[Asterisk-Users] Re: Double sip logins

On 4/8/06, Joe <[EMAIL PROTECTED]> wrote: > Remove the SIP /400 entry from the Asterisk DB. > > Database del At asterisk prompt. > > Or look at the wiki for info on how to remove it. > > Or make sure the SIP/500 uses a different IP address than the old SIP/400. > > Joe > > > > thanks for your

Re: [Asterisk-Users] Cisco 7960 won't dial (sccp)

[EMAIL PROTECTED] ha scritto: On Wed, Apr 12, 2006 at 09:32:12PM +0200, Sergio Chersovani wrote: [EMAIL PROTECTED] ha scritto: context = from-sccp-intenal I guess "intenal" is not the righe context :-) Sergio The from-sccp-internal is almost an exact copy of my from-

[Asterisk-Users] Announcement: New Texas User Group formed

In an effort to bring Asterisk Users from across the state of Texas together, the Texas Asterisk Users Group has been formed. The goal is to help Asterisk users meet other is their area and to help spread the word about the Asterisk community. I anticipate regional meetings of members and look forw

[Asterisk-Users] Problem with Voice Quality

Hi All, We are making a VOIP application for Mobiles (PDA's) and we are using Asterisk for it. We have a setup consisting of both SER and Asterisk. SER acts as a SIP router and routes everything to Asterisk. We also have rtpproxy for SER. Our packet delivery from clients (Mobiles, PDA's)

RE: [Asterisk-Users] freepbx dialing prefix

Submit a bug report to the FreePBX team? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean GarlandSent: Wednesday, April 12, 2006 8:46 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] freepbx dialing prefix I need to put a ‘w’ in t

Re: [Asterisk-Users] How to terminate ringing call before it is answered

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial Checkout options "H" and "h". Darren Wiebe [EMAIL PROTECTED] Obelix wrote: > > Is there a way to terminate a ringing call before it is answered? > > I am speaking of prepaid card appli

[Asterisk-Users] freepbx dialing prefix

I need to put a ‘w’ in the dialing prefix, but it says it isn’t valid.  If I manually modify the extension file, it then affects all calls made over any trunk.  Any ideas?   Sean -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Data

[Asterisk-Users] How to terminate ringing call before it is answered

Is there a way to terminate a ringing call before it is answered? I am speaking of prepaid card application in which you want to make another call, because you current number it is not being answered, and you don't want to hangup before dialling again. /Obelix __

Re: [Asterisk-Users] Re: update - 512 Simultaneous Callswith DigitalRecording

On Wed, 12 Apr 2006, Leo Ann Boon wrote: I'm not sure tmpfs is the right solution for the OP's problem - disk access slowing down the system. My understanding of tmpfs is that it will swap pages in and out to/from disk. Wouldn't that be as bad as directly writing to disk? I can see tmpfs will h

[Asterisk-Users] need help

hi All   I need your help , for used Digium Card TE405P, for setting this Board AS E1 ISDN PRI.   1 .Current for make sure my config its rights or no I inform my configurations in Board Jumper T1/E1 is Closed is that rights or no ? for E1 i closed the Jumper.     2. I Want To seeting E1

Re: [Asterisk-Users] SPA-941/942 Bulk provisioning

[EMAIL PROTECTED] wrote: On Tue, 11 Apr 2006, tracinet wrote: Unfortunately, Linksys is reserving the provisioning tools/info to their official resellers. The idea is that you pay your Linksys reseller to provision your phones (does not make ANY sense to me all). As a service provider, we shou

Re: [Asterisk-Users] SPA-941/942 Bulk provisioning

On Tue, 11 Apr 2006, tracinet wrote: Unfortunately, Linksys is reserving the provisioning tools/info to their official resellers. The idea is that you pay your Linksys reseller to provision your phones (does not make ANY sense to me all). As a service provider, we should be able to buy these ph

