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...empty = yes timeout = 50 member => SIP/1009 member => SIP/1010 sip.conf [1009] username=1009 type=friend secret=XXXX mailbox=779000 context=default host=dynamic call-limit=2 [1010] username=1010 type=friend secret=XXXX mailbox=779000 context=default host=dynamic call-limit=2 -- Regards, Shariq Khan 0333-3501125 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100915/19a431db/attachment.htm

...| / +----+ / | P2 |--+ +----+ When P1 dial P2, P2 hears voice clear but P1 could not hear any voice. My sip.conf is [avaya] type=peer fromdomain=xx.xx.xx.xx host=xx.xx.xx.xx disallow=all allow=ulaw dtmfmode=rfc2833 canreinvite=yes -- Regards, Shariq Khan 0333-3501125 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/7b6890a9/attachment.htm>

Is it possible to start MOH when calling to DAHDI Channel that has ISDN E1 connected with it. When the called party press hold on his phone then asterisk start MOH?? -- Regards, Shariq Khan 0333-3501125 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/ca6cfc50/attachment.htm>

Need Help .......... I am trying to install AreskiCC Calling Card application but each time I tried to login as root -- I recieved this error Fatal error: Call to undefined function: pg_pconnect() in /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 67 Please help me - I am stucked. I will appreciate your response. Thanks, Syed.

Hi Does anyone have experience with Synway cards like SHD-240D-CT/PCI with asterisk and SynAst driver ? Are they any good ? Do they really run on Asterisk ? Thanks. Anita Hall, Simmortel Voice www.simmortel.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100913/c97492c4/attachment.htm

...sers] SIP Listen Multiple Ports To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <Pine.LNX.4.64.1001030840150.12372 at fs.sedwards.com> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > 1 jan 2010 kl. 20.04 skrev Shariq Khan: > >> I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time On Sun, 3 Jan 2010, Olle E. Johansson wrote: > No, Asterisk only supports one port. You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and "forward" to Asterisk...

Hi guys. I am facing this problem here, using a2billing. error: 'Access denied for user 'a2billinguser'@'localhost' (using password: YES)' I am following this step by step http://www.asterisk2billing.org/cgi-bin/trac.cgi/wiki/Installation%20Guideand wend i get into the point that i have to Create a2billing database i am getting this message above. I even try to remove the

Dear List members- I am trying to configure ASTCC (Asterisk calling card application) but having a hard time to configure it properly. My project deadline is approaching and couldn't figure out how to make ASTCC functional. Here are some details what I have done so far. 1) I have installed ASTCC successfully. 2) I can access astcc-admin.cgi script without any problem. 3) I have created