[asterisk-users] Re: Call center manager for Asterisk (Release 0.3)

just to let you know that i've started a mailing list on sourceforge [EMAIL PROTECTED] You can subscribe here https://lists.sourceforge.net/lists/listinfo/ccmanager-users Other news regarding ccmanager will be posted on this mailing list, i invite interested people to subscribe. Thanks On 3/1

Re: [asterisk-users] voip-info.org status update

shadowym wrote: If I can't be confident enough in an important source of information like this then I can't be confident enough to provide an Asterisk solution to businesses. That's the way I see it. Yea, it's a wiki but it's the best source of info out there. Suggestion: switch to somet

Re: [asterisk-users] voip-info.org status update

On 3/15/07, OCOSA List Acct. <[EMAIL PROTECTED]> wrote: Hi All, and if you all depend on James' site so much then you need to donate some time or contact him about getting a mirror. The so called new site Google didn't go down, and if you had bothered searching the archives of this list you

Re: [asterisk-users] voip-info.org status update

Hi All, Personally all of you who are complaining you need to stop becoming part of the problem and become part of the solution. Everyone makes mistakes and if you all depend on James' site so much then you need to donate some time or contact him about getting a mirror. The so called new site

[asterisk-users] DECT to SIP gateway experiences

G'day. I hope this isn't off-topic for the list. I am looking at an Asterisk setup that includes cordless phones. The three choices I can see, at this stage, are: * wifi phones * an ATA and a cordless analog phone * a DECT to SIP basestation The various wifi phone options don't grab us as s

[asterisk-users] AgentCallBackLogin Help!

Hi List! Im using (or trying to) use AgentCallBackLogin() to have * find roaming users, here is a diagram. Server A (Hq) Server B(Branch Site) Server C (Branch Site) All my que users are on Server A, I have Server B/C dial a extension to call AgentCallBackLog

Re: [asterisk-users] voip-info.org status update

shadowym wrote: > Hard to expect the business community to take Asterisk seriously when > this sort of stuff happens IMHO. I can't understand how 3 of 4 hard > drives could just suddenly fail simultaneously. There must be more too > it. No UPS? Someone spilled their coffee into it? Something! >

Re: [asterisk-users] While the VoIP-Info.org site is down...

Patrick May wrote: > On Wed, Mar 14, 2007 at 10:05:20PM -0400, Matt wrote: >> Yikes.. you'd think a server would be running RAID. >> >> At any rate.. Please feel free to visit http://www.voip-wiki.us >> >> I have set this up to be able to hold information for the Asterisk >> community. I will also

RE: [asterisk-users] voip-info.org status update

Shadowym, Errr... Please explain to me what on earth (or even your own planet) voip-info.org has do do with the quality/reliability/etc of Asterisk? You do realise that a little company called Digium is the developer of Asterisk, and *someone else entirely* runs the server that the abovely-ment

Re: [asterisk-users] voip-info.org status update

On Mar 15, 2007, at 12:32 AM, shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. It is drifting off topic, bu

RE: [asterisk-users] voip-info.org status update

Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! Either way, it's amateu

[asterisk-users] DNIS/DNID

Hi i have an asterisk pbx with E1 port connected to another PBX. Im trying to send the DNID/DNIS to the PBX here's my dialplan exten => 888111,1,Dial(ZAP/g2) exten => 888111,n,Hangup() The PBX just get the number 2 as it's DNIS when i change it to ZAP/1 or ZAP/g1 the PBX get the number 1

Re: [asterisk-users] While the VoIP-Info.org site is down...

On Wed, Mar 14, 2007 at 10:05:20PM -0400, Matt wrote: > Yikes.. you'd think a server would be running RAID. > > At any rate.. Please feel free to visit http://www.voip-wiki.us > > I have set this up to be able to hold information for the Asterisk > community. I will also gladly allow others to m

Re: [asterisk-users] Packetization Rate

Oh? I thought the default was 30ms. Ok, if it's ALREADY at 20ms, then I have nothing to modify. On 3/14/07, Luki <[EMAIL PROTECTED]> wrote: > Obviously somewhere in the asterisk code 30ms must be coded... is it set in > just one place, and if so can I set that to 20ms? The default is 20 ms f

Re: [asterisk-users] Cisco 7912

Tom Lynn wrote: Do they appear to have failed as a result of Daylight Savings time? DST for 7905/7912 are set inside the lddefault/gkdefault - or the individual config file (ldMAC / gkMAC), but can't be set in advance like 7940/7960. DST is not the reason here...

