similar to: Anyone Installed a Digium TE110P or TE120P card in Canada?

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

I have a brand new TE120P card that I have installed and asterisk is not starting as I am getting the error: ERROR[5054] chan_zap.c: Unknown signalling method 'pri_cpe' It seems it does not matter what I change the vaule for signalling= to, it always returns it as invalid. I have tried the config from my other 2 servers running TE110P cards and the config from AusTechPartniships

Do the Digium cards have a built in CSU? Is a CSU an FCC requirement? or just a carrier requirement? TIA

I want the ability to setup DIDs in a variety of different remote locations in Canada. There are various providers that have DIDs in major cities, but none that focus on the "smaller" cities. The question is how do I actually setup these DIDs? Thanks, Geoff

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, 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, I recently installed a TE120P in my lab with a full voice PRI (23 channels + 1 D channel). Everything is working well except echo cancellation; for the most part this isn't an issue unless one of the users is in a conference. I'm getting the following error when a call is picked up (incoming or outgoing): [Feb 25 12:54:01] WARNING[8661]: chan_zap.c:1437 zt_enable_ec: Unable to

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

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

Dear all I got one more error my asterisk E1 card connected with avaya E1 card [avaya]-------E1-----[asterisk] i got this 2 error what is start asteris on consol mode asterisk -vvvvc [Jul 27 09:51:29] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jul 27 09:51:30] WARNING[737] chan_zap.c: PRI Error on

Hello, I have some Inf values in a matrix, but I don't want to replace them with some value but rather remove the rows that contain the Inf values. Also I would like to record the rows which were removed. Is there an easy way to do this instead of writing loops over the matrix? Thanks. Ita Cirovic Donev

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:

Hello List, We purchased a TE120P card from Digium and it works great. The only problem is that we are still experiencing echo on some calls. I've tried various echo cancellers (right now we are using OSLEC) and still no luck. My question has anyone gone from the TE120P to a Sangoma A101D-X Single Port T1/E1/J1 w/ echo cancellation? Have you noticed a difference? Also I called

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).

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.

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:

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

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 there, I have a data.frame (pwt6) which I would like to transform: country year gdp MEX 1950 2 MEX 1951 5 BOL 1950 4 BOL 1951 12 ITA 1950 45 ITA 1951 2 This should be the result: year MEX.gdp BOL.gdp ITA.gdp 1950 2 4 45 1951 5 12 2 Right now I have this code (better - no code): country.label<-names(table(pwt6$country)) result<-data.frame(year=NULL) for(i in country.label) ?