similar to: interfacing asterisk with a legacy PBX

just three "stupid" questions I need to ask .. 1. what's the sequence to press on a SIP phone to transfer a call to another extension. 2. what's the same thing if you want to hold an incoming call, speak to the other extension, then pass the call? 3. what's the extensions.conf syntax to dial two SIP extensions at once? many thanks Dave

I have just compiled & installed the latest oh323, on a fresh asterisk installation however using a previously working oh323.conf file. When I try to dial an outbound oh323 call I get the following error : -- Going to extension s|1 because of immediate=yes -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's'

i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the

what is the format for an h323 entry in the dialplan? can I use chan_h323 without compiling anything else or should I compile oh323? basically what's the best way :) cheers Dave

i've just lost $2000 dollars or so on my first commercial asterisk installation .. i'm running a PIV class server, three Digium Wildcard FXO cards, and 10 Grandstream Budgettone SIP phones. The system was to be a PBX for a small company. After over 2 months of pissing about, the client has had his fill of asterisk problems, and asked me to take my equipment out of the building. Obviously,

Greetings all - I'm interested in using an asterisk box to supplement and add VoIP capabilities to our legacy InterTel Axxess PBX. After searching through the list archives and through google, it seems that the best way to go about this is to connect the two systems via a T1. Is this correct? The PBX currently doesn't have any VoIP capabilities, so that's not an option for

I want to interface asterisk with a legacy pbx that has around 23 extensions through my 8 fxs card, how do i work around this? Hint: I have already terminated 8 extensions from the legacy PBX, i was thinking whether i can peer the extensions from the PBX i.e like 5 extensions be peered to one extension connecting to the fxs? How can i do this? Thanks in advance, -------------- next part

Hi guys, Can I use a Sangoma a101 to interface a legacy pbx to an Asterisk server? The pbx doesn't have sip and I want to come in off of a sip trunk and interface with the older system. I know I can use a pri card to hook in to the phone network, but can I use this same card to feed back the signaling as if I were the phone company to the older system? Thanks, Tom

I've done a quick search through the old mailing list archives to see if I could dial an outside line through * using a quicknet linejack setup as an FXO. Is it posible to do so? It appears from the old mailinglist from marko.net that this not possible? Thanks. Bradley Greep

I've been reading the Linejack strikes again messages, and have another Newbie question is it possible to use a Voip Product as a Dialout line for * ? I have a Vegastream 100 Voip to PRI. box. With * can I use that as a Dialout / dialin box? The Vega100 does either sip or h.323. Thanks. Bradley Greep

Greetings to one and all on this fine list; We have the current system: Meridian 1 Option 11 +-------------------+ | | ISDN/30 (DASS/2) ===> |NTAK79BB (2MB Pri) | | |<-->4x16 port Digital / 1x16 port Analogue ISDN/30 (EUROIDSN) ===> |NTBK50AA (2MB Pri)

Hi, I would like to know what is the easiest way to compile two or more set of vectors or data frame, according to their index. They are interrelated to one another by their assigned index. for example: #data set 1 abc #output: X403 X408 X410 X415 X418 X419 X420 X423 X424 X425 X426 X427 549.58 541.91 544.18 549.37 555.54 540.83 543.26 544.26 546.85 548.98 553.10 556.49

hi, can someone who has used Budgettone phones tell me how to do the following: an incoming call comes in and is answered by the receptionist. she need to put the call on hold, speak to whoever the call is for, and either (after that) pass on the call, otherwise speak again to whoever was on the call and hang up .. so far i've got as far as a blind transfer by pressing transfer button and

Anyone interfaced * to the Nortel option 61c pbx via T1's, pri, etc? Need to begin planning the implementation, purchase cards, etc. Any recommended approaches, configuration problems, etc? Off-list is fine if you'd like. Rich radamson @ routers . com

Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The VM box, itself, is beginning to show its age. Big-time. We're thinking it might be time to look for a replacement. I'd love to install Asterisk with an FXO card or something, but I don't think it supports whatever protocol legacy PBX's used to speak to VM systems. If someone can tell me I'm

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 have two t100p cards installed. One card connected to the PSTN, the other card is hooked to the Definity PBX with a T1 cross over cable. The card connected to the PSTN works fine. The problem is with the card hooked up to the PBX. The light behind the card is green, but when I dial from this card I get an error. Its says it could start the D-Channel. Could someone help me out with this

Hi, We are currently in the process of installing an Asterisk box between our legacy PBX (NEC Xen Master) and the Telco (Telstra). Originally the PBX was connected to the telco via 8x BRIs. Our new Asterisk box has been setup with 2 OpenVox B800P cards. Each card has the 4 ports in TE mode and 4 ports in NT mode. The NEC PBX is connected to the Asterisk boxes NT ports and the Asterisk box is

Hi Everyone After speaking with my local telco service provider, Telkom, I am aware that either we don't have E1 services in South Africa or the technical skills of our the engineers leaves much to be desired. Has/does anyone know if the Digium E1 (E100P) card is certified/works in South Africa? Thanks for the help so far! Tielman /* Tielman Koekemoer Unix and

I'm looking at getting the Dev light applications from digium and I have 2 Createive Labs voip blasters. The voip blaster supports the G.723.1 codec. After looking at Gnome meeting it does not talk unless you have a quicknet card for it. Can I make calls using asterisk and the digium cards to the Voip blaster. ie. From PSTN to through the X100P through * into the Voip Blaster. (Voip Blaster