec, but the CLI appear this massage:

vitor-pc*CLI> sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Using SIP CoS mark 4 == Parsing '/etc/asterisk/sip_notify.conf': Found == Using SIP RTP CoS mark 5 [Mar 2 14:39:10] NOTICE[23544][C-00000009]: chan_sip.c:10556 process_sdp: No compatible codecs, not accepting this offer! == Using SIP RTP CoS mark 5 [Mar 2 14:39:18] NOTICE[23544][C-0000000a]: chan_sip.c:10556 process_sdp: No compatible codecs, not accepting this offer!

But I had installed g729 see below

vitor-pc*CLI> core show translation
         Translation times between formats (in microseconds) for one second of data
          Source Format (Rows) Destination Format (Columns)

            gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex speex16 g726aal2  g722 slin16 testlaw speex32 slin12 slin24 slin32 slin44 slin48 slin96 slin192
      gsm     - 15000 15000 15000 15000  9000 15000 15000 15000   23000    15000 17250  17000   15000   23000  17000  17000  17000  17000  17000  17000   17000
     ulaw 15000     -  9150 15000 15000  9000 15000 15000 15000   23000    15000 17250  17000   15000   23000  17000  17000  17000  17000  17000  17000   17000
     alaw 15000  9150     - 15000 15000  9000 15000 15000 15000   23000    15000 17250  17000   15000   23000  17000  17000  17000  17000  17000  17000   17000
     g726 15000 15000 15000     - 15000  9000 15000 15000 15000   23000    15000 17250  17000   15000   23000  17000  17000  17000  17000  17000  17000   17000
    adpcm 15000 15000 15000 15000     -  9000 15000 15000 15000   23000    15000 17250  17000   15000   23000  17000  17000  17000  17000  17000  17000   17000
     slin  6000  6000  6000  6000  6000     -  6000  6000  6000   14000     6000  8250   8000    6000   14000   8000   8000   8000   8000   8000   8000    8000
    lpc10 15000 15000 15000 15000 15000  9000     - 15000 15000   23000    15000 17250  17000   15000   23000  17000  17000  17000  17000  17000  17000   17000
     g729 15000 15000 15000 15000 15000  9000 15000     - 15000   23000    15000 17250  17000   15000   23000  17000  17000  17000  17000  17000  17000   17000
    speex 15000 15000 15000 15000 15000  9000 15000 15000     -   23000    15000 17250  17000   15000   23000  17000  17000  17000  17000  17000  17000   17000
  speex16 23500 23500 23500 23500 23500 17500 23500 23500 23500       -    23500 15000   9000   23500   23000  17500  17000  17000  17000  17000  17000   17000
 g726aal2 15000 15000 15000 15000 15000  9000 15000 15000 15000   23000        - 17250  17000   15000   23000  17000  17000  17000  17000  17000  17000   17000
     g722 15600 15600 15600 15600 15600  9600 15600 15600 15600   15000    15600     -   9000   15600   23000  17500  17000  17000  17000  17000  17000   17000
   slin16 14500 14500 14500 14500 14500  8500 14500 14500 14500    6000    14500  6000      -   14500   14000   8500   8000   8000   8000   8000   8000    8000
  testlaw 15000 15000 15000 15000 15000  9000 15000 15000 15000   23000    15000 17250  17000       -   23000  17000  17000  17000  17000  17000  17000   17000
  speex32 23500 23500 23500 23500 23500 17500 23500 23500 23500   23500    23500 23500  17500   23500       -  17500  17500   9000  17000  17000  17000   17000
   slin12 14500 14500 14500 14500 14500  8500 14500 14500 14500   14000    14500 14000   8000   14500   14000      -   8000   8000   8000   8000   8000    8000
   slin24 14500 14500 14500 14500 14500  8500 14500 14500 14500   14500    14500 14500   8500   14500   14000   8500      -   8000   8000   8000   8000    8000
   slin32 14500 14500 14500 14500 14500  8500 14500 14500 14500   14500    14500 14500   8500   14500    6000   8500   8500      -   8000   8000   8000    8000
   slin44 14500 14500 14500 14500 14500  8500 14500 14500 14500   14500    14500 14500   8500   14500   14500   8500   8500   8500      -   8000   8000    8000
   slin48 14500 14500 14500 14500 14500  8500 14500 14500 14500   14500    14500 14500   8500   14500   14500   8500   8500   8500   8500      -   8000    8000
   slin96 14500 14500 14500 14500 14500  8500 14500 14500 14500   14500    14500 14500   8500   14500   14500   8500   8500   8500   8500   8500      -    8000
  slin192 14500 14500 14500 14500 14500  8500 14500 14500 14500   14500    14500 14500   8500   14500   14500   8500   8500   8500   8500   8500   8500       -

