similar to: ZAP channel on TE410P doesn't hang up (Plain Text this time)

Hello * users I've have a rather disturbing problem, which I don't know how to debug or how to solve, but first a brief description of the set up. One Asterisk server with a TE410P card installed (first line used on this only), and a number of Wellgate 3504A (4 port FXS devices with SIP firmware). There is no connection from the Asterisk server to the outside world or any other

Hi I have kind of an odd problem. When dialing in from an outside line via a TE410P card it seems like MusicOnHold and MP3Player doesn't work properly (for me anyway). The remote end which is calling * doesn't hear the music but just keeps ringing. But if I insert a Playback("file_which_dont_exist") just before the Moh or MP3Player I can hear the music. Actually I observed the

Hi there I have some problems in setting up a connection between * and a Wellgate 1501 (1FXS). Either I can dial from * to my Wellgate or I can dial from the Wellgate to *. On the Wellgate you have to setup both "Line number" (which is the number it answers) and "Line account" (which is the account it registers as). Should I expect any problems if these are set the same?

Hello there I'm quite a newbie in the IP Telephony area. I'm playing a little around with a setup with one linux box with a e100 p card installed, which functions as an Asterisk gateway and a oh323 GK(Gnu Gatekeeper). I have two h323 phones, Welltech WellGate 1501 and 3502. So far I've managed to get the two IP phones and Asterisk connected to the GK. I can place calls from one

Hi everybody I've just installed a new Redhat 8.0 and configured it with Asterisk, zaptel and libpri. Afterwards I installed a TE410P and configured this as well. But when starting Asterisk I get the following error message: ------------------------------------------------------- -- Registered channel 1, PRI Signalling signalling ..... -- Registered channel 15, PRI Signalling

Hi there I have a Welltech Wellgate SIP device and I want to be able to do a supervised transfer. I've read that in order to do that I have to use flash hook. The problem is just that I can't flash hook with this device. I'm in contact with the developer of the SIP device but don't know what to tell him in order to get him to fix this. What is happening when you flash hook, I

I have a problem with my Welltech Wellgates. I can't call any extension which starts with or includes * or #. When dialing it responds fine but after some seconds I just get a busy tone and on the Asterisk console it says "SIP/2.0 484 Address Incomplete". Don't know if it connects to the DTMF payload type. Yesterday I made som different tests and observed that if DTMF payload

Hi there How do I change the dialplan runtime, if I for example wants all calls on the main number to be answered by a voicemail (when it is out-of-office hours). I want to be able to change the configuration by pressing a DTMF combination e.g. *82. Can't figure out whether it is necessary to change contexts or how to do it. I have read a lot of examples and config documentation, but I

Hello all I've discovered that SIP channels sometimes get stuck in *. I've read some posts from Fri 29 Aug 2003 which mentions this issue, but there doesn't seem to be any final answers I don't know if this is related to the 0001604 bug? Below is a list from one of the incidents: I know the (d) means that it is scheduled for destruction but the 10.1.1.45 channel hasn't

Hi I have an Asterisk installation with some SIP and MGPC devices, and I also have a TE410P on a E1 line. If I make an outside ISDN data call to asterisk the phone rings as usual and if I answer it, I just hear some clicks. I've read that the D-Channel has information about the call, if its voice or data. Is it somehow possible to end/ignore this call already before it is ringing?

Hello fellow * users Hope this isn't a stupid question; I've done my research but could not find a proper answer. I have 8 different destinations which I want to match. The numbers are: ###### 00 ###### 20 ###### 30 ###### 40 ###### 15 ###### 35 ###### 12 ###### 44 Right now I've solved it by doing this: exten => _######[0234]0,1,HangUp exten => _######[13]5,1,HangUp exten

Hi there I've installed a 12 port MGCP gateway, (Hitron MDU-5612), which works ok most of the time. Sometimes when talking to the outside world, (via a TE410P), the line gets disconnected. I think its related to MGCP because I've also setup some SIP devices which doesn't behave like this. I've examined the logs but can't find anything useful, it looks like its the MGCP device

Hi all, I'm using following software and equipment and I have very strange behavior: Asterisk CVS-NHEAD-05/30/05-16:42:41 H323 gatekeeper - GnuGK 2.2.2 IAX2 client - Firefly 1.9.8 build 3945 H323 client - SJPhone Build 1.50.271d H323 gateway - Welltech Wellgate 3504A When I dial from Firefly (IAX2) -> SJPhone (H323) everything works as expected. When I dial from SJPhone (H323) ->

I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a "Nothing to pick up" answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to

Hello there Yesterday I updated my pwlib, openh323 and Asterisk from CVS. After making "clean opt" in pwlib and openh323 and make "clean install" in Asterisk i get an "Undefined symbol" error when I try to start Asterisk. As far as I can see its when loading the h323 channel driver the error occurs. Do I have to update other things as well, by reading the various

Hello * users, I have a problem with authenticating my SIP gateway endpoints with *. The gateway I'm using is an AudioCodes MP-124 (24 port) If I setup my sip.conf with an empty secret= option everything is working ok and I can initiate and receive calls, but if I want to be able to authenticate my SIP gateway with * something goes wrong. Below is a little snip of my sip.conf file: ---

Hello there I've got kind of an odd problem. I have a setup with a lot of SIP channels and a 30 channel PRI which is working perfectly. In order to bill the customers I fetch cdr files from the PRI provider, those files are generated every two hours. If I compare the CDR files from Asterisk and the CDR files from my provider there are large differences in the billsec fields in some of the

Hi, I still cant dial out on Zap and I really have no clue why. I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card 4 ports, 31 channels each and able to receive incoming calls and fax perfectly. I've done this in my dial plan. exten => 111,1,Answer() exten => 111,n,Ringing() exten => 111,n,Wait(2) exten => 111,n,AbsoluteTimeout(30) exten =>

I'm not sure if it's the same problem but your error message likely the same. after i additing pridialplan=local in the zapata.conf i'm able to make outboundcalls (located in germany) marcus -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Gesendet: Dienstag, 11. April 2006 16:33 An:

hey * folk, am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls [1235] host = dynamic secret = somepass context = default type