Re: [asterisk-users] Cisco 7912

Also prior to the crash the phones were loosing registration every hour on the hour like clockwork and it was just these three. I dont know if it might just be a flaw in the hardware on these phones or if they were just used to hard but its a little strange that these three would just crash while

Re: [asterisk-users] Cisco 7912

Thats what i would have though but we have two others that are working just fine and they were all identical in firmware and all manually configured. If was on sunday but i dont think it was because of DST. On 3/14/07, Tom Lynn <[EMAIL PROTECTED]> wrote: Do they appear to have failed as a resul

Re: [asterisk-users] Linksys not Ringing

shouldn't there be an answer in there somewhere?... like... No... you can (and probably should) Dial() an extension before answering the incoming call. Do a sip debug and see if the Sipura is getting the INVITE message (and responding with an ACK), and if it sends back a RINGING message. Somet

Re: [asterisk-users] TDM-400, Polycom SIP phones, and echo problems

Ira wrote: > At 02:47 PM 3/14/2007, you wrote: >> - How is the HPEC in dealing with this echo? > > Well, I don't have much of a system, but since I installed the HPEC > there has been no echo on my POTs lines. That makes three positives. HPEC it is, then. I'll let you know how it goes. -Stephen

Re: [asterisk-users] Cisco 7912

Do they appear to have failed as a result of Daylight Savings time? On 3/14/07, Matt Putnam <[EMAIL PROTECTED]> wrote: I didnt have them on tftp files they were all manualy configured. They are not trying to request anything they have the tftp server address but are not requesting any files. It

[asterisk-users] Voip-Wiki Site Information

Community, I have put up www.voip-wiki.us My apologies to our fellow Asteristians outside the us... this was the only easy domain available. At any rate, feel free to populate the database / wiki, and I will be more then happy to have and help others mirror this site so we can have duplicates, He

Re: [asterisk-users] Packetization Rate

Obviously somewhere in the asterisk code 30ms must be coded... is it set in just one place, and if so can I set that to 20ms? The default is 20 ms for most (all?) codecs. It's in rtp.c, where ast_rtp_write() creates a new smoother. --Luki ___ --Bandwi

Re: [asterisk-users] Cisco 7912

I didnt have them on tftp files they were all manualy configured. They are not trying to request anything they have the tftp server address but are not requesting any files. It should start up and look for a vlan but its not even doing that it does nothing when i plug it in just a blank screen and

Re: [asterisk-users] Linksys not Ringing

jason, shouldn't there be an answer in there somewhere?... like... [inbound-sip] exten => 300,1,Wait(1) exten => 300,n,Answer() exten => 300,n,NoOp(${EXTEN}) exten => 300,n,NoOp(${CALLERID}) exten => 300,n,Dial(SIP/300,15) exten => 300,n,VoiceMailMain exten => 300,n,Hangup() dave

Re: [asterisk-users] While the VoIP-Info.org site is down...

Yikes.. you'd think a server would be running RAID. At any rate.. Please feel free to visit http://www.voip-wiki.us I have set this up to be able to hold information for the Asterisk community. I will also gladly allow others to mirror it. It is sitting in a climate controlled data center in C

Re: [asterisk-users] Cisco 7912

Matt Putnam wrote: anything useful any sugestions? Are they requesting anything via TFTP? Do you have the full tftp files ready? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

Re: [asterisk-users] While the VoIP-Info.org site is down...

[Al Bochter wrote on 15/03/2007 12:25 PM]: > So does anyone know when Voip-info.org will be back up? There is a message on the list from James Thompson with the subject "voip-info.org status update" saying it suffered a major hard drive crash and should be back tomorrow. Looking at the headers th

Re: [asterisk-users] Packetization Rate

I am currently using 1.2 and can not upgrade to 1.4 until it becomes stable and we have done much testing with it. Obviously somewhere in the asterisk code 30ms must be coded... is it set in just one place, and if so can I set that to 20ms? On 3/14/07, Dan Austin <[EMAIL PROTECTED]> wrote: Mat

Re: [asterisk-users] Zaptel version for asterisk 1.2.16

Kevin P. Fleming wrote: John Novack wrote: In Digium's infinite wisdom, they have seen fir to have version numbers no longer match, and also NOT provide any sort of map to give the rest of us a clue as to what goes with what. Probably to discourage wider use of the product? I must s

Re: [asterisk-users] Re: Asterisknow with video and X-Lite not quite working

benedikt, try putting these (or your version of these) in the sip.conf [general] heading.   it was suggested to me before, that what is general should go in the general section, what is specific to a particular extension should go in the specific extension section.  also, I put a ton of options

Re: [asterisk-users] While the VoIP-Info.org site is down...

