similar to: Re: Braodvoice - UK Non Geographic Numbers

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

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: > 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, 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

I finally got Solaris to successfully make asterisk, using these instructions: http://sunfreeware.com/programlistsparc10.html#gcc33 Now though, when I issue the make install, I get this error: mkdir -p /var/opt/asterisk/spool/system mkdir -p /var/opt/asterisk/spool/tmp mkdir -p /var/opt/asterisk/spool/meetme install -m 755 asterisk /opt/asterisk/usr/sbin/ install: asterisk was not found

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

Has anyone else had problems with users being able to press key tones during a voice prompt? I have a few users complaining that some systems will not recognize key presses during them. using current CVS-HEAD, linksys PAP2 UA's, rfc2833 dtmf mode. Thanks Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

> I have seen the following effect in Asterisk, though: where > it converts > an inband DTMF (eg coming off a Zap channel) into an > indication, it mutes > the audio where that tone is. But sometimes it leaves a > teeny bit of the > tone behind. > > If you take such a call over say IAX to somewhere and then > back out a Zap > channel, you end up with the

Heya list, I need some advice/experience. Some of our customers are asking us about DECT solutions for their asterisk install. Some others will not go to asterisk if there won't be a DECT solution. They now have a Siemens or a Samsung PBX. Those PBX-es come with a DECT basestation and optionally repeaters etc. All those basestations speak some own protocol to the PBX, so we cannot use them

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

What is CFU and CFNR? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michel Koenen > Sent: Tuesday, August 30, 2005 1:46 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] How to use * and # as part of > numberindialcommand > > > From: "Damon

Hi :) I hope someone has a hint concerning Early Media. The situation: My Asterisk is connected to small local carrier who works with several SIP servers. I traced some SIP headers and find something like this: Via: SIP/2.0 UDP sip1.provider1.de In the SDP part I found something like this: o=- 2268929 0 IN IP4 sip2.provider1.de c=IN IP4 sip2.provider1.de If I send

I have set up extension.conf and sip.con with default parameter of UNIVOICE server, but Asterisk show this message when I call a number: Sep 29 11:34:52 WARNING[4179]: chan_sip.c:1899 create_addr: No such host: univoice,Ttr Sep 29 11:34:52 NOTICE[4179]: app_dial.c:1109 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is

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

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

Can anyone comment or share experences with using Dell 2850's with Asterisk. Proposed config is 2850, 2 x 3.6g procs, 2 g's of ram, 4 x 36g 15k rpm drives raid 10, Digium TE411P ( the echo cancelling cards ). Expected load is 1 or 2 pri's (most likely 1 ) 100 Polycom phones on the local network, 15 phone on a remote T1. 6 phone remote via the internet using IAX, Voicemail for

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