Re: [Asterisk-Users] app_queue on 1.2 ?

Philipp von Klitzing wrote: Hi Florian, have you check that this is not connected to bug 5810? Just a guess. Checked and verified, the patch from 5810 is properly applied in my 1.2.1 checkout and the issue remains with and without the /n. Any hints ? Thanks, Florian ___

Re: [Asterisk-Users] app_queue on 1.2 ?

Hi Philipp. Philipp von Klitzing wrote: Hi Florian, have you check that this is not connected to bug 5810? Just a guess. Thanks for the suggestion, but I don't think so - this is fresh a 1.2.1 svn checkout. I will see if it gets cleared without the /n Florian __

RE: [Asterisk-Users] Asterisk Hardware recomendation

I will be using IP Hard and soft phones all the way, so everything will be on Ethernet, for this I want 1Gbit incoming and 1Gigabit outgoing, looking for atleast 500 simultaneous calls, with 2 3.6Ghz processors I think I could squize out more then that. For codec I want to use g711 on the outgoin

[Asterisk-Users] Asterisk as a gatekeeper

Hello,   Right now i'm trying to set-up a gatekeeper and i'm having a hardtime doing it, what i'm thinking is instead of having a gatekeeper i'll use the asterisk to be a gatekeeper. Can the asterisk be a gatekeeper?   Thanks a lot, Rommel _

Re: [Asterisk-Users] asterisk with EWSD v16

I am using EWSD's PRIs and I am not having this problem my configs are Zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone = us defaultzone=us Zapata.conf [channels] language=en context=ext-acd switchtype=euroisdn signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes g

[Asterisk-Users] Recording Volume on Zap Channel

Hi All I have a call center working on 8 FXO Channels, everything working fine except one little problem, I am using asterisk queues with monitor-format = wav49 and monitot-join = yes asterisk is recording all conversations but the problem is that the volume of Zap Channel is too low in most o

[Asterisk-Users] asterisk with EWSD v16

if any EWSD guru out there..please help ??? Hello all, I am running Asterisk with Digium E1 card with zaptel, libpri, asterisk cvs v1-2. My server is interfaced with EWSD v16 using a PRI on E1. I am running into a problem that at my telco's end alot of trunks are getting BPRM (Block perm

Re: [Asterisk-Users] Sip behind the NAT

Hi: i added these two lines to my general context ,but nothing happened the same result the sound came in one way for 3 seconds and stopped but it didnt hangup. --- Jeffery Chen <[EMAIL PROTECTED]> wrote: > If your Astersik server behind NAT too, your need > modify SIP.conf like > this > > e

[Asterisk-Users] RE:how to listen voicemail messages

 hi all,   I have made two voice mail boxes for 2 users on asterisk server (/var/spool/asterisk/voicemail/testmail/inside this 2 boxes for 2 users). i have made following settings in voicemail.conf : [testmail] vipul=>,vipul patel, [EMAIL PROTECTED] tejas=>,tejas shah,[EMAIL

RE: [Asterisk-Users] Asterisk (OH323) - gnugk connection

hi Leandro Tenorio, What will be the gnugk configuration to route the call to Asterisk? -- Thank You, Code Lover ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://l

[Asterisk-Users] Door Phones

Hola Anton Nosotros distribuimos los porteros de ITS (pancode, pancam, etc.) en México los cuales funcionan bien con Asterisk, si te interesa enviame un mensaje. Héctor Garza Cantú Nordata S.A. de C.V. Tel. (81) 8372-5982 / 8375-4871 Cel. (81) 8010-7960 BEGIN:VCARD VERSION:2.1 N:Garza Cantú;Héct

[Asterisk-Users] ZT_CHANCONFIG failed on channel 1: No such device or address

Hi, I have my TDM2412E card installed on CentOS, with the driver up & running. This card has 1xQuad FXS at the front (ports 1-4) and  2 x Quad FXOs at the back (ports 17-24) I'm editing the /etc/zaptel.conf file for the fxoks and fxsks parameters, but it seems that whatever values I give fo

Re: [Asterisk-Users] Aastra 9133i Configurations - are the file names to be lower case or upper case or does it matter?