[Asterisk-Users] RE: Asterisk-Users Digest, Vol 21, Issue 70

Ok, I did check it before and nothings related to this "#" key, if it's then system will announce that "Please key in the extensions", but not in this case. By default, the blind transfer is #1. Some one can help? >Check your features.conf file for conflicting key set. # is the default >k

[Asterisk-Users] Multiple phones in same call

Hi, all This is what I would like to do: Someone is on the phone and nother person ant to join in. Like in house wheer all phones are connected to same line. I can do it with MeetMe, but my understanding is that all parties have to call "meeting room" number. What I want instead is to have some "

Re: [Asterisk-Users] help -- voicemail

chan (Alpha Trilogies Networks) wrote: Hi, Did someone experience that Asterisk OS 1.2.5 voicemail issues? Problem description: Some one call to the extensions 200, After 10 sec ring then go to voicemail [EMAIL PROTECTED] Announcement "Please leave me a messages.blar blar.." When I completed

[Asterisk-Users] Text labels on incoming call appearances?

Hi All, I'm looking at an Asterix system for a small home office, 3 incoming lines and eight phones scattered about the house, probably with a GUI wrapper like SwitchVox to help with the administration. I'm still looking at phones, and am getting pretty confused. It looks like the phones ha

[Asterisk-Users] ASterisk Back2back

hi All   I need your help , for used Digium Card TE405P, for setting this Board AS E1 ISDN PRI.   1 .Current for make sure my config its rights or no I inform my configurations in Board Jumper T1/E1 is Closed is that rights or no ? for E1 i closed the Jumper.     2. I Want To seeting E1 in

Re: [Asterisk-Users] Asterisk BRI in the USA

Next, I got a Eicon Diva board and tried to get the hisax kernel driver working. It's ni-1 implementation, the only one I could find, isn't very complete. It was written by a guy in Australia using only an isdn simulator, a significant accomplishment. It appears that it's intent was to jus

[Asterisk-Users] Asterisk 1.2.7 Released!

The Asterisk Development Team is pleased to announce the release of Asterisk 1.2.7. This release include a number of important bug fixes, including SIP presence (subscriptions) handling and MixMonitor call recording, and users are encouraged to upgrade their systems when possible. See the included

RE: [Asterisk-Users] Call Forward and AGI

I'm sure there is more than 1 way to do this, but the first thing that comes to my mind is to set a channel variable with the exten # at the top of your extensions macro. Then use that channel var instead of CLID. Josh McAllister -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL

Re: [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)

On Wed, 2006-04-12 at 17:18 -0400, Andrew Kohlsmith wrote: > How are we to know what you tried and didn't try? I didn't think it necessary at the time. Had I have documented the process and included config files and log files and tcpdump traces, wouldn't I have received the TMI lecture instead?

Re: [Asterisk-Users] Texas User Group

I will be out of the office until Wednesday, 09/21/2005. If you need immediate assistance, please contact Phill Miller at 412.262.8503 or [EMAIL PROTECTED] Thank you. >>> asterisk-users 04/12/06 18:25 >>> I setup www.txaug.net with a temporary page and have a working mailing list at list.txaug.ne

Re: [Asterisk-Users] Company List

That is a perfect name since it is a customer to this company.On 4/12/06, C F <[EMAIL PROTECTED]> wrote: I know Halliburton is implementing Asterisk, their techincalrequirement for the proposal alone is 38 pages in French. On 4/12/06, Aaron Daniel <[EMAIL PROTECTED]> wrote:> I know Digium has a few

Re: [Asterisk-Users] Texas User Group

I setup www.txaug.net with a temporary page and have a working mailing list at list.txaug.net. Look forward to comments and suggestions.JR, you definately have a great setup for user group meetings. -- Bruce Nortex NetworksOn 4/12/06, JR Richardson <[EMAIL PROTECTED]> wrote: Hi, I'm in the Dallas

Re: [Asterisk-Users] Cisco 7960 won't dial (sccp)

