Re: [Asterisk-Users] RE: [Asterisk-Dev] SetVarCDR

You have this: ? exten => _X.,1,Dial(SIP/mysipdevice,30) And you are getting _X. In your cdr as DST? That is fux0red. -Matthew > From: "William M. Sandiford" <[EMAIL PROTECTED]> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > Date: Sat, 12 Mar 2005 23:19:43 -0500 > To:

RE: [Asterisk-Users] playing "invalid" to an internal user

> [invalid] > exten => _.,1,Answer > exten => _.,2,Playback(pbx-invalid) > exten => _.,3,Hangup() > > asterisk is playing "invalid" message twice, WHY? > > -- Executing Answer("SIP/11-df84", "") in new stack > -- Executing Playback("SIP/11-df84", "pbx-invalid") in new stack > -- Playing '

RE: [Asterisk-Users] ASTCC or should I use something elsefor different rates, depending on the calling card?

> I use the latest ASTCC version, but I do not see how you can > use different cards / different prices. Can you explain that for me, > please? The cards have a field called "markup", that would mark up the price specified in the routes table by a certain percentage. So it's a card-specific, non-r

Re: [Asterisk-Users] VoIPJet and g.711

Me too facing same problem . some times it work some times not. i am using ilbc . - Original Message - From: "Justin Richards" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, March 12, 2005 8:36 PM Subject: Re: [Asterisk-Users] VoIPJet a

[Asterisk-Users] vars for transfered calls

Hi when i try and transfer a call to another party the setvars I have placed in sip.conf do not get passed to my dialplan at all is there a work around for this so I can use them in my dialplan but when i place a normal outbound call from that device the vars get passwd fine I am passing an areaco

Re: [Asterisk-Users] VoIPJet and g.711

well now i'm confused, because its working again. so i'm not sure if it really had a problem or not. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

Re: DISREGARD!![Asterisk-Users] Broadvoice outgoing problems

[EMAIL PROTECTED] (Brian Dingman) writes: > I doubt that was the problem. I would be interested in hearing what > else you did besides that to get it working. If he had a bad entry in the /etc/hosts file that could have been a problem. The hosts files does require periodic maintenance. Automati

RE: [Asterisk-Users] Vonage a provider?

Thanks. I am not jilted by any means. But I did not want to cause any problems with the established community. Who knows, I might stick around long enough to stop being such a noob! :) BTW, to ALL that responded to my initial question, Thanks a bunch! Frank... -Original Message- Fro

[Asterisk-Users] RE: [Asterisk-Dev] SetVarCDR

I don't know...now I have a _X. in my CDR. -Original Message- From: Matthew Boehm [mailto:[EMAIL PROTECTED] Sent: Saturday, March 12, 2005 8:05 PM To: Asterisk Developers Mailing List Subject: Re: [Asterisk-Dev] SetVarCDR You must have some fux0red config 'cause using _X. works fine h

Re: [Asterisk-Users] VoIPJet and g.711

I am having problem with voipjet and g.711 (ulaw) as well. I tried ilbc with no luck. basically my outbound call connects, i can hear them talk, but they can't hear me. i am not getting errors in console with either ulaw or ilbc, just no audio to the called party. it worked great yesterday, and

Re: DISREGARD!![Asterisk-Users] Broadvoice outgoing problems

I doubt that was the problem. I would be interested in hearing what else you did besides that to get it working. On Sat, 12 Mar 2005 17:46:58 -0500, Jay Carter <[EMAIL PROTECTED]> wrote: > ... I just tried again after removing my hosts file entry (again) and > outbound is now working! I had taken

[Asterisk-Users] chat line

Anyone done a chat line app? -- James Taylor MetroTel 3505 Summerihll 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

[Asterisk-Users] Hang on "making progrogress passing" when dialing out

I am getting the following on dial-out via Sipphone to a 1-800 number (numbers obscured): - == Spawn extension (macro-sipphone, s, 3) exited non-zero on 'SIP/eric-9546' in macro 'sipphone' == Spawn extension (default, 1747xxx, 1) exited non-ze

[Asterisk-Users] Voice Based Bulletin Board.