I don't know, but, I can put a server for mirror this page. This page is a very good tool. I can put a mirror for this on Europe. regards Al Bochter escribió: So does anyone know when Voip-info.org will be back up? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Em

[asterisk-users] Inbound PSTN CLID irratic with A200

I use Trixbox 2.0 with a Sangoma A200 I also have echo so bought the HPEC and yes it works brilliantly. The problem I have is I used to use Trixbox 1.? with this sam hardware and had a few inbound CLID issues on my UK BT lines, Sangoma support suggested changing the RXGAIN in zapata.conf and it

Re: [asterisk-users] While the VoIP-Info.org site is down...

So does anyone know when Voip-info.org will be back up? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Steve Totaro wrote: Is it wise to use an outage to promote your business, not on the user's list and not multiple times? Put it in your signature or som

Re: [asterisk-users] TDM-400, Polycom SIP phones, and echo problems

At 02:47 PM 3/14/2007, you wrote: - How is the HPEC in dealing with this echo? Well, I don't have much of a system, but since I installed the HPEC there has been no echo on my POTs lines. Ira ___ --Bandwidth and Colocation provided by Easynews.c

RE: [asterisk-users] While the VoIP-Info.org site is down...

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Stephen Bosch > Sent: Wednesday, March 14, 2007 5:51 PM > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [asterisk-users] While the VoIP-

[asterisk-users] Cisco 7912

I have 3 cisco 7912 that all stoped working at the same time on sunday. There is nothing on the display and the menu and hold buttons are lit. Resteing produces the same results the phone dosent respond. Anyone have an idea how to fix this or if it can even be fixed. Ive done some searching online

[asterisk-users] voip-info.org status update

A short status update: Yesterday 3 of the 4 disk drives in the RAID array on the server that hosts voip-info.org failed. The coloprovider is currently working to replace the drives and I'm hoping that the site returns to service soon. Tomorrow is looking most likely. I'd like to thank all thos

[asterisk-users] Current voip-info.org Status

News for everyone on this. Recently found out that this site is down due to a hardware failure. Hard Drives in the RAID array failed and currently the problem is being addressed as fast as possible. Hopes are that voip-info.org will be operational again sometime tomorrow afternoon. Hope t

Re: [asterisk-users] what happened to asterisk wiki???

We also would like to offer hosting and/or mirroring until VOIP-INFO hopefully gets back on it's feet. We have MediaWiki setup at www.xmc.com/wiki and will also temp in dns voipwiki.xmc.com direct. It is there if the community wants to use it also. On Wed, 2007-03-14 at 18:26 -0500, Andy Brezins

Re: [asterisk-users] Patton 1400

Hi, I managed to install my patton gateway but i did not succeed to pass the caller id to the sip phone on incoming calls ... instead i see call from 105 ( which is the sip client extension of the patton ) do you know the way to pass the caller id of the caller to the ip phone iso the gw extensio

Re: [asterisk-users] what happened to asterisk wiki???

I work for a conferencing and telecom company and have used voip-info.org more times than I can count for both VoIP and asterisk reasons. We've used asterisk internally and have greatly benefited from it. We're willing to host the Wiki with database dumps available for people to download on a reg

Re: [asterisk-users] what happened to asterisk wiki???

Steve Totaro wrote: > I think Digium should host a wiki (keeping if vendor neutral of course). It escapes me why they haven't done this sooner. It would only help them sell product. > This seems to be the most complete backup of voip-info.org but it is > fairly old http://web.archive.org/web/2005

Re: [asterisk-users] While the VoIP-Info.org site is down...

Davis Sylvester III wrote: > If needed I can put up a wiki for asterisk support, and allow > mirroring. I know we have to start over from ground zero, but it gets > us back online with our life line. Let me know if you guys want me to > proceed. There would be no charge for us hosting the site.

Re: [asterisk-users] While the VoIP-Info.org site is down...

