=====================================================================
Sip read:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5
From: "Marvin Horst" <sip:[EMAIL PROTECTED]:5060>;tag=099422b3d98a1e89
To: <sip:[EMAIL PROTECTED]:5060>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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: <sip:[EMAIL PROTECTED]>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5
From: "Marvin Horst" <sip:[EMAIL PROTECTED]:5060>;tag=099422b3d98a1e89
To: <sip:[EMAIL PROTECTED]:5060>;tag=as6c82465a
Call-ID: [EMAIL PROTECTED]
CSeq: 57341 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
to 192.168.10.2:5060
-- Executing Macro("SIP/mhorst-5fd0", "ext|IAX2/[EMAIL PROTECTED]") in new stack
-- Executing DBget("SIP/mhorst-5fd0", "caller=CF/8030") in new stack
-- DBget: varname=caller, family=CF, key=8030
-- DBget: Value not found in database.
-- Executing DBget("SIP/mhorst-5fd0", "dnd=DND/8030") in new stack
-- DBget: varname=dnd, family=DND, key=8030
-- DBget: Value not found in database.
-- Executing Dial("SIP/mhorst-5fd0", "IAX2/[EMAIL PROTECTED]|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
-- 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" <sip:[EMAIL PROTECTED]:5060>;tag=099422b3d98a1e89
To: <sip:[EMAIL PROTECTED]:5060>;tag=as6c82465a
Call-ID: [EMAIL PROTECTED]
CSeq: 57341 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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
-- Playing 'vm-theperson' (language 'en')
Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5
From: "Marvin Horst" <sip:[EMAIL PROTECTED]:5060>;tag=099422b3d98a1e89
To: <sip:[EMAIL PROTECTED]:5060>;tag=as6c82465a
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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:
strange, do a iax2 debug to see what codecs firefly is asking for. ----- Original Message ----- From: "Paul Zimm" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, February 26, 2004 11:42 PM Subject: [Asterisk-Users] Grandstream -> firefly call translator problemWhen 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 [EMAIL PROTECTED] callerid="marv" <8030> ****** sip.conf ******* [mhorst] type=friend disallow=all allow=ulaw allow=alaw host=dynamic username=mhorst [EMAIL PROTECTED] 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/[EMAIL PROTECTED]) exten => 8040,1,Macro(ext,IAX2/[EMAIL PROTECTED]) exten => 8050,1,Macro(ext,SIP/roger-gs) _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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