[Asterisk-Users] One voicemail -> multiple boxes?

I don't want to re-invent the wheel if someone has already hacked a way to do this. One of my customers has a number of stores, and he wants to leave one voicemail that would be delivered to all the managers at once. Each has a voicemail account on his server. I have googled around and looked

Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

Tony Buser wrote: I looked through your code to see if I could make some changes, unfortunatly I can't speak Italian! :) Not that unfortunate; the comments are all in Spanish, not Italian :-) B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http:

Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

Justin Carlson wrote: just type it in it will remain until you restart your browser. ( it does not disappear and you do not have to hit enter or anything like that) I cut and pasted it right from the source code file, but no matter what I do, I get the following line in debug: La clave no coinci

Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

I downloaded the app and for the most part have it going. I have not yet managed to get it to accept the password in the flash widget that appears as if it wants to accept it. I wonder about browser-related problems in that respect: I'm running fairly recent Mozilla. I have also hacked the thi

Re: [Asterisk-Users] Virbiage Phones - Vapourware??

Adam Hart wrote: I'm not sure who's been telling you 2 weeks away but I've posted previously that the phones were 8 weeks away. How long ago was that :-) B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/ast

Re: [Asterisk-Users] Cisco 7960 SIP Images

Rich Adamson wrote: Wanta take a guess what would happen if Cisco decide to really enforce the legal rules? I'll bite: Their market share would plummet in all their markets, and then smaller, more innovative companies would become more able to compete with them, and the overall marketplace woul

Re: [Asterisk-Users] Cisco 7960 SIP Images

Roderick Montgomery wrote: * Have you purchased used hardware off eBay and need the latest software? There's an inspection and relicensing process that can make your gear fully legit. Ummm, with a mountain of red tape surrounding the whole deal, and basically at a cost that ensures most people wi

Re: [Asterisk-Users] Codec Voodoo

Andres wrote: Are you making calls out to Nufone or simply from one of your servers to another? We noticed this problem when we upgraded one of our servers to the latest CVS and left another one with an older version. Seems that the latest changes with rtp.c need to be applied everywhere. When

Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php

A comment and a question about the latest version: First, the question: What is the organizing principle of the "Asterisk Minutes" chart, with respect to the ordering of the various days? It seems random. . . I also assume the last column is the average number of minutes per call? As to the f

[Asterisk-Users] Voice versus data T1s: Balance of power

I hope this question isn't flamebait. I don't know anything about voice T1. What are the tradeoffs in terms of asterisk's design and performance whether traffic is handled by one type or the other? I wonder about the economics, too. Thanks. B. ___ A

Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php

It might be helpful for us all if the author could let us know more about the environment in which this application was built. . I'm getting all kinds of errors when I try to run it, and I suspect that either my Postgres or PHP installations are incompatible with yours. Thx. B. ___

Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php

It might be helpful for us all if the author could let us know more about the environment in which this application was built. . I'm getting all kinds of errors when I try to run it, and I suspect that either my Postgres or PHP installations are incompatible with yours. Thx. B. ___

Re: [Asterisk-Users] Inbound Toll-Free Providers

Brady Alleman wrote: Are there any Asterisk-supported VoIP providers that have cheap inbound toll-free (ie, 800-number) rates? I know you can have your toll-free numbers pointed at traditional inbound VoIP providers (Voicepulse, etc) but I am looking for an integrated service that can do both part

Re: [Asterisk-Users] Cisco 7960 vs 7905

Eric Wieling wrote: The 7905G (but not the non-G) supports SIP. It does NOT support XML. So only the non-SIP phone can use the XML functionality? That s*cks. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

Re: [Asterisk-Users] Important: The Asterisk Mailing list (new subject)

Tilghman Lesher wrote: On Friday 19 March 2004 02:16, Brian Capouch wrote: Olle E. Johansson wrote: Do *not* send out personal replies on the list. Yes! Yes!! Yes!!! Let's change the way the list software works so people won't get hammered by replying and rid this list of that pox on

Re: [Asterisk-Users] Important: The Asterisk Mailing list (new subject)

WipeOut wrote: The biggest problem with having replies go back to the original poster instead of the list is that on one else will be able to learn from the answers to question that was asked, and the mailing list archives will become useless as a source for information.. That's why Internet e

Re: [Asterisk-Users] Important: The Asterisk Mailing list (new subject)

Olle E. Johansson wrote: Do *not* send out personal replies on the list. Yes! Yes!! Yes!!! Let's change the way the list software works so people won't get hammered by replying and rid this list of that pox once and for all. B. ___ Asterisk-Users ma

[Asterisk-Users] MOH: Copyright issues?

