similar to: Silly questions due to ingrained knowledge of analog phone use.

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-users/attachments/20130511/a6d32bfe/attachment.htm>

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, Some odd behaviour here. A friend and I were talking tonight, and it seems we have both seen the same problem. We are both using aastra phones (I am using 9113is). We have a connection to and from providers via SIP and IAX. When I place a call on the local hold of the phone, and then pick them back up I can hear them, but they can not hear me. However, if I park the call,

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 reviews organized in one place.

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 conferencing to communicate with

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 Kbs. At the server

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 need to fix in order to get a clean make?

I have asterisk 1.0.5 installed via apt on a debian system. It's a custom distrobution called Voyage Linux that runs from a flash card and I have a hard drive installed with mysql installed on it as well as apache. I have been though the AMP install guide (asterisk management portal) and in the interface it has a place for me to record new IVR menus. I have to dial *77 to begin recording

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

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

hy Hope everyone is fine, I have one issue coming in asterisk , What i am doing 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

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.com

Hi there, I was wondering if anyone has successfully used Asterisk as a dedicated Analog PSTN gateway to take the place of, for example, a Mediatrix 1204 or an 8 port model? Basically, I am thinking of using a Linksys SPA9000 as the PBX and just need an Analog PSTN gateway for 4 to 8 FXO lines. It does not sound like the Mediatrix 1204 does a very good job and I figure I can build a much more

hi, all after my installation of asterisk and adds-on . when start astrisk, error accours as follow: [Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available what's wrong with me ? Thanks. -- Best regards, Sucan

The last thing that I read about it was: Steve Underwood [steveu@coppice.org] wrote on Sep 3: >> Is E&M designed to work with the E1 driver code? I think probably not. I >> had to fix some things to get proper access to the CAS signaling bits >> when I implemented MFC/R2... So, apparently he implemented it. I was trying to contact Steve, but he isn't answering me. Does

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/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c

Hi, how I can connect Astrisk to my iptel account??? I have try to this configuration, but it doesn't work: In sip.conf: register => my_account_name:xxxx@iptel.org [iptel.org] type=friend host=iptel.org fromuser=my_account_name secret=xxxx nat=yes in extensions.conf: [fromiptel] exten => my_iptel_number,1,Dial(SIP/phone1,20,r) [toiptel] exten =>

Does anybody has a php code for using DBput (DBget, DBdel) from a web interface, which database is used for astrisk? bye Ronald

Has anyone tried to set Message Waiting Indicators up when public network access and voicemail service are managed by an Asterisk server TDM-connected to a legacy PBX serving analog and digital phones ? For instance: Location 1: - 200 users on a legacy PBX - among those users, 50 have access to voicemail service - TDM trunk to Location 2 Location 2: - 100 users on Asterisk - PSTN access - TDM

Dear folks, I have problem in fax reception. The astrisk detects the fax tone and jusmps to the fax extension and rxfax application starts and the max machine starts the fax but saddenly stops and seems the rxfax have died. It doesnt returns, not files in the output dir and .......... Anyone have any idea or help how could i get whats wrong here. Regards. M. Shokuie Nia