similar to: T100P Incoming Calls Drop

I have recently begun experimenting with Asterisk, and have been mightily impressed by its capabilities and flexibility. I have run across one problem, however, that challenges my ability to use it as a production system. My Asterisk box has a public Internet IP, and works great with SIP (ATA 186) clients that also have public IP addresses. Unfortunately, most of the locations that I would

I ran into a couple of odd problems with a new Asterisk box I have set up... I have a T100P connected to a channelized T-1 (24ch) to the PSTN. The T-1 is set up properly so far as I know and seems to work perfectly with one exception: When a call comes in, asterisk starts a 'simple switch' for it, goes through my 's' extension progression until it gets to the first Playback

I'm looking for anyone who has some vendors for rackmount PC hardware in a fairly decent configuration for Asterisk/Digium T1 termination. I am looking for well-built hardware from an integrator that understands correct component choice. Preferably, I'd like SCSI but I might be able to do Ultra-IDE platforms. I need at least two PCI slots in the chassis (so 2u/3u size factor is OK.)

I am trying to receive both CID and DNIS from the telco through a non-pri T1. Currently I have the T1 setup and operational both outbound and inbound calls are completed as should be expected. The calls came in and were placed in the context specified in zapata.conf on exten => s,1. I have requested that the telco provide callerid (they call it ANI) along with 10 digit dnis for my 800

Is there a way to record your own voice messages ("Welcome to my PBX, Press 1 for ...") using asterisk and an analog phone, or do they need to be recorded using traditional voice recording software? Also, I am confused as to why my replies to the message board are never indented. Call this the ultimate newbie question, but how should the reply be worded so that I don't screw up

I was wondering if anyone had seen this problem before and/or could offer any insight into what the trouble might be: I have an Asterisk machine that it set up as a mutual friend with another one (in another state... about 150ms away). Calls between the two fail to get established approximately 50% of the time. When a call works, everything is fine. When one fails, however, I see a large

I have two Adtran 750's connecting our analog phones to asterisk. On occasion, I get a channel that gets "stuck" off hook. 'show channels' shows: Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None) And will just stay like that until the phone is manually picked up and hung up again (or asterisk is stopped/started). I guess this is a function of an unclean hangup (being

I've been having some problems getting a channelized T1 working with a T100P card. Perhaps someone can help: I have an Eschelon T1 coming into a Vina Integrator box. This box splits out the T1 into an ethernet plug for bandwidth and a secondary T1 which I plug into the T100P card. I've connected the two with a T1 crossover cable. I get a green light on the t100p card and can make

Upgraded from 16.2.14 to 1.6.2.15 on Fedora 13, with aastra 9133i and 57i. On 9133i and 57i: #<extension># works for a blind transfer. Xfer<extension>Xfer doesn't! All this worked on 1.6.2.14. Nothing useful on cli, verbose 3, DEBUG. Here extension 169 answers an outside call, and tries to transfer it to 145 using the Xfer button: -- SIP/169-0000009c answered

I appologize for the long, new-ish question, but after a few days of trying to work a solution by reading through the list archives and WIKI and coming up with what I thought would work, I think I'm just not getting a fine detail. I titled this thread "Setting up Security Groups" because I'm trying to set up some sip user groups with certain calling rights, e.g., one group of

Does anyone have an R Syntax Highlighting file {userDefineLang.xml} for NotePad++?? I've started one, but I'm not so happy with it. Robert Farley Metro 1 Gateway Plaza Mail Stop 99-23-7 Los Angeles, CA 90012-2952 Voice: (213)922-2532 Fax: (213)922-2868 www.Metro.net [[alternative HTML version deleted]]

All, Alright, I've looked around the internet, the voip-info.org wiki, and browsed the contents of this mailing list. While I've found a couple of scenarios that are close to this one, I haven't found one that uses my particular card (T100P). Without further delay -- I have successfully configured internal SIP services between a Snom 200 and a Windows X-Lite client and have

I'm trying to create several contexts for extentions with different levels of access to features and I'm wondering how the heck do I include all the contexts so that you can call internal to any extention in another context without giving the features of the higher level context to the lower level context? ie..... [admin] include => local include => longdistance include =>

I have an interesting dilemma with sequencing in the dialplan. Up to now, I have assumed that the extensions in the dial plan were tested in the order that they appear in extensions.conf. In other words, I have the following fragment which was designed to dial toll free on the PSTN and all other long distance on VoIP: [longdistance] include => local

Hi, I know this is slightly off topic but I figured the knowlege here is probably the best on the subject.. I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box.. These phone will be behind an ADSL router using NAT... I don't want to setup another Asterisk system in each office so IAX is not an option.. I could use

Still trying to upgrade to 1.6.0.9 for 1.4. It worked - it worked all day yesterday, but this morning: -- Executing [646xxxyyyy at longdistance:1] Answer("SIP/172-08276a60", "") in new stack .......... -- Executing [646xxxyyy at longdistance:6] Dial("SIP/172-08276a60", ""DAHDI/g2"/1646xxxyyyy") in new stack May 27 09:56:57]

I can't figure out the syntax I need to get subset to work. I'm trying to split my dataframe into two parts. I'm sure this is a simple issue, but I'm stumped. I either get all or none of the original "rows". > XTTable <- xtabs( ~ direction_ , SurveyData) > XTTable direction_ EASTBOUND

I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed the WIKI page on setting it up but I can't seem to get it to work. Here is my Asterisk version: pbx1*CLI> core show version Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on 2008-01-10 12:08:48 UTC Here is a log of when the FollowMe is being called: NOTE: I've tried to use the AstDB as

Can someone please explain what I am doing wrong here? I only want the extensions listed in long-users to be able to access the longdistance context. If I do this, I get a congestion tone no matter what I dial. If I add a [default] context and include => longdistance, then the local callers can call the long distance number fine, which is not what I want, but I still want long-users to be

First the R question. I have the results of a rather large survey (thousands of forms, each with dozens of questions) with some existing weights and expansion factors. I wish to add additional weighting factors, based on new information that elements of certain variables should appear in certain proportions. Where should I look in R to develop what is essentially an N-dimensional balancing