First, I did my noob homework and found a thread where this had been discussed to some degree in a thread titled "asterisk based bbs" in 2004. I've got a question or two that that thread didn't address. I want to set up a voice based BBS. This will barely stray from the text based bulletin boar

[Asterisk-Users] playing "invalid" to an internal user

* is playing "invalid" message twice. I have: [internal] include => outgoing ; include the outgoing context include => voicemail ... include => invalid [invalid] exten => _.,1,Answer exten => _.,2,Playback(pbx-invalid) exten => _.,3,Hangup() asterisk is playing "invalid" message twice, WHY? --

Re: [Asterisk-Users] Realtime does not work yet, ...

You may not have most recent CVS. You should have this in your sip.conf: rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis. rtnoupdate=yes ; do not send the update request over realtime.

[Asterisk-Users] Does zapateller work in Australia?

as asked. -- Howard. LANNet Computing Associates; Your Linux people -- "When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft." ---

[Asterisk-Users] Question on phones with asterisk

Hi, I've read the wiki... but would like some input from users here (not implying that wiki writers aren't users). I'm looking for a cheap (sub 60$) wired phone, or ATA device.. can anyone recommend one (or several), and possibly a source? ___ Asterisk-

[Asterisk-Users] Problem with ability to dial out when a channel is used from an external equipment in a point to multi point configuration

I have grouped two capi controllers using * as the outgoing msn. when the first two channels are busy normally and I try to use a third channel the channels from the second controller are used. But when a channel is occupied by the capisuite fax and we need a third channel asterisk responds with

[Asterisk-Users] Re: Signaling on PRI channels

> > On Sat, 12 Mar 2005, Laurent Tostain wrote: > >> Hi, >> >> We did an interconnection with our carrier few days ago. But, I noticed >> that there was a signaling problem on our trnuk. In fact, Asterisk indicates >> that the call is answered when we received ALTERTING message from our >> c

Re: [Asterisk-Users] checking active SIP members of a queue?

Anton Krall wrote: You mean that if on a certain queue, your agents are using SIP or IAX phones, and you want to do a check so that when a cllers tryies to get into the queue, if no agent is logged in, do something else with the caller instead of hanging up? Actually, I think he wants to go one ste

RE: [Asterisk-Users] Advanced conference features, meetme2?

Title: RE: [Asterisk-Users] Advanced conference features, meetme2? I’d like to know this to, I’m prepared to kick in $50 to start off a bounty if no one else has done so. Cheers, Dean _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Beha

[Asterisk-Users] Re: Asterisk with Skype

Never mind.. answered my own question looks like their is a bounty on the ability to do this :P On Sat, 12 Mar 2005 18:28:00 -0500, Matt <[EMAIL PROTECTED]> wrote: > Hi, > Does anyone know if it's possible to hook an asterisk PBX up to skype? > And if so, any config examples? > _

RE: [Asterisk-Users] Advanced conference features, meetme2?

Conference lock and member name been recorded and announced when they get in and out of a conference is already available. Check the wiki and look for meetme, you will see they have some parametes like m,a,s that will help you control this features. Anton Krall

RE: [Asterisk-Users] checking active SIP members of a queue?