Shane Breen wrote: > Sorry about that. I figured since it is a FREE site it was no biggie. > > Thanks, > > Shane Breen Well, Shane, nothing is ever really FREE, and when someone offers something FREE, there is almost always a catch, and on the Internet that catch tends to come in the form of SPA

Re: [asterisk-users] While the VoIP-Info.org site is down...

Shane Breen wrote: Sorry about that. I figured since it is a FREE site it was no biggie. Thanks, Shane Breen *The Telecom Directory * Office: 404-797-6633 [EMAIL PROTECTED] * www.TheTelecomDirectory.com The Telecom Directory is the only complete A to Z interactive website for all telecommunic

Re: [asterisk-users] Polycom call parking feature and Asterisk call parking

Hi Stephen - I want to make parking calls easier for my hard-working users. Is there a way to make the Polycom call park feature work with Asterisk? In case it just works out of the box, I haven't tried it yet; but the "call park" feature isn't enabled on the Polycom phones by default. If you

RE: [asterisk-users] Packetization Rate

Matt wrote: > To my knowledge, Asterisk's packetization rate is hard > coded at 30ms.  If I wanted to, where in the code could > I go to change it to 20ms.   Is there anything bad that > might happen if I change it (asterisk related)? You don't mention what version you are using, but 1.4 does su

Re: [asterisk-users] While the VoIP-Info.org site is down...

Sorry about that. I figured since it is a FREE site it was no biggie. Thanks, Shane Breen *The Telecom Directory * Office: 404-797-6633 [EMAIL PROTECTED] * www.TheTelecomDirectory.com The Telecom Directory is the only complete A to Z interactive website for all telecommunication related inform

Re: [asterisk-users] what happened to asterisk wiki???

On 3/14/07, Stephen Bosch <[EMAIL PROTECTED]> wrote: It would be nice to make this information available in a friendlier medium: - a Mediawiki-based wiki engine; - allowing mirroring - Keep it focused just on Asterisk, rather than the entire field Well put. I made the (unfortunate) decision

Re: [asterisk-users] [G.729] Input Gain

Victor Mateevitsi wrote: > Can I configure the input gain of a context on sip.conf or maybe can I > configure the input gain of g.729 encoded streams. In zapata.conf (or > misdn.conf) I know there is the above option (rxgain and txgain). No. Asterisk does not provide any facilities to adjust volum

Re: [asterisk-users] Zaptel version for asterisk 1.2.16

John Novack wrote: > In Digium's infinite wisdom, they have seen fir to have version numbers > no longer match, and also NOT provide any sort of map to give the rest > of us a clue as to what goes with what. > > Probably to discourage wider use of the product? I must say that is an excellent atti

RE: [asterisk-users] While the VoIP-Info.org site is down...

Is it wise to use an outage to promote your business, not on the user's list and not multiple times? Put it in your signature or something ;-) Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PRO

RE: [asterisk-users] what happened to asterisk wiki???

I think Digium should host a wiki (keeping if vendor neutral of course). This seems to be the most complete backup of voip-info.org but it is fairly old http://web.archive.org/web/20051013074214/voip-info.org/wiki/ unless someone else spidered it more completely and recently. Even if VoIP-info

Re: [asterisk-users] TDM-400, Polycom SIP phones, and echo problems

On Mar 12, 2007, at 7:12 PM, Stephen Bosch wrote: Hi: I am working on a new system with a TDM-400P card with three FXO modules and one FXS module. The system has been in place for a week. Users are complaining of echo problems. I have noticed this echo myself. It varies in severity. It is s

Re: [asterisk-users] Polycom call parking feature and Asterisk call parking

Mojo with Horan & Company, LLC wrote: > IIRC, you need an extension named 'callpark' in your extensions.conf > that calls the ParkAndAnnounce application. > > This should get you started: > > exten => > callpark,1,ParkAndAnnounce(PARKED|600|Local/4${BRIDGEPEER:5:[EMAIL > PROTECTED]|incoming,s,1

[asterisk-users] While the VoIP-Info.org site is down...

Feel free to use: http://www.thetelecomdirectory.com/forum If you register your company here as well: http://www.thetelecomdirectory.com You will be able to upload white papers, list your company in our directory, release press releases all for FREE. Here is where you do all of the above: h

[asterisk-users] While the VoIP-Info.org site is down...

Feel free to use: http://www.thetelecomdirectory.com/forum If you register your company here as well: http://www.thetelecomdirectory.com You will be able to upload white papers, list your company in our directory, release press releases all for FREE. Here is where you do all of the above: h

Re: [asterisk-users] what happened to asterisk wiki???

