similar to: Asterisk doesn't detect when the caller hangs up

Hi. I'm getting new lines for using with Asterisk. In my Telco they said I could choose between Analogic lines and RDSI lines... I've already bought a TDM400P with FXO modules. Can you give some hints on the differences between RDSI and normal Analogic lines? Would I have problems for using a RDSI line with the TDM? Any other issue in general? Thanks in advance, RODOLFO --- avast!

who can tell me how to do TDM over enthernet ? pc a connect pc b only use TDM card? thank you John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?23? 11:47 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5, Issue 336 Send Asterisk-Users mailing list

Hi all. I've being reading posts from the list since yesterday and I feel this question was answered a lot time ago, but the list archives are a mess (yet). I hope some one is willing to help me out. I want to set up this: caller ----- PSTN ---- (SOMETHING1) ------ VoIP --------- (SOMETHING2) ---- PSTN I think this must be a very basic architecture, but I'm not sure wat SOMETHING1

Hi: I'm using a TDM-400 to terminate PSTN lines at my Asterisk server with kewlstart signalling. When an outside caller calls the server, the TDM-400 goes off-hook and provides a ringing tone to the caller. If the caller hangs up before the receiving party answers the phone, Asterisk fails to detect the hang-up. The TDM-400 stays off-hook, hogging the line, while Asterisk rings the

Rodolfo, I haven't looked up how to do this with sip phones, but the zap channels can be configured in groups that will hunt through the group until a non-busy line is found. http://www.voip-info.org/tiki-index.php?page=Asterisk%20ZAP%20channels#comments Here is a link to PBX hunting with the dial plan. http://www.voip-info.org/tiki-index.php?page=PBX+Hunt+Groups I haven't tried

Hi. Can you give me some hints on how I can create a rotational board? I dont even know how to spell it in english. What I want is to have more than one line reserved, but with a single phone number, so that people can call to the same number and get a ringing signal if any of the lines is available, instead of having to dial 5 different numbers in order to get a free line. This is done

Hi list, I have search for a clear explanation about this mensage " WARNING T.30 ECM carrier not found", but until now I dont succed on it.Anybody know how can I handle with this problem? I have Asterisk 1.6.2.13 with TDM 410p and the Fax is connected to Dlink FXO dvg 2032s. Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda

Hi Philip, Unfortunately, * speaks MGCP only as the Call Agent, rather than as the Media Gateway. MGCP is a master/slave protocol, and it would take some effort to make * work as the slave. I have the same problem: Free Telecom here in Paris includes MGCP service with their DSL. You can call any fixed phone in France at no charge! Rates to mobiles and international are quite aggressive, too.

Hello, I'm a newbie user of Asterisk, i'm sucessfully install it it's great but, i get some problems with echo in a adsl line. My system is a TDM440P with 3 FXO Ports and 1 FXS. Asterisk 1.2.13 Zaptel 1.2.11 Line 1 - Analog Line 2 - Analog with ADSL It's installed with two analog lines one of them have a ADSL splitter. The location are Spain, i load the tdm driver with opermode

Hi. I gave up with the IBM NetFinity, so I'm going to buy new hardware. I'm going to install: 1-)One X100P (1 FXO module) 2-)One TDM03B (3 FXO modules) I'll have the 4 FXO channels busy almost all the time, and I would like quality to be as good as possible without going to the high-level hardware. I would like to learn of some tested configurations (I've heard of problems

Hi again. My last post was incomplete, so I' reposting it. I've seen that each country has its own PSTN qualities. I would like to know the minimal characteristics needed in PSTN to use Asterisk and also if some body knows which are Spain's PSTN's. I'm interested in buying a TDM card (probably a TDM11B card) and I need to know if it will fully work on Spanish PSTN and what

Hi all! I am in the same situation here in Sweden, but I just got the X100 card and don t get it to work with DTMF caller ID. Is the only way to solve this to buy a TDM400 card w FXO modules? Is it just a driver question or a HW issue with the X100 card? Where to find dok about the new keywords in conf files like "cidstart=polarity" in zapata.conf? rgds Gunnar >>>

Hi. I have installed a x100p (THE x100p for those who have seen my former post). Now I just want to connect a "normal" phone (not an IP phone) to the card and use it as a sip extension (I have a FWD account)... more clearly: I want to be able to pick up the phone and call any FWD user using my FWD account... receive the FWD calls in that phone, and also to be able to make normal

Hy, I'm writing from Spain. I have the 1.2.4 asterisk version and 1.2.3 zaptel version. I've heart that this asterisk's version detects correctly de hang up of PSTN, but in my case this thing doesn't happen. Moreover, my asterisk sends the next messages in the CLI: Feb 21 15:03:13 WARNING[10363]: chan_zap.c:10876 setup_zap: Ignoring signalling Feb 21 15:03:13 WARNING[10363]:

Hey all, I recently noticed that DTMF caller-ID was implemented in CVS, so I requested the service from my telco (the Dutch KPN) and tried to get it going in Asterisk (current CVS), without success so far. This system has 1 X100P, 2 TDM400P's with 4 FXS-modules each and 2 HFC-PCI ISDN-cards (zaphfc-driver) in it. The analog line I'm trying to get caller-ID working on is obviously on the

I have an asterisk 1.2.9.1 machine with zaptel 1.2.6 running. On the TDM400P, I have 1 FXS port and 3 FXO ports. dmesg reveals: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.6 Echo Canceller: KB1 PCI: Found IRQ 10 for device 01:01.0 PCI: Sharing IRQ 10 with 01:05.0 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1:

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to

For those using the new tdm fxo modules, please review and add your comments to: Bug Description 1898 tdm fxo does not sense broken pstn line 1899 tdm fxo senses pstn line disturbances as ringing 1902 tdm fxo echo when calling from sip phone There is at least a suspicion on my part that bug 1902 might be related to specific "systems" that have not yet been diagnosed or

Greetings all, I'm new to asterisk... Finally got two IP phones and two TDM400P's (one with four FXO modules and one with four FXS modules)... Began experimenting a couple days ago and have ran into a very aggrevating problem. My system consists of the following hardware: Abit BL7-RAID motherboard Pentium 4 1.6GHz processor 512MB SDRAM 2 TDM400P cards 2 3Com 3c905

Hi On an old IBM Netvista thinclient, the TDM card doesn't detect incoming calls, although the card seems to be detected, and correctly configured: pbx asterisk # lspci 00:03.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface # cat /etc/zaptel.conf fxsks=1 loadzone=fr defaultzone=fr # ztcfg -vv Zaptel Configuration ====================== Channel map: Channel