Shawn,   What Sergio meant was you misspelled internal under [lines].  Not sure if it is that way in your file, of if it was mistyped here.   context     = from-sccp-intenal   That's listed under the lines, note the missing 'r'.   On 4/12/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: On Wed, Apr

Re: [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)

Jim Rice wrote: I only asked this list as a last resort, having already exhausted many other avenues. I even mentioned that it was OT, but have seen numerous postings for phones of all kinds. A thought: I had similar problems with one phone of mine after I power-cycled it during the provision

Re: [Asterisk-Users] Company List

I know Halliburton is implementing Asterisk, their techincal requirement for the proposal alone is 38 pages in French. On 4/12/06, Aaron Daniel <[EMAIL PROTECTED]> wrote: > I know Digium has a few case studies on their website. > > http://www.digium.com/en/asteriskbusinesses/casestudies/ > > Aaron

Re: [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)

On Wednesday 12 April 2006 17:03, Jim Rice wrote: > What, beyond RTFM, Google, list archives, Polycom support (yeah, right), > and literally hours of trial and error? How are we to know what you tried and didn't try? > My intent was that if someone had seen this specific error, they would > reply

Re: [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)

On Wed, 2006-04-12 at 16:35 -0400, Andrew Kohlsmith wrote: > On Wednesday 12 April 2006 16:06, Jim Rice wrote: > > > Then it displays: > > > Config file error > > > Error is 0x4020 > > > I cannot be the only one with Polycom 501s to have seen this error?! > > Surely when presented with a problem

RE: [Asterisk-Users] Asterisk BRI in the USA

I was reading that Junghanns was palnning on supporting National (Q.931) if they do, all you should need is a NT1 to turn the U insterface in an S/T. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Walt Reed > Sent: Wednesday, April 12, 2006 4:39

Re: [Asterisk-Users] Company List

I know Digium has a few case studies on their website. http://www.digium.com/en/asteriskbusinesses/casestudies/ Aaron On Wed, 12 Apr 2006, Bruce Reeves wrote: That's exactly right, I had hoped that Digium had something like this, and may at their booth at conferences. I just need something to

Re: [Asterisk-Users] Asterisk BRI in the USA

On Wed, Apr 12, 2006 at 09:10:09AM -1000, Mark Coccimiglio said: > I guess what I need to find out first if there is anyone out there using > Asterisk & BRI in the USA? If so what hardware have they been able to > use. I no longer want to hack around with analog circuits. BRI has the > potential

Re: [Asterisk-Users] CAPI Installation Eicon Diva Server

[EMAIL PROTECTED] wrote: Asterisk says it has 30 capi channels available, but my mistake may be in configuring the trunks... When I was debugging my Eicon Diva 4-BRI board, I found it useful to play with extensions_custom.conf (in AMP) just to ensure I got the Custom Dial String absolutely co

Re: [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)

On Wednesday 12 April 2006 16:06, Jim Rice wrote: > > Then it displays: > > Config file error > > Error is 0x4020 > I cannot be the only one with Polycom 501s to have seen this error?! Surely when presented with a problem you don't throw your hands up in the air and expect others to fix it. The

Re: [Asterisk-Users] Company List

That's exactly right, I had hoped that Digium had something like this, and may at their booth at conferences. I just need something to prove credibility. If you haven't seen is the Forbes article http://www.forbes.com/free_forbes/2006/0410/063.html does some of that with the mention of 3 customers

RE: [Asterisk-Users] Company List

I disagree a bit. A lot of companies publish their "customer list" for reasons of advertisement. If I have a client that is joe blow fortune 500 company, I'm gonna publish that for my credibility. I think that is what we are looking for (I think I can safely speak for both of us on this). Curt --

Re: [Asterisk-Users] Cisco 7960 won't dial (sccp)

On Wed, Apr 12, 2006 at 09:32:12PM +0200, Sergio Chersovani wrote: > [EMAIL PROTECTED] ha scritto: > >context = from-sccp-intenal > > > I guess "intenal" is not the righe context :-) > > Sergio The from-sccp-internal is almost an exact copy of my from-sip-internal context, which works fine.

