[asterisk-users] Re: Playtones

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... > It looked promising so I tried it. Unfortunately it didn't help. Calling > person doesn't hear ringing. I don't know why this application didn't work as > it should. I have tried with and without "wait" command. > > -- Executing Pla

Re: [asterisk-users] Digium GUI?

On Tue, Sep 19, 2006 at 09:58:45PM -0700, mitcheloc wrote: > You are incorrect. The GUI you are referring to is the framework I already > mentioned. The webpages are static html & javascript (AJAX functionality). > Asterisk has a simple built in HTTP server in trunk now which will be used > to serv

Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

Kristian Kielhofner wrote: Benjamin Jacob wrote: Rushowr wrote: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-user

Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

Benjamin Jacob wrote: Rushowr wrote: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --

RE: [asterisk-users] Polycom default handset volume

You are right,   You can change the default volume play with thoses parameter :     Those seem to be dB value.   David   De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Damon Estep Envoyé : 19 septembre 2006 11:24 À : Asterisk Users Mailing List - Non-Comm

RE: [asterisk-users] Problem with # locking up call

A bug has been opened for this :   http://bugs.digium.com/view.php?id=7982   You should had this comment to the bugs   David De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de marvin horst Envoyé : 19 septembre 2006 09:18 À : asterisk-users@lists.digium.com Objet 

Re: [asterisk-users] Digium GUI?

You are incorrect. The GUI you are referring to is the framework I already mentioned. The webpages are static html & _javascript_ (AJAX functionality). Asterisk has a simple built in HTTP server in trunk now which will be used to serve the webpages up and keep the footprint on the server to a minim

Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

Rushowr wrote: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -

RE: [asterisk-users] Digium GUI?

There is the underlying framework for developers to do their own thing but Digium has also made their own GUI.  It's a GUI, a REAL GUI!  it's in the FAQ's and press releases.  In other words it's public knowledge.  A GUI like FreePBX.  In other words, a GUI!  Did I mention it's a GUI!  Not j

Re: [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out

On Tue, Sep 19, 2006 at 10:33:28PM -0400, Jay R. Ashworth wrote: > So, that wiki page not having answered my question: is there in fact > enough power in the not-really-a-programming-language that is the > dialplanning syntax anyway to capture the appropriate information in a > "variable" so it can

[asterisk-users] How to setup multiple iax2 trunks between two asterisk server?

Hi, all   How to setup multiple iax2 trunks between two asterisk server?   thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/ma

Re: [asterisk-users] SkypeOut with Asterisk?

I have successful link skype with asterisk with http://www.nch.com.au/skypetosip/index.html but not sure whether you need this. here is another link http://www.voip-info.org/wiki/index.php?page=Skype%20Gateways. Good luck!On 9/20/06, Devraj Mukherjee < [EMAIL PROTECTED]> wrote:Has anyone managed t

Re: [asterisk-users] Digium GUI?

just wait until 1.4 is released and we can all find out if there really is or is not one From: mitcheloc <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Digium GU

Re: [asterisk-users] SPA 3102 does not even attempt to register

Start up a sip debug ip x.x.x.x (x=ip address of the device) and see if it is trying to register. You should be able to tell from that if you need to mess with the NAT settings.  If nothing is coming in at all, it sounds like a networking issue.   bp  On 9/19/06, Allan Kamau <[EMAIL PROTECTED]> w

Re: [asterisk-users] Pri Event 6 and 8

jra i think you are correct, because the timing source being the hardware or channel psuedo can get "bogged down" i believe, or at least from my experience i have seen this on more than one occasion dealing with the timing source being either locked(terminology?) or under use, or just bogged do

Re: [asterisk-users] MOH distorted on Pound Key Linux on asterisk 1.2.8

Remove mpg123. In the Asterisk source directory type "make mpg123" I believe that "make install" is required to install it. Jeronimo Romero wrote: Running Asterisk 1.2.8 on Pound Key linux which I downloaded from Digium site. Uname output: Linux localhost 2.6.13.4-1.x86.i686.cmov #1 Wed

Re: [asterisk-users] Cisco 7960 part numbers ...