After reading a (hopefully) joke web news article today that said the RIAA was thinking about asking automobile owners to pay extra royalties when there's more than one passenger in the car, I began to worry about putting the classic 1974 Pointer Sisters' tune, "Little Pony" in my mohmp3 direc

Re: [Asterisk-Users] Asterisk, X100P and AT&T PBX

Carlos Chavez wrote: Yesterday I tried to connect an * server with an X100P card to an extension of an AT&T PBX. The X100P never could detect the line and always gave an alarm. Is there some special type of config that must be done to connect an FXO port to an extension of a PBX? I had that

Re: [Asterisk-Users] NuFone?

Eric Wieling wrote: Michigan only, but I believe they have decent coverage within Michigan. I seem to recall they were planning on Chicago DIDs, but I don't know the status of that. Jeremy from Nufone and Mark Spencer were both at this week's WISPCON in Chicago. From the smell of it we won't h

[Asterisk-Users] WiSip SIP settings locked?

I got my new WiSip, and it works for the most part with FWD. I *thought* I had read that some of you were using it with asterisk, but so far I can't figure out how to get the SIP settings changed. . . changing them causes the phone to reboot, and when it does the settings are back pointing at F

Re: [Asterisk-Users] PCI front mount chassis?

Jeff Stohl wrote: I am running six reliably right now. Surely I am not the only one doing a large capacity single site? I too am running 6 cards in my system, although not in a "high traffic capacity" load environment. So far my (limited) high-load simulations have shown no problems. B. __

Re: [Asterisk-Users] Hotel wake-up

David Hickman wrote: man I am tired. I did not mean for my last email to go out to the whole list :( Don't worry about it man; the list is set up to cause that sort of thing to happen, and if you read it for a while, you'll see that it's an almost-daily occurrence. Sort of like a free soap op

Re: [Asterisk-Users] Best Budgetone firmware?

Matthew Marlowe wrote: You may also obviously check Grandstreams site instead of another providers site. http://www.grandstream.com/BETATEST/ A free windows TFTP server is available from solarwinds.net which works great. Obviously linux can use tftpd Well, maybe. The Grandstreams use an extende

Re: [Asterisk-Users] Best Budgetone firmware?

Philipp von Klitzing wrote: Hi! I'm still running 1.0.3.81 because I read that once you move up to 1.0.4.x you can't go back again, and my experience isn't *that* crappy. You probably want to start with 1.0.4.26 although also 1.0.4.17 seems to have been relatively stable. Then there is also 1

[Asterisk-Users] Best Budgetone firmware?

I wonder if those who have the nerve to piddle with Grandstream's somewhat chaotic firmware release methodology could give me any idea as to which version of the firmware I want to be running for a maximally functional experience? I'm still running 1.0.3.81 because I read that once you move up

Re: [Asterisk-Users] Supervised transfer (almost) with GS phone

Stephen R. Besch wrote: I have now tested a (previously suggested) method for doing supervised transfers using the Grandstream SIP phone. It isn't perfect, but it works and is very functional. Here are the steps: When I try this, all goes well until, after putting the original caller on hold an

Re: [Asterisk-Users] TDM400 hardware problems?

John Morris wrote: Hi! I didn't get an answer on this, so I'm going to be annoying and try again. For some reason, it looks like two of my channels on my TDM400 stopped working for no good reason. Asterisk stopped working on my main extension today (it does this every week or so). I usually reme

Re: [Asterisk-Users] SCO finds someone to pay!!!

John Bittner wrote: Does anyone know if they took out SCO's code in Linux 2.6 kernel ? Hahaha. SCO won't tell anyone which code it is!! B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UN

[Asterisk-Users] Testing ENUM

Is there any way to "prove" that an EnumLookup is actually being done? I've been trying to get ENUM working, and have gotten to the point where I'm pretty sure the NAPTR records are resolving the way they ought to be, and "manual" lookups using dig return just what they "ought" to. But asterisk

[Asterisk-Users] A working number at enum.fierymoon.com?