Re: [Asterisk-Users] Company List

I would doubt that anyone is going to share their "customer list" for obvious reasons. I'd have to guess that in access of 80% of the production implementations are sold by resellers (of various sizes), and maybe 20% are actual in-house implementations by those that frequent this list. The 80%

Re: [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)

On Tue, 2006-04-11 at 04:56 -0700, Jim Rice wrote: > Using FTP to configure 501. > Gets past "Running... App = sip.ld" > and: Welcome! Processing configuration... > "This may take a few seconds." > > Then it displays: > > Config file error > > Error is 0x4020 > > and reboots continuously, re

Re: [Asterisk-Users] Company List

>> Bruce wrote: >> >> >> The question was raised by a CFO who is looking at Asterisk if there is a list of >> companies using Asterisk. I have not found one yet, has anyone seen anything like >> this I can give him. > > Curt Shaffer wrote: > > I have not but if you find one, please pass it on b

[Asterisk-Users] Texas User Group

Hi, I'm in the Dallas area, my office is in Irving.  I would be willing to host a user group.  I have a decent size conference room, big whiteboard and on-site lab with 5 asterisk servers dedicated for testing/development.  I can host web space and e-mail list also.  I do like the wiki idea though,

RE: [Asterisk-Users] Company List

I have not but if you find one, please pass it on because I have the same requirement.   Curt   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Wednesday, April 12, 2006 3:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Compa

[Asterisk-Users] Playback sound file while on-line

Hi, I am looking for a way to play a sound file (wav, gsm or whatever) while a SIP client (extension) is on-line with a Zap channel. Ideally both ends would hear the sound file. Any hints or pointers appreciated. Andre ___ --Bandwidth and Coloc

[Asterisk-Users] Company List

The question was raised by a CFO who is looking at Asterisk if there is a list of companies using Asterisk. I have not found one yet, has anyone seen anything like this I can give him.-- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easy

Re: [Asterisk-Users] SIP MWI

12 apr 2006 kl. 21.30 skrev David Gomillion: If it's already been covered, please forgive the repetition. I searched Mantis, but couldn't come up with anything. We upgraded to Asterisk 1.2.6, and suddenly the Polycom MWI stopped working on SP IP 300s and 600s. All of them. I tried splitti

Re: [Asterisk-Users] Recording queue transfers

Hello, you can use the veryu same technique to turn recording on in any context - what matters is that you bring along the uniqueid of the original call, so you know how to match the different recordings you may find on your hard disk! l. In data Wed, 12 Apr 2006 19:24:56 +0200, Maximili

Re: [Asterisk-Users] SIP conections, with RTP not going trough Asterisk

12 apr 2006 kl. 14.58 skrev Ronald Wiplinger: Tiago Stein D`Agostini wrote: Hi, Ie been looking for some time how to use asterisk to initiate SIP connections between 2 IP phones, but afetr initiated the communication making the RTP go directly from one telephone to the other, withou

[Asterisk-Users] Call Forward and AGI

Hi i have a agi script that gets called when a user wants to dialout externally. it gets passed in the exten number and the number dialled and looks up in a db to see if they are allowed to dial the number. the problem is if someone forwards their phone to a external number the CALLERIDNUM is the

Re: [Asterisk-Users] Cisco 7960 won't dial (sccp)

[EMAIL PROTECTED] ha scritto: context = from-sccp-intenal I guess "intenal" is not the righe context :-) Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: ht

[Asterisk-Users] SIP MWI

If it's already been covered, please forgive the repetition. I searched Mantis, but couldn't come up with anything. We upgraded to Asterisk 1.2.6, and suddenly the Polycom MWI stopped working on SP IP 300s and 600s. All of them. I tried splitting the friend entries in sip.conf into user and p

