search for: astrisk

...audio is fine. Tonight I tried placing a call on hold using a Sipura/Linksys ATA (that is just hitting 'flash', which basically puts the call on local hold and starts music). The problem did not manifest itself. Has anyone else had this issue? Do you have a fix for it? It is an astrisk issue or an aastra issue?

I am trying to compile the astrisk-1.0.9 tarball on a RedHat 9 linux box with dev environment. I get a lot of the following as a result of a make /usr/bin/ld /usr/lib/crtn.o: invalid string offset 1>0 for section `.shstrtab' and final show stopper ./gentone busy 480 620 make[1]:***[busy.h] segmentation fault What do I n...

Greetings all! I've got some really silly questions. I'm a technical guy, and I understand how the astrisk server works and how VOIP works, etc... The problem I have is that at my small company we have a phone system with analog lines and everyone here is comfortable with the concept of using them. I've never seen IP phones in action so I don't know how they work from a users point of view....

Hi Once I installed astrisk , how do we connect with SIP interface ? Can somebody guide me how to integrate SIP interface with asterisk ? I want to use Astrisk just for IVR purpose. Thank you Luke -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-...

Hi, All! Hopefully someone can point me in the right direction. I have a Mitel SX2000 Light that I am interfacing with my Asterisk box over a T1. I can receive calls on the Asterisk box without any problems, but when I call an extension on the Mitel, I heard the ringing from the IP phone, but I don't see anything on the Mitel, other than channel 1 of the T1 showing as "Locked

Here's my planed setup : PRI from telco <--> (port 1 of A104d) * (port 2 of A104d) <--> PM3 The PM3, for those who don't know is lucent's portmaster RAS dial-up router. I had setup asterisk, zaptel, libpri, wanpipe (as I have sangoma's cards). In wancfg, I have port 1 as TDM_VOICE, with hardware echo on, span 1. Port 2 is TDM_VOICE, without hw echo, Clock as

Dear Members, I am searching for a new PBX for the company. My choice is Astrisk. My Boss wants background music via all the telephones. This is done in a conventional PBX that he wants, but I can use the Asterisk PBX if it can do this also. As I said he needs background music on every telephone this is not to be mistaken with music on hold. The bit stream is an MP3 file of 8...

Hi.., we have two asterisk server interconnected to each other through IAX2 trunk in two separate office. with this bellow configuration do we need to have Licensing for using G729 codec???? Office A --------T1 ----- Astrisk TE05P----------------IAX2----------------Astrisk Box -2 | | | |...

Hi All, I have installed asteriskNow with Asterisk 1.6.2.11 and FreePBX 2.7.0.10. I have configured x100p fxo card in my asterisk box. But in my cli mode i am not getting the command *"dahdi show status"* Output of CLI : astrisks*CLI> *dahdi show status* No such command 'dahdi show status' (type 'core show help dahdi show' for other possible commands) I really not understand why this command is not avilable in CLI mode. Please help me to resolve this issue Michael.k -------------- next part ---------...

I have question thats not really about astrisk but I figure you guys are doing this sort of thing. We use Aastra 6757i phones. there is some support for XML. the question is how would i go about learning to customize these phones? _________________________________________________________________ Bing brings you maps, menus, and re...

Hi, I am thinking of starting a project at Work to pilot to use of Astrisk and VOIP, and would like to tie in several projects together. Currently we are looking at purchasing an ISDN and H.323 based video conferencing system from Polycom. It suggests the use of 3-4 BRI lines (ie 6-8 B channels, but in the form of individual BRIs). We need ISDN for the video conferenci...

...ng is i am generating a callback if some one calls at a specif access number on asterisk, Asterisk sends a busy signal to the calling party that he received a request from party and then sends the call back to the person from where asterisk received a request but in From field as you can see below astrisk is sending the calling ID as "asterisk" and username same , What i want is that it should forward some CLI in From Field , I have done my best effort but still not resolved i am adding a callerid in script still same please help me if some one can IP1:5060 -> IP2:5060 INVITE sip...

...gives me a beep-beep-beep like I have an invalid line when attemping to dial it. *77 exists in my extensions.conf and I am able to make and recieve calls from the extension I am using. I didn't think this sounded like an AMP problem as the conf files has the entries in there. I looked in the astrisk "full" logfile but saw no errors pertaining to this nor did I see anything in the console when I connected with asterisk -rvvvvv and did a sip debug ip 192.168.5.114:5060 (my extensions's ip and port). Where should I begin with troubleshooting?

After searching the archives, I came acrross a few people mentioning this, but I never saw anything about what became of it. Has anyone tried to make a virtual modem that could be directly handled by astrisk, I saw a while ago that someone was going to try and make one using the same DSP libaries that the WinModems use, but then nothing. Would do this even be possible, and if so, what kind of connection speeds could one hope to achive, with the compression and such? I don't need a 56k connect (...

Hi All, I am wondering if I will be breaking the GPL, if I write for example, a channel driver or make some modifications to the astrisk source code, to interface at RUN TIME, through sockets, with a proprietary system. Eg. 1. I write chan_xxx + modify asterisk source (make changes + new code publicly available) 2. chan_xxx supports hardware by XXX Corp. 3, XXX Corps interface is proprietary. 4. I write a layer over XXX C...

I'm about to download Festival source, apply the astrisk diff's, and initiate basic testing. Thoughts are to download v1.4.3 (latest per the fesitval website. If anyone has an existing how-to, install notes, tips, or any suggestions I'd greatly appreciate it. Direct email is fine if you'd rather not post them. Thanks, Rich radamson@routers...

...Posted To: Asterisk >> Conversation: [Asterisk-Users] Is there any MFC-R2 implementation for >> asterisk? >> Subject: [Asterisk-Users] Is there any MFC-R2 implementation for >> asterisk? >> >> >> Hi guys, >> Is there anyone has implemented MFC-R2 for astrisk? >> >> Regards >> Herry Sitepu >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >>

Hi, When I make a call from my asterisk and it is passed thru another astrisk eg. iaxtel, I got 'Answered' in my astrisk, and bill-sec is start counting as soon I get connected to the other asterisk, and not if the party on the other asterisk server pick up the phone. So IF the other party are not aswering the call at all, I still get Answerd and billsec in cdr. Wha...

Hi together, i try to compile astrisk-oh323 like described in the Readme - pwlib V1.6.6 (janus) - openh323 V1.13.5 (janus) with make-patch - asterisk V0.9.0 i got the following error gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/src/redhat/BUILD/asterisk/inclu...