similar to: Weird..bridging to Zap channel FXS instead of bridging to PSTN FXO on outgoing group

Hellow everyone, here's an other newby question. I've got a * configured with the card in the subject line. At times Asterisk fails to notice a disconet from the incoming line going into one of the FXO ports. Consequently it just keeps the line off-hook for ever and that causes my provider to mark the line aut of order. Is there any way to "help" Asterisk notice the disconect?

Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell

Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683&nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Can someone please help. Regards, Sanjay Rajdev

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

I have googled this to death, and all I get are reference to the MoH needing a Zaptel timing source, and then people saying no they don't any more. -- Set Response Timeout to 2 -- Executing BackGround("SIP/211-57ba", "my-greeting") in new stack -- Playing 'my-greeting' (language 'en') == CDR updated on SIP/211-57ba -- Executing

I'm using Asterisk 1.0.9, a Digium TE210P dual T1 card, and two Rhino channel banks (one 12FXO/12FXS, the other 24 FXS). So it's an analog phone on the inside connected to one of the FXS ports, and PSTN line connected to one of the FXO ports. My problem is that as soon as I hear the _first_ ring when I dial out through the PSTN line, I hear a tiny echo on the phone (I estimate between

Wow ! No reply... May be I must talk about X100P instead of X101P ? Is someone has yet encountered this kind of "no sound" problem when bridging two FXO lines like this (first as input, second as output) ? Any idea ? TIA. Best Regards, Francois BERGERET, France. ----- Original Message ----- From: "Francois BERGERET" <f6hqz-m@hamwlan.net> To: "Asterisk Users

Hi, I thought I remember seeing somewhere on the Asterisk website a card that had 16 ports fxo or fxs, that was user selectable with straps on the card. Am I going crazy, I can't seem to find it now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040930/86c8ae69/attachment.htm

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

Dear List members, After succesfully installing the * on a couple of systems, and putting them on test, I observed that there is an intermittent call drop on PSTN line. The systems are - Dell Optiplex P3/500MHz/128MB - Built-in ethernet - 1 X100P (Motorolla chip) card on PCI - 10G HDD etc. - Asterisk April 17 CVS. - 2 Mediatrix FXS ATA (4 phones) - 2 Grandstream phones. - sip.conf, zaptel.comnf

Hi, folks: Okay, so here's an idea. I have a TDM-400 card with an FXO card in it connected to the PSTN and a Polycom IP 501 phone. Observe the following simple dialplan for illustration: > [incoming] > ; incoming calls from the FXO port are directed to this context from zapata.conf > > exten => s,1,Answer() > exten => s,2,Dial(SIP/polycom) And zapata.conf: >

I have an eight channel fxo setup (2 TDM400P cards) and I have them setup. Here are my configs: Zaptel.conf: fxsks=1-8 loadzone=us defaultzone=us Zapata.conf: [trunkgroups] [channels] musiconhold=default rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callerid=xxxxxx callwaiting=yes busydetect=no callprogress=no usecallingpres=yes callwaitingcallerid=yes

Hi, I installed a 1-port FXO on my Ubuntu 8.4. I was earlier only hearing a fast clicking sound and now I am not hearing any dial-tone. The FXO card has 2 slots: [phone | line ]. I hve connected the wall-phone-input to the "line" slot and "phone" to my home-phone. I do not hear any dial-tone on my home-phone. Asterisk seems to recognize my hardware....here are the relevant

I've been having trouble with making outbound calls to my TELCO from a TDM400 card (FXS KS signalling) after upgrading from 1.6-beta9 to 1.6.0. The problem is completely intermittent. When it fails, I get this message: [Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) At some point, it starts working, but I don't know what

Hi I have been having a rough time getting a Sangoma A200/Remora FXO/ FXS Analog AFT card set up properly. The main issue is that the card has four ports and as far as I can tell Asterisk is only seeing two. On the two that it recognizes the "Green" FXS ports are not green, they just are not lit. The "RED" FXO ports are indeed red, but from what I have read your not

I have a TMD400 card installed in a PC with one fxs (installed in slot 2) and two fxos (installed in slots 3 & 4).? fxos work fine but I am unable to get a dial tone for any devices connected to the fxs.? I have correctly connected the power supply to the card and I have even tried moving the card from slot 1 to 2 on the board. Below is from the console when I try to route a call from FXO on

Hi, I'm installing an Asterisk box with a TDM2400P in Brazil. I can make analog phones work while lines are not working. Since I do not know anything about brazilian lines, is there anybody who can tell me what is wrong/missing in my conf files (below)? TIA Giorgio _zaptel.conf:_ fxoks=9-16 fxsks=17-24 defaultzone=br loadzone=br* * _zapata.conf:_ context = inbound_zap echocancel = 128

Please explain the relationship between modules from the driver (wctdm), the /etc/zaptel.conf file and zapata.conf. Specifically, if I have a FXS module 0 and FXO module 1, what should be used in zaptel.conf and what should be used in zapata.conf? Then finally, in extensions.conf, what is the Zap channel for dialing out? Zap/? % dmesg Module 0: Installed -- AUTO FXS/DPO Module 1:

I am trying to get distinctive ring to work on my PSTN with no luck. I can get 2 different ring codes but it skips the context assigned... here is my complete zapata.conf: [channels] signalling=fxs_ks usecallerid=yes rxgain=1.0 txgain=1.0 language=en context=default usedistinctiveringdetection=yes dring1=134,0,0 dring2=137,0,0 dring1context=internal2 dring2context=default

The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card. Here is my /etc/zaptel.conf loadzone=us fxsks=1 and here is my /etc/Zapata.conf [channels] language=en #include