I can't answer definitively in regard to the Smartnet, but I only see the US and ASIAPAC versions SKU'd up with US Cisco distributors (Ingram, Comstor, TechData) and I do not see an EU version on the GPL. Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225

RE: [asterisk-users] MOH distorted on Pound Key Linux on asterisk 1.2.8

is this open source poundkey? and can i see your moh conf?? im guessing its open source pk since you mentioned the asterisk 1.2.8 part. and also is it the default MOH or your own cooked up version?? also i recommend, if not necessary the EXACT mpg version described in the conf (321 0.9?) somet

Re: [asterisk-users] Pri Event 6 and 8

On Tue, Sep 19, 2006 at 07:02:28PM -0400, Doug Lytle wrote: > Dave Wise wrote: > >I was looking through my log files and keep seeing: > > > >PRI got event: 8 on Primary D-channel of span 3 > >on different spans. sometimes it is event 6. > > A quick Google on this showed a post by Mark on bugs

Re: [asterisk-users] Cisco 7960 part numbers ...

On Tue, 2006-09-19 at 19:58 -0400, Cory Andrews wrote: [snip] > What you would need would be the following. > > (1) Cisco CP-7960G= (Global Spare) > (1) Cisco SW-SM-UL-7960= (SIP & MGCP License for Single 7960 IP Phone) > (1) CON-SNT-7960 (Smarnet 8X5 NBD IP Phone 7960 MGR Set) > > You will al

Re: [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out

On Wed, Sep 20, 2006 at 12:12:58AM +0200, Michael Neuhauser wrote: > On Tue, 2006-09-19 at 16:46 -0500, Lacy Moore - Aspendora wrote: > > It does matter if they are using Asterisk. I think we'd all like to > > know how they are doing it. > > > > When I was using the chan_sccp driver, I was able

Re: [asterisk-users] When does Scalability requests Asterisk to Use SER ?

- Original Message - From: "Rushowr" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, September 19, 2006 10:38 AM Subject: Re: [asterisk-users] When does Scalability requests Asterisk to Use SER ? Thanks for the info. So it was reall

[asterisk-users] SkypeOut with Asterisk?

Has anyone managed to use SkypeOut as your VoIP provider? -- "I never look back, it distracts from the now", Edna Mode (The Incredibles) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update opt

[asterisk-users] MOH distorted on Pound Key Linux on asterisk 1.2.8

Running Asterisk 1.2.8 on Pound Key linux which I downloaded from Digium site. Uname output: Linux localhost 2.6.13.4-1.x86.i686.cmov #1 Wed Nov 23 11:31:48 EST 2005 i686 athlon i386 GNU/Linux     It didn’t come with mpg123 so I downloaded it from the internet.  MOH works, but it is

Re: [asterisk-users] Polycom 500 power supply

sure thing, 12VDC, 400mA, center pin positive :) Moj Forum wrote: I have a Polycom 500 and I have misplaced the power supply. I can not find the specs anywhere – can someone please send me the voltage and amps? Steve !DSPAM:500,45108432185952288320433! --

[asterisk-users] Polycom 500 power supply

I have a Polycom 500 and I have misplaced the power supply. I can not find the specs anywhere – can someone please send me the voltage and amps?   Steve     ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users maili

Re: [asterisk-users] Cisco 7960 part numbers ...

Neither is technically the product you need.   The CP-7960G-CH1 is a Cisco 7960G phone, with a CallManager client license, preloaded with SCCP firmware.   The CP-7960G-CCME is a Cisco 7960G phone, with a CallManager Express client license, preloaded with SCCP firmware.   To my knowledge, Cisc

Re: [asterisk-users] IAX or SIP termination provider that reaches 6421xxxxxxx?