You mean that if on a certain queue, your agents are using SIP or IAX phones, and you want to do a check so that when a cllers tryies to get into the queue, if no agent is logged in, do something else with the caller instead of hanging up? Is that what you are trying to do? -Original Messag

[Asterisk-Users] Asterisk with Skype

Hi, Does anyone know if it's possible to hook an asterisk PBX up to skype? And if so, any config examples? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update op

[Asterisk-Users] Sjphone call quality: free phone vs. commercial

Hello, Could anyone say if there is any significant boost in voice quality with the commercial SJPhone (payed for) vs. their free version? Also, any reports on SIPPS free vs commercial? It is worth to buy the licenced SJPhone/SIPPS to increase the voice quality (they want $95 for it, pretty exp

[Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue 88

These allow and disallow work with NuFone for me disallow=all allow=ulaw allow=alaw allow=gsm Jeff Message: 11 Date: Fri, 11 Mar 2005 11:15:51 +0100 From: "Edward Banfa" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] NuFone Configuration [problem] To: "'Asterisk Users Mailing List - Non-Commerci

[Asterisk-Users] after *-1.0.6 upgrade error: vm_execmain: Unable to read password

After upgrading to asterisk 1.0.6 on Gentoo when I try to log-in to check the voice mail I get: app_voicemail.c:3389 vm_execmain: Unable to read password -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/

Re: [Asterisk-Users] CNAM for Asterisk

Kevin Nguyen wrote: Thank you for your help. Kevin N. I already replied to your first message with a great deal of information; did you not receive it? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listin

Re: [Asterisk-Users] OT: Best DB

On Fri, Mar 11, 2005 at 04:25:59PM -0500, David Filion wrote: > > Maybe I miss read, but doesn't the licensing of the newer releases of > MySQL require companies to purchase a license? No. The license is GPL. Originally it was LGPL for the client libraries but this got changed recently. So yo

[Asterisk-Users] Video Conference

Hi All;     I know Asterisk can support video calls over sip or h323 but I need to know if it can be used in Video Conferencing? Can I use "meet me" for that purpose?       Regards Mohammad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.c

[Asterisk-Users] CNAM for Asterisk

I am working for Accudata Technologies. We provide CNAM via http request or raw TCP/IP connection. We would like to provide the same capability to Asterisk. I installed Asterisk on Fedora 2.0 and did reading about AGI and AGI application at http://www.voip-info.org/tiki-index.php?page=Asterisk+AG

RE: DISREGARD!![Asterisk-Users] Broadvoice outgoing problems

... I just tried again after removing my hosts file entry (again) and outbound is now working! I had taken it out before, but I think I was getting a different error at the time. Sometimes it seems like asking for help is itself a cure! Thanks anyway! JDC

Re: [Asterisk-Users] OT: Best DB

On Fri, Mar 11, 2005 at 01:56:47PM -0500, Giudice, Salvatore wrote: > As for the "production recommendation" you refer to, I would > respectufully disagree. If you are an enterprise comapny looking to > deploy an open-source DB, you will pick the one that has an established > support company to co

RE: [Asterisk-Users] what is best free softphone.

Pulver.communicator (FWD) ? > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > FCG ZHAO Zigang > Sent: Freitag, 11. MÃrz 2005 06:17 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] what is best free softphone. > > > > I use xli

[Asterisk-Users] Broadvoice outgoing problems

Hello All, I'm just getting into *, and trying to use a Broadvoice account. It works inbound, but Outbound fails no matter what sip.conf parameters I try. From the recent posts here I think it could be: A bad CVS release - I will try to download and build from a new one Broadvoice

Re: [Asterisk-Users] Voicemail to UK mobile

Hi Can u let me know how you are doing this, please Iqbal On 3/12/2005, "Rafal Kaniewski" <[EMAIL PROTECTED]> wrote: >What issues/options are there when forwarding voicemail to uk mobile >voicemail? > >ta > > >Rafal Kaniewski > > >-- >No virus found in this outgoing message. >Checked by AV

Re: [Asterisk-Users] Signaling on PRI channels

On Sat, 12 Mar 2005, Laurent Tostain wrote: > Hi, > > We did an interconnection with our carrier few days ago. But, I noticed > that there was a signaling problem on our trnuk. In fact, Asterisk indicates > that the call is answered when we received ALTERTING message from our > carrier. This

Re: [Asterisk-Users] Advanced conference features, meetme2?