I’m use Zoiper but I can’t call.

How can I fix then?

2

Show us the sip debug of the failed call.

3

Hi, see below

== Using SIP RTP CoS mark 5
[Mar  2 15:24:39] NOTICE[23544][C-0000000f]: chan_sip.c:10556 process_sdp: No compatible codecs, not accepting this offer!
vitor-pc*CLI>
4

This is not a sip debug. You need to enter sip set debug on in the Asterisk-CLI, afterwards make a new test of an Call and post the result. And: There should be no sip reload between setting the debugging on and the test.

5

Ah sorry, please see below

<------------->
--- (17 headers 0 lines) ---
Creating new subscription
Sending to 192.168.1.33:57685 (no NAT)
Found peer '2001' for '2001' from 192.168.1.33:57685

<--- Transmitting (no NAT) to 192.168.1.33:57685 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.1.33:57685;branch=z9hG4bK-d8754z-8ff5e369323759a0-1---d8754z-;received=192.168.1.33
From: "vitor2"<sip:2001@192.168.1.35:6050;transport=UDP>;tag=b71f4566
To: "vitor2"<sip:2001@192.168.1.35:6050;transport=UDP>;tag=as1f0a504c
Call-ID: YzUyYmE4OTcyZTVjN2MxYTJlMWJhYjU3MzkyYmRiMzA.
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'YzUyYmE4OTcyZTVjN2MxYTJlMWJhYjU3MzkyYmRiMzA.' Method: SUBSCRIBE
Really destroying SIP dialog 'ZjRhMGQ1ZTUwNjdlYTFhNWQ4MWExNzdhMzQ0N2IyZGY.' Method: ACK
6

He asked for the SIP debug of the call, not an unrelated SUBSCRIBE!

7

Sorry I don’t know what part is this, but see below full result

<--- SIP read from UDP:192.168.1.35:47841 --->
INVITE sip:2001@192.168.1.35:6050;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-c7028be622895ab9-1---d8754z-
Max-Forwards: 70
Contact: <sip:2000@192.168.1.35:47841;transport=UDP>
To: <sip:2001@192.168.1.35:6050;transport=UDP>
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=45ee091a
Call-ID: NjdlN2JkNzQ3MmU5NGViN2FmZmVkMGZmMTllNTViZTY.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Allow-Events: presence, kpml
Content-Length: 239

v=0
o=Z 0 0 IN IP4 192.168.1.35
s=Z
c=IN IP4 192.168.1.35
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.1.35:47841 (no NAT)
Sending to 192.168.1.35:47841 (no NAT)
Using INVITE request as basis request - NjdlN2JkNzQ3MmU5NGViN2FmZmVkMGZmMTllNTViZTY.
Found peer '2000' for '2000' from 192.168.1.35:47841

<--- Reliably Transmitting (no NAT) to 192.168.1.35:47841 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-c7028be622895ab9-1---d8754z-;received=192.168.1.35
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=45ee091a
To: <sip:2001@192.168.1.35:6050;transport=UDP>;tag=as68c5da70
Call-ID: NjdlN2JkNzQ3MmU5NGViN2FmZmVkMGZmMTllNTViZTY.
CSeq: 1 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5ece4300"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NjdlN2JkNzQ3MmU5NGViN2FmZmVkMGZmMTllNTViZTY.' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.35:47841 --->
PUBLISH sip:2000@192.168.1.35:6050;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-e167a6ae0b118ace-1---d8754z-
Max-Forwards: 70
Contact: <sip:2000@192.168.1.35:47841;transport=UDP>
To: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=32989b73
Call-ID: NTIwMTIxMzQxODRjOWE2MjEzOGIyMTQ2MDdmM2YzNjA.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence
Allow-Events: presence, kpml
Content-Length: 271

