similar to: TDM400P FXO callprogress doesn't detect remote answer

My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes

Hi, all I have couple of SIP phones and they are in [from-sip] context. I also have an IAX2 phone. I have put this one in [iax-user] context. I want to make calls between SIP and IAX2 phones. If I put them all in same context all is fine, however when they are in different contexts they will not call each other and I will get message (in * CLI) that particular extension does not exist in a

>From this list I found that I could use SetGroup and CheckGroup to do what I wanted. But I'm not quite sure how I do it. The case is that I have 3 user groups, and one main group. The main group is for all the incoming calls from external phones. The main group should be allowed to have 3 calls at the time. The 3 user groups are internal groups that I want to limit by ony having one

Here is some more stuff to add to the confusion about the "callprogress" option. I currently have my * system operating with a T100P talking to an ADTRAN TSU600 channel bank with 8 FXO ports connecting to the outside world and Grandstream SIP phones as handset extensions. At first I naively set "callprogress=yes" in zapata.conf. The results were typical of what many

I'm running Asterisk 1.6.2.1 with DAHDI 2.2.1 using a TDM410P. I have callprogress=yes in chan_dahdi.conf because, from everything I've read, it is needed when using call files over PSTN, which I DO use occasionally. I know that callprogress=yes is "experimental" and causes some issues. We've never experienced any problems when making local calls over PSTN with callprogress

Hi, Would anyone happen to have experience with the callprogress option? What I'm trying to do is use a couple normal POTS lines in a queue setup where it will call the queue members to pass the call to them. Of course the problem I'm running into is that under normal conditions the lines register as answered immediately and the caller gets transferred to the ringing line which

Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I live in Sweden and I am having problems getting asterisk to properly detect when a caller hangs up. And yes, I DO have disconnect-supervision on my line. Also asterisk sometimes misinterprets the disconnect-signal as another incoming call. This usually happens if I hang up first and then when the caller hangs up, asterisk treats it as a new

Hi Guys, I recently got a TDM400P 4 FXO for use in the UK, this at the time seemed like a good idea as I had good results with an X100P clone. Installation went great and call clarity is excellent no echo like I had on the clone card. My problems start with detecting hanging up the line. If a person calls into the system and speaks to me on a SIP phone when I hang up the call clears

Hi All, This must have been asked before, but despite my best googling efforts, I can't find a solution or reason why it won't work. Please accept my Apologies. Is it possible for an outgoing call on the FXO port (PSTN line) to not always register in the call logs as being answered? I've tried adding the 'r' option to the end of the Dial command, but that does not appear to

Hello, I've got an Asterisk server with a TDM400P with 2 FXO modules and 2 FXS modules. This server is connected to 2 PSTN lines and 2 analog phones. In my Zaptel configuration, I've defined 2 groups : one for the 2 FXO's and one for the 2 FXS. The asterisk server is just used to add a little IVR and Voicemail service. Eveything works fine, but sometimes the conversation is

>From what I've been reading about the callprogress option, it seems like it will work properly only with a T1 or PRI in the US. Is that correct or are there still issues with call progress detection even if those qualifications are met? Thanks, Shaun Tierney

Howdy, 1 x TDM400P card with 1 x fxo module. 1 x BT Pots line. Location - UK Calls work fine outbound but i'm unable to pickup the inbound calls. Asterisk debug: Asterisk -vvvvvvvvvvcg *CLI> -- Starting simple switch on 'Zap/1-1' -- Executing Wait("Zap/1-1", "1") in new stack -- Executing Answer("Zap/1-1", "") in new stack

I have been having some troubles with the zaptel channel on what appears to be the inbound process. The box is running the stable CVS code and has a TDM400P 4-port FXO card in it for analog connectivity. Channel 1 is the only active port on the card at the moment as we only have one analog line. What has been happening is that it looks like Asterisk has been detecting an inbound call even though

Hi- I'm totally stumped configuring my TDM400P with one FXS and one FXO module. Before I got the FXO module, I used to have an X101P, and everything was working very well. Now * doesn't seem to recognize the FXO channel. I've searched the wiki and the list archives. Stock Debian 3.0 stable installation. Any advice? Thanks. -- David Here's my configuration: modprobe zaptel

Hi all, I can't figure out why my TDM400P (with one FXO plugin) won't answer any calls. There are no messages in the Asterisk console when a call is placed to the FXO line from the PSTN. Any suggestions would be most appreciated. The wctdm and zaptel modules are loaded: [root@steerpike asterisk]# lsmod | grep wc wctdm 37952 0 zaptel 189700 1 wctdm

Just added a new Grandstream BT102 to my network. Its running new firmware (Ver 1.0.5.22 of 2005-01-21). I could NOT get the damn thing to (SIP) register.... Gripe 1: The New Firmware does NOT show the current version of all the firmware. You have to ask the phone manually with its menu button. Gripe 2: It does not show '****' in the the two password fields... This is what caught me - I

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

Hello all! I have tried and tried to resolve this one to no avail. Hopefully one of you can help... The system in question is a Compaq Evo D600 (iirc) business desktop, with a 1.4GHz Pentium 4 and 512mb of RAM, running a stock install of PoundKey 1.2. It has two Digium cards installed: a TDM400P with four FXO modules, and a TE110P hooked to a Carrier Access Adit 600 which serves 8

Just installed the new 4-port FXO card and moved two pstn lines from existing x100p cards to ports on the FXO card. All zapata.conf entries that were functional on the x100p's were copied to the new FXO channels (including callprogress=no). Observations thus far: 1. asterisk will spontanously decide a pstn call has arrived, and ring the sip phone designated in the dailplan. Verified