I'm trying to play around with ENUM, and John Todd helped me last night on the IRC channel in terms of finding this site and other docs to get me going. But now I wonder: how can I test it? I started by trying to randomly try every possible number in the US, but soon tired of that approach. .

Re: [Asterisk-Users] Sorry, OT (NuFone)

Joseph Finley wrote: Is anyone having problems registering with NuFone? My system has not been able to register over the last couple hours. I've sent a support email in without any answe as of yet. Seems to be working just fine here. B. ___ Asterisk-Us

Re: [Asterisk-Users] Simulating the "lighted line in use" type of phone

Chris Clifton wrote: That's fine for outbound lines, but what if I want to call the guy in the next office ? I have to call him and get redirected to his busy vm just to know that he's on the phone. This is a huge issue with the recepetionist with the 'master console'. How does he/she know whether

[Asterisk-Users] Simulating the "lighted line in use" type of phone

I'd like to see if anyone out there might have some ideas on this. I have a customer who wants to move to VoIP, but who has an office full of people who are very conservative about their telephones. They would like the asterisk system that I am proposing to have something analogous to what they

Re: [Asterisk-Users] Anyone working with NUFONE?

Tri Tu wrote: Hi Brian, By looking at the www.nufone.net, it doesn't have any much details of the services. What is the current rate that you have for domestic long distance rate? 2.9c/min domestic. International varies by country, of course, but is very competitive according to those of my cust

Re: [Asterisk-Users] Anyone working with NUFONE?

Sales wrote: Curious if anyone has any feedback on Nufone voip pbx. Perfectly happy customer. Most of my customers use NuFone as well--perhaps a dozen of us all told. Excellent uptime, reasonable rates. Email-based customer service for the most part. Others are bothered by that; I am not. B

Re: [Asterisk-Users] cannot find -lXext when building * ?

Tilghman Lesher wrote: You have GTK installed, but not X? If you don't intend to run X applications on the server, then deinstall GTK (as the X libraries are required to run GTK apps). This leads to a question that has been bugging us for a while. Is X required to build asterisk? When we trie

[Asterisk-Users] Speex == Screech using version 1.1.4

I just downloaded and built the latest "beta" version of Speex, and it appears to be the case that the 1.1.x versions do not work with asterisk. Just to be sure, I built the 1.1.4 (both with and without the fixed-point option) on two servers so I could test using the same codecs. All calls mad

[Asterisk-Users] CallWaiting CallerID: Available on all channel types?

I have some phones that purport to handle this properly but am having quite a time figuring out just when to expect it to work and when not to. Limited to Zap channels? Zap and SIP, but in different manners? Sieving the list archives yielded more questions than answers. Pointers or discussion

Re: [Asterisk-Users] Cepstral TTS Code

Your website is refusing connections at the moment. Or more properly I should say I get "Connection refused" when I try to access the Cepstral link you posted earlier today to the Asterisk-users list. FYI. Thx. B. ___ Asterisk-Users mailing list [EM

Re: [Asterisk-Users] Cepstral TTS Code

I'm prolly showing my ignorance here, but where *is* this code? I've done a search at the bugs site and it came up dry. It's not in the CVS contrib tree. Don't know where else to look. Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http:/

Re: [Asterisk-Users] Cepstral TTS Code

[EMAIL PROTECTED] wrote: Feedback for the list. I compiled Andy's code. Installation went well (except for me misspellng something in the dialplan) with no problems. The Application works great. Will run down Brian's and give it a try too. Hope you can do us a "HOWTO." Cepstral would be a majo

Re: [Asterisk-Users] Can audio streams go client to cleint with IAX?

Reading this thread leads me to chance asking a somewhat broad question of the gurus: is there a place in VoIP for multicasting? Streaming scenarios, as well as conferencing, would seem to be ripe for that sort of integration, but I know nothing more about it beyond the fact that multicasting i

Re: [Asterisk-Users] Introducing Firefly

Kris Stark wrote: Wow! That seems very reasonable for a phone that looks like it will *work* I wish I could say I was a reseller to get an early sample, but I guess I'll just have to wait with baited breath like most of us here... :) I would bate (i.e. hold) my breath, instead of baiting it, such

Re: [Asterisk-Users] Re: ultra-cheap asterisk box -> Small Biz Robust Asterisk Solution - SBRAS

