Re: [asterisk-users] Urgent: DTMF does not work with, rtpmap:101 telephone-event/8000

ct: Re: [asterisk-users] Urgent: DTMF does not work with rtpmap:101 telephone-event/8000 To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii"; format=flowed At 23:40 5/24/2007, JK wrote: Hello aster

Re: [asterisk-users] Urgent: DTMF does not work with, rtpmap:101 telephone-event/8000

ct: Re: [asterisk-users] Urgent: DTMF does not work with rtpmap:101 telephone-event/8000 To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii"; format=flowed At 23:40 5/24/2007, JK wrote: Hello aster

Re: [asterisk-users] Urgent: DTMF does not work with, rtpmap:101 telephone-event/8000

Alex thank you for your response. In this case we are USING INBAND, though I have tried both. Nothing works. Yes ser is configured with mediaproxy. Thank you, -JK JK, In-band or RFC2833 DTMF signaling? Also, unless you have SER configured with a media proxy, the actual "call&quo

[asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 114

I am running asterisk 1.2.12.1 JK, Message: 26 Date: Thu, 24 May 2007 21:40:31 -0700 From: JK <[EMAIL PROTECTED]> Subject: [asterisk-users] Urgent: DTMF does not work with rtpmap:101 telephone-event/8000 To: asterisk-users@lists.digium.com Message-ID: <[EMAIL PROTECTED]> Conten

[asterisk-users] Urgent: DTMF does not work with rtpmap:101 telephone-event/8000

Hello asterisk-users list. I have been scratching my head for almost a week. We are trying to set a service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not working. In our scenario the SP is sending call to our ser server and ser is forwarding the call to asterisk. In the asterisk debug I c

[asterisk-users] isup-oli or ani2

Hello, I am using asterisk and a2billing. Can some one tell me how can I get callingani2 field in a2billing. That way I will be able to identify if the call is from a pay phone. My telco provider is sending me isup-oli in the the from field. Or if there is another way to get the information if

Re: [asterisk-users] Call Forwarding Using Asterisk

Thank you Ram, Can you give me some example, how can I do that. -Jk ram wrote: Hi its possible you need mention in the config Ram On 10/17/06, *jk* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: Can I do this with Asterisk, Call comes to Asterisk Se

[asterisk-users] Call Forwarding Using Asterisk

Can I do this with Asterisk, Call comes to Asterisk Server (Master), Then master just forwards calls to other slave asterisk servers one by one. Like this Master forward 1st call to Slave 1, Second call to Slave 2, Third call to slave 1 Fourth call to slave 2. Is it possible? I will appreciat

[asterisk-users] Load balance Asterisk server, when it is a SIP client.

I am little confused on load balancing, when asterisk server is also a sip client. Based on these, XO Communications one of the largest US DID Provider, now offer SIP Orignation Services for wholesale. Verizon Communications One of the largest US Teleco, now offer SIP Orignation Services. That mea

[asterisk-users] Asterisk Load balancing

. But I am not getting how a CO knows where to send the call. Can anyone give me some lead to get around this load balancing issue? Thank you, -JK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or