search for: webaccount

Hello Everyone, We are looking for solutions where the transcoding is abstracted away from our * box (i.e., to the network layer) using some carrier grade gateway, or router. The reason for my post is to know about solutions people have used in the past, and how it fits into their overall architecture. Our transcoding needs consists mainly of u/alaw <-> g729, and gsm would also be good....

Thanks a lot Dona and jg for your inputs. I'll try to find some way to do this from Dialplan or AMI and let you guys know soon. Please share if you have some more ideas. Regards, Rajib Date: Tue, 11 Jun 2013 18:34:46 +0200 From: jg <webaccounts at jgoettgens.de> Subject: Re: [asterisk-users] announcement to be played for attended transfer call To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <51B751A6.5000401 at jgoettgens.de> Content-Type: text/pl...

When I set CONNECTEDLINE() info for an incoming ISDN call, the calling party sees only CONNECTEDLINE(num) and the name does not get displayed. Some time ago I called a number, where I did get back a name and a number and everything was displayed correctly. So I think the calling site should basically be able to handle all connected line info. Looking at a pcap trace of the D-channel data, I

...Thanks, -Justin -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130708/a09901d8/attachment-0001.htm> ------------------------------ Message: 2 Date: Mon, 08 Jul 2013 19:21:10 +0200 From: jg <webaccounts at jgoettgens.de> Subject: Re: [asterisk-users] analog phone digit delay To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <51DAF506.5070900 at jgoettgens.de> Content-Type: text/plain; charset=UTF-8; format=flowed Have a l...

...Thanks, -Justin -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130708/a09901d8/attachment-0001.htm> ------------------------------ Message: 2 Date: Mon, 08 Jul 2013 19:21:10 +0200 From: jg <webaccounts at jgoettgens.de> Subject: Re: [asterisk-users] analog phone digit delay To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <51DAF506.5070900 at jgoettgens.de> Content-Type: text/plain; charset=UTF-8; format=flowed Have a l...

Is there a recommended way to find out the cause of DIALSTATUS = CONGESTION for PRI/BRI channels? Currently I am evaluating the DIALSTATUS variable and I also count the active ISDN channels for the ISDN trunk in question. Counting the active ISDN channels seems somewhat clumsy as the mapping to a specific trunk must be done by hand (or write even more code). I have a setup where outgoing calls

I have 2 ISDN BRI boxes, each with 4 spans, where the first one is configured as CPE, the second one as NET(so I don't need real lines for developing and testing). Once in a while I do see the following libpri error messages simultaneously on both boxes: PRI Span: 1 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state 7(Multi-frame established) PRI Span: 2 TEI=0 MDL-ERROR (A): Got

...------- > An HTML attachment was scrubbed... > URL: < > http://lists.digium.com/pipermail/asterisk-users/attachments/20130429/5eba090f/attachment-0001.htm > > > > ------------------------------ > > Message: 3 > Date: Mon, 29 Apr 2013 22:35:08 +0200 > From: jg <webaccounts at jgoettgens.de> > Subject: Re: [asterisk-users] Gateway? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <517ED97C.8090606 at jgoettgens.de> > Content-Type: text/plain; charset=UTF-8; format=...

It seems that initial audio for SIP channels does not get transmitted for a period of varying length, typically about 1 second. This also applies to bridged SIP calls as well to one-legged calls where only Playback() gets called. The Definitive Asterisk Guide uses constructs like "silence/1" or "Wait()" extensively and the explanation given in the text is "to

I have a setup where there are occasional problems with attended transfers. I have already checked the devices as well as the relevant DTMF modes (SIP INFO and rfc2833). I could not find any problems here. The setup is a follows: The front desk (F) accepts calls from customers (C). In some cases F needs to transfer C to a specific department (D). If D cannot handle the problem, D tries to

Maybe this is a stupid question. Are the files in "Extra Sound Packages" related to any product or are they just supplemental material? I searched the source files for some of the file names and didn't find any reference. jg

I have a problem transferring calls multiple times using DTMF sequences (#, *2). The scenario is: Transferee calls Transferor 1 Transferor 1 transfers to Transferor 2 Transferor 2 transfers to Transfer Target When Transferor 2 enters '#' or '*2', Asterisk no longer reacts and the call remains with Transferor 2. I have tested this with Asterisk 11.2 and 11.5 and

Is there a general recipe to avoid fraudulent calls under the following conditions? A receptionist transfers calls as a callee (customers are calling) and as a caller (boss asks to call and then transfer to him), i.e. the Dial cmd for the internal context contains "Tt". Then an outside call would operate as a Local channel in an internal context after the first transfer. If the

Recently I asked a question about possibly unwanted calls due to extended transfer rights after attended transfers using DTMF sequences (http://lists.digium.com/pipermail/asterisk-users/2013-September/280536.html). Obviously, transferring with SIP INVITEs (hold + transfer keys) is not immediately affected by the this, but it is not always possible to enforce this. Meanwhile I have changed the

Hi, I am using the XML-browser and Call-Info header features for some SIP phones. SIPAddHeader(Call-Info: ...) seems to work only in the outgoing direction. Does somebody know a way to send a Call-Info header to the originating SIP device by using only the dial plan? Currently, I am using the XML-browsers to update callee info, but I'd like to use the icon purpose to do that. It's