[Asterisk-Users] Asterisk with G729 codec does not want to connect with mediatrix SIP device

[Wes Marderness]

You need a G729 license for asterisk to make a connection. You have to get
them from diguim, they are $10 a channel. They do give you a single channel
demo license, you just have to get it from them.

Wes

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Marko Rakar
Sent: Thursday, March 25, 2004 8:23 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Asterisk with G729 codec does not want to
connect with mediatrix SIP device


I have tried to connect asterisk (which I use through hisax isdn4linux
device) with mediatrix sip device with g729 codec

asterisk can not connect with mediatrix (it connects when ulaw/alaw are
used) when g729 is forced

any ides what to do?



Sip read:
SIP/2.0 200 OK
Call-ID: 1714ebf049da1da918d54b84725aeedb at 192.168.3.6
CSeq: 102 INVITE
From: 0 <sip:0 at 192.168.3.6>;tag=as01323dfd
To: <sip:301 at 192.168.3.211>;tag=674991B479A02CF7-370B96C56CAF118
Via: SIP/2.0/UDP 192.168.3.6;branch=z9hG4bK247473b9
Content-Length: 178
Content-Type: application/sdp
Contact: <sip:301 at 192.168.3.211>
Allow: INVITE, ACK, BYE, CANCEL, REFER

v=0
o=MxSIP 0 0 IN IP4 192.168.3.211
s=SIP Call
c=IN IP4 192.168.3.211
t=0 0
m=audio 5004 RTP/AVP 18 8 0
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000

10 headers, 9 lines
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found description format G729
Found description format PCMA
Found description format PCMU
Capabilities: us - 268, them - 268/0, combined - 268
Non-codec capabilities: us - 1, them - 0, combined - 0
list_route: hop: <sip:301 at 192.168.3.211>
set_destination: Parsing <sip:301 at 192.168.3.211> for address/port to
send to
set_destination: set destination to 192.168.3.211, port 5060
Transmitting:
ACK sip:301 at 192.168.3.211 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK247473b9
From: "0" <sip:0 at 192.168.3.6>;tag=as01323dfd
To: <sip:301 at 192.168.3.211>;tag=674991B479A02CF7-370B96C56CAF118
Contact: <sip:0 at 192.168.3.6>
Call-ID: 1714ebf049da1da918d54b84725aeedb at 192.168.3.6
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.3.211:5060
Mar 25 15:24:26 NOTICE[1225991360]: channel.c:1513 ast_set_write_format:
Unable to find a path from UNKN to SLINR
Mar 25 15:24:26 WARNING[1225991360]: channel.c:1920
ast_channel_make_compatible: No path to translate from
Modem[i4l]/ttyI0(64) to SIP/301-3309(256)
Mar 25 15:24:26 WARNING[1225991360]: app_dial.c:702 dial_exec: Had to
drop call because I couldn't make Modem[i4l]/ttyI0 compatible with
SIP/301-3309
set_destination: Parsing <sip:301 at 192.168.3.211> for address/port to
send to
set_destination: set destination to 192.168.3.211, port 5060
Reliably Transmitting:
BYE sip:301 at 192.168.3.211 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK247473b9
From: "0" <sip:0 at 192.168.3.6>;tag=as01323dfd
To: <sip:301 at 192.168.3.211>;tag=674991B479A02CF7-370B96C56CAF118
Contact: <sip:0 at 192.168.3.6>
Call-ID: 1714ebf049da1da918d54b84725aeedb at 192.168.3.6
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.3.211:5060
asterisk*CLI>

Sip read:
SIP/2.0 200 OK
Call-ID: 1714ebf049da1da918d54b84725aeedb at 192.168.3.6
CSeq: 103 BYE
From: 0 <sip:0 at 192.168.3.6>;tag=as01323dfd
To: <sip:301 at 192.168.3.211>;tag=674991B479A02CF7-370B96C56CAF118
Via: SIP/2.0/UDP 192.168.3.6;branch=z9hG4bK247473b9
Content-Length: 0


7 headers, 0 lines


----
Sometimes you're the bug, sometimes you're the windshield.

mailto:marko at printel.hr
http://printel.hr
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