RE: [Asterisk-Users] G.729? Worth it?
I do not experience the same issues as below. I personally think g729 is more then worth the $10 per licensed channel. I have 10 licensed channels and they are worth every penny. The call quality is far superior to gsm or g726 and is nearly identical to a g711 call. My application is probably different then yours, I convert everything to g711 to my terminators to ensure pristine call quality. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Rodan Sent: Wednesday, January 19, 2005 1:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] G.729? Worth it? Funny thing about g729, when I setup my server to ONLY allow g729, and then put my Cisco 7960's phone into g729 mode, the first call went through fine as g729. If I put that party on hold, and dialed another number, that too would be g729. But if I tried to call 1 party, and conference in another party, I got codec warning messages, the phone(s) were trying to make the 2nd call as ULaw, and what's worst, it try to transform the original stream that was g729 into ULaw as well. I've gotten some strange behaviour of my Sipura and using g729 with multiple streams as well. Certain calls try to revert back to ULaw. For the most part, I've given up on g729. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Schulte Sent: Wednesday, January 19, 2005 12:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] G.729? Worth it? There's a MOS scale for this kind of stuff -Original Message- From: Paul Fielding [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 19, 2005 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] G.729? Worth it? Low bandwidth Low CPU utilization Best audio quality I think you might want to clarify that Best audio quality is in relation to other highly compressed codecs. Certainly my (albeit limited) experience is that g711 is much more clear than g729. Compared against gsm, for example, however, the audio quality is quite good regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Service contract for * in NYC area
I'll make my entry into this Flamewar BKW I believe the term is pot, as there are many colors of felines... As to the content... None of my business. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Monday, January 03, 2005 12:03 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Service contract for * in NYC area Karl, You talk about unprofessional? Isn't that the cat calling the kettle black? Your post was way beyond unprofessional it's not even funny Thanks, Brian -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Karl Brose Sent: Sunday, January 02, 2005 10:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Service contract for * in NYC area This was posted in a public forum asking for business relationship. But the intent was for disguised market research which could have been stated clearly upfront and there would have been overwhelming positive response attesting to the facts and assisting in his quest. But communicating with colleagues, so to speak, and luring response by offering established business (which hasn't been ascertained) is totally unprofessional and I don't want to do business with this guy no matter how big the contract is. There is plenty of business out there. Been in business for a long time and learned to stay away from manners like that. The public post is to warn others of these tactics. My response to the original query and a couple of private follow-ups were in good faith, and no professional deserves to be told finally oh, well, I just wanted to see what the response would be Note that the original post stated I have been contracted.. C F wrote: Mr Brose. I don't see why you did this? why was this needed? All I wanted was to show my client that * is popular. Without this none of us can sell this if we want. Nobody buys a pbx they can't get support on (we are talking about a $50,000 installation). Therefore I had to show my client that it is popular enough in this area that he will be able to get a service contract in the event that I can't service it. So I posted this question as I did. I did not mislead you in any way, if my client decides to use me to install * and then wants a service contract I will have to give him someone local. Now up until this email it could have been you, you just lost this contract (if/when my client decides to take * from me). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
I do about 500 calls per day on average volume and about 750 on heavy volume and find it necessary to run a logger rotate every other day... other then that I can go on for a couple weeks until I need a full reboot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
I just make it a habit, the only issues I run into are after an IAX2 gridlock and my log files get filled up quickly... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Thursday, December 30, 2004 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Is asterisk that unstable On Thu, 2004-12-30 at 09:50 -0500, Luke Catranis wrote: I do about 500 calls per day on average volume and about 750 on heavy volume and find it necessary to run a logger rotate every other day... other then that I can go on for a couple weeks until I need a full reboot. Oddly enough, My logs are approaching a year or more back and don't need to be rotated for size yet. I will do it the next time I have to do something with asterisk that time. My debug file is 418megs for over 1 year of logging. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
Logger rotate from cli -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Randy MacKay Sent: Thursday, December 30, 2004 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Is asterisk that unstable I do about 500 calls per day on average volume and about 750 on heavy volume and find it necessary to run a logger rotate every other day... other then that I can go on for a couple weeks until I need a full reboot. How do you rotate your logs? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 12/28/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Another Asterisk Certification
How much time did you waste on that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Taylor Sent: Sunday, August 22, 2004 10:24 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Another Asterisk Certification Alternate Certification For those of you who can't (or won't) shell-out the $3000+ for the 5 day certification class, here's a quicker way AND IT'S HALF THE MONEY! www.metrotel.net/asterisk.htm Asterisk is a good product. Some people need certification. A mature product needs certified professionals. Asterisk is maturing. Remember the Certified Novell Engineers? There a a lot of people that know everything about Novell who never got the white lab coat. There is a place for cetification. It helps all of us, even those who never become certified. -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] My Boss wants background music!!!!
