similar to: possible solution to PRI T100P dropped call issue

Hi All. I'm working on an * configuration. We require 8 inbound POTS lines, and CT1 or PRI seems like it will be quite expensive at that level. I've read that a T1 Channelbank plus the T100P would be a (the?) way to go for this situation. What is the recommended channelbank for use in this scenario? From searching the archives I see a lot of suggestions to get "a

I've been working with an adtran ta 850 hooked to a t100p pretty much all day today, and I haven't gotten past configuring zaptel.conf and zapata.conf. For some reason, when I pick up analog phone hooked up to the first module of a quad fxs card in the second slot of the ta 850, asterisk thinks that all four of the fxs modules in that card are going off hook. If I pick up a phone hooked

In the next week or so, we'll be turning up a new T1 Pri from Cox Cable in Omaha using a T100P (installed but not yet configured). While discussing the interface parameters with a very knowledgable Cox engineer, we decided on some of the basics including b8zs, esf, dchan=24, callerid, 5 digits of called number forwarded to *, and switchtype=national (NI2). They indicated their CO switch is a

Hi List, After reading a bunch of the docs, list post archives, it still seems that a clear definition of how to clock the T100P card is muddy. zttool says that the link is "INTERNALLY CLOCKED", does this mean the T100P is providing clock, or does this mean the T100P is getting clock from the T1 line side (ergo getting clock from the Telco) ?? If you have sync = 0 then zttool

Should FAX transmission generally work through Asterisk and a TDM400P connected through a PSTN gateway? At first blush I'd think that if they're all g.711uLaw encoded that it would work. But experience shows otherwise. Is there a better way to do FAX? -brian

Hi Last week I installed Asterisk (release1) with digium t100p single span T1 (wct1xxp) board on Dell GX270 pc configured for PRI. Asterisk/t100p is currently the only user of the t1 line. All worked well for about a half a day, PSTN to SIP phones to non-SIP IP phones etc. Alas, since then I consistently get multitudes of blue alarms on all b-channels followed by a loss of d-channel: Aug 31

Due to some failed hardware on another platform, I've had to move a T100P card to a different chassis. After this move was completed, I am seeing some strange results on the T100P card that do not display to me any failure mode with which I am familiar. The card comes up, and shows "good" carrier and status, but Asterisk refuses to bring the D-channel up. Calls of course do

I am trying to figure out the correct syntax for the CLI command "SIP SHOW CHANNELS". I have tried SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is connected such as: -- Zap/15-1 is ringing -- Zap/15-1 answered SIP/206-4299 asterisk*CLI> sip show channel SIP/206-4299 No such SIP Call ID 'SIP/206-4299' I always get the "No such SIP

hi, i saw the cdr file called Master.csv and i want to know what these represent. examples "","","4","incoming","","Zap/1-1","Zap/4-1","Voicemail","u8888","2003-11-07 17:43:04","2003-11-07 17:43:04","2003-11-07 17:43:22","ANSWERED","DOCUMENTATION"

I'm trying to get a Carrier Access Corp. Channel Bank I working with a Digium T100P without success. What is stranger is that the status lights on the channel bank and T100P seem to change almost each time I power cycle the channel bank or reset the T100P. The channel bank has three status lights: T1, Framing, Status. T1 is green, Status is yellow, and Framing is usually red but sometime

Does anyone know how to monitor * to see if they are receiving timing slips on a span connected to a T100P card? I am seeing b-channel restarts quite often and also getting "No D-channels available" warnings from time to time. Yesterday I had all the b-channels crash during a MeetMe Conference. Not good! This PRI is connected to an Avaya Definity PBX that is onsite and located in the

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)

In my * server I have a T100P, connected with a T1 crossover cable to an Adtran TA750 with 2 FXO cards. After I do 'modprobe wct1xxp;ztcfg' the alarm on the card starts blinking red. Running zttool confirmed that yes, it is a red alarm. I've read a few previous posts on this type of setup and I'm not too much closer. What should I be checking next? A different cable? The

Hi, i just received an E100P, this is the first one I have ever seen, and notice that the board reads T100P. Is this right ? The antistatic bag had a small label that has E100P written on it, and the card is a bit different than the T100P I already have, Does Digium use the same boards for both cards ? I don't have an E1 link here, can I test the card just by loading the driver and run

I have a dialplan that requires that we use * to send the long distance access code to AT&T. I have found in the list that the `w` command can be used to inject a pause, I have tried the following: exten => _91NXXXXXXXXX,1,Dial(ZAP/g1/${EXTEN}www5555555,70) There `5555555` is the ld access code. I tried various quantities of `w`s but I never got * to dial the ld access code. Allof the

I am having trouble getting my T100P and ADIT 600 talking. I have 2 FXO 8 port cards in the ADIT and a T1 controller card. I have a T1 x-over cable connecting two together. Here are copies of the relevant bits of my zaptel.conf and Zapata.conf Zaptel.conf span=1,0,0,esf,b8zs loadzone=us defaultzone=us Zapata.cfg context=default signalling=fxs_ks group=1 channel => 1-24 My

We've got one of the Budgetone phones here, and we can call from any SIP phone, or an outside line TO this phone and the conversation sounds great for bothways, not a bad delay, no echo problem, etc. But when we pick up the Budgetone and dial an outside line or another SIP phone the person on the Budgeton just sounds really choppy and there is a slight delay. We've messed with

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040112/e9dceeb4/attachment.htm -------------- next part -------------- Hello One of the option in VoicemailMain() is "Adavance Options". Could anyone explain what are these ?. Because whenever I select Advance Options, it repeats the same process of asking "Change Folders,Advance

Does anyone know how to send a message to a Cisco 7940/7960 phone running SIP images 6.3 telling it to light up one of its LED's when new voice mail arrives? I found alot of web based solutions http://www.voip-info.org/wiki-Asterisk+GUI and easy ways of getting email or getting paged of a new voice mail - but nothing where you can just look at the phone and see a blinking light or

Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box? Thanks in advance Robb