On Sat, 12 Mar 2005, C. Tomlinson wrote: > I have been playing about with meetme as a conference bridge, and find it > lacking in some features which I believe are out their somewhere. > > Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design > it looks like a plan happened

[Asterisk-Users] InterVivo and MusicOnHold()

Hi All, I've been trying for a while to get * to play MusicOnHold with my SIP connection. I can hear it when I call a test extension from my local X-Lite phone, but when I dial in via InterVivo, I just hear silence. I have a Gentoo box with kernel 2.4.28-gentoo. I have no sound card or speakers

[Asterisk-Users] Tracking/Billing Incoming & Outgoing Minutes?

Hello, I found a way to do a callback service using call files however, only call leg B gets recorded in the CDR so I would only be able to accurately bill for one leg? Does anyone have any suggestions on how to get numbers for both call legs? Or a way to bridge two seperate SIP calls and have bo

[Asterisk-Users] Zapping around

Dear list, I am trying to learn how to use Zap-things in Asterisk. While loading Asterisk verbosely I get this error: [chan_zap.so]Warning, flexibel rate not heavily tested! => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Mar 12 17:19:01 WARNING[5563]: chan_zap.c:76

Re: [Asterisk-Users] OT: Best DB

On March 10, 2005 07:14 pm, Giudice, Salvatore wrote: > I vote for MySQL. PostgreSQL is fine, but MySQL handles much better > under extreme load. MySQL is also usually touted as being generally You *gotta* be kidding me. MySQL can't hold a candle to PostgreSQL for high load, high volume or compl

RE: [Asterisk-Users] No ringback over IAX - LiveVoip

Title: RE: [Asterisk-Users] No ringback over IAX - LiveVoip Excellent thing to hear.  I am glad there are positives on this site as well as teh warnings.   Now to get the ringback issue resolved   Using m switch to get MOH is OK but there has to be alogical reason this is occuring adn

Re: [Asterisk-Users] ASTCC and NuFone billing is different!!

NuFone service bills in industry standard billing increments, which are: six (6) seconds for the US48, sixty (60) seconds to Mexico and fifteen (15) seconds to the remainder of the world. From: http://www.nufone.net/tac.html ___ Asterisk-Users mailing li

Re: [Asterisk-Users] ASTCC and NuFone billing is different!!

On March 11, 2005 11:13 am, Ronald Wiplinger wrote: > I have ASTCC installed, and compare it with NuFone, however, I find that > the billing of NuFone is always a few secondes more (6 to 24 seconds) > > Does anybody has an explanation / solution for it? Have you emailled [EMAIL PROTECTED] and aske

Re: [Asterisk-Users] VoipJet Terms of Service

On March 11, 2005 10:49 am, John Goerzen wrote: > * Then there's the NDA: People are specifically prohibited from >telling anyone that they use VoipJet, including "end users". Also, >we can't tell people what we pay for it, even though the prices are >right there on their website. >

Re: [Asterisk-Users] Re: Incoming echo cancel

On March 11, 2005 04:51 pm, Eric Wieling wrote: > echocancel=yes > echotraining=yes or 600 or 800 I absolutely *despise* echotraining. A half second (or in your case 8/10 of a second) delay before hearing anything is unacceptable in almost all situations. Maybe if you've got a physical disabil

[Asterisk-Users] Where to download the asterisk-oh323?