Xml services for cisco 7960, setup a broadcast stream. Check the wiki. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Friday, December 17, 2004 3:28 AM To: Wilson Pickett; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] My Boss wants background music On Fri, 17 Dec 2004, Wilson Pickett wrote: I am searching for a new PBX for the company. My choice is Astrisk. My Boss wants background music via all the telephones. This is done in a conventional PBX that he wants, but I can use the Asterisk PBX if it can do What a waste of resources though, like installing video games on the asterisk server... Ther must be a powerline intercom that would handle this (adding a speaker per music distribution point.) The requirement of the original poster was to mute the music at the desk when a call is in progress. It would be really nice if there was a hardphone capable of accepting a multicast high-quality stream when no call was in progress. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: cisco 7940 help
Solar winds... free tftp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk users Sent: Thursday, December 02, 2004 10:18 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: cisco 7940 help Hi, I spent 3 days trying to upgrade my 7960's with similar probs to what you say. My problem was that my TFTP server didnt follow the RFC's properly and didnt respond with 'file not found' when a file didnt exist. the phone kept on looking for the same file over and over and never moved on to the next step in the upgrade process. the trick for me was to use a different TFTP server. Klever Pumpkin, Windows 2000, and the newer 3Cd all worked. my original broken TFTP server was a 3Com one. so perhps you should try pumpkin as its prolly the easiest for a temp solution. cheers, Mick Andrei (MPI) [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Rich Adamson wrote: I've successfuly converted 7940 from call manager firmware version 3 to SIP 7.3. just last week. You need to upgrade to firmware version 6 first, then upgrade to 7. Also once you've upgraded the phone, you should remove firmware config file from tftp server, otherwise the phone would be in constant upgrade loop. There is couple of tricks in between. The phone does a version check before attempting an upgrade. If the same, it doesn't bother upgrading again. No need to remove anything. In fact, my server has v2.3, 3.3, 5.1, 6.1, etc on it at all times. Some of those are required (in steps) to upgrade the older 7960's. Rich, I am telling what I saw on my 7940: it was in a constant loop trying to upgrade again and again. You may say that Cisco phones are easy upgradable and all one need is just follow instructions, but that won't help the man who was pulling his hair off. Andrei ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] After setting up my FXO card, what should I now order from my telco?
Get a vonage line unlimited for $24.99 per month and get a softline and have vonage set it up as a rollover line. Grand total $34.98 per month. Route outbound to the ZAP/VONAGE first, then to the SIP/VONAGE NEXT, if need be setup an IAX termination account with voicepulse or iax.cc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Clements Sent: Wednesday, December 01, 2004 1:00 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] After setting up my FXO card,what should I now order from my telco? Ok, so I'm setting up my small office. I have my asterisk machine setup and I have 3 sip phones connected as my stations and a 4 port FXO card ready to go(planning on only using 2 lines currently). What should I now order from my telco(sbc in this case) Everytime I call, they want to sell me this expensive $50 package that bundles everything and that's for a single line. Is there a specific type of line I should request? What is everyone else doing? How much is everyone else paying for single line PSTN access for their small/medium setup's? Thanks, Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP to SIP Wait
Question?
Is it possible to pause during a dial statement?