Thank you! looking into it :) kjcsb wrote: We offer NZ mobile termination for NZD0.35/minute (incl GST). If you're based outside New Zealand GST will not be charged so the rate would be NZD0.3111/minute. If you're interested, get a web login at www.conversant.co.nz

[asterisk-users] Cisco 7960 part numbers ...

Hi,I requested a quote from a cisco reseller (or something like this) for 2 cisco 7960 phones, ideally preloaded with SIP firmware ... and i got the quote back with: 1x CP-7960-CH1 and 1x CP-7960-CCME. My question is, what is the difference between the two? If these are not the part number for the

Re: [asterisk-users] Digium GUI?

No it's not, it's supposed to just be a framework for developers and resellers to create GUIs that can go on the appliance. On 9/19/06, shadowym <[EMAIL PROTECTED]> wrote: I am talking about the GUI that was announced as part of the new AsteriskAppliance.Sounds like it is going to be a full feature

Re: [asterisk-users] Pri Event 6 and 8

Dave Wise wrote: I was looking through my log files and keep seeing: PRI got event: 8 on Primary D-channel of span 3 on different spans. sometimes it is event 6. A quick Google on this showed a post by Mark on bugs.digium.com: >> Generally 6 and 8 errors are not a problem, but can be ind

[asterisk-users] IAX or SIP termination provider that reaches 6421xxxxxxx?

Hi, my asterisk is set up with a pay-as-you-go Teliax account, and can dial out just fine to most numbers, but this cell phone number in New Zealand, 6421xxx, just rings and rings. Teliax support says: "Unfortunately, not all International Cell numbers can be dialed by Teliax users. Ther

Re: [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out

On Tue, Sep 19, 2006 at 04:46:13PM -0500, Lacy Moore - Aspendora wrote: >On 9/19/06, Jay R. Ashworth <[EMAIL PROTECTED]> wrote: > On Tue, Sep 19, 2006 at 04:59:37PM -0400, Mailing List wrote: > > What phones/system do they use? > I'm not sure it matters. > >It does matter if

Re: [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out

On Tue, 2006-09-19 at 16:46 -0500, Lacy Moore - Aspendora wrote: > It does matter if they are using Asterisk. I think we'd all like to > know how they are doing it. > > When I was using the chan_sccp driver, I was able to display that > information on the screen. I'm not sure the SIP protocol i

Re: [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out

On Tue, 2006-09-19 at 17:24 -0400, Jay R. Ashworth wrote: > On Tue, Sep 19, 2006 at 04:59:37PM -0400, Mailing List wrote: > > What phones/system do they use? > > I'm not sure it matters. > > Many commercial key systems display the destination extension user's > name on the originating set on inte

Re: [asterisk-users] DTMF Detection Problems with certain phones incoming zap channels

I did turn it on and off as it does not seem to make a difference. On 9/19/06, Jay R. Ashworth <[EMAIL PROTECTED] > wrote:On Tue, Sep 19, 2006 at 05:29:17PM -0400, asterisk asterisk wrote:> I'm having a problem with the autoattendant. It won't recognize the > DTMF signals from certain  people that

re: [asterisk-users] g729 and polycoms problem

Make sure the codec used by the Polycom will be only g729 via the phone's web interface, as far as I remember Polycom will try always to use ulaw or alaw first unless it is configured to use only or as first choice the g729 codec.Alyed Return-Path: <[EMAIL PROTECTED]> Tue S

Re: [asterisk-users] grandstream gxp 2000 does not display names when calling out

On Tue, 2006-09-19 at 13:45 -0700, Christopher Corn wrote: > michael, > at my real job, the phones display peoples names when calling out from > your phone. how is this done? And the use Asterisk there? SIP phones? Legacy PBX? Cisco Call Manager? -- Dr. Michael Neuhauser mailto:[EMA

Re: [asterisk-users] DTMF Detection Problems with certain phones incoming zap channels