Dear ALL: Where can I find the oh323 module on CVS or anywhere? I want to implement the SIP(ser) to Asterisk to H323(gnugk). Thank you. Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailm

Re: [Asterisk-Users] digium card

On March 11, 2005 10:49 pm, M.N.A.Smadi wrote: > does any body know what are the physical dimension of a digium care > 400pm for example? It's a half-length PCI card. It's maybe an inch or so longer than the PCI slot itself. It is full-height though (i.e. the circuit board is as tall as the ba

Re: [Asterisk-Users] EADS6550 and asterisk - echo on PSTN call

administrator tootai wrote: If you're telling that I have to pass parameters to module when loading, I checked with modinfo wctdm (at office I have head version) and options I have are those: [EMAIL PROTECTED] asterisk]# /sbin/modinfo -p wctdm debug int loopcurrent int robust int _opermode int

Re: [Asterisk-Users] No ringback over IAX - LiveVoip

- Original Message - If asterisk is going to be modified to support LiveVoip expectations, then yet another Dial option would need to be implemented to force ringback to occur as an audio stream for iax only. Guess one could open a bug report for both LiveVoip and Asterisk, but not likely

Re: [Asterisk-Users] Vonage a provider?

Don't sweat it - it just so happens that you came into the fray just moments after this list has had a big long drawn out argument about newbie etiquete (sp?). You've just managed to get caught in the middle. Don't let it be indicative of how everyone feels, and don't let it scare you away...

RE: [Asterisk-Users] Parked Call

Hello Justin, dtmfmode should be inband for broadvoice either way because that's what they support. Now for the extensions.conf do you have: exten => ,1,Dial([SIP|IAX2|..]/, , t) <-- t for transfers? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J

Re: [Asterisk-Users] Parked Call

Thanks for the reply. Here is my sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default disallow=all allow=ulaw dtmfmode=rfc2833 register => [EMAIL PROTECTED]:password:[EMAIL PROTECTED]/ canreinvite = no [sip.broadvoice.com

[Asterisk-Users] ANNOUNCEMENT: Updates for app_cbmysql and MeetMe2 gui (out of tree modules)

New PHP web interface at www.fitawi.com/Asterisk There is also a sample cbmysql config file and the database Tables description. -New in the interface: Added schedule conflict detection for Add and Update conference functions. -New in app_cbmysql: No chang

[Asterisk-Users] How do I pick up a trailing number in extensions.conf?

Hi, my SIP provider sends me all the numbers that are dialed, i.e. when a number is appended to the phone number proper, it gets appended to the incoming number, like (my number is 0123456789, and I append a "5"): To: ;tag=as1174b008 Question is, how can I use the trailing number in extensions.

[Asterisk-Users] Advanced conference features, meetme2?

Hi, I have been playing about with meetme as a conference bridge, and find it lacking in some features which I believe are out their somewhere. Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design it looks like a plan happened, but where is meetme2 at now? Things like rec

Re: [Asterisk-Users] asterisk and Broadvoice Outgoing Again :(

Thanks, I have that already in my /etc/hosts But it's still not working :( On Saturday 12 March 2005 03:48, Rich Adamson wrote: > For everyone that's trying to get BV to work, you'all might want to > edit your /etc/hosts file and insert something like: > > 147.135.8.128 sip.broadvoice.com > > Th

Re: [Asterisk-Users] asterisk and Broadvoice Outgoing Again :(

It doesn't try to authenticate the incoming call. On Friday 11 March 2005 03:56, Randy Johnson wrote: > What does insecure=very do? > > Dan Weber wrote: > >> Here is my sip.conf: > >> === > >> register => > >> [EMAIL PROTECTED]::[EMAIL

Re: [Asterisk-Users] asterisk and Broadvoice Outgoing Again :(

Hi, I tried that but same error Specially I didn't find people posting about "Bad Request" or "Unknown Dialog" -- Got SIP response 400 "Bad request" back from 147.135.8.128 -- SIP/5092321848-ccd4 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Conge

Re: [Asterisk-Users] Realtime does not work yet, ...

Matthew Boehm wrote: Are you running any NAT anywhere? I see that your NAT value is set to '1'. It should be 'yes' or 'never'. That might be your problem. The database field for nat it int(1) I have NAT, ... 601 & 602 are on a ATA (public IP) 615 is a SNOM on the LAN (private IP) 621 is a Budg

Re: [Asterisk-Users] Unable to create channel of type 'IAX2'

I do not undertand how to tell to Asterisk to use the X100P to dial an external number instead the internal one. It always calls the internal extension. Could someone give me a valid iax.conf and extension.conf examples files. Regards, chan_iax2 says you have no section in your iax.conf telling

Re: [Asterisk-Users] ASTCC or should I use something else for different rates, depending on the calling card?