The Issue I have is that I have to enter an IVR bypass code before I
enter the dialed number from my phone. The SIP client I'm sending my
calls too is either not getting the dialed number {EXTEN} or it's coming
through wrong...
Anybody?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Motorola Vt1000
Yes go restore it to a factory config and your all set... Via the web interface. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Halverson Sent: Wednesday, October 27, 2004 3:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Motorola Vt1000 Okay so here's the deal..I canceled my Vonage account today. And they are going to charge my credit card cancellation a fee if I don't return their device, a Motorola VT-1000. Any way to re-flash or use this with asterisk? $39.99 cheap ATA if I am able to use it. Otherwise I'll return it and get my cancellation fee back. -Mark Halverson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trabas Radius
Any tips, tricks or treats out there? I'm building a new system and would like to move away from my SQL based call rating solution... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] modem question
http://store.yahoo.com/asteriskpbx/witd1pofxs.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of christophe de coninck Sent: Thursday, October 21, 2004 1:17 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] modem question Hey, I've got asterisk running fine in my internal network now but it is not yet connected to the telephone line, now I'm wondering, would it be possible to use a: US Robotics 56K faxmodem V92 - Internal - PCI to make phonecalls to normal telephones and receive them ? or wich other brands/models do you recommend that are affordable and easy to maintain in belgium/europe because i cant find the intel IA92 card or any of the wildcard's (wich are pretty expensive) -- Christophe De Coninck | Zarek K http://www.zarekk.be mailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] image001.gif___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Control access to external dialing
Wondering if anyone could give me a tip on controlling access under the following scenario. I have an ATA connected to a legacy pbx as a trunk line. I want to control who can make calls on this trunk. I cannot set restrictions on the users via the pbx, so I would like to be able to assign a passcode for people so they can dial out using this trunk line... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Control access to external dialing
That would work, but I have multiple people and I my customer needs to be able to track who is using the line and for what. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David J Carter Sent: Wednesday, October 20, 2004 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Control access to external dialing Luke, I have a situation like yours, mine is to enable an IAX2 call between two servers and then break out to a trunk. All I have done is added a six digit code in front of the number (eg Birthdate ,210573 or 052173 if in US), and then stripted the six digits before dialing. You only tell the people you want to be able to dial out the six digit code. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Luke Catranis Sent: 20 October 2004 15:55 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Control access to external dialing Wondering if anyone could give me a tip on controlling access under the following scenario. I have an ATA connected to a legacy pbx as a trunk line. I want to control who can make calls on this trunk. I cannot set restrictions on the users via the pbx, so I would like to be able to assign a passcode for people so they can dial out using this trunk line... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sccp cisco 12sp HELP !!!
The cnf files are in plain text, I don't know where you got the firmware upgrades, but you should be able to get the SEPDefault.cnf and the SEPMAC.cnf off the CISCO site. You also need the O97XX.txt or whatever it is... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: Friday, October 15, 2004 9:31 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] sccp cisco 12sp HELP !!! SCCP Files are dynamically generated by Cisco routers and CME, They are in a binary format. I haven't been able to upgrade my sccp phones due to not knowing anyone with a CME software to generate the files for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Price Sent: Friday, October 15, 2004 8:02 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] sccp cisco 12sp HELP !!! ok guys, ive been trying to get this to work for 6 hrs now ive got a cisco 12 sp and i am trying to get it to work with sccp. The phone boots and is looking for the SEPDefault.cfg or the one below, BUT i cant find anywere on the net what the content of this file is im guessing that its the ip of the * box. im riping my hair out on this one please help... 20:54:47.793156 192.168.1.15.51216 apollo.tftp: 28 RRQ SEP00D0BA848162.cnf [tos 0x10] Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sccp cisco 12sp HELP !!!