Thanks for the pointers, Steve. In general terms, it has always made me uncomfortable that the bulk of Asterisk "documentation" is on a wiki operated by some rarely named person only peripherally related to the project. It would be nice to make this information available in a friendlier medium:

Re: [asterisk-users] RE: what happened to asterisk wiki???

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 JR Richardson wrote: > A friend of mine was on the site yesterday, late morning, when he > refreshed his screen, a banner came across the web page "VOIP SUCKS" > and then the site was no longer available. I'm pretty sure the site > was compromised by

Re: [asterisk-users] IAX2 and Faxing

Davis Sylvester III wrote: Thanks for the quick response. But how do other VoIP Service Providers like vonage provide fax capabilities? Inevitably what I say here is going to be countered from some Vonage customer saying, "You're wrong! It always has worked for me." Vonage even advertise

Re: [asterisk-users] IAX2 and Faxing

Lee Howard wrote: Davis Sylvester III wrote: Can someone tell me how to get faxing to work with an IAX2 remote client? Wish upon a falling star, throw a penny into a fountain, find a four-leaf clover, and pat your head and rub your stomach all at the same time. :-) http://hylafax.source

[asterisk-users] Packetization Rate

To my knowledge, Asterisk's packetization rate is hard coded at 30ms. If I wanted to, where in the code could I go to change it to 20ms. Is there anything bad that might happen if I change it (asterisk related)? ___ --Bandwidth and Colocation provided

[asterisk-users] RE: what happened to asterisk wiki???

A friend of mine was on the site yesterday, late morning, when he refreshed his screen, a banner came across the web page "VOIP SUCKS" and then the site was no longer available. I'm pretty sure the site was compromised by some hacker trying to prove a point or make a statement. Not to throw stin

Re: [asterisk-users] IAX2 and Faxing

Davis Sylvester III wrote: Can someone tell me how to get faxing to work with an IAX2 remote client? Wish upon a falling star, throw a penny into a fountain, find a four-leaf clover, and pat your head and rub your stomach all at the same time. :-) http://hylafax.sourceforge.net/docs/fax-

[asterisk-users] IAX2 and Faxing

Can someone tell me how to get faxing to work with an IAX2 remote client? I have two customers both are connected via Digium's ATA's S101I both can place and receive calls without any problems. However when they connect there line to a fax machine it never connects to the remote fax machine.

Re: [asterisk-users] Zaptel version for asterisk 1.2.16

Wilson Pickett wrote: I'm used to seeing the same versioning (maybe I've been gone too long) Is zaptel 1.2.15 the right one for asterisk 1.2.16 ? In Digium's infinite wisdom, they have seen fir to have version numbers no longer match, and also NOT provide any sort of map to give the rest of

[asterisk-users] Sped up recordings with 1.4.1

Normally I manage to figure stuff out on my own, but this one is driving me absolutely bonkers... I'm working with an Asterisk box at work, running 1.4.1 on a Core2 Duo machine with kernel 2.6.19.1. There's a quad T1 card (older tor2) in the machine, though this problem happens if I just use ztdu

[asterisk-users] Which SIP method/option to display a short text message ?

Hi, Using SIP methods and options, is there any way for a callee to send the caller a short text message when the call is establishing ? Scenario is : Alice and Bob's SIP phones are registered to an Asterisk server. Alice calls Bob : an INVITE message is sent to Bob's phone Bob is replying : a

[asterisk-users] IAX2 - Congestion

Hy all! Your Asterisk server is return this log : *CLI> -- Executing Dial("Khomp/B0C0", "IAX2/*.*.*.*/9834|30|r") in new stack -- Called *.*.*.*/9834 Mar 14 15:35:40 NOTICE[4212]: chan_iax2.c:2836 auto_congest: Auto-congesting call due to slow response -- IAX2/*.*.*.*:4569-1 is circu

Re: [asterisk-users] Polycom call parking feature and Asterisk call parking

IIRC, you need an extension named 'callpark' in your extensions.conf that calls the ParkAndAnnounce application. This should get you started: exten => callpark,1,ParkAndAnnounce(PARKED|600|Local/4${BRIDGEPEER:5:[EMAIL PROTECTED]|incoming,s,1) in the CLI: Show Application ParkAndAnnounce for

[asterisk-users] Call center manager for Asterisk (Release 0.3)

Hi i just want to let you know that is available a new release of ccmanager. I've added the possibility to import queue_log information in a mysql database and to generate reports using this information. The software is in a beta state and provides this functionality: - users management - call

[asterisk-users] ${EXTEN} is limited to 17 characters under IAX ?

