similar to: android won't play wav49: how to change format

Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should

On 8/7/07, raviprakash sunkara <sunkara.raviprakash.feb14 at gmail.com> wrote: > > Hello Russell, > Nice To meet U and Good Morning. I got u r mail-Id from > http://www.asterisk.org/node/48325 > Recently i started the SLA configuration. But i didn't understand the > Flow of its Functionality > One of the My Client Ask to have do deploySLA feature > He Using

I understand that the Asterisk SLA implementation is somewhat different from most key systems and PBX systems. I also understand that in Asterisk, one does not put an SLA line on hold since it is just a MeetMe conference. However, is there any way to make the BLF flash when the answering party on the Asterisk system presses the hold key on their set and leaves the calling party alone in the

The Asterisk Development Team has announced the release of Asterisk 1.8.9.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.9.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * ---

Someone submitted a bug about wav49 voicemail problems with the Windows Media Player here http://bugs.digium.com/bug_view_page.php?bug_id=0000254 bkw918 changed the status of the bug to resolved because he could not reproduce the error with his version of Windows Media Player. I am having the same problem as the original bug poster. I am using WMP 9.00.00.3075 running on Windows XP and using

Does anyone have Shared (bridged) Line Appearance working in Asterisk 1.4? Specifically with the Aastra 55i. Specifically, I am using the Aastra 55i with the expansion module. We want to see if other handsets are being used. (BLF) Getting BLF to work would be a great start. It sounds like setting up the hints properly will achieve this. right? Not totally sure how this should be configured.

There's a been a long standing issue with voicemail volume levels for files saved in WAV49 format as compared to WAV format. WAV49 is much smaller in emails and that's great, but it's also less than half the volume level than the exact same voicemail saved in WAV format. I've seen this mentioned by several others over the years in the mailing list -- has there been any

Hi, I tried to find a reference in terms of size but got back a bunch of tech documents and couldn't get the idea of wav49 format. wav49 format is supposed to be half the size of a normal wav right? so, how much disk space takes to save one minute of audio in wav49? I trying to do some capacity planning for a voicemail server. -- ------------------------------------------------------------

Hello, I can do simple, "yum install asterisk18-*" and it installs Asterisk and Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and smooth. However, if I want to compile dahdi-linux on the same openvz then I get the error, *"You do not appear to have the source for the 2.6.32-4-pve kernel installed".* * * 1- Based on above error and Google search I have

I have the voicemail format set to wav49 in my voicemail.conf file. When retrieving voicemails, the first message plays back ok - but then Asterisk hangs up and the log shows the following error. Any idea what's up? May 19 12:57:24 VERBOSE[7860]: Asterisk Ready. May 19 13:48:51 WARNING[7860]: Not a wav file 49 May 19 13:48:51 WARNING[7860]: Unable to open fd on

I would like to convert a file to WAV49 for use with Asterisk using the linux command line. Specifically I would like to upload sounds to use for unavail.wav and busy.wav, but I'd like them to be compressed so that space is not wasted. I tried using SOX but havent found the correct command-line options to make it or Asterisk happy. Ideas?

How can I play wav49 or gsm voicemail files on FC6? Nothing seems to play them (hxplay, xine, mplayer, etc). I think I have all the normal codec packages installed. I can play regular wav files, but they're too big. - Mike

Hello everyone. I'm struggling to get T.38 faxing to work in Asterisk 1.6.1.13 with a SIP DID provider here in Brazil (GVT - Vox IP service). Here's my scenario: When faxes arrive by a specific DID, they are routed thru this simple macro: [macro-recebefax] exten => s,1,Set(DB(fax/count)=$[${DB(fax/count)} + 1]) exten => s,n,Set(FAXCOUNT=${DB(fax/count)}) exten =>

On Mon, Sep 21, 2015 at 9:45 AM, D'Arcy J.M. Cain <darcy at vex.net> wrote: > On Mon, 21 Sep 2015 06:48:52 +0000 > Emil Ohlsson <emo at svep.se> wrote: >> [sip-im] >> exten _X!, 1, NoOp(Got message) >> exten _X!, n, Answer() >> exten _X!, n, Agi(agi://localhost/messagehandler.agi?...) >> exten _X!, n, SendText(Message received) > > I am not

Does anyone know what options you need to use with "sox" to output the audio in the WAV49 format that Asterisk uses.

Can anyone point me in the direction of a utility which will let me determine the length (in seconds) of a wav49 file created by Asterisk? Many thanks, Malcolm

re-posting the question. ----------- use case: when some one in my pbx calls 100.200, I have translations well defined, Media also (media via asterisk) --Works. when some one calls bob, or for any names I am adding Domain and call is been sent to the other party -- Works, no media... For the cases when it is talking to the external work, I want Astersik not to do anything with the SDP/payload.

Sorry for the delay here. For some reason the mail from Joshua Colp failed to deliver to my mailbox. So, anyway, I've set up a local scenario on my computer a PJSIP client and Asterisk 11.17.1 (On a fedora linux workstation) with the settings listed below. In this scenario I've used UDP, but I want a configuration that can be used with any transport protocol. I can see that the context

Hello, We are trying to send faxes by T.38 protocol to a remote SIP proxy from a local extension. The local extension sends the INVITE, Asterisk sends the call to the Proxy the call is connected with a regular audio codec. After a few seconds the remote proxy sends an INVITE with UDPTL and the Asterisk sends it to the local extension and it's accepted, but (here the problem starts) just

Hello! I have a sip device that is sending in the SDP: rtpmap:98 g729a It does not seem like Asterisk is negotiating the codec properly, because while the call rings, the rtp lines fail. However, on other sip devices that have "rtpmap:18 g729" in their SDP, things work fine with Digium's commercial g729 license. How do I get "98 g729a" recognized by Asterisk? Thanks,