I actually found samples of them on the cisco site on a standard html page when I was looking for the SIP.cnf files. I copied the text off the page and made the files from scratch in notepad. I only have the SIP files, which if you were converting to sip you could use and rename as SEP, even though some of the data would be incorrect. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: Friday, October 15, 2004 9:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] sccp cisco 12sp HELP !!! Luke, Would you please point me to where those files are on the cisco site? All I have been able to find are the sipdefault.cnf and the sipmac.cnf files on Cisco's site. According to a cisco engineer they are not plain text for sccp. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luke Catranis Sent: Friday, October 15, 2004 8:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] sccp cisco 12sp HELP !!! The cnf files are in plain text, I don't know where you got the firmware upgrades, but you should be able to get the SEPDefault.cnf and the SEPMAC.cnf off the CISCO site. You also need the O97XX.txt or whatever it is... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: Friday, October 15, 2004 9:31 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] sccp cisco 12sp HELP !!! SCCP Files are dynamically generated by Cisco routers and CME, They are in a binary format. I haven't been able to upgrade my sccp phones due to not knowing anyone with a CME software to generate the files for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Price Sent: Friday, October 15, 2004 8:02 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] sccp cisco 12sp HELP !!! ok guys, ive been trying to get this to work for 6 hrs now ive got a cisco 12 sp and i am trying to get it to work with sccp. The phone boots and is looking for the SEPDefault.cfg or the one below, BUT i cant find anywere on the net what the content of this file is im guessing that its the ip of the * box. im riping my hair out on this one please help... 20:54:47.793156 192.168.1.15.51216 apollo.tftp: 28 RRQ SEP00D0BA848162.cnf [tos 0x10] Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voice choppy
Nearly everyone of my clients is between 600 and 700 ms and there is no choppyness to the calls. They are on VSAT uplinks with SCPC return channels, but I would say that latency is certainly not your problem. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dana Nowell Sent: Friday, August 13, 2004 2:22 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] voice choppy OK, background/config. running * (show version reports 0.9.0) on Mandrake 9.2 (kernel: 2.4.22-32mdk) with a dual 800mhz PIII with 256M Ram 4port FXO digium card, no IRQ sharing I can find (cat /proc/pci cat /proc/interrupts), vmstat reports a minimum of 80+% CPU idle when problem occurs. connect to a Grandstream 101 (GS) via vpn (no nat). Link has 100ms - 150ms ROUND TRIP latency (constant 'ping' during test). Codec is alaw or ulaw. TDM card is plugged to an NEC PBX (old NEAX 2000 IVS) via the analog station card in the PBX. PROBLEM: Establish a call from the GS to an NEC phone (Dterm III) connected to the PBX. The voice quality on the GS sounds good. The voice quality on the NEC gets very choppy (random). A call from the same GS to an internal GS here (same latency, same IP path) is much better (some chop). SOLUTIONS tried to date: Tried a local network test, sound excellent. Tried a 'low latency vpn' test bed to reduce latency (is latency the issue?). Latency down to ~70ms round trip, better but still noticeably choppy. Tried different codecs (ulaw, alaw, iLBC), no impact. Tried various echo cancel/train on/off values, cancel and train on seem best. Tried modifing Samples/TX packet changes, 2 and 64 tried as 'ends of spectrum' values. little noticable impact (2 seems marginally better, still clips, MAY be less often). Tried changing TXgain and RXgain under the assumption audio level clipping occurs between tdm and NEC, some improvement. Substituted an SJPhone on a PC instead of the GS, better, still choppy. Tried the 'demo speech' on an extension, called from Dterm, sound is excellent, tried GS over low latency vpn, choppy, tried SJ over low latency vpn, choppy. (h, does that eliminate the tdm to NEC link as an issue?) Tried internal GS over low latency vpn to SJ (out thru vpn to * back thru vpn to sj), choppy. Current state: I've improved from 'crappy cell call about to disconnect' to 'average to crappy cell call' quality. Not exactly ready for customer use. ASSUMPTIONS: Original assumption was latency, but latency was still below the '150MS one way' values I see quoted for good sound quality. In fact over the low latency vpn the values were less than a third the quoted value. Sometimes I get a high latency and get a drop out, OK expected. Sometimes I get chop at a lower latency than I was getting with 'good' sound. Second assumption was interface between NEC and TDM was clipping, after changes to tx/rx gain the 'demo' speech sounds fine on the Dterm. (Does this remove the interface from the possible bad guys list?) HELP Needed: So given that latency is in the 70 - 100 ms round trip. Given that I've diddled rx/tx gain. Given that I've tried basic echo on/off settings. What's next? Is the 'common' 150 ms one way (150+ms round trip) value a bunch of crap? Do I need some other magic latency goal? Would a 729 codec help? Is there a test I missed? Some other values to twiddle? Is this a 'known issue' I don't know about, fixed in some more recent version? (Yeah, I can just try the CVS-HEAD and hope to get lucky and I probably will as I'm out of options/ideas but KNOWING it is fixed is better than hoping it is fixed) I'm stuck and my forehead is getting flat from pounding it on the wall. Anyone handing out clues? -- Dana Nowell Cornerstone Software Inc. Voice: 603-595-7480 Fax: 603-882-7313 email: DanaNowell_at_CornerstoneSoftware.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How Many Calls On This Config