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:2000@192.168.1.35:6050;transport=UDP"> <tuple id="2000" > <status><basic>open</basic></status> <note>On the phone</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 192.168.1.35:47841 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.35:47841 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-e167a6ae0b118ace-1---d8754z-;received=192.168.1.35
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=32989b73
To: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=as0e14541e
Call-ID: NTIwMTIxMzQxODRjOWE2MjEzOGIyMTQ2MDdmM2YzNjA.
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'NTIwMTIxMzQxODRjOWE2MjEzOGIyMTQ2MDdmM2YzNjA.' Method: PUBLISH

<--- SIP read from UDP:192.168.1.35:47841 --->
SUBSCRIBE sip:2000@192.168.1.35:6050;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-05eee551a5e8dbb1-1---d8754z-
Max-Forwards: 70
Contact: <sip:2000@192.168.1.35:47841;transport=UDP>
To: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=a953b30f
Call-ID: MTQ2YWE4ZWIyZTVmN2ViN2I4ZTBlNmQ0ZDBlZTQ5Zjg.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to 192.168.1.35:47841 (no NAT)
Creating new subscription
Sending to 192.168.1.35:47841 (no NAT)
list_route: hop: <sip:2000@192.168.1.35:47841;transport=UDP>
Found peer '2000' for '2000' from 192.168.1.35:47841

<--- Transmitting (no NAT) to 192.168.1.35:47841 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-05eee551a5e8dbb1-1---d8754z-;received=192.168.1.35
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=a953b30f
To: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=as3a6370ed
Call-ID: MTQ2YWE4ZWIyZTVmN2ViN2I4ZTBlNmQ0ZDBlZTQ5Zjg.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4218bad9"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'MTQ2YWE4ZWIyZTVmN2ViN2I4ZTBlNmQ0ZDBlZTQ5Zjg.' in 32000 ms (Method: SUBSCRIBE)

<--- SIP read from UDP:192.168.1.35:47841 --->
ACK sip:2001@192.168.1.35:6050;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-c7028be622895ab9-1---d8754z-
Max-Forwards: 70
To: <sip:2001@192.168.1.35:6050;transport=UDP>;tag=as68c5da70
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=45ee091a
Call-ID: NjdlN2JkNzQ3MmU5NGViN2FmZmVkMGZmMTllNTViZTY.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.35:47841 --->
INVITE sip:2001@192.168.1.35:6050;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-ac62471c50912fb4-1---d8754z-
Max-Forwards: 70
Contact: <sip:2000@192.168.1.35:47841;transport=UDP>
To: <sip:2001@192.168.1.35:6050;transport=UDP>
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=45ee091a
Call-ID: NjdlN2JkNzQ3MmU5NGViN2FmZmVkMGZmMTllNTViZTY.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="2000",realm="asterisk",nonce="5ece4300",uri="sip:2001@192.168.1.35:6050;transport=UDP",response="662e51b54174062fa42f517ad1c383ec",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 239

v=0
o=Z 0 0 IN IP4 192.168.1.35
s=Z
c=IN IP4 192.168.1.35
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 12 lines) ---
Sending to 192.168.1.35:47841 (no NAT)
Using INVITE request as basis request - NjdlN2JkNzQ3MmU5NGViN2FmZmVkMGZmMTllNTViZTY.
Found peer '2000' for '2000' from 192.168.1.35:47841
  == 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 98
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
[Mar  2 17:23:16] NOTICE[23544][C-0000001a]: chan_sip.c:10556 process_sdp: No compatible codecs, not accepting this offer!

