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Paul Zimm

2004-Feb-26 13:11 UTC

[Asterisk-Users] Grandstream -> firefly call translator problem

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I also ran sip debug. The output is listed below.<br>
<br>
=====================================================================<br>
Sip read:<br>
INVITE <a class="moz-txt-link-abbreviated"
 href="mailto:sip:8030@asterisk.pbzinc.loc:5060">sip:8030@asterisk.pbzinc.loc:5060</a>
SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5<br>
From: "Marvin Horst"
<a class="moz-txt-link-rfc2396E"
 href="mailto:sip:mhorst@asterisk.pbzinc.loc:5060"><sip:mhorst@asterisk.pbzinc.loc:5060></a>;tag=099422b3d98a1e89<br>
To: <a class="moz-txt-link-rfc2396E"
 href="mailto:sip:8030@asterisk.pbzinc.loc:5060"><sip:8030@asterisk.pbzinc.loc:5060></a><br>
Contact: <a class="moz-txt-link-rfc2396E"
 href="mailto:sip:mhorst@192.168.10.2"><sip:mhorst@192.168.10.2></a><br>
Call-ID: <a class="moz-txt-link-abbreviated"
 href="mailto:dfbe4a29ebddd5a9@192.168.10.2">dfbe4a29ebddd5a9@192.168.10.2</a><br>
CSeq: 57341 INVITE<br>
User-Agent: Grandstream SIP UA 1.0.4.26<br>
Max-Forwards: 70<br>
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO,
SUBSCRIBE<br>
Content-Type: application/sdp<br>
Content-Length: 272<br>
<br>
v=0<br>
o=mhorst 8000 8000 IN IP4 192.168.10.2<br>
s=SIP Call<br>
c=IN IP4 192.168.10.2<br>
t=0 0<br>
m=audio 5004 RTP/AVP 0 8 4 18 2 15<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:4 G723/8000<br>
a=rtpmap:18 G729/8000<br>
a=rtpmap:2 G726-32/8000<br>
a=rtpmap:15 G728/8000<br>
a=ptime:20<br>
<br>
12 headers, 13 lines<br>
Using latest request as basis request<br>
Sending to 192.168.10.2 : 5060 (non-NAT)<br>
Found audio format UNKN<br>
Found audio format ALAW<br>
Found audio format ULAW<br>
Found audio format UNKN<br>
Found audio format GSM<br>
Found audio format UNKN<br>
Found description format PCMU<br>
Found description format PCMA<br>
Found description format G723<br>
Found description format G729<br>
Found description format G726-32<br>
Found description format G728<br>
Capabilities: us - 2147483647, them - 285/0, combined - 285<br>
Non-codec capabilities: us - 1, them - 0, combined - 0<br>
Looking for 8030 in home<br>
list_route: hop: <a class="moz-txt-link-rfc2396E"
 href="mailto:sip:mhorst@192.168.10.2"><sip:mhorst@192.168.10.2></a><br>
Transmitting (no NAT):<br>
SIP/2.0 100 Trying<br>
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5<br>
From: "Marvin Horst"
<a class="moz-txt-link-rfc2396E"
 href="mailto:sip:mhorst@asterisk.pbzinc.loc:5060"><sip:mhorst@asterisk.pbzinc.loc:5060></a>;tag=099422b3d98a1e89<br>
To: <a class="moz-txt-link-rfc2396E"
 href="mailto:sip:8030@asterisk.pbzinc.loc:5060"><sip:8030@asterisk.pbzinc.loc:5060></a>;tag=as6c82465a<br>
Call-ID: <a class="moz-txt-link-abbreviated"
 href="mailto:dfbe4a29ebddd5a9@192.168.10.2">dfbe4a29ebddd5a9@192.168.10.2</a><br>
CSeq: 57341 INVITE<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<br>
Contact: <a class="moz-txt-link-rfc2396E"
 href="mailto:sip:8030@192.168.10.205"><sip:8030@192.168.10.205></a><br>
Content-Length: 0<br>
<br>
&nbsp;to 192.168.10.2:5060<br>
&nbsp;&nbsp;&nbsp; -- Executing Macro("SIP/mhorst-5fd0",
"ext|IAX2/marvcomp@marvcomp")
