I have been battling this problem for 2 months with no resolution as of
yet with TelaSIP. I am told that it is a provider problem(Level
3) because all TelaSIP is doing is passing the call directly to them
once the call comes through.<br>
<br>
Anyone else having this issue with TelaSIP or Level3?<br><br><div><span class="gmail_quote">On 10/10/05, <b class="gmail_sendername">John Millican</b> <<a href="mailto:john@millican.us">john@millican.us</a>> wrote:
</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hello all,<br>yes there is a lot of information about this on the wiki and in past posts on
<br>this list but have not found anything that has solved my problem.<br>setup is:<br>phone-->PAP2-na-->asterisk v1.0.9(in house on local subnet dtmf is<br>inband)--->PSTN--->Telisip---->asterisk box at colo
v1.0.9 VoIP only. I have<br>only access to dial up so can not go VoIP out of the house.<br>In extensions.conf on colo * i have some logic that based on callerid lets me<br>hit a single digit to get to DISA, this work every time.
<br>the problem is that when i enter a number for DISA to dial i get duplicate<br>digits, example i enter 6037862111 and disa tries to dial 6003778621. I have<br>tried setting relaxdtmf=yes in sip.conf with no luck. I have read on the
<br>wiki that RFC2833 should work, but alas its a no go. I am also using ulaw<br>which should not be distorting the dtmf through compresion, correct? Also<br>with RFC2833 it should not matter? Everything works great otherwise.
sip.conf<br>for colo * is posted below:<br>[general]<br>context=telasip<br>port=5060<br>bindaddr=<a href="http://0.0.0.0">0.0.0.0</a><br>srvlookup=yes<br><br>disallow=all ;
First disallow all codecs<br>allow=ulaw<br><br>register => <a href="mailto:username:password@gw3.telasip.com">username:password@gw3.telasip.com</a><br><br>[telasip]<br>type=peer<br>username=*****<br>fromuser=*****<br>
authname=*****<br>secret=*****<br>host=<a href="http://gw3.telasip.com">gw3.telasip.com</a><br>context=default<br>dtmfmode=RFC2833<br>disallow=all<br>allow=ulaw<br>canreinvite=no<br>nat=no<br><br>Thanks in advance for any help
<br>John Millican<br>_______________________________________________<br>--Bandwidth and Colocation sponsored by <a href="http://Easynews.com">Easynews.com</a> --<br><br>Asterisk-Users mailing list<br><a href="mailto:Asterisk-Users@lists.digium.com">
Asterisk-Users@lists.digium.com</a><br><a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">
http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br>-- <br>Tom Vile<br>Baldwin Technology Solutions, Inc<br>Consulting - Web Design - VoIP Telephony<br><a href="http://www.baldwintechsolutions.com">
www.baldwintechsolutions.com</a><br>Phone: 518-631-2855 x205<br>Fax: 518-631-2856