similar to: I don't listen first seconds of audio from call - Asterisk integration with old PBX

Hi all, My scenario is this one: LandLine------------------E1---------------|-------------| |-------------------| |OLDPBX|-------E1-----------|Asterisk1.2.5|-----VoIPusers GSMGateway---------Analogue------ |-------------| |-------------------| What is happening: 1- SipUserAgent "A" Dials a call to a Local Extension "B" in the OldPbx 2- "B" , the called party

Hi, I've been debuging the call disconnection problem in our architecture: PSTN---E1---OldPBX---E1---Asterisk This is our problem: -SIP user agent "A" calls a pstn phone "B". -"B" hangs up the call. -SIP user agent "A" starts listenning busytones... But the call still on. (and being payed). - Call only ends when it is correctly hanged up in the

I'm sorry, i send mail in asterisk-bug, but asterisk-users is better for my problem Hello, During a call with canreinvite = no, at the beginning of the call I lose 2 seconds of audio. is obvious when I call autoattendant. schema: SipPhone --> Centrex (asterisk 1.4.24.1) --> Voip1 (Asterisk 1.4.24.1) --> Operator SIP capture of voip1: - Executing [0825387205 at

Calling all members of the asterisk community, I am posting about an old issue that has been reported many places and times online, To my amazement, there has yet to be anyone that has reported any solutions to the following problem. "Initially when putting callers on hold, it plays between 30 and 60 seconds of old audio that was on the stream in the past. Then after that 30-60 seconds, it

Could anyone help me set up Asterisk in such a way that it makes SIP-SIP transfers using the REFER / NOTIFY method according to RFC-3515 ? SCANARIO: - Asterisk registers with PSTN<->SIP VoIP provider "V" (Vonage) as a friend - Asterisk is located in Europe, Vonage in located US. - Asterisk acts as an autoattendant located in Europe. - Asterisk answers and incoming call from

Hi all I'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX

I've tried to transfer a call using the Flash command, but with my configuration it doesn't work. I have a traditional PBX connected with a zap channel to Asterisk that acts like an IVR: TELCO line --> traditional PBX (FXS) --> (FXO) Asterisk >From the TELCO line I can make a call to the traditional PBX and reach Asterisk, the IVR system on Asterisk answers the call and I can

Hello All, Figured I'd ask this. I have an asterisk running with some X100P clones. The system runs fine. My question is, the lines are tied into a panasonic dbs analog extension. When you call through, all goes well except that the call does not know it is connected. This causes problems if lets say, I want to do a 3 way conference, since my cisco 7960 does not see the call as connected,

Folks, For the last few days I've been trying to experiment with a Panasonic PBX and an X100P but have run into quite a few problems which I am not sure if they can be solved with this type of card (how about TDM01B?) 1) I wanted to use *'s IVR capabilities, so I routed the calls to the extension where the x100p was connected to. Asterisk should answer the call, playback a message,

So I have whose autoattendant is colliding with their extensions... Quick fix anyone? Second someone presses say a person's extension (101) ... Autoattendant sends them to the first context... [companyx-main-aa] exten => s,1,Background(companyx/companyx-main) exten => s,2,Background(silence/10) exten => s,3,Background(companyx/companyx-main) exten => s,4,Background(silence/10)

Hi all, I've bought a TE110P, and received it today. So i decided to install Asterisk@home 2.7 with this card. In the past i had experiencies with X100P (clone card) and it never take me so long to reboot the machine.... Machine: P4- 2,8Ghz 1GRAM TE110P What could be wrong? Best regards, Marco Mouta

Hi, At my house, I have two POTS lines. Both are connected to my * server on a TDM400P card. As an example, say the phone numbers are (919)555-1212 and (919)555-1213. I also have four SIP extensions, an ATA with a fax machine, and a DID coming in from an ITSP. I have an autoattendant configured that talks and allows users to forward to the extension they choose, but my family doesn't like

Consider this dialplan fragment, where the call is being dialed into [macro-process-routing] over an iax2 channel from another (same build) Asterisk server: [macro-process-routing] ; This is the entrypoint of the debug call but is also refered to by Macro(process-routing) elsewhere in the dialplan ; XXX-NNN-6800 exten => _6800,1,Macro(6800-interceptor) ; This is matched when 8 is

Just to make sure this isn't a typo in your original email... Is this example from your zapata.conf? Also, the extension you have shown are in extensions.conf not zapata.conf correct? Here is an example of a good zapata.conf.... [channels] language=en busydetect=yes faxdetect=both busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes useincomingcalleridonzaptransfer=yes

---------- Forwarded message ---------- From: Marco Mouta <marco.mouta@gmail.com> Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: asterisk-users-request@lists.digium.com Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me

Hi all, I was thinking about integrating an old PBX with Asterisk and I was wondering some possible configurations. The question is: which is the best way to let the 2 systems interact ? I can imagine some possible scenarios: - scenario 1: I want to use other then old PBX terminations (ie I have to link the 2 systems with some internal number line) In this scenario I could think to give each user

> Hi all, Hi Marco, > I was thinking about integrating an old PBX with Asterisk and I was wondering > some possible configurations. You didn't mention the number of lines your PBX uses, but think of a third scenario: Install an asterisk with twice the number of BRI/PRI-Ports your current PBX has. Connect half of them to your carrier, the other ones to your old PBX (Some sort of

All, Has anyone ever seen a problem where the autoattend detects the first digit twice? What I am seeing is this: My extensions are 421-468. When a caller calls in and dials exten 433 from the autoattendant, they get exten 443. This is happen for any extension that is valid in the 44x range (i.e. 42x -> 442, 43x -> 443, 44x -> 444, etc.). I am seeing this problem about 1/3 of the

I've spent the past several weeks reading up and playing around with Asterisk while I've been waiting for an ISDN card I got on ebay to arrive so I can really get to business. I'd just like to run my project ideaa by some of you to hopefully get a little feedback. I aplogize if this ends up being a somewhat long message. In the Marine Corps we've somewhat recently started using

I'm trying to get into the world of Asterisk in order to use the voicemail and autoattendat features (and more stuff later) with a Redcom switch. But, I've only started and haven't gotten to that yet. At this point my solitary goal is to talk to the autoattendant via an SIP phone (SJPhone). I've spent countless hours trying to find the documentation I need to accomplish my goals