Tilghman Lesher wrote: On Sunday 18 January 2004 12:01, Adthrawn wrote: Spell out RAID - Redundant Array of Inexpensive Disks. Bingo. That's "Independent Disks". It's the independence of each spindle that is valued, not the cost proposition. If one spindle goes, it's not all of your data which

Re: [Asterisk-Users] Zone Paging

Alfred R. Nurnberger wrote: There are a number of paging interfaces available which connect to a regular phone line on one side and to a paging amplifier on the other side. Could you provide a pointer? The search terms "pager" and "telephone" together are giving me a heck of a lot of noise. . .

Re: [Asterisk-Users] Voicemail Sequence Bug?

days or even weeks later, there the daggone thing is again." I haven't been able to get onsite to play, so I don't know if the voicemailmain app just presents the messages ahead of the numeric gap. But that would be my guess, given her explanation. Thx. B. On Thu, 15 Jan 2004,

[Asterisk-Users] Voicemail Sequence Bug?

I have a user, running CVS a/o 11/23/03, who has complained about "phantom" messages showing up days or even weeks after she has deleted them. So I asked her to let me know when it happened again, and she called a few minutes ago. The directory listing below shows a listing of the /var/spool/a

[Asterisk-Users] Nufone.net net wackiness?

I can't send mail to any addresses in nufone.net; they all get rejected by a spam blocker. And their website is gone, too!! The URL leads to a "parking site." My accounts still seem to work, but I wonder WTH is going on? Thx. B. ___ Asterisk-Users m

Re: [Asterisk-Users] More words for Allison

Andrew Thompson wrote: Original Message - From: "John Todd" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, January 11, 2004 7:36 PM Subject: [Asterisk-Users] More words for Allison As usual, donations to what will be a ~$110 USD expense would be appreciated, as I am paying

Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

[EMAIL PROTECTED] wrote: And why is this unnecessary cruft included in the source tree? So that Digium can leverage the Free Software community into developing proprietary software for them. Am I way off the mark? I think you're unfairly impugning Digium's motives. And I also think you're--ag

Re: [Asterisk-Users] New to asterisk? RUN... don't walk.

Justin Sinclair wrote: Your complaints about the Asterisk Community remind me very much of complaints often made about the Linux Community. Judging an entire community (and even quality of the software) based on the actions of a few people is a big mistake. It was trolling, plain and simple, and u

Re: [Asterisk-Users] Grandstream Quality Survey.... :P

Miguel Cavazos wrote: On Wed, 2003-12-24 at 08:18, Brian Capouch wrote: I have about a dozen Budgetone 101s and I'm pretty much satisfied with them. Sorry, Brian; they've got some rough edges, but they're $65, for God's sake. They are $65 yes, but you can get the best

Re: [Asterisk-Users] Grandstream Quality Survey.... :P

I have about a dozen Budgetone 101s and I'm pretty much satisfied with them. Sorry, Brian; they've got some rough edges, but they're $65, for God's sake. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aste

Re: [Asterisk-Users] Grandstream Early Dial

Andres wrote: On Wednesday 17 December 2003 23:03, Brian West wrote: Stop using beta firmware... I honestly think that GrandStream needs to either fix the phones or stop making them.. THEY SUCKS! I think I would rather eat glass than work with a grandstream phone. We are starting to feel the same

[Asterisk-Users] Share a line with a modem?

I've been on a search to find out whether it is possible to somehow rig up a modem onto a line that also has an X100P on it, and (the rub) allow the line to be used for both inbound calling *and* answering calls on the modem. I would imagine it would have to be in the dialplan somehow, but I ca

[Asterisk-Users] MeetMe: Zap channels don't ever disconnect. . .

I was playing around with conferencing tonight. I was able to place a bunch of SIP phones and a couple of my Zap FXS phones into a conference. So I thought, "Let's see what it's like when people come in from outside." So I called a friend and had him call in on one of my Zap channels, WHICH I

[Asterisk-Users] Wrong voicemail after transfer?