On Tue, Sep 19, 2006 at 05:29:17PM -0400, asterisk asterisk wrote: > I'm having a problem with the autoattendant. It won't recognize the > DTMF signals from certain people that call in. I have relaxed DTMF, Are you sure? > Zapata.conf > > [channels] > signalling=fxs_ks [ ... ] > ;relaxdtmf=yes

Re: [asterisk-users] Problem with # locking up call

Likewise.On 9/19/06, marvin horst <[EMAIL PROTECTED]> wrote: On 9/19/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: Your phones do not have a native transfer feature?Our phones are a mix of Zap and SIP channels ___--Bandwidth and Colocation provi

Re: [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out

It does matter if they are using Asterisk.  I think we'd all like to know how they are doing it.   When I was using the chan_sccp driver, I was able to display that information on the screen.  I'm not sure the SIP protocol in Asterisk supports this (and don't know if the SIP protocol itself support

[asterisk-users] g729 and polycoms problem

Hi, I'm experiencing some problems with polycom phones, asterisk and g729 codec. As I understand, between polycom and polycom i can use g729 without license at all as long as I'm using codec_g729.so module (i'm using the Open Source Implementation ( http://www.readytechnology.co.uk/open/ipp-codec

[asterisk-users] Grandstream SX2000 attended tranfer

Hi all, could anyone share how to perform attended transfers with Asterisk and Grandstream SX2000's - we are able to perform blind transfers with no problem, but attended transfers fail - is it necessary to set two line identities on the phones to be able to do this? Appreciate all input, thanks -

[asterisk-users] DTMF Detection Problems with certain phones incoming zap channels

Hello, I'm having a problem with the autoattendant. It won't recognize the DTMF signals from certain people that call in. I have relaxed DTMF, upgraded Asterisk from 1.2 to 1.2.12 to 1.2.12.1 as well as the zaptel drivers. I have stopped X from running then only thing I didn't do that was on Dig

Re: [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out

On Tue, Sep 19, 2006 at 04:59:37PM -0400, Mailing List wrote: > What phones/system do they use? I'm not sure it matters. Many commercial key systems display the destination extension user's name on the originating set on internal/intercom calls, and some display the destination user's name on pho

Re: [asterisk-users] pickup call little complicated

See here:  http://linux.thorsten-knabe.de/asterisk/pickup.jsp On 9/19/06, Miloš Kocbek <[EMAIL PROTECTED]> wrote: The problem is that i want to be able steal a channel without any action at extension 100. I want to be able to dial a number and talk to whoever was exten 100 talking toI hope you und

Re: [asterisk-users] pickup call little complicated

The problem is that i want to be able steal a channel without any action at extension 100. I want to be able to dial a number and talk to whoever was exten 100 talking toI hope you understand what i want greetingsmk ___ --Bandwidth and Colocation provided

Re: [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out

- Original Message - From: Christopher Corn To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, September 19, 2006 4:48 PM Subject: Re: [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out im sorry if im not being clear. when im calling f

Re: [asterisk-users] Re: Mediatrix 1204 trix

Erik, I have tried it and it did NOT work, can you tell me where to enter that info? Have done it and it worked? On 9/19/06, Erik <[EMAIL PROTECTED]> wrote: mediatrix DOES support SIP Register, just enter authentication details and a registar server C F wrote: > Keep in mind that the Mediatrix

Re: [asterisk-users] grandstream gxp 2000 does not display names whencalling out

im sorry if im not being clear. when im calling from my gxp to the grandstream 100.the gxp doesn't pullup the users name from grandstream 100.   someone in another mail mentioned that this is not the way its supposted to work.   my office does this, when i dial someones number, it displays thei

Re: [asterisk-users] grandstream gxp 2000 does not display names when calling out

michael, at my real job, the phones display peoples names when calling out from your phone. how is this done?Michael Neuhauser <[EMAIL PROTECTED]> wrote: On Tue, 2006-09-19 at 11:53 -0700, Christopher Corn wrote:> ... i dial now from SIP phone to SIP phone. one being a grandstream> gxp >>> grands