I've seen a couple posts about SMP, and I'm wondering if anyone has an idea what I can handle with this box: Dual P2 Xeon 450 MHZ with 2MB L2 Cache 1 GB PC100 SDRAM RAID 5 Running SuSE 9.1 2.6 SMP Kernel... Currently I have 16 users, 10 CISCO 7912G Using ULAW 3 Sipura 1000's Using GSM or g726 2 Motorola MTA VT1005's Using ULAW 1 Iaxy Using ULAW and g726 So far I've had 7 concurrent calls with mixed codecs and no issues on call quality. I'm just wondering if anyone has had any experience with this type of config? Luke ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 831 Santa Cruz/Watsoncille, Calif. DIDs
gafachi This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of lists-jmhunter Sent: Tuesday, August 10, 2004 1:38 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 831 Santa Cruz/Watsoncille, Calif. DIDs Hey there, I don't know who else has suffered broadvoices terrible service, but I am about to my end with them. The lack of a LBR codec, the outages, the changing of servers without notifying subscribers haspushed me to my end. Now most incoming calls are abbruptly cut off within a minute of the call starting. Anyone know of any other * friendly providers that have DID, besides Voicepulse, Nufone, broadvoice. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Inbound not working with iconnect
Did you change the voicepulse context to user in you iax.conf? This is what they told me to do when I had the same problem 3 weeks ago: [voicepulse] type=user context=voicepulse-in ;auth=md5 ;secret=mysecret host=gw5.voicepulse.com qualify=yes This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raj Sent: Saturday, August 07, 2004 11:35 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Inbound not working with iconnect You are right Greg, I do get a 407 error when I sniff the packets and it's the same problem now. I'm able to receive calls from Broadvoice but not from iConnect. Any help would be appreciated in this regard. Thanks, Raj Greg Blakely [EMAIL PROTECTED] wrote:This may be what I experienced in my thread New Head Appears to break SIP to iConnect. Maybe it WASN'T the fact that I upgraded my asterisk software. But, yes. I noticed the problem day before yesterday. - From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raj Sent: Friday, August 06, 2004 12:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Inbound not working with iconnect Hi there, Since last 2 days iconnect's incoming is not working. Is it the same with everybody? For the past 5 months I've been using this service perfectly in two boxes and suddenly it stopped functioning. I'm able to call out, the version is 0.9.1. Any help is appreciated Thanks, Raj - Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! __ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Inbound not working with iconnect
Oops... mybad This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luke Catranis Sent: Saturday, August 07, 2004 11:53 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Inbound not working with iconnect Did you change the voicepulse context to user in you iax.conf? This is what they told me to do when I had the same problem 3 weeks ago: [voicepulse] type=user context=voicepulse-in ;auth=md5 ;secret=mysecret host=gw5.voicepulse.com qualify=yes This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raj Sent: Saturday, August 07, 2004 11:35 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Inbound not working with iconnect You are right Greg, I do get a 407 error when I sniff the packets and it's the same problem now. I'm able to receive calls from Broadvoice but not from iConnect. Any help would be appreciated in this regard. Thanks, Raj Greg Blakely [EMAIL PROTECTED] wrote:This may be what I experienced in my thread New Head Appears to break SIP to iConnect. Maybe it WASN'T the fact that I upgraded my asterisk software. But, yes. I noticed the problem day before yesterday. - From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raj Sent: Friday, August 06, 2004 12:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Inbound not working with iconnect Hi there, Since last 2 days iconnect's incoming is not working. Is it the same with everybody? For the past 5 months I've been using this service perfectly in two boxes and suddenly it stopped functioning. I'm able to call out, the version is 0.9.1. Any help is appreciated Thanks, Raj - Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! __ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone use AdvancedVOIP ?