Re: [Asterisk-Users] Polycom VLANs

Use VLANs on Ploys all the time, but manually set also. Of course switches and routers all need to be setup for the proper vlan config also. On Apr 12, 2006, at 12:18 PM, BJ Weschke wrote: On 4/12/06, Rob Terhaar <[EMAIL PROTECTED]> wrote: So has anyone had any experience working with the

Re: [Asterisk-Users] Asterisk BRI in the USA

I guess what I need to find out first if there is anyone out there using Asterisk & BRI in the USA? If so what hardware have they been able to use. I no longer want to hack around with analog circuits. BRI has the potential of PRI with only 2 B channels. A great idea for a small office such as

Re: [Asterisk-Users] URL in Queue App / Determining the DID/Queue at Agent's Phone

Kyle, That's bloody brilliant Thanks so much! -Steve Feinstein GatherWorks, Inc. Kyle Sexton wrote: Have you tried something like: exten => 2,1,SetCIDName(QUEUENAME: ${CALLERIDNAME}) exten => 2,n,Queue(QUEUENAME) On 4/12/06, * Steve Feinstein* <[EMAIL PROTECTED]

Re: [Asterisk-Users] URL in Queue App / Determining the DID/Queue at Agent's Phone

Have you tried something like:exten => 2,1,SetCIDName(QUEUENAME: ${CALLERIDNAME})exten => 2,n,Queue(QUEUENAME)On 4/12/06, Steve Feinstein <[EMAIL PROTECTED]> wrote: Thanks!, I will definitely take a look at that.  We were hoping not tohave to do AGI in the client, but if we have to, we have to.  I

[Asterisk-Users] DUNDi with SIP

Anyone out there have a functional DUNDi configuration using SIP for the inter-Asterisk transport? I've gotten it to work with IAX2, but if I change it to SIP it does not pass the call over even though it knows where to send it. Thanks. The contents of this email message and any attachments are

Re: [Asterisk-Users] free video (soft) phone available?

We use Neos from neosmt.com to connect to our interoffice jabber server and I noticed recently that it can do video and audio via a h.323 gatekeeper. Haven't tried it out yet but you might. Ronald Wiplinger wrote: I am using eyebeam and I am happy with it. However, it is boring just to talk t

RE: [Asterisk-Users] playback soundfile stored in mysql database

.want to playback a "raw" binary file without writing into an intermediate file which would increase latency From: "Alexander Lopez" <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: "Asterisk Users Mailing List - Non-Commercial Discussion" Su

[Asterisk-Users] Callback Agents and Dial 'g' option

I'm unable to get the Dial option 'g' to work with callback agents. The plan is to use it so that I can redirect a customer to a menu so they can rate the call they just had with the agent. However, when the agent hangs up the call does not continue in the dialplan. I login with the agent.

Re: [Asterisk-Users] * 1.2.4 & 1.2.6: "Ringing" anamoly

Ronald Lewis wrote: I was alerted the other day by of all people, my mom, that she wasn't hearing a "ring" when she dialed my number. Puzzled, I tried calling myself. The call connects, but there's dead silence until voicemail picks up. Calling internally, extensions worked perfectly. So, I fi

[Asterisk-Users] Cisco 7960 won't dial (sccp)

I'm trying to setup a couple of Cisco 7960's in asterisk. I have asterisk working fine for sip clients, and can call the 7960's just fine, but I can't seem to dial out on them. As soon as I enter the first digit, the phone attempts to dial it without waiting for the rest. I've changed timeout

RE: [Asterisk-Users] call center running Asterisk-sound quality-critical!

Yes. That's is the one. It is resolved now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tamas Sent: Wednesday, April 12, 2006 1:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asteris

Re: [Asterisk-Users] call center running Asterisk -sound quality-critical!

>>> Matt Roth wrote: >>> >>> These statements seem contradictory. I know of no way (short of a >>> custom patch) to tell Monitor() to mix the in and out legs prior to >>> writing them to disk. On the other hand, MixMonitor() does just that >>> and I believe it also buffers the writes in a way th

[Asterisk-Users] Config with TE210P, Asterisk and Legacy PBX and FreePBX?