Re: [asterisk-users] Asterisk AGI question

David R. wrote: > Can AGI be used to have a web application talk back and forth between > Asterisk and itself? What if the web application is on a separate box? Yes, if its on another box you should use FastAGI (you should do this anyway for performance reasons ;), see http://www.voip-info.org/wi

[asterisk-users] Asterisk AGI question

So, STDIN, STDOUT, and STDERR can be used to talk with .agi scripts to control the dialplan.Can AGI be used to have a web application talk back and forth between Asterisk and itself?  What if the web application is on a separate box? Thanks,David ___ --Ba

[asterisk-users] clustering asterisk is possible ?

Hello all, one of our customers is interested in deploy various asterisk in cluster environment. In a first look I don't see special problems for this (if I can provide a duplicate E1 input...), but I don't see problem for clusterizing asterisk. Is this ok ? Anyone has experience in cluster's of

Re: [asterisk-users] grandstream gxp 2000 does not display names whencalling out

- Original Message - From: Christopher Corn marco, can you explain what it is your recommending? i dial now from SIP phone to SIP phone. one being a grandstream gxp >>> grandstream 100. but the gxp, when dialing, doesn't see the name of the person at the grandstream 100. i believe thi

Re: [asterisk-users] Asterisk Appliance, will Asterisk Business Edition be mandatory?

Distribution channels aren´t being made to the public yet, other then direct from digium. Looks like they will be waiting 2 weeks before we hear anything else. (After VON) The questions you are asking, I dont believe have yet been confirmed by asterisk or digium. Though I am sure it is on there min

Re: [asterisk-users] grandstream gxp 2000 does not display names when calling out

On Tue, 2006-09-19 at 11:53 -0700, Christopher Corn wrote: > ... i dial now from SIP phone to SIP phone. one being a grandstream > gxp >>> grandstream 100. but the gxp, when dialing, doesn't see the > name of the person at the grandstream 100. i believe this should be > picked up from the asterisk

[asterisk-users] SPA 3102 does not even attempt to register

I don't see a "-- Saved useragent " line for the SPA 3102 device am trying to connect to Asterisk. I have similar configuration for the SPA 3102 as I have for another hard phone in the sip.conf file but the device (SPA 3102) does not even attempt to register. I have configured the device to re

Re: [asterisk-users] Problem with # locking up call

On 9/19/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: Your phones do not have a native transfer feature?Our phones are a mix of Zap and SIP channels ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSC

[asterisk-users] Pri Event 6 and 8

I was looking through my log files and keep seeing: PRI got event: 8 on Primary D-channel of span 3 on different spans. sometimes it is event 6. Does anyone know what causes this? ___ --Bandwidth and Colocation provided by Easynews.com -- asteri

Re: [asterisk-users] grandstream gxp 2000 does not display names when calling out

marco, can you explain what it is your recommending? i dial now from SIP phone to SIP phone. one being a grandstream gxp >>> grandstream 100. but the gxp, when dialing, doesn't see the name of the person at the grandstream 100. i believe this should be picked up from the asterisk server, but im no

Re: [asterisk-users] grandstream gxp 2000 does not display names when calling out

test it with someextension and using a sipphone like xlite or something else and you will be able to understand if it is a grandstream issue!hope it helpsOn 9/19/06, Christopher Corn <[EMAIL PROTECTED]> wrote: i have trixbox running, the latest version and when i make an outgoing call from this ph

[asterisk-users] grandstream gxp 2000 does not display names when calling out

i have trixbox running, the latest version and when i make an outgoing call from this phone it doesn't pick up the user's name, but instead display the number. is this a grandstream problem? or asterisk?   i did specify the user name from 'extension"  within trixbox. in matter of face, if i call

Re: [asterisk-users] Problem with # locking up call

Your phones do not have a native transfer feature? Anthony Cennami wrote: Which we use on all calls so that if someone wants to transfer said in/out call they can (such as a secretary placing a call for an executive, or a regular user transfering a call to a verification service/person.) On 9