in new stack<br>
&nbsp;&nbsp;&nbsp; -- Executing DBget("SIP/mhorst-5fd0",
"caller=CF/8030") in new stack<br>
&nbsp;&nbsp;&nbsp; -- DBget: varname=caller, family=CF,
key=8030<br>
&nbsp;&nbsp;&nbsp; -- DBget: Value not found in database.<br>
&nbsp;&nbsp;&nbsp; -- Executing DBget("SIP/mhorst-5fd0",
"dnd=DND/8030") in new stack<br>
&nbsp;&nbsp;&nbsp; -- DBget: varname=dnd, family=DND,
key=8030<br>
&nbsp;&nbsp;&nbsp; -- DBget: Value not found in database.<br>
&nbsp;&nbsp;&nbsp; -- Executing Dial("SIP/mhorst-5fd0",
"IAX2/marvcomp@marvcomp|15|Tt") in new stack<br>
Feb 26 14:58:51 WARNING[-1242121296]: chan_iax2.c:5112 iax2_request:
Unable to create translator path for UNKN to G723 on IAX2[marvcomp]/1<br>
&nbsp;&nbsp;&nbsp; -- Hungup 'IAX2[marvcomp]/1'<br>
Feb 26 14:58:51 NOTICE[-1242121296]: app_dial.c:527 dial_exec: Unable
to create channel of type 'IAX2'<br>
&nbsp; == Everyone is busy at this time<br>
&nbsp;&nbsp;&nbsp; -- Executing
VoiceMail("SIP/mhorst-5fd0", "b8030") in new stack<br>
We're at 192.168.10.205 port 10514<br>
Answering with preferred capability 2147483647<br>
Reliably Transmitting (no NAT):<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5<br>
From: "Marvin Horst"
<a class="moz-txt-link-rfc2396E"
 href="mailto:sip:mhorst@asterisk.pbzinc.loc:5060"><sip:mhorst@asterisk.pbzinc.loc:5060></a>;tag=099422b3d98a1e89<br>
To: <a class="moz-txt-link-rfc2396E"
 href="mailto:sip:8030@asterisk.pbzinc.loc:5060"><sip:8030@asterisk.pbzinc.loc:5060></a>;tag=as6c82465a<br>
Call-ID: <a class="moz-txt-link-abbreviated"
 href="mailto:dfbe4a29ebddd5a9@192.168.10.2">dfbe4a29ebddd5a9@192.168.10.2</a><br>
CSeq: 57341 INVITE<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<br>
Contact: <a class="moz-txt-link-rfc2396E"
 href="mailto:sip:8030@192.168.10.205"><sip:8030@192.168.10.205></a><br>
Content-Type: application/sdp<br>
Content-Length: 111<br>
<br>
v=0<br>
o=root 5520 5520 IN IP4 192.168.10.205<br>
s=session<br>
c=IN IP4 192.168.10.205<br>
t=0 0<br>
m=audio 10514 RTP/AVP<br>
<br>
&nbsp;to 192.168.10.2:5060<br>
&nbsp;&nbsp;&nbsp; -- Playing 'vm-theperson' (language
'en')<br>
<br>
<br>
Sip read:<br>
ACK <a class="moz-txt-link-abbreviated"
 href="mailto:sip:8030@192.168.10.205">sip:8030@192.168.10.205</a>
SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5<br>
From: "Marvin Horst"
<a class="moz-txt-link-rfc2396E"
 href="mailto:sip:mhorst@asterisk.pbzinc.loc:5060"><sip:mhorst@asterisk.pbzinc.loc:5060></a>;tag=099422b3d98a1e89<br>
To: <a class="moz-txt-link-rfc2396E"
 href="mailto:sip:8030@asterisk.pbzinc.loc:5060"><sip:8030@asterisk.pbzinc.loc:5060></a>;tag=as6c82465a<br>
Contact: <a class="moz-txt-link-rfc2396E"
 href="mailto:sip:mhorst@192.168.10.2"><sip:mhorst@192.168.10.2></a><br>
Call-ID: <a class="moz-txt-link-abbreviated"
 href="mailto:dfbe4a29ebddd5a9@192.168.10.2">dfbe4a29ebddd5a9@192.168.10.2</a><br>
CSeq: 57341 ACK<br>
User-Agent: Grandstream SIP UA 1.0.4.26<br>
Max-Forwards: 70<br>
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO,
SUBSCRIBE<br>
Content-Length: 0<br>
<br>
<br>
11 headers, 0 lines<br>
<br>
<br>
Adam Hart wrote:<br>
<blockquote cite="mid000701c3fc68$ec568f80$11ee22cb@PacMan"
type="cite">
  <pre wrap="">strange, do a iax2 debug to see what codecs
firefly is asking for.

