similar to: Difficulty evaluating the return value of PlayBack (or any other extensions.conf 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, 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-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 /

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

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

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

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

Hi, all!!! What will Asterisk do in the following case: For example, we have 4 licenses, and have 4 simultaneous calls, using G729. Will asterisk allow incoming calls from peer, that can talk G729 and ulaw, and will it force it somehow to use ulaw in this case? All phones there in LAN behind Asterisk prefer GSM codec, so it does transcoding. So, what I mean is will Asterisk fall back to use

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

I have thsi code in a dial plan. The purpose of which is to set distinctive ring tones for internal and transferred calls. exten => _.,1,Noop(CALLERID_ALL=${CALLERID(all)}) exten => _.,n,Set(CallerIDNum=${CALLERID(num)}) ; This just shows a list of interesting variables and their values ; Comment it out when finished debugging ;include => macro-dumpvars ;exten =>

Hi, need a quick help ... it should be easy but ... exten =>_9898,1,Answer exten =>_9898,2,VoiceMailMain(${CALLERID}@domain) Accepting overlap call from '342' to '9' on channel 0/2, span 3 -- Executing Answer("Zap/8-1", "") in new stack -- Executing VoiceMailMain("Zap/8-1", "@domain") in new stack As you can see there

Hi, Our gateway has asked that we send DTMF as RFC 2833. Although chan_oh323 seems to do this, it doesn't specify the DTMF mode during the H323 setup headers. Is there an easy way around this? Thanks, Andrew

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use

Hi, I run asterisk with oh323 plugins.It runs correctly with sjphone H323 Gatekeeper. But When i run openphone it doesn't recognize my asterisk server like a gatekeeper !! What is the problem ? Thx

I am looking at building an IVR product with a few interesting features and need some more information about how asterisk and VoIP work and what I can get from them. As far as I can tell when I use ISDN/GSM telephone networks the DTMF information travels as data representing 'start tone' and 'stop tone' for each button pressed, it is then generated at the other end if an

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

[thread moved from -dev due to non-dev content] At 6:40 PM +0200 on 8/13/05, Andreas Sikkema wrote: >On Sat, 2005-08-13 at 12:44 +0800, Steve Underwood wrote: > >> He doesn't seem to really understand the strengths and weaknesses of >> IAX. IAX has drawbacks, but none of the problems he lists actually exist. > >OK, I'll bite ;-) > >How would IAX2 solve