Lists wrote: According to the wiki page http://www.voip-info.org/tiki-index.php?page=Aastra+480i+Configuration it shows lowercase file name and then there is a comment at the bottom that it needs to be capitalized. I have tried it both ways with no luck. Could someone comment on which way the c

[Asterisk-Users] local not ring,,,,

hello, guyes..   Why i dial(SIP/EXT ,r),,, the IAXOMMM can not hear ring my config file as below...       [sip-ext]include => oh323 exten => _11,1,Answerexten => _11,n(dial),Dial(SIP/${EXTEN},15,r) exten => _11,n,Goto(s-${DIALSTATUS},1) exten => s-CHANUNAVAIL,1,Playback(invalid) ex

Re: [Asterisk-Users] ADIT 600 T1 with DNIS digits problem

Just to follow up in case others run into this, the problem was with the zaptel 1.0.10 drivers, I had to add: options wct4xxp vpmsupport=0 to /etc/modprobe.conf, with vpmsupport enabled the last DNIS DTMF digit kept getting delayed (it could/would show up on the next call or as the extension dialed

[Asterisk-Users] London DID 30 Cents a Number available - 60 Channels, ULAW

Dear All,DIDX.net  is pleased to announceLondon DID's starting 30 cents eachChannels 60Codec: Ulaw or any other   Visit www.didx.netFor a complete list, visit www.didx.org/did/36 hrs money back gurantee.Rehan  Super Technologies Inc., Pensacola, Floridahttp://www.SuperTec.com - Technologies from to

[Asterisk-Users] Aastra 9133i Configurations - are the file names to be lower case or upper case or does it matter?

According to the wiki page http://www.voip-info.org/tiki-index.php?page=Aastra+480i+Configuration it shows lowercase file name and then there is a comment at the bottom that it needs to be capitalized. I have tried it both ways with no luck. Could someone comment on which way the cfg files need t

[Asterisk-Users] RE:IConnecthere dial out problems

On Wed, 7 Dec 2005 13:32:09 -0600 Dennis Gilmore wrote: >Im having no issues with my outgoing calls  but my incoming call registration >keeps locking up.  it seems after awhile it stops sending reregistration >packets. >my extentions.conf has >; ** Dial Out iconnecthere

[Asterisk-Users] HOW TO: CDR Customer IP address where call came in from

  HI Does any one know how to record the IP address of the call that is coming in on the asterisk.   I dont see the ip in the Master.csv but you can view the IP when the call comes in on the CLI Window.   I am guessing there must be a command or a way to record this ip in your CDR using AGI, we are

Re: [Asterisk-Users] Recording a call

Noah Silverman wrote: Moj, It is set as the default. *1 When I dial "*1" I actually see "user pressed *1 to start recording". I then hear a beep. The system DOES create in and out files and then combines them to a single file when the call is done. The problem is that the file is SILEN

Re: [Asterisk-Users] IAX2: Don't know any of 0xf800 formats

Ryan Courtnage wrote: [guest] type=user context=incoming ; Asterlink [1234567] context=incoming type=user disallow=all allow=g726 If you have two users defined that don't require authentication, IP addresses or IP address masks, you cannot expect Asterisk to predictably choose one or the ot

Re: [Asterisk-Users] Asterisk Hardware recomendation

Krystian - what kind of port density are you aiming for? Will you be running analog or digital? Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Krystian Filiks wrote:

Re: [Asterisk-Users] Recording a call

Moj, It is set as the default. *1 When I dial "*1" I actually see "user pressed *1 to start recording". I then hear a beep. The system DOES create in and out files and then combines them to a single file when the call is done. The problem is that the file is SILENT Thanks, -N On Dec

Re: [Asterisk-Users] Recording a call

Does your features.conf specify a custom setting for automon? If it does, is that what you were dialing? ie. [featuremap] automon => *# Moj Noah Silverman wrote: Tried that, Doesn't seem to do anything... -N On Dec 7, 2005, at 3:38 PM, Time Bandit wrote: I'm trying to figure out how

[Asterisk-Users] Re: Call Recording

OK, The plot thickens. I've managed to get everything configured so that the system WILL create a file. The problem is that the file just contains silence. If I have a 10 second call that I record, I just get a wav file with 10 seconds of silence. Anybody have an clues?? Thanks, -N

Re: [Asterisk-Users] Announcement only Devices that work with Asterisk for Dedicated 24/7 Conferencing

