similar to: TE120P and Unknown Signalling Method

Hi All, I'm having problems trying to get a TE120P operational in Canada. I keep getting a congestion error when I try to make a call. I'm not sure if my switching, parity, etc is correct. I'm hoping that someone will be able to verify my config. The Telco is SaskTel, with a 10 channel 50 DDI service. Zap show channels show and ztcfg -vv looks ok and the zttool show

The Telco in Canada is been real painful. I was wondering if anyone has installed a Digium TE1X0P card in Canada and if their Telco was so difficult. The Telco will not provide us a service until they see a FCC or DOC number for the equipment ware are connecting to their service. If have found "FCC Part 68, ANSI/ITA-968-A, Including Amendment A1 and A2 Industry Canada CS-03"

Hi guys, Hope someone can help me with this... I got an Asterisk with TE120P card... using dahdi 2.1.0.4 and asterisk 1.4.24. The problem is the link between the card and the telco... suddenly through the day looses connection... and the people on site just power cycle the PC and the problem got fixed.... for a while.... I've already replace the cable between the card and the telco. Telco

Hi, My asterisk server was working with a 4-FXO analog card (TDM400P). I recently added two digital cards: a TE120P (1 PRI) and a B410P (4 BRI). The B410P is still unconfigured but inserted in a PCI slot. The TE120P's jumper is set to E1 as it will connect to a commercial PBX's PRI card also configured as E1. My analog channels used to be 1-4 but since I added the new cards I changed

Hi, I have successfully configured DIGIUM card and successfully communicated through it to the another E1 card running application. Can anybody tell me does TE120P support MFC/R2 protocol. Thanks and Regards sanchal singh

Hi All, I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 I'm not sure if the above is correct.

All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch,

Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David

Hi, I have successfully configured DIGIUM card and successfully communicated through it to the another E1 card running application. Can anybody tell me does TE120P support MFC/R2 protocol. Thanks and Regards sanchal singh

Hi all I use asterisk (1.2 brach) from debian official packages and it works fine. Now I need to install and configure a Digium TE120P card, but I cannot find any guide to install it using debian packages. I would like to know if anyone of you knows about packages that would include the necessary kernel modules or any other method that won't be broken when the asterisk packages are updated.

Hi Folks, I have a T100P ordered from Digium in the past. Its working perfectly fine but I now need to more the server to India (out-sourcing) and we will be getting an E1 line there. Does anyone know if T100P was capable (upgradeable) or swappable so that I could use it with the E1 lines instead? Any pointers to driver/firmware update would be highly appreciated. Thanks. R -------------- next

Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I hope this doesn't already exist, but I couldn't find anything to help. I am installing a brand new Asterisk server, and want to use the Polycom IP331 phones. Does anyone have any steps on how to configure these? I have softphones working just fine, but for some reason I can't find a clear step by step on provisioning the Polycoms. Any help is greatly appreciated! Mark J.

Hi All, I can not install the asterisk-addons as it thinks there is no mysqlclient installed. I have installed mysql, mysql-server and mysql-devel and I am still unable to install the addons. I am running CentOS 5.2 i386. Please somebody help. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and I've ran into an issue. After a call placed any DTMF tone causes the server to lock up entirely. Calls placed work just fine (except for a problem with echo cancellation). The phone registered to the server is a Linksys SPA-942. I am seeing in zttool some IRQ misses, but it never seems to go above 74 (below).

All, I have a Polycom 650 phone, when turned on displays "Checking application". Can any give me some information as to what is wrong? I have copied the CFG files from a 601 phone to work with this 650. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070429/daadda2a/attachment.htm

Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David. -------------- next part

I'm not sure if I have a configuration problem or not. I am unable to dial out. When I try to dial in I can hear the phone ring on the dialling phone but Asterisk does not register anything. In zaptel.conf I have loadzone = au defaultzone=au fxsks=1 In zapata.conf language=au context=from-pstn When I do: zap show channels I get: Chan Extension Context Language

Hi All, I have installed newt and newt_devel but dahdi_tool will not compile/install. I'm trying this with dahdi-linux-complete-2.5.0.2+2.5.0.2. does anyone have any suggestions as to what I am doing wrong? Regards David. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys, I currently have about 10 Asterisk servers scattered around the place each hosting their own dynamic conference centre. Is there any way that when people join these conference centres on each server that somehow Asterisk bridges the conference centres on each server to form one large conference? Many Thanks David. -------------- next part -------------- An HTML attachment was