I tried to get it working, even opened several tickets with them, but eventually hired a programmer friend to build something similar for me... I'm sure with the proper programming and time you could get it working, but it won't have nearly the same level of integration as a it would with a CISCO AS53XX This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Bentley Sent: Thursday, August 05, 2004 1:32 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Anyone use AdvancedVOIP ? Has anyone used the Voip Billing System from http://advancedvoip.com/ ? They seem to also offer a billing solution for Interconnections. I'm curious if anyone has some experience using their software? Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail attachment setup per user
Yes in the voicemail.conf Set the default to no and follow the instructions in the voicemail.conf file i.e (not sure if this is exact X = ,Joe Foo,[EMAIL PROTECTED],,||attach=yes This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Carr Sent: Thursday, August 05, 2004 4:34 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] voicemail attachment setup per user Is it possible to set the attach= setting on a per user or per context basis? We want to give our users the choice of no email notfiication, email notification with no attachment, or notification with attachment. Thanks, Gary ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PSTN Access Providers for Asterisk
Try voicepulse connect connect.voicepulse.com This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William R. Lorenz Sent: Wednesday, August 04, 2004 3:23 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PSTN Access Providers for Asterisk Asterisk Users, I'm looking for U.S. providers that will provide access to the PSTN and allow me to easily use my Asterisk box with their services. I would prefer a provider that supports number portability, so that I can park my existing cell number on their network and later move it again, but I'm open to doing some funky stuff with call forwarding if I have to do that. Can anyone provide their recommendations or experience in using a VoIP provider, as opposed to a LEC, to provide Asterisk with PSTN access? Thanks, in advance, for your ideas. -- _ __ __ ___ _| | William R. Lorenz [EMAIL PROTECTED] \ V V / '_| | http://www.ohiolinux.org/ ; Free conference and event hosting \./\./|_| |_| Linux and OSS-related topics. October 2, 2004 - Columbus, OH. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: No incoming audio on incoming SIP calls
I must do the same with the proxy... one note... the t stands for transfer per the wiki: t : Allow the called user to transfer the call http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Dial -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gurr Sent: Wednesday, August 04, 2004 3:56 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: No incoming audio on incoming SIP calls Solved my own problem ... thought I'd record it here for any others who come across it. The problem arises since Asterisk is trying to get out of the way of the media stream, by doing a SIP re-INVITE to get the two ends of the conversation to talk directly. This won't work, as Asterisk is telling the calling party that the IP address to talk to is the private IP address of the softphone on the internal network. Adding canreinvite=no to the softphone's stanza in sip.conf solves the problem. It would be helpful if Asterisk noticed that it's about to tell the other end to use a private IP address ... the ranges are well known, and Asterisk could do an implicit canreinvite=no in this situation. The same problem didn't occur on outgoing calls as the Dial string includes a t for timeout - as per the wiki, this means that Asterisk must stay in the stream to be able to implement this. Of course, the other way to solve this would be to use a proper SIP proxy server which handles RTP stream port forwarding ... something I must get around to. -- David Gurr Congruity Ltd. Hemel Hempstead, UK -Original Message- From: David Gurr [mailto:[EMAIL PROTECTED] Sent: 04 August 2004 14:05 To: [EMAIL PROTECTED] Subject: No incoming audio on incoming SIP calls Now this is really frustrating. Everything was working fine, and now it isn't ... I don't think I've changed anything that would affect this, but I guess you never can be too sure. My setup is as follows: SIP softphone (SJphone) connected to Asterisk running my Linux NAT firewall box. This is all on the internal network. Asterisk then dialing out through various means - SIP to Stanaphone, FWD, Gossiptel and PSTN via an X100P. For incoming calls, an 0870 number from CallUK routes to my FWD account, and an 0870 number from Gossiptel routing to my Gossiptel account. Outbound calls all work fine ... I get audio in both directions, no problem. Incoming calls on either 0870 number connect fine, and audio goes from the softphone to the caller, but not the other way ... I hear no audio on the softphone from the caller's phone. I'm getting no alerts from my firewall that it's dropping anything. I know my way around packet sniffers, but I don't know what to look for here. What should the inbound audio packets look like? Thanks -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gafachi?
