similar to: Zap channels - help

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?

Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josu?

Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josu?

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

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

Hi All, I would like to explain the layout that i am trying to achive. I am so helpless on this regard. So here is the story ........ " This is with regard to the setup which you can find at the "Asterisk The Future of Telephony" , chapter 11, page # 196-197, I am attaching the picture for your information. Now I am taking a challenging step to of integrate IP PBX with our

Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (>5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to

http://voip-info.org/wiki/view/Easy+PABX With Easy PABX you can create your own virtual PABX online in just minutes. Easy PABX is based on Asterisk and best of all - it's completely free. Regards thorben.dk

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

Guys, I need to make asterisk connect to another device over a pri using QSIG signalling (master/slave). Is this feasible and has it been done before? Thank you! al. _________________________________________________________________ Get ready for school! Find articles, homework help and more in the Back to School Guide! http://special.msn.com/network/04backtoschool.armx

Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam -------------- next part

How do I get * to take an incoming oh323 call and let it dial a number? I.E. if my boss sets up netmeeting with the gateway as my.pabxbox.com, whenever he enters a number to dial it always just dials into the pabx rather than calling that number i.e. he wants to call 12345 he types it in and presses dial but it just goes to the message.... also I have developed an H323 client app which I

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello I am running an * box with just 8 extensions connected to our old Alcatel BCN 5200 PABX. The requirement is that we now scale it up to handle about 300 lines and get rid of our old PABX. Is there a way of hooking up 300 phones to asterisk without going via the PABX. I am more of a network person than a telecomms one so i may not fully

Trying again... *Summary:* I need to have 2 machines with 4 BRI connections, 2 in NT mode, 2 in TE mode and 1 machines with 2 BRI connections, 1 in NT mode, 1 in TE mode; What card(s) should I put in to these servers? *The long story:* I have 3 locations I want to connect using (*) servers. 1 of those has a single BRI with a Siemens DECT PABX. 1 of those has two BRI's with 2 Siemens DECT

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

Hi all, I am using Asterisk CVS, and I am getting a huge delay in dialing SIP. This Asterisk box is taking calls from a PABX over ZAP, then dialing SIP users. So, a user '0251' dials from his phone, the PABX sends it the my Asterisk box, no delay, then I get a 15 sec delay, before it actually dials the end SIP user. 1 -- Accepting call from '0251' to '0834541083' on

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

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