similar to: TDM400P and Caller ID on Call Waiting

Hi All I have problem with LIBMFCR2 for once Exchange I using Sangoma card, the firstly. my ssystem run successful with MFCR2, connected to E10 (Acatel Exchange), after that, i move connection connect to EWSD (Siemens), my system don't work. error protocol R2. my system: Asterisk CVS 1.1.X LibMFCR2 pre.005 and unicall-pre.005 this's my setting and my setting wanpipe1.conf

Hi I have an asterisk box with digium hardware connected to a Siemens EWSD version 15 using a crossed E1 cable. The asterisk is serving as a h323 media gateway. When i start asterisk, the Siemens gives me this alarm: REPORTES GENERADOS EN LA EWSD AES/V15SBOL/BOLCBK1V51327079/013 05-06-06 11:25:38 7773 3062/03728 HF.ARCHIVE-80040

Hi all, Has anyone got any experience with hooking Asterisk up with a Siemens EWSD 13 switch over a E1/PRI ? We're located in Belgium (Europe) and one of our telecom partners uses this switch. We connected one of our TE410P ports with their switch, but the status light on the TE410P card keeps blinking red. On their side they are getting a DSA (distance service alarm) error, so this

Can anybody recommend a good web interface for asterisk that actually works. I am looking for a web interface that can show how many callers are on the phone, should be able to transfer the calls and disconnect. I have tried using the flash operator but has been unsuccessful in making it work. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com

Hi, I have Asterisk 1.4.0 using Queue App. I use PRI connecting my Asterisk with Siemens EWSD This was working OK but since two days I have this error: [Jun 22 10:53:09] WARNING[8050] chan_zap.c: Ring requested on channel 0/3 already in use on span 1. Hanging up owner. [Jun 22 10:53:09] WARNING[8050] chan_zap.c: Ring requested on channel 0/3 already in use on span 1. Hanging up owner. [Jun 22

Hello that might would be an easy question for someone, but im in doubt Is there any possibility to pass a call from one asterisk to another and then to ZAP channel. For instance I have "A" asterisk with numbering 45670 "B" asterisk with numbering 45680 second asterisk has TE110P card with single PRI port connected to Siemens EWSD. When I originate call from asterisk

Hello All. I'm using sangoma card A-101. tested successful with E10 (ACATEL) Exchange, connection with E1, CAS, (using unicall-0.0.3pre4). my system run success, incoming call and call out are good. when i switch to EWSD (SIEMENS) R-15 . my asterisk faill, cannot connect with EWSD. (E10 and EWSD exchange store in two provinces difference. in Vietnam) this is my logfile of E10

Hi all, We're trying to hook up our Asterisk config (Card: TE410P) with a Siemens EWSD switch. The link is ok on both ends (green), with no errors. The problem is when we try to make a call from our side (via call files), we get the pri/E1 error Ext: 1 Cause: Temporary failure (41), class = Network Congestion (2) Our Telecom partner (they checked with Siemens) mentioned that we need to

Dear all, a newbie question... I have two external lines (PSTN & SIP) and two internal lines. When receiving an incoming call, I correctly get the CID, but it's not propagated to the internal lines. My analog phones shows "External call" instead of the CID. My analog device is a TDM400P (2 FXO + 2 FXS) with two analog phones attached to it (Siemens Gigaset and a wired

I am sure that quite a lot of people would like to have Caller ID working with their X100P and TDM400P cards outside of USA. Judging from previous threads this is just a matter of implementing this support in the software driver! So, I was thinking, if we get together and put few $(USA DOLLARS) into a basket, we could then ask Digium to actually properly implement Caller ID for non USA

I have a Digium TDM400p (1 FXO, 1 FXS) with the FXS module connected to a standard analog handset with caller id display (US caller ID). Although it appears that caller id information is coming into asterisk (it shows up in voicemail), I can not get it to display on the analog handset. Is there anything special I need to do to send the caller id info out the FXS port? I've tried a few

Has anyone here managed to get UK Caller ID (BT) working using a TDM400P card? . I've got the latest drivers from CVS, but can't find clarification if UK caller ID is supported and if so what the settings should be. Cheers Graham -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I am trying to use Asterisk as a backend to send and receive faxes over analog channels connected to a Siemens HiPath 3550 switch and a TDM400P card. Receiving faxes, including multipage ones, works really fine, we have no issues at all. But when it comes to send faxes using the app_txfax application, spandsp can't send multipage TIFF files for some reason, it stops sending in the

Hello All. I'm using sangoma card A-101. tested successful with E10 (ACATEL) Exchange, connection with E1, CAS, (using unicall-0.0.3pre4). my system run success, incoming call and call out are good. when i switch to EWSD (SIEMENS) R-15 . my asterisk faill, cannot connect with EWSD. (E10 and EWSD exchange store in two provinces difference. in Vietnam) this is my logfile of E10

I have been looking for the answer to this question for a while. Google-ing and reading the archives of Asterisk-Users has not enlightened me. It seems that this question has been asked many times, and many times it has gone unanswered. I have call waiting and three way calling on my PSTN line from Verizon (the local telco). This is connected to a FXO port on a TDM400P. I also have

I had a similar issue with sending a flash to the PSTN for call waiting. I found that my dial plan in the extensions.conf file was not allowing me to dial *xx. Once I corrected my dial plan I was able to dial *0, *69, *78, *79, etc. Training the wife on how to actually use it was an entirely different issue. I have not tried to enable 3-way calling via the PSTN. "Jeff Otterson"

Hi, I'm using Asterisk 1.2.1 on a Debian Sarge with a TDM400P and a monoBRI using chan-mISDN from beronet site. It seems to work all right except for autodial calls, monoBRI ISDN channel behaves differently waiting for the caller to answer and then continue. Asterisk console says: analog: -- Attempting call on Zap/2/3391818250 for 104@inbound_originate:1 (Retry 1) > Channel

Anyone have any experience using gr303? May have a need to interface * to a Siemens Class-5 CO for pstn trunking (inbound and outbound). Rich

Hi! Can I use my Linksys PAP2 with asterisk and an analog CLIP phone to show the Caller number on the phone. There's a "Caller ID Method:" option on Regional settings, but I tested all options, and my CLIP phone never shows the Caller number... :( Any idea?

Hi, I bought the Digium Asterisk Developers kit a few weeks ago, got the x101p and a tdm400p cards, both seemed to be working fine up until today, now the tdm400p seems to have gone to the great PBX in the sky ...... If I left the reciever on the analog phone I don't get a dial tone and nothing appears on the asterisk console, If I use my cellphone and dial in I see Asterisk Ready.