Anybody use them... I signed up for $20 to see how there system works.. They're at $.02 per minute for US Termination and their other ITX rates aren't too shabby. Sadly my IAX registration is rejected... maybe a glitch, wondering if anyone's had a similar issue. Luke ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Gafachi?
Nevermind... the microfiche username was wrong... PBAK Works... but call quality is a little weak... This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luke Catranis Sent: Tuesday, August 03, 2004 10:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Gafachi? Anybody use them... I signed up for $20 to see how there system works.. They're at $.02 per minute for US Termination and their other ITX rates aren't too shabby. Sadly my IAX registration is rejected... maybe a glitch, wondering if anyone's had a similar issue. Luke ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Motorola MTA RFC3389 Problems
Testing MTA VT1005, one specific issue, RFC3389 support incomplete, please turn off on client... I 've tried all sorts of RTP settings (there's only 4 possible options). And no luck... Any help? Luke ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Motorola MTA RFC3389 Problems
I fixed it...at least within my capacity... if someone knows better, please advise. 4FR: IN THE MTA CONFIG BTIOPT TeleSipOobDtmf = FALSE BTIOPT TelSipVadEnabled = FALSE BTIOPT TelSipRtpPortBase= UINT16(3) BTIOPT TeleSipPacketPeriodSelection[0] = UINT8(30) IN ASTERISK SIP.CONF dtmfmode=inbound This got rid of the RFC3389 ERRORS. If you do not set the SIP.CONF for dtmfmode=inbound, you will generate the RFC3389 ERRORS, and you will not be able to login to VOICEMAIL. This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luke Catranis Sent: Thursday, July 29, 2004 12:12 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Motorola MTA RFC3389 Problems Testing MTA VT1005, one specific issue, RFC3389 support incomplete, please turn off on client... I 've tried all sorts of RTP settings (there's only 4 possible options). And no luck... Any help? Luke ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voice mail problem
Set the minimum voicemail length in the voicemail.conf file. This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Keding Sent: Thursday, July 29, 2004 1:09 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voice mail problem I am having a problem with getting voice mails, even when the caller hangs up before getting to the recording prompt. If I call my number, even if I hang up the second I get the I'm not in recording, it still generates a voicemail. Is there a way around this? Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Aastra 480e phone ADSI config
The Scripts are in GSM format... locate *.gsm or look in /etc/sounds, /usr/src/asterisk/sounds This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Woolley Sent: Thursday, July 29, 2004 1:55 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Aastra 480e phone ADSI config Couple of things 1) My phone seemed to either be pre-programmed with the Comedian mail scripts or asterisk did not need to be patched for my phone to have functional Comedian vmail access screens. I have however noticed that a number of the Comedian vmail screens are incomplete (saying things such as Options Menu Not Done. I assume the Comedian vmail ADSI scripts are burned into the phone (by manufacturer) or something similar because I have not been able to find any Comedian scripts within Asterisk source. I wonder if there are more current Comedian vmail ADSI scripts available? 2) The Asterisk wiki is noticeably absent in any ADSI information. I would like enjoy working on these ADSI scripts with you (and others) for the 480e. Maybe we can get these added to the Asterisk CVS tree (maybe something like /etc/asterisk/aastra480e.adsi). Unfortunately, my supplier really did not have any documentation other that what was on their web site (very little). Let me know. -- Steve Woolley IT Manager ADS Telecom, Inc. 59 Skyline Drive Suite 1250 Lake Mary, Florida 32746 Phone: (407)682-6226 x1110 Fax:(407)682-3455 IAXtel: (700)682-6226 x1110 Cell: (321)229-5311 [EMAIL PROTECTED] www.adstelecom.