----- Original Message ----- 
From: "Paul Zimm" <a
 class="moz-txt-link-rfc2396E"
href="mailto:pbzinc@dejazzd.com"><pbzinc@dejazzd.com></a>
To: <a
 class="moz-txt-link-rfc2396E"
 href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></a>
Sent: Thursday, February 26, 2004 11:42 PM
Subject: [Asterisk-Users] Grandstream -> firefly call translator problem


  </pre>
  <blockquote type="cite">
    <pre wrap="">When I try to initiate a call from my
Grandstream phone (ext 8010) to my
firefly softphone (ext 8030) I get the following error messages, but  I 
have no problem calling from firefly ext to grandstream ext. Calling 
from a Zap phone to firefly works fine also.

Feb 26 07:25:47 WARNING[-1242334288]: chan_iax2.c:5112 iax2_request: 
Unable to create translator path for UNKN to G723 on IAX2[marvcomp]/3
    -- Hungup 'IAX2[marvcomp]/3'
Feb 26 07:25:47 NOTICE[-1242334288]: app_dial.c:527 dial_exec: Unable to 
create channel of type 'IAX2'
  == Everyone is busy at this time

I have ULAW, ALAW, and GSM enabled on the firefly softphone.

here are relevant configs.

***** iax.conf ********
[marvcomp]
disallow=all
allow=ulaw
allow=alaw
type=friend
host=dynamic
username=marvcomp
secret=mayhem
context=home
mailbox=8030@bell
callerid="marv" <8030>

****** sip.conf *******
[mhorst]
type=friend
disallow=all
allow=ulaw
allow=alaw
host=dynamic
username=mhorst
mailbox=8010@bell
context=home
callerid="mhorst" <8010>

****** extensions.conf **********
exten => 8010,1,Macro(ext,SIP/mhorst)
exten => 8020,1,Macro(ext,Zap/2)
exten => 8030,1,Macro(ext,IAX2/marvcomp@marvcomp)
exten => 8040,1,Macro(ext,IAX2/roger@roger)
exten => 8050,1,Macro(ext,SIP/roger-gs)

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    </pre>
  </blockquote>
  <pre
wrap=""><!---->_______________________________________________
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Adam Hart

2004-Feb-26 15:30 UTC

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[Asterisk-Users] Grandstream -> firefly call translator problem

I'd suggest disabling g723 in grandstream or disallowing it in asterisk.

----- Original Message ----- 
From: Paul Zimm
To: asterisk-users@lists.digium.com
Sent: Friday, February 27, 2004 7:11 AM
Subject: Re: [Asterisk-Users] Grandstream -> firefly call translator
problem>I also ran sip debug. The output is listed below.

asterisk users - Feb 2004 - Grandstream -> firefly call translator problem

[Asterisk-Users] Grandstream -> firefly call translator problem

[Asterisk-Users] Grandstream -> firefly call translator problem