Hi list. We have a problem when dialing over IAX to another Asterisk server: we've setup an extension named 'f19dffb971b93746d73ec46d5f1d4b36c199f48c-g1' in a specific context (its large because it needs to be unique). I've read in past discussions on asterisk-dev list that the extension length

RE: [asterisk-users] Zaptel version for asterisk 1.2.16

From: "Wilson Pickett" <[EMAIL PROTECTED]> Date: Wed, 14 Mar 2007 15:18:35 +0100 I'm used to seeing the same versioning (maybe I've been gone too long) Is zaptel 1.2.15 the right one for asterisk 1.2.16 ? It works. I've tried some other mixes and they also work. Yuan Liu __

RE: RE: [asterisk-users] Re: Asterisknow with video and X-Lite notquiteworking

Hi, Codec details you need to put only in the sip.conf Biju.V.P -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benedikt Franz Sent: Wednesday, March 14, 2007 3:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: RE: [asteris

[asterisk-users] Linksys not Ringing

I have 2 linksys SIP phones SPA-942 I have a dialplan of exten => 144,1,Wait(1) exten => 144,2,Dial(Sip/phil,20) exten => 144,3,Voicemail([EMAIL PROTECTED],u) The CLI looks like this when I dial 144 -- Executing Wait("IAX2/JASONSERVER-9", "1") in new stack -- Executing Dial("IAX2/JASONSERVE

[asterisk-users] IVR after hangup

I have a rather interesting issue with catching calls which have been hung up by the callee, but not by the caller. I would like those calls to return to an IVR, and it almost works: [incoming] exten => 12345678,1,Goto(testIVR,s,1) [testIVR] exten => s,1,Answer exten => s,n(play),Background(be

RE: [asterisk-users] What happend to voip-info?

On Wed, 14 Mar 2007, Jonathan k. Creasy wrote: I would be willing to mirror it also?. At the risk of sounding like an AOLer, Me Too ... (UK based mirror?) The site is pingable, so I'd suggest it's either crashed in some awkward way and just needs resetting, but you never know... Gordon

[asterisk-users] [G.729] Input Gain

Hello, I have recently buyed the g.729 license from the Digium site and have the following issue: I have two voip providers (SIP). The first uses g.726 and the second g.729. The problem is that the input gain from the second provider is a little lower than the first one. I usually use both, so c

Re: [asterisk-users] Asterisknow with video and X-Lite not quite working

I do not think that this is a specifically *now related issue, but I would also welcome such a mailing list. Regards Original-Nachricht Datum: Wed, 14 Mar 2007 10:22:34 -0500 Von: Pari Nannapaneni <[EMAIL PROTECTED]> An: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING

nivlekch, nice to hear that :) I hope more people can test this. On 3/14/07, nivlekch <[EMAIL PROTECTED]> wrote: nice job moises, the hardwork you and steve put into chan_unicall is remarkable. with a little editing and tweaking, i was able to make the port to 1.4 here in the philippines witho

Re: [asterisk-users] Asterisknow with video and X-Lite not quite working

I request every one to post AsteriskNOW specific questions on the asteriskNOW forums - http://forums.digium.com/ I will talk to our administrator to see if i can get a seperate mailing list created for AsteriskNOW. thanks Pari Pari Nannapaneni GUI Developer Digium I

RE: [asterisk-users] what happened to asterisk wiki???

It is down. Two options, search in Google, click on cached and then click on cached text only at the top. This may also be helpful http://web.archive.org/web/*/http://www.voip-info.org Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB _ From: [EMAIL PROTECTED] [mailto

Re: [asterisk-users] what happened to asterisk wiki???

Rizwan Hisham wrote: Hi im trying access the www.voip-info.org website since yesterday but i cant open it. google search diaplay correct search results but it doesnt open when i click the link. it displays a message about tcp error which says -->"There was a problem

Re: [asterisk-users] What is the best phone to get when using a headset?

Cory, Are the Polycom phones able to detect that the headset is off-hook? We have had problems with 2 different brands of headsets working fine but the phones seem unaware of the headset's hook state. On 3/14/07, Cory Andrews <[EMAIL PROTECTED]> wrote: I like any of the Polycom Soundpoint IP Se

[asterisk-users] what happened to asterisk wiki???

