similar to: Sip and ISDN problem

Hi, Did anyone try to set up alarmreceiver application over IP network? Which ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck. Maybe I did something wrong with alarmreceiver.conf (I tried diverse settings, but nothing worked). Sometimes alarmreceiver is able to get some events but sometimes not. I think Linksys PAP-2 has a problem with recognizing digits in appropriate

Hello, I'm familiar with Pickup/PickupChan for taking a ringing call, but does anyone know how a phone can "steal" an already answered call from another phone? Our users have decided that call parking is too long-winded and don't want to use that. For example: phone A calls phone B, phone B answers the call, phone C dials something to "steal" the call from B, and

I have a strange problem when calling some numbers with asterisk, I get an hangup for busy condition even if the phone at the other end isn't busy. I can route the calls via SIP to another carrier and then I have a SIP code 486 or I can terminate them on digium cards (E1) and I have an Hangup code 17 I know for sure that one of the numbers is hosted by a different provider than the one

Hello everyone, I have been asked for "directed pickup" and saw that both "PickupChan" from bristuff and "Intercept" applications do the dirty work. I have tried both on asterisk-1.0.9 ( BRIstuffed-0.2.0-RC8o ) but I always got an error when trying to pick the ringing call. the debug says: SIP/marco-73a0 is ringing -- SIP/marco-73a0 is ringing --

Hi All, I've installed Enhanced PickupChan on asterisk 1.2.14 using howto from http://www.thorsten-knabe.de/linux/asterisk/pickup.jsp . from extensions.conf: exten => 0,1,Dial(SIP/eosoiris|20|tTrR) exten => 200,1,Dial(SIP/dzalewski|20|tTrR) exten => _7.,1,Pickup2(${EXTEN:1}) When I try to pickup ringing SIP channel from other IP headset I go disconnected. here is debug from

Paul Madley wrote > >Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 >release, and therefore I don't think any config changes will fix it. >We've been told to roll back to our previous 1.4.13 installation. It >also seems to manifest itself in "ghost ringing" as I've called it; >place a call to a SIP extension, then put down the

I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729

Hello everybody. I have 100+ sip extensions over 100+ different geographical places. Many times, the phone is intentionally left off hook, or the user change the phone from the sip line to the land line, simply because they want to call and the sip phone is the "new" one... other times, the sip line, which are linksys' pap, tell me that is in congestion or "circuit

Hi, I have configured my SNOM 360 to monitor another extension by setting the following: [default] exten => user1,hint,SIP/user1 The next step was to define a function key on the phone as an extension with the value <sip:user1 at 192.168.0.101> and later with <sip:user1 at 192.168.0.101|*8> When someone now calls extension 97 (which is the number of the corresponding phone),

Hi, I'm looking forward to install an asterisk-server to perform ISDN-to-SIP- bridging. But for the times when I'm not available via SIP I also need call- forwarding-features. Afaik it is possible to directly forward a call on ISDN instead of opening a second ISDN-channel additional to the incoming-ISDN- channel and forward the call "by hand", right? What if I need features

Hi, I am just moving from Asterisk 1.2+bristuff up to 1.6.2, a huge leap :) I was wondering if someone could point me at 3 things that I appear to have "lost"? 1) ZapEC(off) - Is there an equivalent dialplan command to request no EC on a channel before dialling in DAHDI? 2) rxfax(file.tiff) - I have found ReceiveFax(), but I am aware that much has happened in the faxing stakes

Hi all! I have the following setup: Phone lines -> traditional PBX -> Welltech 3802 -> VPN -> Asterisk -> Linksys PAP2/Welltech ATA-151 -> phone There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2) PBX extensions. Asterisk is a proxy here. Each device successfully register itself. I tried the setup above with Linksys and Welltech devices as well. I setup

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to pick up a ringing SIP phone (203) across the office with exten => *9,1,Pickup(783743) where 783743 is the local part of the number that our ISDN works on. I tried all of these first: exten => *9,1,Pickup(203) exten => *9,1,Pickup(SIP/203) exten => *9,1,Pickup(203@internal) and got a "declined" message back from my phone (snom 300), so I then

All, I need to run this by everyone and see if someone has any idea's. I have a asterisk server setup and currently am receiving the inbound calling number where the name should be. My setup is.... One pri terminating into a Cisco 2431 router Sip messages from the Cisco get sent to a asterisk server linksys ata's a each remote end. I can receive the calling name if the call originates

I'm working through my list of features people will expect, and Hold Pickup is at the top at the moment -- has anyone done any work on this? We've had some unpleasant experiences with call parking, and everyone seems to like the Hold Pickup model. If you don't know what I mean by Hold Pickup, it's sort of a reverse transfer; pick up the nearest phone and dial

Should setting a namedcallgroup & namedpickupgroup supersede numeric callgroups and pickupgroup ? I've got 5 peers on my 13.7.0 box, Three of them have a namedcallgroup & namedpickupgroup of 'kiniston' and Two of them have a namedcallgroup & namedpickupgroup of 'sanday'. I'm not specifying a numeric callgroup or pickupgroup so all the peers are defaulting to

Hi, I have built another Asterisk box using one ISDN HFC-S card and Bristuff-0.2.0-RC8p. But this time it behaves very strangely. Asterisk simply hangs and in logs I receive something like this: --NOTICE [1197]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 --NOTICE [1197]: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 And when I want to call a ZAP channel I get

All, I know the question has been asked before, but any of the solutions posted in the past have not solved my problem. I have got a Asterisk setup using a P4 1.8 / 512mb server running Redhat Enterprise 3 and 3 grandstream budgetone phones (plus a couple of xten clients on windows) and I'm at advanced stage of testing to see if asterisk will fill our needs as a PBX using voice over IP

hi, how to set caller ID for internal SIP users when dialing out on telco ISDN p2p (hfc card) line? I need to setup numbers from 0 - > 9 (10 sip users).internal caller id is working correctly .. from 0 to 9 ,but when I dial on isdn telco line -> gsm show only our prefix number ( xxxxxxY ) , only 6 (x) numbers of 7 (Y). here is extension 1 dialing on isdn line but on gsm is only 6