[asterisk-users] Re: gTalk no audio issue

I should have posted the logs when the call is acceptedhere it is:-- SIP/5001-081ef020 answered Gtalk/guan.alex-e086[Sep 19 12:13:47] DEBUG[31226]: rtp.c:1402 ast_rtp_early_bridge: Channel 'Gtalk/guan.alex-e086' has no RTP, not doing anything[Sep 19 12:13:47] NOTICE[31226]: chan_gtalk.c:1331 g

[asterisk-users] gTalk no audio issue

Gang,With the latest code I am having audio issue with the chan_gtalk and wondering if anyone else had the same problem- I was able to set up the call between Asterisk and my gTalk account, but there was no audio - Looking closer, I am seeing these messages for an incoming call:    -- SIP/5001-081e

Re: [Asterisk-Users] How to learn or teach VoIP QoE

Mojo with Horan & Company, LLC wrote: I've heard a MOS of 4.4 is ulaw/alaw, presumably like a local exchange call through the pstn. testyourvoip.com tells me that the highest score available with G.729 is 4.2, which is pretty darn close to 4.4. Alaw and ulaw are about 4.4. ulaw on a robbed

Re: [asterisk-users] codecs/voicemail/DTMF

Use type=user for inbound and type=peer for outbound. Have different codec settings for each of them. Mr. Jones wrote: Hi Folks, We're trying to roll Asterisk out to production and are having a few complications. Most specifically we have G711 for our inbound origination, but would prefer G7

[asterisk-users] Repost: Register message received from realtime peer crashes Asterisk

When Asterisk (1.2.12.1) receives a SIP register message for a realtimepeer, the CLI reports "Disconnected from Asterisk server". Asterisk hasdisappeared:asterisk -rUnable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctlexist?)A look at the full log doesn't reveal much:Sep 17 06:1

[asterisk-users] Re: Semi-OT: SIP or IAX provider in the Boston area?

On 9/19/06, Erik Anderson <[EMAIL PROTECTED]> wrote: Hello - I'll be heading out to the Boston area next week to start up a branch office for my company. I'll be implementing an Asterisk box as part of their network infrastructure...so...does anyone have any recommendations on a good reliable SI

Re: [asterisk-users] When does Scalability requests Asterisk

On Tue, Sep 19, 2006 at 07:01:41PM +0200, Roy Sigurd Karlsbakk wrote: > The tests we've done shows that asterisk doing RTP bridging SIP/SIP > calls can handle up to approxmately 4-500 calls for a single Xeon 3.0 > before locking up, spending approx 60-70% system/kernel time, _not_ > usertime.

[asterisk-users] 404 not Found

Hello, I wish asterisk to forward the none local uri to an outbound proxy instead of send back 404 . I add this in extension.conf and outbound is a peer with an outbounproxy set to SER : [sip] exten => _.,1,NoOp(Incoming Call from house extension ${CALLERID} for [EMAIL PROTECTED]) exten => _.,

[asterisk-users] SIP "Lines" Example Citel

Anyone know how to setup the SIP lines on a Citel box so it can register with Asterisk. I keep getting "Unauthorized" and I have tried every different combination of settings that I can think of. I am not sure what fields are required or what information goes where in the Citel interface. T

[asterisk-users] codecs/voicemail/DTMF

Hi Folks, We're trying to roll Asterisk out to production and are having a few complications. Most specifically we have G711 for our inbound origination, but would prefer G729 for outbound termination, so far so good - it appears that dtmfmode=auto works in both cases. The area I'm having troub

Re: [asterisk-users] When does Scalability requests Asterisk

SM > Sorry, should have been a little more specific. I've had Asterisk running realtime SIP users/peers and SM > realtime sql calls from the dialplan (all with MySQL), and have had around 2.5k registered users and a SM > peak (that I recall) of around 500 concurrent calls. Wow that sounds pre