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Keding Sent: Thursday, July 29, 2004 11:04 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Aastra 480e phone ADSI config Thanks for your input I managed to get what you suggested going last night and it works fine. I also got a note from Sayson on Commedian Mail. Once I did what they suggested, I got a full Voice mail interface on my phone. Pretty cool! From Sayson If you are using ADSI phones and trying to access Commedian Mail, CM tries to do an FDM download (it's own ADSI script) to the phone first. If you don't change the FDN and secur. code in the CM app, you will get and error. In the app_voicemail.c file (for me it was located in /usr/src/asterisk/apps ), the adsi_begin_download is evoked as follows: if (adsi_begin_download(chan, addesc, adapp, adsec, adver)) Where addapp (fdn) and adsec are hardcoded as follows: static char *adapp = CoMa; static char *adsec = _AST; They need to be changed to the correct FDN and Security numbers for the slot you wish to download. So you don't overwrite your own programming, use slot 3 or four. (I used slot 3 for my sayson 480e) static char *adapp = \xFB\xC6\x45\x0C static char *adsec = \x9B\x60\x94\x30 Then recompile and press the Vmail button on your phone. It should automatically download the script and then you have a bunch of new buttons to play with! On a side note, I am tring to enhance the ADSI programing in the orignal script. Did your supplier give you any help with additional commands etc. I have not found any docs. So far. Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Woolley Sent: Thursday, July 29, 2004 9:14 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Aastra 480e phone ADSI config There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): snip 3. I only had to tune the SENDDTMF 8500 values to properly send it to the right voicemain extention 4. Added the following to my /etc/asterisk/extensions.conf file in a local only context so that the phone could only be programmed locally: [adsi-program] exten = 9666,1,Authenticate(1234) exten = 9666,2,ADSIProg(asterisk.adsi) exten = 9666,3,Hangup 5. Called extension 9666 from the 480e. It asks for my password and then I am off to the races. Good luck! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Keding Sent: Wednesday, July 28, 2004 5:44 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Aastra 480e phone ADSI config Greetings Does anyone have a ADSI config file for an Astra (Sayson) 480e phone. I am using the sample asterisk.adsi file but
RE: [Asterisk-Users] Voice mail problem
; Should the email contain the voicemail as an attachment attach=no ; Maximum length of a voicemail message in seconds ;maxmessage=180 ; Minimum length of a voicemail message in seconds minmessage=5 ; Maximum length of greetings in seconds ;maxgreet=60 ; How many miliseconds to skip forward/back when rew/ff in message playback skipms=3000 ; How many seconds of silence before we end the recording maxsilence=10 ; Silence threshold (what we consider silence, the lower, the more sensitive) silencethreshold=128 This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luke Catranis Sent: Thursday, July 29, 2004 2:43 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Voice mail problem Set the minimum voicemail length in the voicemail.conf file. This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Keding Sent: Thursday, July 29, 2004 1:09 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voice mail problem I am having a problem with getting voice mails, even when the caller hangs up before getting to the recording prompt. If I call my number, even if I hang up the second I get the I'm not in recording, it still generates a voicemail. Is there a way around this? Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MS SQL Free TDS
Help! I've been using mysql for cdr storage, I need to switch to MS SQL. I must be stupid or something but I cannot figure out how to setup the cdr_tds. I have FreeTDS configured properly, but my unixodbc is not working properly either... I'd be happy with either solution, but I'm in need of assistance. Luke Catranis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unix ODBC Segmentation Fault
I've successfully connected to my SQL server with tsql (freetds) and isql (unixodbc). I've followed the wiki, and everytime I try and run * shutdowns with a segmentation fault. Any help? Luke Catranis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