Stephen Misel wrote: Ronald Wiplinger wrote: New developments in our business plan make a change necessary. We would like to offer different prices, depending on the user/calling card. How can we use that with ASTCC? or should we use something else? ASTCC allows for multiple brands and different

Re: [Asterisk-Users] EADS6550 and asterisk - echo on PSTN call

> would like to know if some of you have tested asterisk connected to an > EADS 6550 analogique PBX (also know as Nexpan50). > > Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no > other card, each of them have their own IRQ) all ports connected to the >

Re: [Asterisk-Users] ASTCC or should I use something else for different rates, depending on the calling card?

Ronald Wiplinger wrote: Stephen Misel wrote: Ronald Wiplinger wrote: New developments in our business plan make a change necessary. We would like to offer different prices, depending on the user/calling card. How can we use that with ASTCC? or should we use something else? ASTCC allows for multi

Re: [Asterisk-Users] checking active SIP members of a queue?

Roy Sigurd Karlsbakk wrote: having a queue with some SIP members, is there a way to check how many of them are connected to asterisk, and if none are, go to a different context? Not at the moment, no. ___ Asterisk-Users mailing list Asterisk-Users@lists

Re: [Asterisk-Users] Unable to create channel of type 'IAX2'

On Sat, Mar 12, 2005 at 04:04:29PM +0100, Androtech wrote: > Hi all, > I'm a newbie and I have a configuration problem with Asterisk. > Seems that I'm not able to call an outbound number. I'm quite sure that it > is a configuration problem, but I'm not able to find out where is the > mistake, eve

[Asterisk-Users] checking active SIP members of a queue?

hi having a queue with some SIP members, is there a way to check how many of them are connected to asterisk, and if none are, go to a different context? thanks roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/

Re: [Asterisk-Users] Realtime does not work yet, ...

> sipfriends => mysql,astconf,sip_buddies Yes. Remove that line. This was done a few weeks ago to better split peers/users. > "sip show users" and "sip show peers" does not show the phone, but Go into sip.conf and enable the 3 RealTime cacheing variables. This will make them show up in

[Asterisk-Users] ASTCC - Regex: How to "Country" but "special City" different?

I am trying to figure out a way to add something like: 61 100 pennies (Everything what is not listed below) 61 78150 61 5 130 61 342 180 How could I do these (four) regex? bye Ronald ___

Re: [Asterisk-Users] Realtime does not work yet, ...

Are you running any NAT anywhere? I see that your NAT value is set to '1'. It should be 'yes' or 'never'. That might be your problem. Have you tried adding this user into the sip.conf first to verify that this is truly an ARA problem? -Matthew > From: Ronald Wiplinger <[EMAIL PROTECTED]> > Repl

RE: [Asterisk-Users] Droping calls

Will give it a try. Thx! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Sábado, 12 de Marzo de 2005 07:15 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Droping calls > Guys, this is wei

RE: [Asterisk-Users] Droping calls

So, should I just disable busydetect then? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Sábado, 12 de Marzo de 2005 04:13 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Droping calls I

[Asterisk-Users] Looking for an Asterisk Expert/Partner for project

Hello All, How are you all doing today? Good I hope. Well, I have been learning a lot about Asterisk and must say that it is GREAT I have been able to configure and compile various ideas that I have been working on in my test Asterisk PBX and everything worked just fine. The reason for this