<--- Reliably Transmitting (no NAT) to 192.168.1.35:47841 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-ac62471c50912fb4-1---d8754z-;received=192.168.1.35
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=45ee091a
To: <sip:2001@192.168.1.35:6050;transport=UDP>;tag=as68c5da70
Call-ID: NjdlN2JkNzQ3MmU5NGViN2FmZmVkMGZmMTllNTViZTY.
CSeq: 2 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NjdlN2JkNzQ3MmU5NGViN2FmZmVkMGZmMTllNTViZTY.' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.35:47841 --->
SUBSCRIBE sip:2000@192.168.1.35:6050;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-09cafc9b591fa032-1---d8754z-
Max-Forwards: 70
Contact: <sip:2000@192.168.1.35:47841;transport=UDP>
To: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=a953b30f
Call-ID: MTQ2YWE4ZWIyZTVmN2ViN2I4ZTBlNmQ0ZDBlZTQ5Zjg.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="2000",realm="asterisk",nonce="4218bad9",uri="sip:2000@192.168.1.35:6050;transport=UDP",response="968f635b67eccc7c32231a1d9ba3f359",algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (17 headers 0 lines) ---
Creating new subscription
Sending to 192.168.1.35:47841 (no NAT)
Found peer '2000' for '2000' from 192.168.1.35:47841

<--- Transmitting (no NAT) to 192.168.1.35:47841 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-09cafc9b591fa032-1---d8754z-;received=192.168.1.35
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=a953b30f
To: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=as3a6370ed
Call-ID: MTQ2YWE4ZWIyZTVmN2ViN2I4ZTBlNmQ0ZDBlZTQ5Zjg.
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'MTQ2YWE4ZWIyZTVmN2ViN2I4ZTBlNmQ0ZDBlZTQ5Zjg.' Method: SUBSCRIBE

<--- SIP read from UDP:192.168.1.35:47841 --->
ACK sip:2001@192.168.1.35:6050;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-ac62471c50912fb4-1---d8754z-
Max-Forwards: 70
To: <sip:2001@192.168.1.35:6050;transport=UDP>;tag=as68c5da70
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=45ee091a
Call-ID: NjdlN2JkNzQ3MmU5NGViN2FmZmVkMGZmMTllNTViZTY.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.35:47841 --->
PUBLISH sip:2000@192.168.1.35:6050;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-518633ab70580c22-1---d8754z-
Max-Forwards: 70
Contact: <sip:2000@192.168.1.35:47841;transport=UDP>
To: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=1ddbbc19
Call-ID: MjhiNDU5YTAyNmM4MjFhNTg0M2ExZmRlMGNkMDU2Mjc.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence
Allow-Events: presence, kpml
Content-Length: 265

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:2000@192.168.1.35:6050;transport=UDP"> <tuple id="2000" > <status><basic>open</basic></status> <note>Online</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 192.168.1.35:47841 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.35:47841 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-518633ab70580c22-1---d8754z-;received=192.168.1.35
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=1ddbbc19
To: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=as2162c635
Call-ID: MjhiNDU5YTAyNmM4MjFhNTg0M2ExZmRlMGNkMDU2Mjc.
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'MjhiNDU5YTAyNmM4MjFhNTg0M2ExZmRlMGNkMDU2Mjc.' Method: PUBLISH

<--- SIP read from UDP:192.168.1.35:47841 --->
SUBSCRIBE sip:2000@192.168.1.35:6050;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-21ffccf5e820e540-1---d8754z-
Max-Forwards: 70
Contact: <sip:2000@192.168.1.35:47841;transport=UDP>
To: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=deede572
Call-ID: NmI2MTcwMmM5NjE2ZWYyNDU5MjliNDY0ODEwYTNjMjA.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to 192.168.1.35:47841 (no NAT)
Creating new subscription
Sending to 192.168.1.35:47841 (no NAT)
list_route: hop: <sip:2000@192.168.1.35:47841;transport=UDP>
Found peer '2000' for '2000' from 192.168.1.35:47841