Re: [asterisk-users] Problem with # locking up call

Which we use on all calls so that if someone wants to transfer said in/out call they can (such as a secretary placing a call for an executive, or a regular user transfering a call to a verification service/person.) On 9/19/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: Anthony Cennami wrote:

[asterisk-users] mpg123

Hi all, I'm using * 1.0.9 which use mpg123 for music on hold. But sometimes starts eating up a lot of CPU. I sthere any alternative method to use moh without use mpg123? I tryied this http://astrecipes.net/?n=152 but i doesn't wotks for me.   Anyone can help me pls ?   Thanks in advance.

Re: [Asterisk-Users] How to learn or teach VoIP QoE

I've heard a MOS of 4.4 is ulaw/alaw, presumably like a local exchange call through the pstn. testyourvoip.com tells me that the highest score available with G.729 is 4.2, which is pretty darn close to 4.4. I don't know why I think this (or why I've heard it (or if it's right)) but I think g

Re: [asterisk-users] Alcatel OXO Sip

On Tue, Sep 19, 2006 at 02:53:17PM +0200, Christian Gatti wrote: > It the question why does asterisk has problems with SIP/2.0/udp or > SIP/2.0/UDP > > if (strcasecmp(via, "SIP/2.0/UDP")) { > ast_log(LOG_WARNING, "Don't know how to respond via '%s'\n", via); > return -1; > } > > This

Re: [asterisk-users] Problem with # locking up call

Anthony Cennami wrote: I've also found similar problems with blindxfer -- such as when someone is attempting to interact with an IVR using a '#' option. By default # seems to transfer a call, but if you have blindxfer enabled with '#1' or ##, then Asterisk hears the first # and waits for a follo

Re: [asterisk-users] Polycom default handset volume

I think the following should resolve your problem, if I'm understanding it correctly:In the sip.cfg file, look at setting the following:1= remember last setting, 0=return to defaultThis will make the phone remember your setting and will not reset the setting every time you need to make a new call.

RE: [asterisk-users] Digium GUI?

I am talking about the GUI that was announced as part of the new Asterisk Appliance. Sounds like it is going to be a full featured GUI like FreePBX. -Original Message- From: Noah Miller [mailto:[EMAIL PROTECTED] Sent: Monday, September 18, 2006 8:11 PM To: Asterisk Users Mailing List

RE: [asterisk-users] Polycom default handset volume

I don't think you can set a default volume, but you can configure the handset (and headset) volume to persist between calls. Look for 'persist' in sip.cfg or phone1.cfg. Doug. -Original Message- From: Damon Estep [mailto:[EMAIL PROTECTED] Sent: Tue 9/19/2006 9:23 AM To: Asteris

Re: [asterisk-users] Problem with # locking up call

I've also found similar problems with blindxfer -- such as when someone is attempting to interact with an IVR using a '#' option.  By default # seems to transfer a call, but if you have blindxfer enabled with '#1' or ##, then Asterisk hears the first # and waits for a following DTMF. At the moment

[asterisk-users] Call forward with CFU?

Hi all I've got the following message from the telco regarding call forward number presentation. Can someone please help me decipher this? I don't understand shit about this :P roy Roy , Below is an extract taken from a working scenario of the CFU (A B - B C) functionality problem :

Re: [asterisk-users] When does Scalability requests Asterisk

Never mind my previous question. I see that Ryan already asked that and you responded. It does sound like a pretty neat project. Would love to hear more once you finish the project that you are working on currently. - AK ___ --Bandwidth and Colocation pro

Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

[EMAIL PROTECTED] wrote: > Can you explain your design in a little more detail? What kind of hardware > did you use to get over 1k users on a single box and 500 concurrent calls? > Sounds like a very interesting medium-large scale implementation that > others could learn from. > > thanks, > Ryan

[asterisk-users] Semi-OT: SIP or IAX provider in the Boston area?