Re: [Asterisk-Users] Trouble with Realtime

On Fri, 11 Mar 2005 16:38:38 -0600, Nathan Bowyer <[EMAIL PROTECTED]> wrote: > On Fri, 11 Mar 2005 15:04:03 -0600, Matthew Boehm <[EMAIL PROTECTED]> wrote: > > You can't have this: > > > > [from-sip] > > switch => Realtime/[EMAIL PROTECTED] > > > > The context in your extensions.conf must be differ

Re: [Asterisk-Users] EADS6550 and asterisk - echo on PSTN call

Rich Adamson a écrit : would like to know if some of you have tested asterisk connected to an EADS 6550 analogique PBX (also know as Nexpan50). Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no other card, each of them have their own IRQ) all ports connected to the EADS. We

Re: [Asterisk-Users] ASTCC or should I use something else for different rates, depending on the calling card?

Stephen Misel wrote: Ronald Wiplinger wrote: New developments in our business plan make a change necessary. We would like to offer different prices, depending on the user/calling card. How can we use that with ASTCC? or should we use something else? ASTCC allows for multiple brands and different

Re: [Asterisk-Users] Realtime does not work yet, ...

Mark & Matthew, I know how frustrating it may be, ... I can imagine your feelings, ... HOWEVER, with all respect, it does not help me to fix my problem! Can we come back to the subject, please? I apologies for the missing words "for me" in the Subject! I tried to follow (and may made some mistakes

Re: [Asterisk-Users] X-Lite and * SIP Problem

> [200] > type=friend > username=richard change this to username=200 > And my X-lite Default SIP Proxy config is as follows: > > Enabled: Yes > Display name: richard > Username: richard Change this to Username: 200 and this one > Authorisation User: richard to Authorisation User: 200 and it s

[Asterisk-Users] Unable to create channel of type 'IAX2'

Hi all, I'm a newbie and I have a configuration problem with Asterisk. Seems that I'm not able to call an outbound number. I'm quite sure that it is a configuration problem, but I'm not able to find out where is the mistake, even reading several docs to www.voip-info.org. I do not have a good kno

Re: [Asterisk-Users] EADS6550 and asterisk - echo on PSTN call

> >>would like to know if some of you have tested asterisk connected to an > >>EADS 6550 analogique PBX (also know as Nexpan50). > >> > >>Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no > >>other card, each of them have their own IRQ) all ports connected to the > >>EADS. W

Re: [Asterisk-Users] Wireless VoIP

we have a full line of WiFi phone that work great. Works at any hot spot also for more info contact me at [EMAIL PROTECTED] - Original Message - From: "Sylvain COUTANT" <[EMAIL PROTECTED]> To: Sent: Friday, March 11, 2005 12:42 PM Subject: [Asterisk-Users] Wireless VoIP Hi all, I have

[Asterisk-Users] Sipura 2100 and Asterisk one-way audio

Hello People, I have a Sipura SPA-2100 with default configuration and the last software upgrade, and a * from Debian Sarge with the simple configuration: [general] port = 5060 bindaddr = 0.0.0.0 [103] username=103 type=friend secret=qaz123wsx qualify=no port=5060 nat=yes host=dynamic dtmfmode=

Re: [Asterisk-Users] No ringback over IAX - LiveVoip

> >> I saw some coverage of this in the list archive but no > >>one seems to have > >> posted a resolution. > >> > >> I am using [EMAIL PROTECTED] 0.06 and when I get a call from > >>LiveVoip over > >> IAX I dump it into my IVR. > >> >From there the call is routed to groups based upon > >>input

Re: [Asterisk-Users] EADS6550 and asterisk - echo on PSTN call

Rich Adamson a écrit : would like to know if some of you have tested asterisk connected to an EADS 6550 analogique PBX (also know as Nexpan50). Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no other card, each of them have their own IRQ) all ports connected to the EADS. We

Re: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server]

How did you go? On Tue, 8 Mar 2005 11:28:59 +1030, Peter Childs <[EMAIL PROTECTED]> wrote: > > Digium shipped me a replacement card, but they sent the wrong one, so they > fedex'd another and its just arrived. > > Should be testing in the next two days (the box is in another state...) > > The

