similar to: logger reload/restart hanging

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

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

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

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, 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, 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.

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-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 /

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

Hi, I am implementing T.38, and finding a problem getting boxes that work with T.38 for testing. A lot (maybe most) ATAs now claim to support T.38, but I'm finding a lot of these lie. I have one box here that just crashes when it hears a fax tone. :-) I'm looking for boxes known to implement T.38 properly, and which really work in the real world. Regards, Steve

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

Hi there, I'm using ser and asterisk together. Asterisk for voice mail etc and ser for registration of the users usig database. I can restrict forwarding calls from another sip proxy to ser (using proxy_authorize) but how can I restrict access to asterisk ... Now everyone can forward calls to my asterisk and can place pstn calls. Thanks in advance, Pavel -------------- next part

Hello, I would like to use Intel Blade machine for running Asterisk. Is there anyone who already use Intel Blade server for running Asterisk? Can you please explain, how perform Asterisk with Intel Blade machine? I would appreciate for giving me feedback regarding this issue. Regards Nahid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi When I register a SIP or IAX client to asterisk and I dial to it from another UA then there is no problem at all But, when I register two or more clients to the SAME peer (with the same user/pass) and I call to this peer.. Then only the UA which registered the last will ring.. Others don't ring... What can I do about this?? I would like to register for example 10 UA's to the same

I get seg. fault with my * box. at the crash time i had about 35 Bridged Channel. i have: - dual xeon box (3.2Ghz) - 2Gb of memory - E7501 chipset motherboard. - U320 scsi disks - intel Gb ethernet device. - i only use sip for clients (no fxs in box) - TE405P for fxo (with 4 atran TA750). - ulaw is used as codec and echo cancellationo is enabled. but the core dump file has nothing to show with

Hello All, I'm having problems with international calling via Global Crossing. I'm told I need to send a true ani versus a sudo ani. What is the difference and how can I configure asterisk to do this. Global Crossing is denying calls with sudo anis. Thanks, Keith

I've successfully got inbound/outbound calling working with our Asterisk using the Asterisk-OH323 channel driver. We are using a parent gatekeeper and the NuFone H323 channel driver would not work with the parent gatekeeper... I'm trying to determine a way to ensure that the line used for outbound calling is always available i.e. like trunking.. >From what I can tell when I place an