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

Paul Zimm
Thu, 26 Feb 2004 13:11:33 -0000

<meta content="text/html;charset=ISO-8859-1"
http-equiv="Content-Type">



<meta content="text/html;charset=ISO-8859-1"
http-equiv="Content-Type">

I also ran sip debug. The output is listed below.

=====================================================================
Sip read:
INVITE <a class="moz-txt-link-abbreviated"
href="mailto:sip:8030@asterisk.pbzinc.loc:5060">sip:8030@asterisk.pbzinc.loc:5060
SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5
From: "Marvin Horst"
<a class="moz-txt-link-rfc2396E"
href="mailto:sip:mhorst@asterisk.pbzinc.loc:5060"><sip:mhorst@asterisk.pbzinc.loc:5060>;tag=099422b3d98a1e89
To: <a class="moz-txt-link-rfc2396E"
href="mailto:sip:8030@asterisk.pbzinc.loc:5060"><sip:8030@asterisk.pbzinc.loc:5060>
Contact: <a class="moz-txt-link-rfc2396E"
href="mailto:sip:mhorst@192.168.10.2"><sip:mhorst@192.168.10.2>
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:dfbe4a29ebddd5a9@192.168.10.2">dfbe4a29ebddd5a9@192.168.10.2
CSeq: 57341 INVITE
User-Agent: Grandstream SIP UA 1.0.4.26
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO,
SUBSCRIBE
Content-Type: application/sdp
Content-Length: 272

v=0
o=mhorst 8000 8000 IN IP4 192.168.10.2
s=SIP Call
c=IN IP4 192.168.10.2
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 15
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=ptime:20

12 headers, 13 lines
Using latest request as basis request
Sending to 192.168.10.2 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format ULAW
Found audio format UNKN
Found audio format GSM
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G726-32
Found description format G728
Capabilities: us - 2147483647, them - 285/0, combined - 285
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for 8030 in home
list_route: hop: <a class="moz-txt-link-rfc2396E"
href="mailto:sip:mhorst@192.168.10.2"><sip:mhorst@192.168.10.2>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5
From: "Marvin Horst"
<a class="moz-txt-link-rfc2396E"
href="mailto:sip:mhorst@asterisk.pbzinc.loc:5060"><sip:mhorst@asterisk.pbzinc.loc:5060>;tag=099422b3d98a1e89
To: <a class="moz-txt-link-rfc2396E"
href="mailto:sip:8030@asterisk.pbzinc.loc:5060"><sip:8030@asterisk.pbzinc.loc:5060>;tag=as6c82465a
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:dfbe4a29ebddd5a9@192.168.10.2">dfbe4a29ebddd5a9@192.168.10.2
CSeq: 57341 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <a class="moz-txt-link-rfc2396E"
href="mailto:sip:8030@192.168.10.205"><sip:8030@192.168.10.205>
Content-Length: 0

to 192.168.10.2:5060
&nbsp;&nbsp;&nbsp; -- Executing Macro("SIP/mhorst-5fd0",
"ext|IAX2/marvcomp@marvcomp")
in new stack
&nbsp;&nbsp;&nbsp; -- Executing DBget("SIP/mhorst-5fd0",
"caller=CF/8030") in new stack
&nbsp;&nbsp;&nbsp; -- DBget: varname=caller, family=CF,
key=8030
-- DBget: Value not found in database.
&nbsp;&nbsp;&nbsp; -- Executing DBget("SIP/mhorst-5fd0",
"dnd=DND/8030") in new stack
&nbsp;&nbsp;&nbsp; -- DBget: varname=dnd, family=DND,
key=8030
-- DBget: Value not found in database.
&nbsp;&nbsp;&nbsp; -- Executing Dial("SIP/mhorst-5fd0",
"IAX2/marvcomp@marvcomp|15|Tt") in new stack
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
-- Hungup 'IAX2[marvcomp]/1'
Feb 26 14:58:51 NOTICE[-1242121296]: app_dial.c:527 dial_exec: Unable
to create channel of type 'IAX2'
== Everyone is busy at this time
&nbsp;&nbsp;&nbsp; -- Executing
VoiceMail("SIP/mhorst-5fd0", "b8030") in new stack
We're at 192.168.10.205 port 10514
Answering with preferred capability 2147483647
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5
From: "Marvin Horst"
<a class="moz-txt-link-rfc2396E"
href="mailto:sip:mhorst@asterisk.pbzinc.loc:5060"><sip:mhorst@asterisk.pbzinc.loc:5060>;tag=099422b3d98a1e89
To: <a class="moz-txt-link-rfc2396E"
href="mailto:sip:8030@asterisk.pbzinc.loc:5060"><sip:8030@asterisk.pbzinc.loc:5060>;tag=as6c82465a
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:dfbe4a29ebddd5a9@192.168.10.2">dfbe4a29ebddd5a9@192.168.10.2
CSeq: 57341 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <a class="moz-txt-link-rfc2396E"
href="mailto:sip:8030@192.168.10.205"><sip:8030@192.168.10.205>
Content-Type: application/sdp
Content-Length: 111

v=0
o=root 5520 5520 IN IP4 192.168.10.205
s=session
c=IN IP4 192.168.10.205
t=0 0
m=audio 10514 RTP/AVP

to 192.168.10.2:5060
&nbsp;&nbsp;&nbsp; -- Playing 'vm-theperson' (language
'en')


Sip read:
ACK <a class="moz-txt-link-abbreviated"
href="mailto:sip:8030@192.168.10.205">sip:8030@192.168.10.205
SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5
From: "Marvin Horst"
<a class="moz-txt-link-rfc2396E"
href="mailto:sip:mhorst@asterisk.pbzinc.loc:5060"><sip:mhorst@asterisk.pbzinc.loc:5060>;tag=099422b3d98a1e89
To: <a class="moz-txt-link-rfc2396E"
href="mailto:sip:8030@asterisk.pbzinc.loc:5060"><sip:8030@asterisk.pbzinc.loc:5060>;tag=as6c82465a
Contact: <a class="moz-txt-link-rfc2396E"
href="mailto:sip:mhorst@192.168.10.2"><sip:mhorst@192.168.10.2>
Call-ID: <a class="moz-txt-link-abbreviated"
href="mailto:dfbe4a29ebddd5a9@192.168.10.2">dfbe4a29ebddd5a9@192.168.10.2
CSeq: 57341 ACK
User-Agent: Grandstream SIP UA 1.0.4.26
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO,
SUBSCRIBE
Content-Length: 0


11 headers, 0 lines


Adam Hart wrote:
<blockquote cite="mid000701c3fc68$ec568f80$11ee22cb@PacMan"
type="cite">
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>
To: <a
 class="moz-txt-link-rfc2396E"
href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com>
Sent: Thursday, February 26, 2004 11:42 PM
Subject: [Asterisk-Users] Grandstream -> firefly call translator problem




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)

_______________________________________________
Asterisk-Users mailing list
<a class="moz-txt-link-abbreviated"
href="mailto:Asterisk-Users@lists.digium.com">Asterisk-Users@lists.digium.com
<a class="moz-txt-link-freetext"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   <a
 class="moz-txt-link-freetext"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users


  <pre
wrap=""> _______________________________________________
Asterisk-Users mailing list
<a class="moz-txt-link-abbreviated"
href="mailto:Asterisk-Users@lists.digium.com">Asterisk-Users@lists.digium.com
<a class="moz-txt-link-freetext"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   <a
 class="moz-txt-link-freetext"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users