I'm using a modified "default config" file for extensions.conf, the one that uses macro-stdexten to handle the stations. We use a TDM30 card for our stations. When a call that has been rung in using that macro transfers the call things work just fine as far as the "other" instrument ringing. B

Re: [Asterisk-Users] SIPURA Breaches Contract

John Breeden wrote: These are just claims. Post a pdf of the contract. It is in NO ONE'S best interest that he do so. Legal matters belong in courts of law, not email discussion lists. I daresay most lawyers would have advised against the original posting. I am not a lawyer, but my layperson's

Re: [Asterisk-Users] John Brown from Chagres!

John Brown (CV) wrote: Hi List, Just a quick note that we have cleared all back logs of Grandstream product. If you have been awaiting shipment, its shipped. Everyone should be getting tracking numbers shortly. We also have NEW STOCK that can ship within 2 to 3 days of order BT-101 BT-102 HT-

Re: [Asterisk-Users] Erratic DTMF on E1/PRI (continuation of Strage bip on ISDN/PRI)

Paulo Mannheimer wrote: At the same site, DTMF recognition is functioning badly, sometimes duplicating digits and sometimes totally missing others. We have checked already /proc/interrups, there is no interrupt being shared. FWIW right now this problem is the most vexing one we're facing, actually

Re: [Asterisk-Users] Asterisk freezing HELP

TeleSIP wrote: And by the way, do not use firmware > 1.0.4.18 on GS phones. It contains a nasty SIP Port bug. Where do you guys come up with this Grandstream firmware that isn't on their website? Are their "back channels" that the rest of us are kept out of? This is the umpteenth reference to

[Asterisk-Users] Vonage sending Motorola gear now?

I got a call from an ISP friend tonight who said he is getting calls from people who are getting signed up with Vonage. Instead of sending them ATA186s, apparently they're receiving something made by Motorola. They apparently work significantly differently than the Cisco units, and there have

Re: [Asterisk-Users] some success with linux 2.6 and wcfxo

Tristan 'Minty' Colgate wrote: I'd like to thank everyone on #asterisk for all the support they gave to a fellow linux enthusiast... absoutely none. That was really a nice post until right at the end here. I hope you understand that cheap shots like this just make *you* look like an asshole,

[Asterisk-Users] SIP behind NAT: NAT'ted end has to talk first?

I am having problems in a couple of installations where I have SIP phones (both GS101 and ATA186) connecting to an asterisk box that has a public IP address, where the stations are behind NAT. I'm still testing to make sure I have all the permutations looked at, but from what I can tell, what i

Re: [Asterisk-Users] Multi-line TTS Outbound Dialer

Michael Devenijn wrote: We are working with realspeak and it is a wonderfull product (even in product) it supports up to 20 languages and has aquired a really good prod. stability ! What kind of money we talking for that product? Thx. B.

Re: [Asterisk-Users] Mailing list archives searchable ?

Greg Hill wrote: On Thu, 27 Nov 2003, Brian West wrote: Come on guys how hard is it to add "site:lists.digium.com" into the google search box along with your keywords? Or is that like too hard? This doesn't always work well. For example, ten days ago a message came through the list with the te

[Asterisk-Users] One voicemail -> multiple recipients?

The subject pretty much says it all. I have a customer who would like to have an option where a caller can leave a voicemail in such a fashion that it would be simultaneously delivered to a set of mailboxes all at once--the idea is "trouble ticket" type operation where multiple technicians wil

Re: [Asterisk-Users] Iconnect (DeltaThree) config on Asterisk

Chris HARIGA wrote: Hi, Is anyone using the iconnect on Asterisk to receive and to place calls? I use it for both incoming and outbound calls. My phones use private IP only; my asterisk server is private but is NATted to public. The only problem I have with it is a very slight delay before it

Re: [Asterisk-Users] Asterisk GUI Client Released!!!

mattf wrote: Hello, I have finished my basic polishing of the Asterisk GUI client I have been writing in Perl/TK and have released a first beta version on sourceforge: http://sourceforge.net/projects/astguiclient/ I am still working on a user manual for the application, but the code works and we

Re: [Asterisk-Users] box for asterisk

Steve Bradwell wrote: Hi All, We are looking at creating our first asterisk box, what type of server requirements do we need to keep in mind? Is there any preferred server that is used for asterisk? How processor intensive is asterisk? And what is the requirements for the sound card? I was just

Re: [Asterisk-Users] OT: Document Control System?