Re: [Asterisk-Users] Droping calls

> Guys, this is weird.. Today I started having some problems with calls been > dropped. Im suing X100p cards (clones) and I have this setting on my zatala > fle: > > busydetect=yes > busycount=4 Try changing busycount to 6 or 8, stop asterisk, and restart. _

[Asterisk-Users] Cisco 7970

Hi, Is there anyone with some uptodate info on the Cisco 7970 and Asterisk skinny / SCCP? I know chan_sccp doesn't support the 7970. Is it true the basics should work with skinny? All info is welcome. Steven ___ Asterisk-Users mailing list Asterisk-Use

RE: [Asterisk-Users] No ringback over IAX - LiveVoip

> Grett. This should be loads of fun then... 8( > > I have noticed what I can only describe a negative undertone with > several VoIP poviders. > Not an easy customer? We don't want you. Things like that. > The LiveVoIP website is in fact like that. > > There are several places on the si

RE: [Asterisk-Users] Asterisk on MS Virtual Server

I think you missed out a *NOT* below... " In short, you *CANNOT* install or otherwise use any hardware cards, like Zaptel, with Asterisk when running on CoLinux and generally, I'll advise you to not use Astwind for anything other then playing. It's a nice toy, but that is all. " -Origina

[Asterisk-Users] Signaling on PRI channels

Hi, We did an interconnection with our carrier few days ago. But, I noticed that there was a signaling problem on our trnuk. In fact, Asterisk indicates that the call is answered when we received ALTERTING message from our carrier. This is PRI debug logs : -- Executing Dial("IAX2/[EMAIL P

Re: [Asterisk-Users] ATA 186 Codec Question.

David Uzzell wrote: I have seen the list of codecs for the ATA 186's but not sure if it was 100% or not. I want to know really is it possible to run GSM or ilbc on them or is a G729 lic the only way to get a low bandwidth codec? This is the list of codecs that I have seen. RxCodec and TxCodec—C

Re: [Asterisk-Users] TDM04B lock up

> I have a strange situation. Once in a while (non-deterministic) the 2 TDM04B > cards lock up at the same time and stop processing incoming and outgoing > calls even though * shows that it is trying to communicate to ZAP channels > (at least on the outgoing). The only cure is to reboot the system

Re: [Asterisk-Users] EADS6550 and asterisk - echo on PSTN call

> would like to know if some of you have tested asterisk connected to an > EADS 6550 analogique PBX (also know as Nexpan50). > > Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no > other card, each of them have their own IRQ) all ports connected to the > EADS. We have GS AT

Re: [Asterisk-Users] Multiple IAX Phones Behind NAT

> Ok, I've seen this question go unanswered on the mailing list, and I assume > it's because no one had the heart to break the bad > news to the guy asking, but be honest with me, I can take it. At this time > it's flat impossible to have multiple IAX phones > behind a NAT without using an * g

Re: [Asterisk-Users] OT: Best DB

[EMAIL PROTECTED] wrote: I know this is a bit off topic but we are using Asterisk :) Since this list is full of tech gurus w/ all different sorts of backgrounds, I thought I would get the best opinions here. We have several different switches and other telecom equipment at our facilities which

[Asterisk-Users] ATA 186 Codec Question.

I have seen the list of codecs for the ATA 186's but not sure if it was 100% or not. I want to know really is it possible to run GSM or ilbc on them or is a G729 lic the only way to get a low bandwidth codec? This is the list of codecs that I have seen. RxCodec and TxCodec—Configure the codec I

Re: [Asterisk-Users] Print-to-Fax client

Florian Overkamp wrote: Hi, -Original Message- You should be able to download one (for WIndows and possibly Mac) from efax or j2.com I think. http://www.efax.com/en/efax/twa/page/download?rqcp=2 http://www.j2.com/jconnect/twa/page/download You might be able to do that, but take a

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