Hello - I'll be heading out to the Boston area next week to start up a branch office for my company. I'll be implementing an Asterisk box as part of their network infrastructure...so...does anyone have any recommendations on a good reliable SIP or IAX provider? I'd need DIDs for incoming calls a

Re: [asterisk-users] fast SIP failover (outgoing sIP requests) wi th 1.2

Michiel van Baak wrote: On 16:44, Tue 19 Sep 06, Klaus Darilion wrote: Hi! I've tried it but apparently chanisavail does not work with "non-local" SIP peers. In sip.conf try to add qualify=yes to the remote office part This will make the sip peer dissapear when it's more then 2 seconds awa

Re: [asterisk-users] When does Scalability requests Asterisk

SM > Sorry, should have been a little more specific. I've had Asterisk running realtime SIP users/peers andSM > realtime sql calls from the dialplan (all with MySQL), and have had around 2.5k registered users and a SM > peak (that I recall) of around 500 concurrent calls.Wow that sounds pretty neat

[asterisk-users] Polycom default handset volume

I had read a post somewhere that there is an XML parameter for the Polycom config files for default handset volume, but I can not locate it again.   Anyone know what it is?   I want to set the default handset volume higher on some phones, despite the ADA hearing aid warning in the admin

RE: [asterisk-users] fast SIP failover (outgoing sIP requests) with 1.2

Try taking to 90 second timeout off Change exten => _[1-9].,4,Dial(SIP/${enumresult},90) to exten => _[1-9].,4,Dial(SIP/${enumresult}) a btter method is to set up each office as a unique peer with qualify = yes and then add the peer name to the dial string, like dial(SIP/[EMAIL PROTECTED]) if

RE: [asterisk-users] Digium GUI?

Could a carrier use it to provide service to it's customers, such that each customer only had access to their own content? -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Tue 9/19/2006 5:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc:

Re: [asterisk-users] transcoding error?

Damon Estep wrote: Anyone encountered this on yet? WARNING[23251]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Started after an upgrade from CVS 8/2005 to current 1.2.12.1 If I had a reference for what frame types 4 and 6

Re: [asterisk-users] fast SIP failover (outgoing sIP requests) wi th 1.2

On 16:44, Tue 19 Sep 06, Klaus Darilion wrote: > Hi! > > I've tried it but apparently chanisavail does not work with "non-local" > SIP peers. > In sip.conf try to add qualify=yes to the remote office part This will make the sip peer dissapear when it's more then 2 seconds away (probably down).

Re: [asterisk-users] Aastra 9133i and Atcom AT-320 - Comments please

We tested a couple 9133i, dont remember the specifics right now but we stopped as there was some inconsistency in provisioning. I was very optimistic as I like the look and feel. We did deploy a couple 480iCT which worked very well - when they worked. But they keep locking up and freezzing

Re: [asterisk-users] RE: FollowMe question

Rushowr wrote: Hall, Eric M. wrote: I got the config working. Not sure if someone has pre-recorded sounds for this app or not. Looked all over for them and I'm unable to locate them.If anyone has sound file they would like to share that would help me greatly. I wouldn't mind a shot a

[asterisk-users] transcoding error?

Anyone encountered this on yet?   WARNING[23251]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)   Started after an upgrade from CVS 8/2005 to current 1.2.12.1   If I had a reference for what frame types 4 and 64 are I might be

Re: [asterisk-users] polycom 501 digitmap

the digitmap only tells the phone when to send the digits it has collected. They have no digit substitution feature. This would be done within your * dialplan On Sep 19, 2006, at 7:57 AM, Jordan Novak wrote: This is really starting to get to me. I have deleted this field in the phones per

Re: [asterisk-users] Polycom Expansion Module

Per 2.0.1 release notes 13315: Increased the maximum number of buddies to 8 for all platforms except SoundPoint IP 600 and 601 which can watch 48 buddies On Sep 18, 2006, at 10:35 PM, Douglas Garstang wrote: 48 was the limit on the number of speed dial entries that you could have in the

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