similar to: Recommendations for * monitoring?

I've got a handful of T1s going into a TE410. When I place calls into the system over these T1s, the system either doesn't decode all of the DTMF digits or it decodes ones that aren't there. When the system places calls out, there is no problem doing the DTMF detection. Everything works great.

I have a couple of questions about Digium's T1 cards, such as the TE410P. Any answers would be greatly appreciated. 1) Do they support standard T1s or are they ISDN-only? 2) Do you know of anyone offering support for configuring T1s for Digium cards, and if so at what cost? Thanks, Matthew Roth http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian

I've got a problem w/ zap calls being dropped right after they are answered. I have a log file: http://pastebin.com/368526 Everything looks OK except for the DEBUG[25563] chan_zap.c: Exception on 9, channel 1 that seems to come up quite often. As soon as the other end of the Zap answers (my cell phone), and I can even hear a half second of noise, the line goes dead and gets hungup. In

I've got an issue w/ 1.2.0beta1, where if I call VoiceMailMain from a sip phone, and then either put in incorrect passwords or just hang up, I never get a Spawn Extension that hangs up the call, and my sip phone is not capable of making any more calls until I restart the daemon. Can anybody help me fix this? -- Jesse Keating GameHouse -- Systems Engineer

When I hang up in voicemail, Asterisk seems to stop responding. (hangup vs pressing # to disconnect). After that, no calls can be made until I restart Asterisk. In IRC, a developer seemed to think it had to do with me using switch => in my dial plan. Basically I never see the calling extension get the -1 signal. Can somebody help me figure out why this is happening and how I can fix it

Hi folks, New to the list. Just updated the bootrom and app firmware on a Soundpoint IP 501 as per: http://www.voip-info.org/wiki-Polycom+Phones Updated from: to: APP 1.4.1.0040 1.5.2.0054 BootROM 2.6.1.0003 2.6.2.0032 After I did this, it appears that the Web interface for the phone won't change the settings, nor will it actually reboot the phone now. What do I

I tried following the Wiki page regarding the Polycom 501 and having the same extension appear on all 3 line buttons (just like my cisco) but I'm having no luck. Has anyone else had success in doing this? Perhaps someone who has been successful can update the wiki? Thanks, Matthew http://www.voip-info.org/tiki-index.php?page=Polycom+Soundpoint+IP+501

Hello, Sorry, I know I read this somewhere but now I can't find it when I need it. I'd like to force a call to go out one line if we dial '9' first and then the number. Same for '8' only I will force it out a different line. There is a parameter or a method to allow the dialtone to come back after pressing the first 9... but I can't remember how to do it. Anyone know?

I have installed Asterisk and i have configured with two SJPhones; i am able to make calls between these two phones.I am planning to develop a application basically web based application from which the administrator able to trace the call logs or call summary, i mean from which user agent to user agent call is going , and what is the staus, if second user tranfers the call to the third

Hello, We just had some new T1s turned up today to replace others that our contract has run out on and we are now getting more echo on the new T1 lines than we had on the old ones. The T1 type is 24-channel, D4/AMI SF Robbed-bit(the same as the T1s they replaced) The problem is that we are getting echo on about 10% of the calls in and out placed on these new T1s compared to less than 1% with

Are you looking for local or Long distance? What will these T1s primarily be used for(inbound/outbound, domestic, local, long distance, international) How important are per minute rates to you? how many minutes do you expect to use per month? We are in Tampa Florida and have 15 T1s from several different providers so I may be able to refer you to one if it's a match to what you're

Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM -> VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features

I have a couple of live T1s sitting around and they are not ISDN(like most of the people that are using Asterisk seem to be using), they are regular old 24 channel, robbed-bit, E&M wink start, D4AMI T1 circuits. Can I get these T1s to work with a T100P Digium card and asterisk? Searching through the lists and the documentation I haven't seen any examples of how to configure this kind

I just installed a TE410P on a Debian Sarge system running kernel 2.6.11-1-686-smp. Zaptel and Asterisk seem to be working fine. However, I have a couple of problems with the TE410P and Zaptel. First, the TE410P is showing me red alarms on 2 of the 4 T1s. This board (the TE410P) was just moved from another machine running REL3 and all 4 T1s were working there. I don't know why only

Is there a way I can have "voice mail check" calls coming from my internal users automatically get to the right extension, without having the user enter their extension? I'm thinking that I could have the local SPA boxes translate, or have each user live in a context where the extension in question exists uniquely per user, but both of these seem kludgey. Thanks in advance for

Hello, I've seen various suggestions thrown around for hardware when people ask, but can we all agree on some basic hardware recommendations for a few basic setups(and post them on a website) to make it easier for new people to avoid some of the hardware/software pitfalls when they are setting up their first systems. Something like this: (THIS IS JUST A PROPOSED LAYOUT SO PLEASE BE GENTLE)

I have asterisk installed on a Dell 2850 dual-Xeon 3.0Ghz box with 2GB of memory. This is serving about 75 sip clients, Polycom500's and 600's. We are running into problems with the memory. Asterisk, right now, is using about 1.8GB of system memory. I am using Asterisk 1.0.7, Zaptel 1.0.7 with Digiums TE410 1xT1 RBS and 1xT1 PRI, Libpri 1.0.7 on Fedora Core 3. My question; is this

Has anyone here used T1s with RBS with asterisk? Cary Fitch

Hi, Please forgive my terminology, still a bit new to T1s and such. I'm looking for a way to have 5 T1s from a carrier terminate into some type of box (multiplexer?), then be able to plug 7 asterisk servers into that box (each with single port T1 card) and be able to have 2 * servers go down at any given time and not actually have the carrier see that anything has happened. Obviously if a *

Hi I plan to setup an asterisk box to function as a SIP gateway forwarding lots of calls to/from a backend of several other asterisk boxes, each with a TE410 card for PSTN connectivity. It will only gateway the calls into the PSTN gateways. No transcoding is planned - only plain ALAW. How many concurrent calls would you think this can handle? I'm asked to plan a system that can handle