Hi list! Has anyone ever tried the following installation : I want to replace our legacy PBX with Asterisk but... I still need the legacy PBX as a 'channel bank' for fax (I need E1 not T1) I will put a dual port PRI card in the Asterisk box, and for incoming and outgoing faxes I want to use

Re: [Asterisk-Users] Macro-hangupcall - has a Wait(5) - [EMAIL PROTECTED] --- why?

On 4/12/06, Marco Mouta <[EMAIL PROTECTED]> wrote: > [macro-hangupcall] > exten => s,1,ResetCDR(w) > exten => s,2,NoCDR() > exten => s,3,Wait(5) > exten => s,4,Hangup > > > Hi all, currently i've been getting troubles with SIpphone Sjphone and Xlite > seems also to get delay but no crash on hanging

Re: [Asterisk-Users] call center running Asterisk -sound quality-critical!

Wai Wu wrote: > Except that mixmonitor still has a bug in it. > What kind of bug? Issue number? FYI: yesterday one issue has been fixed :D http://bugs.digium.com/view.php?id=6457 Did you mean that type of bug? If something else, please let us know... T. > -Original Message- > From: [

[Asterisk-Users] Recording queue transfers

Regarding this article (1) I have one question to make. What can I do to record the call if the agent makes a transfer using the "flash" button instead of "transfer button" or using blindxfer or atxfer defined in features. conf If the agent makes the transfer with "flash", the comunication between

Re: [Asterisk-Users] call center running Asterisk -sound quality-critical!

On 4/12/06, Wai Wu <[EMAIL PROTECTED]> wrote: > Except that mixmonitor still has a bug in it. > Had. Corrected yesterday. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk

Re: [Asterisk-Users] Polycom VLANs

On 4/12/06, Rob Terhaar <[EMAIL PROTECTED]> wrote: > So has anyone had any experience working with the polycom 501 or 301 and > vlans? > > We run dell managed switches here, so we don't have the luxury of running > CDP to force the VOIP vlan. I haven't been able to get the polycom phones to > talk

RE: [Asterisk-Users] playback soundfile stored in mysql database

Look at using EAGI. > > Hi Guys, > > I want to playback a sound file stored in mysql database in > my perl scriptpls can anyone help with an idea? > response would be greatly appreciated > > Rgds ___ --Bandwidth and Colocation provided

RE: [Asterisk-Users] call center running Asterisk -sound quality-critical!

Except that mixmonitor still has a bug in it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 12, 2006 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center r

[Asterisk-Users] playback soundfile stored in mysql database

Hi Guys, I want to playback a sound file stored in mysql database in my perl scriptpls can anyone help with an idea? response would be greatly appreciated Rgds _ Express yourself instantly with MSN Messenger! Downloa

[Asterisk-Users] Polycom VLANs

So has anyone had any experience working with the polycom 501 or 301 and vlans? We run dell managed switches here, so we don't have the luxury of running CDP to force the VOIP vlan. I haven't been able to get the polycom phones to talk on a manually set vlan. I have some junky sipura phones that wo

RE: [Asterisk-Users] Bandwidth Management

Brought over from -users, Please reply to the -dev list. I agree, lets move the discusstion over to that list as it has to be discussed there. After we reach an accord on how it should be done we will open up a issue on Mantis. I see this as being two distinctive parts that would need to be tie

RE: [Asterisk-Users] Setting Codecs on the Fly

Simply check out the READMEs in asterisk/doc/ in your source directory. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Julio Arruda > Sent: Wednesday, April 12, 2006 12:42 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subje

RE: [Asterisk-Users] Setting Codecs on the Fly

Ahhh a variable. I was looking for a command. Thanks, I'll try it out. > -Original Message- > From: Julio Arruda [mailto:[EMAIL PROTECTED] > Sent: Wednesday, April 12, 2006 10:42 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Setting Codecs

