similar to: Trunk calls ring internal analog phone

Hi All, I have a rather peculiar problem. Whenever I dial out over ZAP/g0 the phone will just ring and ring, even if I answer the phone on the other end. Whats strange is that the * phone will continue to ring even after I've answered and (sometimes) hung up the dialed phone. If I make an extension to just directly dial out on ZAP/1, its almost the same behavior, it will continue to

I have a situation when I dial out my Zaptel I am getting a recording that I need to add a 1 or a 0 and the area code with this number. I have tried appending this and the number going out the zap is 1NXXNXXXXXX so it is going out with 1 and the area code. Someone has suggested that maybe the zaptel is dialing too fast. My question is how can I add a pause before dialing to test this out. I am

Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. If i pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if i dial any digit than there is no ring on Analog phone after hangup. Log's looks like this

Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. When I pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if I dial any digit than there is no ring on Analog phone after hangup. Log's looks like this

I have a trixbox 2.2 and Nortel santral that are speak each other. I use digium TDM100M 2 fxs-2fxo. After I made yum update I had met with some problems when I want to make any call from extension of trixbox to extension of nortel. When I attend to log (/var/log/messages) I meet with these messages as you see below. When I try to make any call from trixbox extension the call seems established but

Below is a copy of my log, zapata.conf & extensions.conf that relate to the ZAP lines. Basically when we dial out it takes on 10-12 seconds before the ZAP line actaully picks up. I'm hoping to find out what the cause is for this as it's causing user grief with extremely long connect times, and I believe it may be causing issues of cross lines (an outgoing call gets mixed with an

Hi list, Attempting to get an ISDN-BRI line connected using an HFC-S PCI card together with Asterisk v1.4.14 and Zaptel 1.4.7 on a Debian etch system, I find that I can't access the card's resources because the channels are always be busy. An attempt to call out results in the following CLI output: == Primary D-Channel on span 1 down == Primary D-Channel on span 2 down

Using Digium fax I've tried a simple dialplan: '8447' => 1. Answer() [pbx_config] 2. Set(CALLERID(num)=xxxyyyzzzz) [pbx_config] 3. Dial(DAHDI/g0/1bbbcccdddd,,G(send)) [pbx_config] [send] 4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif) [pbx_config] 5. HangUp() But I doesn't work. It executes

Hello, I have a system with two cards: a HFC-PCI ISDN and a TDM21B (2 FXO and 1 FXS), running Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with freePBX beta5 dialplan. I have connected an analog phone to TDM FXS port, but when I pickup the phone to make a call, Asterisk "hangs up" the call. Let me explain: In another system, when I pickup the phone, Asterisk give me tone to dial: >---

A very good day to you all, We can't get the phones to pick up on an incoming call on analog trunks. We're using the digium products in the box, with snom phones internally. This is the output from the asterisk console: linux*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo pstn-incoming en default 1 pstn-incoming

Yo! I changed callprogress to no, and in wcfxo.c source around line 334 i changed the value 32000 and -32000 to 10000 and -10000 because it had something to do with the DC voltage when it was ringing. I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an interesting diagram of wiring that was incorrect for sending voltage to a phone or something like that. So put it

Hi list, After upgrading from Asterisk v1.2 to v1.4.14, all kinds Zaptel error messages related to my HFC-S PCI card disappeared, but now I can't access the card's resources because it always seems to be busy. Any idea why? Thanks, Jaap PS -- Below is some info regarding my configuration. =========================== Zaptel version: 1.4.7 (incl. firmware and modules). OS: Debian

Hi All, It's broken !! (drat) Asterisk if failing to load with the following error (taken from end of /var/log/asterisk/full) after adding bristuff. Can anyone help please? Jul 17 19:57:54 VERBOSE[2503]: == Registered channel type 'Phone' (Standard Linux Telephony API Driver) Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so]Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so] =>

Hello list. I have a question about how to read the incoming calls' callerid on an FXO interface of a TDM 400 analog card; (it's one of those RED modules). Now -may this is the complexity adding step..- I have a GSM gateway attached to this FXO thing; incoming calls are processed as they should. But both when peeking on the CLI, as well as in the phone display I do not see the caller id.

I have the following in my extensions.conf [ext-local] exten => _53XX,1,Wait(2) exten => _53XX,2,NoOp,Dialing ${EXTEN} from ext-local-custom exten => _53XX,3,Macro(dialout-trunk,2,${EXTEN},,) This is used to match inbound caller-id for my legacy PBX. It works fine for inbound calls, but not for internal SIP calls. If I call from a SIP phone that is also in [ext-local], it looks like it

Dear Gurus, i am facing some unknown problem here.. first, let me describe my case. my office is using a nortel merridian option 11c PBX. and it is connected to the provider thru a E1 card, which is working fine with no problem. my plan it to slot in a server with 2 TE11XP card to intercept the outgoing calls. Provider <--te11xp---> asterisk <---te11xp------> nortel merridian

I am trying to solve the fax problem by installing an E1 channelbank (Megaplex MP-104) It's a box that has 8 x FXS ports and a single E1 port. The plan was to use one of my 4 E1 ports to connect to the Telstra onramp and one to the MP104. I have since discovered however that the MP104 only supports CAS signalling and I am having trouble getting asterisk to work in this mode. Currently we

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've got my developers kit from telappliant and got a machine up and running to become the house phone system. Most things are working now, such as incoming calls, call transfer, call parking, voicemail, etc. The one thing I can't do is make my analog phone ring! I can see the call coming in on the asterisk console and can then pick

I'm running an E400P. Every now and then Asterisk stops receiving incoming calls. This turns up in the messages log: Nov 25 10:49:12 WARNING[65541]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 1. Hanging up owner. Nov 25 10:49:15 WARNING[81926]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on

I'm struggling to get my dialplan to work with a simple analog fax machine. I have TDM400B zaptel card with an FXO and FXS port. I have the FXO port connected to the POTS machine and the FAX machine connected to the FXS port. The FAX machine itself works fine, I can FAX outgoing messages fine. I can also dial the FAX extension from the internal context, the FAX machine answers and I