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Hi, In a VoIP only environment, Asterisk has to use ztdummy to have any chance of playing back understandable audio files (without drops, hickups etc). I have been using ztdummy to some degree of success, but I also have a "Wildcard TDM400P REV E/F Board 1" in the Asterisk machine I'm using. I'm not using this card for anything at all, but I'm wondering how to set it

Hello all, I am running into some problems with a snom 105 phone trying to do a attended transfer . Snom phones are connected to Asterisk. This does not work, it will only do a unattended transfer. I have downloaded the manual from snom and followed the instructions. Has anyone experienced the same problem ? any ideas how to solve the problem. thanks, Arne.

*spews coffee over keyboard* - FUNNIEST - THREAD - EVER - Also one of the most insightful. Teddy, your gmail invite is on the way.

Hello veryone, I have a strange problem. I have an asterisk (latest from CVS) with latest oh323 channel driver. I place calls with DIAX. The H323 gateways only support G711A De DIAX only supports GSM When I perform an inbound call: H323 -> asterisk -> DIAX :: sound is ok. When I perform an outbound call: DIAX -> Asterisk -> h323 :: sound is terrible and CPU load is 80% When I

Hi, I just started to "play" with Asterisk today and while I'm writing some IVR-like functionality in extensions.conf I would like to take a decision based on whether playing a file succeeds: exten => s,2,GotoIf($[Playback(${CALLERIDNUM}_personal) = 0]?3,501) So if Playback succeeds I want to jump to label 3, otherwise to label 500. Unfortunately Asterisk doesn't seem

Hi, We're routing SIP calls through Asterisk and we want to be able to reinvite calls without Asterisk performing codec conversion. We've performed the following test: Asterisk has license for G.729 installed sip.conf [general] context=default autocreatepeer=yes disallow=all allow=alaw allow=g729 canreinvite=yes nat=no We have configured two endpoints: EP1, preferred codec order

Hi, For those interested in using MySQL directly from extensions.conf, there's already a source file floating around for using a MYSQL application to do SELECT queries. We're using the MYSQL app a lot in our exensions.conf, but we missed support for queries that don't return a result like UPDATE or INSERT. Here's an updated app_mysql.c which introduces the Execute command.

Hi, We've got endpoints and gateways who have T.38 fax support. We now use SER and Asterisk to do our routing and other functionality, but fax doesn't seem to work. Asterisk complains like this: Sep 9 09:25:45 WARNING[467828746]: RTP Read too short Sep 9 09:25:45 WARNING[467828746]: Asked to transmit frame type 256, while native formats is 32 (read/write = 256/256) With lots of the

asterisk-users-bounces@lists.digium.com wrote: > Isn't it possible to use T.38 for interconnecting hardware gates > supporting T.38 with asterisk using SIP REINVITE? > I'm not shure but but think its's might be possible because after > reinvite traffic goes directly from one gate to anotger, not over > Asterisk We've seen a problem here with asterisk. Wehn

Hi, We're running a very old version of Asterisk (CVS-HEAD-08/03/04) and we're having some problems with logging. Our logger.conf has the following: full => notice,warning,error,debug,verbose After having started Asterisk, asterisk will hang in "/usr/sbin/asterisk -rx 'logger reload'" unless some output has been sent to the file. I can't find anything on

Hi, What does this line of output mean? Bridge stops because we're zombie or need a soft hangup: I'm seeing this sometimes... I've looked in channel.c, but the code is not much more revealing than the debug line... -- Andreas Sikkema Rits tele.com Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45

asterisk-users-bounces@lists.digium.com wrote: > http://www.serviceview.bt.com/list/current/docs/Call_Charges.boo/00165.htm > Of course these are BT retail rates but I fully expect wholesale > rates based on call prefix will be available for carriers / ITSP In some countries there's a company (companies?) providing access to a database which telcos can use to find the rates on this

Hi, Has anyone written a SPEC file for zaptel, with kernel 2.6 and udev support? I can find some spec files here and there, but from what I can see they're all kernel 2.4 / non udev... -- Andreas Sikkema bbned NV Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45

asterisk-users-bounces@lists.digium.com wrote: > i have installed asterisk on my system and using only g729 > enabled phones. > from what i understand, we would not be needing any g729 > licenses as all my > voicemail prompts are also in g729 and asterisk is not doing any > transcoding. when i use the voicemail function to record, the > message is not recorded (0 byte file is

asterisk-users-bounces@lists.digium.com wrote: > Is there anyone else with the same problem? Yes, we've seen the same problem. We have found a work around, but I'm unable to to look into it today. -- Andreas Sikkema Rits tele.com Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540

asterisk-users-bounces@lists.digium.com wrote: > In the following setup: > call coming from a pstn line -> into FXO card -> asterisk -> SIP > phone > > i get an incredible loud echo in the SIP phone (about 0,5-1s) > (everything i speak into SIP phone microphone i hear in its > speaker). The person calling from PSTN is not getting any echo. Make sure you're not

asterisk-users-admin@lists.digium.com wrote: > It's running Debian Sarge with the stock 2.4.26 kernel (I > know it's still an "unstable" release, but I'd need to jump > through all sorts of hoops to get Woody working properly). I wouldn't make a fuss about this. sarge is at least as good as woody and much more up to date for the stuff asterisk can do /