Re: [Asterisk-Users] Setting Codecs on the Fly

Douglas Garstang wrote: Does anyone know if it's possible to set the codecs for a number via an Asterisk command? Ie, yes you can set the codecs in sip.conf for a user, but I'd like to have a command that can set the same thing so that it can be done without having to change sip.conf. Essent

[Asterisk-Users] * 1.2.4 & 1.2.6: "Ringing" anamoly

I was alerted the other day by of all people, my mom, that she wasn't hearing a "ring" when she dialed my number. Puzzled, I tried calling myself. The call connects, but there's dead silence until voicemail picks up. Calling internally, extensions worked perfectly. So, I figured, "another damned Br

Re: [Asterisk-Users] call center running Asterisk - sound quality-critical!

Tamas wrote: > Kevin, does MixMonitor have buffering? How big is the buffer? Is it > possible to change the size? I guess, we are talking about buffering > voice samples and writing only a bulk of them to disk (e.g. in every 50 > packets - 1second). It buffers the data in memory, there is no fixe

Re: [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording

Hey Henri, Long time no talk. How far have you been able to scale oreka up to? How many simultaneous calls have you been able to record and under what hardware config? Thanks, Waldo On Apr 12, 2006, at 11:12 AM, Henri Herscher wrote: Another solution would be to use a dedicated recording

Re: [Asterisk-Users] call center running Asterisk - sound quality-critical!

Kevin P. Fleming wrote: > Matt Roth wrote: > > >> These statements seem contradictory. I know of no way (short of a >> custom patch) to tell Monitor() to mix the in and out legs prior to >> writing them to disk. On the other hand, MixMonitor() does just that >> and I believe it also buffers th

Re: [Asterisk-Users] TE410P upgrade to TE411P => (solution to) no more fax carrier detection !

Rob Lith wrote: > Kevin - if you stop it from tone detection with 'vpmdtmfsupport=0' will it > detect the fax cgn? Yes, that was the point of my message; with that setting, the software tone detector will be used, just as it was before the OP's VPM got installed. _

Re: [Asterisk-Users] TE410P upgrade to TE411P => (solution to) no more fax carrier detection !

Kevin - if you stop it from tone detection with 'vpmdtmfsupport=0' will it detect the fax cgn?RegardsRobOn 12/04/06, Kevin P. Fleming < [EMAIL PROTECTED]> wrote:[EMAIL PROTECTED] wrote:> I changed from a TE410P to a TE411P and fax carriers weren't detected> anymore !> I have tried everything (reco

Re: [Asterisk-Users] iax2 show netstats

> i've been using iax2 show netstats and i wonder if someone could explain > what all these means, just in case i have them wrong. Because i am looking > for something that tells me that there is delay , and/or packet loss. > > LOCAL - -

Re: [Asterisk-Users] call center running Asterisk - sound quality-critical!

Matt Roth wrote: > These statements seem contradictory. I know of no way (short of a > custom patch) to tell Monitor() to mix the in and out legs prior to > writing them to disk. On the other hand, MixMonitor() does just that > and I believe it also buffers the writes in a way that circumvents t

[Asterisk-Users] Setting Codecs on the Fly

Does anyone know if it's possible to set the codecs for a number via an Asterisk command? Ie, yes you can set the codecs in sip.conf for a user, but I'd like to have a command that can set the same thing so that it can be done without having to change sip.conf. Essentially I want the user to b

[Asterisk-Users] CAPI Installation Eicon Diva Server

Hi   I've got a dell 2550 with an Eicon Diva server PRI card plugged into it. I can call out using the acopy2 test utility.   I'm having trouble with asterisk making calls however... my capi.conf and modules.conf looks correct by the wiki instructions - does anyone have any advice on where

Re: [Asterisk-Users] Texas User Group

That may be the best idea. Unfortunately we're such a huge state that it's going to be pretty hard to get everyone in the same room unless there's some big event going on. Astricon may be a good time to get together in person though. As for the site, a simple wiki may be best, and if everyon

Re: [Asterisk-Users] call center running Asterisk - sound quality-critical!