Hi im trying access the www.voip-info.org website since yesterday but i cant open it. google search diaplay correct search results but it doesnt open when i click the link. it displays a message about tcp error which says -->"There was a problem communicating with the server". I dont know what the

Re: [asterisk-users] How many outgoing phone line/voip account do I need?

Kurt Kuo wrote: Hi list, I have an application which has to automatically dial and send out a voice message to 50 different phone numbers at the same time. Does it mean that I need to sign up 50 phone lines or voip accounts in order to achieve this purpose? Is there a provider(voip prefer) who

[asterisk-users] Zaptel version for asterisk 1.2.16

I'm used to seeing the same versioning (maybe I've been gone too long) Is zaptel 1.2.15 the right one for asterisk 1.2.16 ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

RE: [asterisk-users] Manager connection problems

Correction, it was queue weight that caused the crashes not queue_prio. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, March 14, 2007 10:03 AM To: Asterisk Users Mailing List

[asterisk-users] Re: asterisk on mini-itx

Gordon Henderson wrote: I've built several systems based on this motherboard (the 1GHz fanless one) Compressed codecs are fine - as long as you aren't transcoding ;-) I figured I could push 30 non transcoded calls through one, but I've never had the ability to fully test it out. The max. I had

RE: [asterisk-users] Manager connection problems

What version of Asterisk? I had a lot of problems when hitting the manager interface with any real volume with 1.2.x, it is my understanding that 1.4.x has a newer revamped AMI that should be more robust. I haven't tried it yet so I cannot confirm. Another major thing you should check is your

Re: [asterisk-users] Playback 5% Too Fast?

I've had similar behavior on my own IVR. I moved my sound files to a ram disk and all pops and ticks stopped! David Brazier wrote: Hi All I have a problem with IVR scripts which consist mainly of Playback of audio files, driven from an AGI application. There are clicks every few seconds or mo

[asterisk-users] Manager connection problems

I am wondering how many and how often manager connections can be setup and torn down reasonably. here is the scenerio... I have 10 to 20 agents on two queues one with priority over the other I changed this the day before I also implemented a php program that runs every 8 seconds on an aut

RE: [asterisk-users] What is the best phone to get when using a headset?

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Gareth Blades > Sent: Wednesday, March 14, 2007 7:54 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] What is the best phone to get when using a > headset? > > We c

RE: [asterisk-users] Nomination for Coolest App in 2007

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Brad Templeton > Sent: Monday, March 12, 2007 8:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Nomination for Coolest App in 2007 >

RE: [asterisk-users] What is the best phone to get when using a headset?

I like any of the Polycom Soundpoint IP Series phones (IP301, 430, 501, 601, 650) paired with a VXI Tuffset 37 with Quick Disconnect Cord. IMO VXI makes the best headsets out there for the money, and they are quite inexpensive. You can find more info on VXI at www.vxicorp.com Cory Andrews

[asterisk-users] Autoprovisioning ST2030S

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Anyone tried autoprovisioning for a Thomson ST2030S phone? I did so and are quiet happy with it except for one thing: DTMF settings Whad parameter in the provisioning files do I have to set to transmit DTMF via SIP INFO? I accomplished Inband an

[asterisk-users] What is the best phone to get when using a headset?

We currently have the Grandstream GXP-2000 phones which generally work very well except that we cannot get find a headset which works reliably with them. Either the sound quality is poor or the other party has difficulty in hearing us. We therefore want to get a couple of different phones and head

RE: [asterisk-users] T1 Integrator Birch

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of John Schmerold > Sent: Wednesday, March 14, 2007 1:35 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] T1 Integrator Birch > > I'm thinking about replacing my Birch

Re: [asterisk-users] Earliest dial tone, after boot up.

Thanks. Very reassuring. It really must be too early. joe a. Tzafrir Cohen<[EMAIL PROTECTED]> Wrote: 3/14/2007 8:23 AM: > On Wed, Mar 14, 2007 at 07:53:58AM -0400, joe acquisto wrote: >> New system install. >> >> At what point, in bootup, should I be able to get a dial tone on the >> phone

RE: [asterisk-users] What happend to voip-info?

I would be willing to mirror it also…. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, March 14, 2007 9:39 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] What happend to voi

  1   2   >