similar to: Setting up PABX

Hi U can visit the http://iaxclient.sf.net for some opensource underdevelopment softphones. Take Care Obaid Amin Syed >From: Chris Albertson <chrisalbertson90278@yahoo.com> >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] Good W2K softphone >Date: Fri, 3 Oct 2003 23:00:13 -0700 (PDT) > > >I haven't

Hi All, Can Asterisk dial extension which resides in the PABX? (eg. 2000) Sip Phone <-----> Asterisk <------> ATA (FXS) <------> (CO side) PABX <-----> Extension (eg. 1000) (2100 & 2101) can my sip phone call to pabx extension 1000? What will be my dial plan? I know I can connect to 1000 by

? Dear all, We are connecting the PABX with Application server.What we are trying is that when a nbr 1800 (this is not registered in PABX) is dialled the pabx should route the call to Application server .The PABX should also have intelligence to route the call by itself for its registered clients.For this scenario to work please guide us what are the files we need to change and other necessary

Hi Folks, I am running a SPA-3000 behind a legacy PABX on an analog line. I have been able to set up a dial plan that sends outgoing calls out to the appropriate VSP depending on prefix, and that part and the incoming call handling works fine. I am now trying to implement call pickup (dial 6*) or manual call forwarding (flash, dial extension). On the first of these I have worked out how to get

Hi to all i think this is not a new problem for this forum....but its newest for me as i m a new linux lerner. Even if it is new plzz....reply me ur answer..n if its already asked n have solution..plzz forward the solution. My problem is here mentioned: I have fedora core 4 as a linux server. there r two external links connected to this. the settings are as: eth0 ->for internal (that is

Hello all! I'm VERY new in using VoIP. I'm looking for any tip or trick to connect a physically PABX behind an Asterisk-System(or similar) via an SIP to Analog- or ISDN-Converter. The point is, I _need_ to deliver calls to extensions in the connected PABX directly (in ISDN-speech "DDI" (DirectDialIn)) without intervention of an operator. Is this technically possible, and if

Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil

Hi all, I have an Asterisk server connected in a PABX (TELEDATA) by channel Unicall.. I`m having problem when somebody call from PABX to Asterisk.. Eg: When somebody dial 1234, I received 1111112222333333444444 in the Asterisk CLI... If somebody can help me... or already saw this... Everton Goularth Uberlandia - MG - Brazil _______________________________________________________

Hi guys! I'm having one or two issues with queues hosted by an Asterisk machine where the clients are on a legacy PABX - at least for the interim. I fully expect most of these issues to be non-resolvable, but thought I'd at least ask to find out if there is some way of working around the issues. The legacy PABX is an NEC 7400 ICS connected to Asterisk via an E1 ISDN link. Calls are

I'm new in installing wine in Sled 11.. So plzz could any one give me hint how to start this. I'm Studying the Linux life so that i could migrate from windows to linux.

Hi all I've got a setup where I use a Sangoma PRI card driven via Sangoma WanPipe to connect to a legacy Samsung PABX (I'm not sure which model) form Asterisk 1.8.11.0. The reason is the customer has a large installed base of Samsung phones physically connected to it and on each users desk. They did not want to spring for a complete replace of all their Samsung phones with generic, and

Hi all, I've been lurking here and reading the Wiki for a month or so now, getting information on the suitability of Asterisk for my installation. I'm responsible for the technical stuff at a mostly-inbound call centre in Melbourne, Australia. Due to our rapid expansion, it's getting close to the time to put our old beloved Fujitsu 9600 PABX out to pasture. I'm evaluating

Hi all,as good? It would like to know if already they had had success, in the integration of the functions of asterisk, with one pabx legacy (Avaya)for that pabx avaya, could use the voicemail of asterisk. For sample, user of pabx avaya, it would have its calls directed for not attendance and busy, for asterisk and asterisk, it would send the same one for the voicemail. Best Regards Josu?

Hi it's possible to use a Digium TE110P Single T1 / E1 PCI Interface for supply a E1 link to a old PABX ? Thanks

Good morning all, I have a question: if I want to connect an * to a traditionnal PABX via TDM04B to get PSTN line (incoming and outcoming call), are FXO enough (no analog phones connected to *)? I just want to add PSTN call (in and out) possibility for SIP or IAX UA. From what I saw here http://www.voip-info.org/wiki-FXO I'm right. Thanks. -- Daniel

Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks.

Hello, I'm researching the possibility of using VOIP (SIP) with an existing PABX system. Ideally, the setup would be to dial an outside line through the PABX (that would actually link to the the VOIP gateway). At this point I would prefer not to purchase a hardware-based VOIP gateway. I would prefer to use a software-based gateway for research & testing purposes. Could anyone please

Dear All, Hii, I am student working on a project called Network Radio modelled on the icecast project. Could u plz give me the design of Icecast project>>so that i can model my project accordingly viz by threads or sockets?? Plzz help me. Also could u telll me where i can get the source code for icecast in c++ or java. Thanks, Chetan Yahoo! India Matrimony: Find your life

Hi, I am having some confusion.It has been said that we can only estimate the future values using meanForecast +/- 2*standardDeviation. with 95% confidence.This means using this garch model we can only have a upper and lower limit of the values within which the next actual value is expected to lie.Then how come in research papers they plot the actual and predicted value so neatly.The simple

hy all actually i want to have a setup of five offices having round about 200 extensions ( each office having 35 to 45 ) which will be connected through asterisk. now either i should go for voip phones( hard phones ). or use any interface card to asterisk server to which the analogue phones will be connected. ----- if i use analogue phones in the above case ( we have analogue phones already )