adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX
Dear SIr,
please find attached my sip.conf file
Regards
On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas <danny at debsinc.com>
wrote:
Show us your sip.conf
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 9:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] FAX
Hi all,
When trying to send a FAX I got the following error:
Executing [003228949469 at micho:1] Dial("SIP/028949469-08466918",
"SIP/003228949469 at 80.169.210.181|60") in new stack
[Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format
found to offer. Cancelling call to 003228949469
-- Couldn't call 0032234534534 at 1.1.1.1.1
Where I should define the codec other than the extension in order to succeed
the call?
Regards
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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<p class=3DMsoNormal><font size=3D2 color=3Dnavy
face=3DArial><span style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>From your sip.conf, you are only
allowing
ulaw and alaw codes. I’d try adding gsm or just comment out
the disallow
and the 2 allows. (your recipient is using a codec that
isn’t ulaw or
alaw).<o:p></o:p></span></font></p>
<p class=3DMsoNormal><font size=3D2 color=3Dnavy
face=3DArial><span style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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</span></font></div>
<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font
size=3D2
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] <b><span
style=3D'font-weight:
bold'>On Behalf Of </span></b>michel freiha<br>
<b><span
style=3D'font-weight:bold'>Sent:</span></b> Wednesday,
January 28, 2009
2:21 PM<br>
<b><span
style=3D'font-weight:bold'>To:</span></b>
<st1:PersonName w:st=3D"on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span
style=3D'font-weight:bold'>Subject:</span></b> Re:
[asterisk-users] FAX</span></font><o:p></o:p></p>
</div>
<p class=3DMsoNormal><font size=3D3 face=3D"Times New
Roman"><span style=3D'font-size:
12.0pt'><o:p> </o:p></span></font></p>
<div>
<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font
size=3D3
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>Dear SIr,<br>
<br>
please find attached my sip.conf file<br>
<br>
Regards<o:p></o:p></span></font></p>
<div>
<p class=3DMsoNormal><font size=3D3 face=3D"Times New
Roman"><span style=3D'font-size:
12.0pt'>On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas <<a
href=3D"mailto:danny at debsinc.com">danny at
debsinc.com</a>>
wrote:<o:p></o:p></span></font></p>
<div link=3Dblue vlink=3Dblue>
<div>
<p><font size=3D2 color=3Dnavy face=3DArial><span
style=3D'font-size:10.0pt;font-family:
Arial;color:navy'>Show us your
sip.conf</span></font><o:p></o:p></p>
<p><font size=3D2 color=3Dnavy face=3DArial><span
style=3D'font-size:10.0pt;font-family:
Arial;color:navy'> </span></font><o:p></o:p></p>
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</span></font></div>
<p><b><font size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;
font-weight:bold'>From:</span></font></b><font
size=3D2 face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'> <a
href=3D"mailto:asterisk-users-bounces at lists.digium.com"
target=3D"_blank">asterisk-users-bounces at
lists.digium.com</a>
[mailto:<a href=3D"mailto:asterisk-users-bounces at
lists.digium.com" target=3D"_blank">asterisk-users-bounces at
lists.digium.com</a>]
<b><span style=3D'font-weight:bold'>On Behalf Of
</span></b>michel freiha<br>
<b><span
style=3D'font-weight:bold'>Sent:</span></b> Wednesday,
January 28, 2009
9:30 AM<br>
<b><span
style=3D'font-weight:bold'>To:</span></b>
<st1:PersonName w:st=3D"on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span
style=3D'font-weight:bold'>Subject:</span></b>
[asterisk-users] FAX</span></font><o:p></o:p></p>
</div>
<div>
<div>
<p><font size=3D3 face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'> <o:p></o:p></span></font></p>
<div>
<p><font size=3D3 face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>Hi all,<br>
<br>
When trying to send a FAX I got the following error:<br>
<br>
Executing [003228949469 at micho:1]
Dial("SIP/028949469-08466918",
"SIP/<a href=3D"mailto:003228949469 at 80.169.210.181"
target=3D"_blank">003228949469 at
80.169.210.181</a>|60")
in new stack<br>
[Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format
found to offer. Cancelling call to 003228949469<br>
-- Couldn't call 0032234534534 at
1.1.1.1.1<br>
<br>
Where I should define the codec other than the extension in order to succeed
the call?<br>
<br>
Regards<o:p></o:p></span></font></p>
</div>
</div>
</div>
</div>
</div>
<p class=3DMsoNormal><font size=3D3 face=3D"Times New
Roman"><span style=3D'font-size:
12.0pt'><br>
_______________________________________________<br>
-- Bandwidth and Colocation Provided by <a
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<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
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