similar to: PLEASE respond: how to get Asterisk to change coders on RTP handoff??

So, when I get no comments on this at all, either here or on any of the forums, does that mean nobody knows what I'm talking about?? Or does nobody know the answer?? Or is it just a stupid question and nobody wants to bother telling me where to look?? It *is* a question that I have to answer somehow; I've read all through TFOT and see nothing relevant to this issue. It's silly to

We have an external FXO/FXS, and use Asterisk as a call router. We want to use G723 for the actual phone calls, because we have limited bandwidth on our return direction. This has been working fine so far. However, now we want to set up Asterisk to handle PBX menues and accept extentions. Asterisk, of course, uses GSM for its messages, and cannot terminate G723 calls. So I want to tell

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

----- Original Message ----- From: "Jess Magnaye" <jess@arretni.com> To: <wipe_out@users.sourceforge.net> Sent: Tuesday, January 06, 2004 3:19 PM Subject: Re: [Asterisk-Users] Pls confirm > Is the format "allow=g723.1" in sip.conf valid? > > somehow i cannot get it working to do g723 passthru. also, i've read that > doing g723 will disable

hi everybody, I try to record a call with *1 - one touch record feature in g723 format. exten => _00[1-9].,1,Set(TOUCH_MONITOR_FORMAT=g723) exten => _00[1-9].,n,Dial(SIP/${EXTEN}@ext-sip-account,,wW) I have chosen g723 format because in my CLI> show translation there is no translation between g723 format and wav (default for *1 feature). After pressing *1 sequence I have two

I am configuring a sip-phone, receing calls, excellent voice quality. but it does not place calls, please, can some one sort out. here is my debug output, and below that is sip-debug, Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to 0 Jul 16 11:34:32 DEBUG[163850]: Stopping retransmission on 'iiasPlzFribMJMcW' of Response 1: Found Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to

Sorry if this has already been posted and I have missed it. This is a great little reference I found that even veteran programmers can find useful. http://www.rubyinside.com/19-rails-tricks-most-rails-coders-dont- know-131.html Sunny

Somebody must have! Please, send me a g723 and/or g729 (for Asterisk) to pisac@hotmail.com (antispam subject: codec) Thanks, thanks, thanks... :-)

Can anyone recommend a codec pack with G723 for use under Vista? I have G723 prompts (about 70 prompts totaling 1MB) needing to be converted to G711 uLaw. I tried Audacity but it doesn't have G723 codecs. I tired some google found adware free tools and websites with no success in converting G723. It does appear the old Cool Edit (now Adobe Audition 3.0 for $349USD) can do it -jason

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Hi, does anybody made G723 or G729 to work with a GrandStream Phone ? I've a Cisco here and it works fine with G723, but not with my asterisk. The bandwitdh is very important, since we will have our extensions at home. I know that I have what I pay, but the phone works with cisco. Trying to use G723 or G729 Asterisk says no codec available. Does anybody have it working with any compression

I haven't looked at the code so forgive me if this is already what it's doing.. Considering the problems with the current metadata implementation.. couldn't it be changed to just cache the data that comes from the source in order to update it if it changes, and then just send it down to the listening clients then in the form of an ID3V2 tag when they first connect, and then only

I am setting up Asterisk for use in a low bandwidth environment. As bandwidth is precious and our ATA's support it, the decision was made to use the g723 codec. I have been working on this for a few days and have not been successful. The issue that I am having is garbled noise at the client on calls whose RTP streams are terminated by Asterisk system. This is the case for all the dialplan

Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723

Hey All, I am setting up a network of Asterisk servers using IAX2. I am wondering if it is possible to disable the handoff feature? At the moment I have 4 asterisk machines, 3 are at SOHO offices and 1 is centrally hosted in a data centre. In addition the central machine has an IAX2 link to a VOIP provider (and might be set up with more in the future). All calls are routed through that

Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/1d6c78cb/attachment.htm

Ok, My provider is sending a call to me via ULaw but Asterisk isn't picking up on this, I've only allowed ulaw, I disallow=all and then allow=ulaw in my sip.conf and that's the only thing I allow, but when my provider sends me the requests, I get an error about No Compatible Codecs: 17 headers, 8 lines Using latest request as basis request Sending to 67.19.245.213 : 5060

Hi list, we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk 1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both machines for meetme timing. Doing core show translation give on the Lenny server Translation times between formats (in microseconds) for one second of data

All, Our users have a softphone client that supports the G723 Codec which we want to use for bandwidth reasons, however we do not wish (or have the funds) to license the codec on our Asterisk servers. We have G723 pass-through working between the clients just fine, however calls fail when terminating with Asterisk itself (i.e. Voicemail) or out to the PSTN due to transcoding issues. If it

Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP: ___________ HOME _______________ ____OFFICE ____ SPA2000 <---> Linux Box <--> Asterisk Box 192.168.0.253 192.168.0.1 eth1 200.93.xxx.a 200.93.xxx.b eth0 My problem is when I try to call to any trunk or extention