Wai Wu wrote: > You got to be kidding about 53 calls being recorded at sametime is an > issue. I have done at least twice as many on my dual xeon 3.4Ghz system > and had no problem as clients like to record every call that goes > through the system. Nope. We took our system to MCI's development

Re: [Asterisk-Users] Performance: Xeon or Opteron?

Rich Adamson wrote: While talking with one of the sangoma folks very recently, he was rather emphatic the pci bus was designed to "share" interrupts. I was a little concerned as a test server had the wanpipe driver sharing an interrupt with libata and uhc1_hcd. His comment was "that's the

Re: [Asterisk-Users] call center running Asterisk - sound quality - critical!

If you don't want to worry about * handling the full recording of all traffic, you can potentially do this on a separate server on the RTP path using http://www.oreka.org. Cheers Henri On 10/04/06, Dov Bigio <[EMAIL PROTECTED]> wrote: > > Hi, > > I am using Asterisk for a call center on a Dual Xe

Re: [Asterisk-Users] URL in Queue App / Determining the DID/Queue at Agent's Phone

Thanks!, I will definitely take a look at that. We were hoping not to have to do AGI in the client, but if we have to, we have to. It'll probably be useful for other things down the road. -Steve Feinstein GatherWorks Inc. BJ Weschke wrote: On 4/12/06, Steve Feinstein <[EMAIL PROTECTED]> wro

Re: [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording

Another solution would be to use a dedicated recording server sniffing RTP and signalling packets in the media path using software such as http://www.oreka.org. Oreka automatically mixes both legs of an RTP conversation to disk and GSM encodes the result in a separate thread so that capture always

Re: [Asterisk-Users] Performance: Xeon or Opteron?

Kristian Kielhofner wrote: Rich Adamson wrote: Yep, there is a lot of chatter about how hardware "x" performs with Asterisk and while I/O is the primary mover, most designs today will handle the modest Asterisk install easily. I've got a site where they use 6 lines and 15 users on a 500Mhz C

[Asterisk-Users] Newbie MOH and call transfer question

Hi, I use the AAH2.7 (asterisk version 1.2.5). When someone call me and I pickup the phone. If I want to transfer to another extension. Then I dial the "#" key the system will play the onhold music. After I dial the extension number. The system stop play onhold music and play ringtone. Is it pos

RE: [Asterisk-Users] call center running Asterisk - sound quality-critical!

Just good old monitor with no mixing onto the scsi drive. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tamas Sent: Wednesday, April 12, 2006 4:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center r

RE: [Asterisk-Users] SipXPhone

Mark,   I could not get SipXPhone working either.  We've been using this SDK and really like it: http://www.worksoutsoftware.com/   The pricing is seems decent as well.   Thanks, Greg   From: Mark Hayward [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 12, 2006 3:21

RE: [Asterisk-Users] Bandwidth Management

I think this belongs to the development mail-list. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Wednesday, April 12, 2006 12:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bandwid

Re: [Asterisk-Users] Performance: Xeon or Opteron?

Rich Adamson wrote: Yep, there is a lot of chatter about how hardware "x" performs with Asterisk and while I/O is the primary mover, most designs today will handle the modest Asterisk install easily. I've got a site where they use 6 lines and 15 users on a 500Mhz CPU w/512MB RAM and boot off

Re: [Asterisk-Users] Performance: Xeon or Opteron?

Yep, there is a lot of chatter about how hardware "x" performs with Asterisk and while I/O is the primary mover, most designs today will handle the modest Asterisk install easily. I've got a site where they use 6 lines and 15 users on a 500Mhz CPU w/512MB RAM and boot off a 2GB flash disk.

Re: [Asterisk-Users] Texas User Group

It sounds like what might be best is a Texas User group, since most of us are spread out across our great state. With Astircon 2006 coming to Dallas this year, we could all probably get together at that time. Mainly I would like to see a user group in Texas because I am deploying a wide spread aste

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