<--- Transmitting (no NAT) to 192.168.1.35:47841 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-21ffccf5e820e540-1---d8754z-;received=192.168.1.35
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=deede572
To: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=as61e7a075
Call-ID: NmI2MTcwMmM5NjE2ZWYyNDU5MjliNDY0ODEwYTNjMjA.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2b6cfb94"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NmI2MTcwMmM5NjE2ZWYyNDU5MjliNDY0ODEwYTNjMjA.' in 32000 ms (Method: SUBSCRIBE)

<--- SIP read from UDP:192.168.1.35:47841 --->
SUBSCRIBE sip:2000@192.168.1.35:6050;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-90162680867bdeb2-1---d8754z-
Max-Forwards: 70
Contact: <sip:2000@192.168.1.35:47841;transport=UDP>
To: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=deede572
Call-ID: NmI2MTcwMmM5NjE2ZWYyNDU5MjliNDY0ODEwYTNjMjA.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="2000",realm="asterisk",nonce="2b6cfb94",uri="sip:2000@192.168.1.35:6050;transport=UDP",response="7eff309b6877c5e8efe76fe74f67eb58",algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (17 headers 0 lines) ---
Creating new subscription
Sending to 192.168.1.35:47841 (no NAT)
Found peer '2000' for '2000' from 192.168.1.35:47841

<--- Transmitting (no NAT) to 192.168.1.35:47841 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.1.35:47841;branch=z9hG4bK-d8754z-90162680867bdeb2-1---d8754z-;received=192.168.1.35
From: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=deede572
To: "Vitor"<sip:2000@192.168.1.35:6050;transport=UDP>;tag=as61e7a075
Call-ID: NmI2MTcwMmM5NjE2ZWYyNDU5MjliNDY0ODEwYTNjMjA.
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'NmI2MTcwMmM5NjE2ZWYyNDU5MjliNDY0ODEwYTNjMjA.' Method: SUBSCRIBE
Really destroying SIP dialog 'NjdlN2JkNzQ3MmU5NGViN2FmZmVkMGZmMTllNTViZTY.' Method: ACK
8

Your client (2000 -> Z 3.3.25608 r25552) wants to make a call with either speex or iLBC while Asterisk doesn’t offer these. That’s why the call fails.
There are three possible solutions (as speex is avalable as a codec on Your asterisk based on Your translation table):

A) Add a enable=speex for extension 2000 in sip.conf

B) Change the codec settings within the phone 2000 to allow at least one of the codecs You allowed in Your sip.conf (usually You would allow alaw and ulaw)

C) According to Your initial post and assuming, that You’ve allowed g729 for extension 2000 in sip.conf: Enable this codec within the cofdec settings of the phone.

9

The client is also prepared to to use alaw, ulaw or gsm. It is not prepared to support g729!

10

m=audio 8000 RTP/AVP 3 110 8 0 98 101 This line tell you all the codecs is prepared to use, and other additional info like : media type, protocol , port etc

Here is brief definition just for future reference.

Session attribute lines

a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000

101 = DTMF payload type number the SIP phone supports.

11

[quote=“abw1oim”]Your client (2000 -> Z 3.3.25608 r25552) wants to make a call with either speex or iLBC while Asterisk doesn’t offer these. That’s why the call fails.
There are three possible solutions (as speex is avalable as a codec on Your asterisk based on Your translation table):

A) Add a enable=speex for extension 2000 in sip.conf

B) Change the codec settings within the phone 2000 to allow at least one of the codecs You allowed in Your sip.conf (usually You would allow alaw and ulaw)

C) According to Your initial post and assuming, that You’ve allowed g729 for extension 2000 in sip.conf: Enable this codec within the cofdec settings of the phone.[/quote]

Ok, I use Zoiper, but it do not support, but are there another app for linux and app that support this codecs?

12

Most unlikely. There are very few paid applications on Linux and there would have to be a large number of users of the application to justify the upstream licensing fees. I think first level licences are only available in very large quantities.

13

It seems like Bria, the paid for softphone for which X-Lite is the loss leader, does it. It will cost just under USD 50, per phone.

14

Ok guys, thanks very much