Steven Critchfield wrote: Are you trolling here, or are you just clueless about the people who will be helping contribute to your documentation? I'm sure I am not the only one here that goes weeks on end without touching windows. Screw Word and its largely bloated file formats. Hear hear!! I ha

Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing

WipeOut wrote: So the basic rules where NAT is involved are.. Asterisk server must always be on a public IP address.. You keep saying this, but it is not correct. I have several asterisk servers running behind NAT servers, and they function perfectly. I won't say configuring them was as easy

[Asterisk-Users] Directory App Weirdness

I noticed tonight, when doing a demo of the Directory app, that something mighty odd is going on. I have one Zap FXS channel and a SIP channel (Grandstream B101). When I invoke that app on the Zap phone things work normally. When I invoke it from the GS phone, the CLI shows that it is playing t

Re: [Asterisk-Users] Asterisk behind NAT to SIP provider

rnc Info Lists wrote: Thats great to hear. Can you please share your config files that connect iconnecthere and net2phone via SIP? I think there are a number of people here who have tried and not been able to get it to work. Here's what I'm using for iconnecthere. They provide me with both ori

Re: [Asterisk-Users] Asterisk behind NAT to SIP provider

Jonathan Hogg wrote: At this moment, Asterisk behind a NAT can't connect to an outside SIP provider. If you put asterisk outside your NAT, your inside clients can connect to Asterisk and Asterisk will be able to connect to your providers. Not true. My asterisk server(s) are behind NAT, and I am

[Asterisk-Users] Tonight's CVS breaks Grandstream phone

FYI. Haven't dug enough to be able to report any more, but re-fetched CVS to verify that sometime in the last few days CVS changes now break my GS phone. It appears to be at the RTP level. It seems to set the call up just fine, but no audio is passed back to the instrument. I reverted, and w

Re: [Asterisk-Users] A software FAX modem

Steve Underwood wrote: You can find the software at . It currently consists of two parts: I cannot get that domain to resolve tonight. FYI. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailma

Re: [Asterisk-Users] Survey: Grandstream improvements.........

John Brown (CV) wrote: Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. 1. More volume out of the speakerphone, and better range of the headset volume. I guess it would be

Re: [Asterisk-Users] Creating new voicemail accounts

WipeOut wrote: If you are using VM2 and using something other than the [default] context then you will not create the correct directory structures when you run the addmailbox script to create the mailboxes.. I have attached a copy of my addmailbox2 script which takes into account the context w

[Asterisk-Users] Creating new voicemail accounts

I have googled this one to death, and can't find anything. I added a number of new users to my asterisk (current CVS) system. I am using the "Voicemail2" family. I added entries in extensions.conf and voicemail.conf for my new users, and I have tested leaving and retrieving new voicemails for

[Asterisk-Users] Digium TDM card bad DTMF again

I hesitate to open up another bug ticket on this (my older one just having been closed), but after having worked for some number of CVS iterations, my Zap TDMXX stations seem to be struggling again to send DTMF clear enough to be "understood" by IVR equipment I interact with. I can use the same

Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't:no sound after 5sec

Paul Cheng wrote: Our experience with the Budget Tones 101have been poor as well. The devices seem to die after a day or two (even with the new firmware) and then need to be rebooted. On occasion, the device needs to be literally unplugged and plugged back in as even the reset doesn't work. Min

Re: [Asterisk-Users] Anyone else use Audacity for prompts?

Shaun Ewing wrote: I know that some of this comes with the territory, but I wonder if there is anyone out there who does this ( wav -> gsm) routinely, and who can advise me as to the MO I could use that results in the highest quality in the resulting playback files. What are you using to convert th

[Asterisk-Users] Anyone else use Audacity for prompts?

I am using Audacity to record some voice prompts. The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. I know that some of this comes with the territory, but I wonder if there is anyone out there who does this routinely, and wh

Re: [Asterisk-Users] Iconnect Incomming calls

Glenn Dalgliesh wrote: I have an IconnectHere account with a Inbound number and have setup the sip.conf to register and am recieving the call but When I answer the call it disconnect. I have tried sending the call to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon a

Re: [Asterisk-Users] "New" TDM cards--driver won't load

Mark Spencer wrote: Is it showing up on /proc/pci? It should be a tigerjet. Yes. I put the other card back in (production machine) but over the weekend I'll get the card in there and capture the output of lspci. Does "dmesg" report anything unusual? Nope. Doesn't show any sign of seeing

[Asterisk-Users] "New" TDM cards--driver won't load

I've searched the site with google, but can't think of the magic words I guess. I got a "swap out" TDM30 today to replace my buzzy one. I swapped it with the older one, swapped out the FXS modules, hooked it up to the computer's power supply, and booted, but the wcfxs driver won't load--it giv

Re: [Asterisk-Users] error message 49159

Martin Pycko wrote: We send SIP messages to that device up to 6-7 times and then we stop and this message shows on the console. WARNING[49159]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) So it isn't really an error then, but a

[Fwd: Re: [Asterisk-Users] Voicemail: Timestamp suddenly reverted to GMT!!]

FYI to list. Thx. B. --- Begin Message --- Tilghman Lesher wrote: Please post your entire voicemail.conf, with the general and zonemessages sections, as well as your voicemailbox definition (minus the password). Also, post the output of: bash$ grep app_voicemail2.c /usr/src/asterisk/apps/CVS/Ent

Re: [Asterisk-Users] SIP Registration Difficulties

Dave Cotton wrote: I have SIP registrations working correctly for FWD and Sipphone, but it is impossible to connect to Sipcall or Nikotel, I saw that someone on the list has problems with ICH. Does this imply that it will work even in a NAT environment? I have watched the list like a hawk for ev

[Asterisk-Users] Voicemail: Timestamp suddenly reverted to GMT!!

I wonder if I'm the only one who finds Allison reading the timestamp on my voicemails in GMT, after months of having it done with the local time I have set in voicemail.conf. The timestamps on the message files, and the emails that are sent, are correct. Only the spoken dates appear to be affe

Re: [Asterisk-Users] RE: SIP i.e. Is something broken?

WipeOut wrote: Lists wrote: This is my issue as well, Does anyone know how to fix it? Roll back to the CVS from last Thurdsay, This worked for me.. If you like you could try Friday and see if it works which will help narrow down when the problem started.. :) I'm going to bet that it's codec ne

Re: [Asterisk-Users] RE: SIP i.e. Is something broken?

WipeOut wrote: Your ATA-186 and Bugetone are both behind NAT or both non-NAT?? In otherwords are they both in the same setup in relation to the server?? The setup is identical; both are behind a NAT translating router. B. -- This message has been scanned for viruses and is believed to be cle

Re: [Asterisk-Users] RE: SIP i.e. Is something broken?

Christopher J. Wolff wrote: Is it safe to assume that a fresh CVS build will not have the SIP translation problem described? Regards, Christopher --__--__-- Just FYI: I had similar problems for a while, and then I completely scrapped my CVS directory and did a CVS CHECKOUT (instead of an update).

[Asterisk-Users] Re: Continuing Budgetone woes: asterisk was the culprit!!

Brian Capouch wrote: I have spent the morning on this project, still without success. When I saw the mail on the list tonight from "[EMAIL PROTECTED]" it finally dawned on me to try a CVS-revert and see what happens. It turns out that solved the problem--I can't say when exactly

[Asterisk-Users] Continuing Budgetone woes

I have spent the morning on this project, still without success. Summary: Yesterday I inadvertently unplugged my Grandstream phone. I might add I did a rebuild of my s/w from CVS at the same time. Since then, the Budgetone seems to talk SIP just fine, but the RTP being sent to it by asterisk

[Asterisk-Users] Budgetone + NAT: Firmware Version?

I inadvertently unplugged my Budgetone phone tonight and I think it went out and upgraded its firmware. It is now at 1.0.3.81. Does anyone know how new this might be? Suddenly things that have been working wonderfully before don't work--basically it seems to do the SIP stuff just fine but the

[Asterisk-Users] Oops!!! Current CVS crashes

I don't know whether this ought to go to the bugtracker. I downloaded the current CVS last night and then again just a few minutes ago. In both cases I can crash asterisk very easily by the following method: 1. Call up and leave a voicemail. 2. Log in and listen that I have a new message. 3. Hi

Re: [Asterisk-Users] how many production systems are there?

Steve Totaro wrote: i am excited too. what kind of wireless wan? I operate a wireless WAN that covers approximately 500 square miles of NW Indiana. We are lightly loaded right now, so it is not yet possible to tell whether or not we're going to have QoS problems once traffic picks up. But for

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