OK, sorry if thats what he wanted, then what I suggest to use a paging system, will sure work. In fact any paging system will work, even one that doens't allow input. Just look it up on the wiki. On 12/7/05, Rusty Dekema <[EMAIL PROTECTED]> wrote: > Huh? > > I think the original poster might be t

Re: [Asterisk-Users] Sip behind the NAT

If your Astersik server behind NAT too, your need modify SIP.conf like this   externalIP= x.x.x.x localnet= x.x.x.   hope this can help you  On 12/8/05, Moises Silva <[EMAIL PROTECTED]> wrote: what type of NAT do you have? sync? full cone? cone restricted, port restricted?any messages in as

[Asterisk-Users] Asterisk Hardware recomendation

Hello asterisk people!   I have been running a test * server a P III box for some time now and it’s been rock stable.   Now I’m looking to build a production system with as big capacity as possible on 2 Xeon 3.6Ghz processors.   I’m wondering what you are thinking about Supermicro 601

Re: [Asterisk-Users] Recording a call

Hi, I tried setting verbose to 50 and never got any feedback on the CLI about a pressed key... -N On Dec 7, 2005, at 3:53 PM, Time Bandit wrote: That helps, but I'm still missing one piece. I want to be able to press a button during the call to start and stop recording. I tried using:

Re: [Asterisk-Users] A company that sells Toll Free Number in USA

If only looking for 800 numbers; http://www.nufone.net I use them for Toll Free, and they have been good for me. JR On 12/7/05, Alvaro Parres <[EMAIL PROTECTED]> wrote: > Hi any one can recommend me a company in the USA that can sell me a Toll > Free Number > and send me the call via IP. > >

Re: [Asterisk-Users] Announcement only Devices that work with Asterisk for Dedicated 24/7 Conferencing

On 12/7/05, Kanuri, Seshu (Company IT) <[EMAIL PROTECTED]> wrote: > > Does anyone know an IP Phone or Device that works with Asterisk as an > announcement only device with a loud speaker and that is online forever and > will not hungup for any reason and even if it hangsup, it will reboot itself >

Re: [Asterisk-Users] Announcement only Devices that work with Asterisk for Dedicated 24/7 Conferencing

Huh? I think the original poster might be talking about the opposite of what you are talking about. I understood him to mean that he wants a device that stays connected to an Asterisk extension and plays any audio received over the connection through a loudspeaker. I don't see how a cassette play

Re: [Asterisk-Users] A company that sells Toll Free Number in USA

Many companies can do that including the following: www.teliax.com www.gafachi.com www.broadvoice.com www.sellvoip.net Of these I would recommend Teliax, although I have not had a lot of experience with any of them. -Rusty On 12/7/05, Alvaro Parres <[EMAIL PROTECTED]> wrote: Hi any one can reco

Re: [Asterisk-Users] app_queue on 1.2 ?

Hi Florian, have you check that this is not connected to bug 5810? Just a guess. Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

Re: [Asterisk-Users] Recording a call

> That helps, but I'm still missing one piece. > > I want to be able to press a button during the call to start and stop > recording. > I tried using: > > exten => s,1,Dial(101,20,Ww) > > But it doesn't seem to do anything. On the console, put verbose to something like 50 (set verbose 50) and you s

Re: [Asterisk-Users] Sip behind the NAT

what type of NAT do you have? sync? full cone? cone restricted, port restricted? any messages in asterisk verbose console? best regardsOn 12/7/05, chawki hammoud <[EMAIL PROTECTED]> wrote: Hi list:i have an asterisk box behind the NAT ,when i try tosend calls through Sip to the voip provider serve

Re: [Asterisk-Users] Recording a call

Tried that, Doesn't seem to do anything... -N On Dec 7, 2005, at 3:38 PM, Time Bandit wrote: I'm trying to figure out how to setup "live" recording of a phone call. I've read all the docs at the wiki, but can't seem to figure out how to implement it. I'm running asterisk 1.2 I have the Po

Re: [Asterisk-Users] Recording a call

Thanks, That helps, but I'm still missing one piece. I want to be able to press a button during the call to start and stop recording. I tried using: exten => s,1,Dial(101,20,Ww) But it doesn't seem to do anything. -N On Dec 7, 2005, at 3:29 PM, Philip Edelbrock wrote: Noah Silverman

Re: [Asterisk-Users] Announcement only Devices that work with Asterisk for Dedicated 24/7 Conferencing

For starters, an old fashioned cassette player that holds the button in with auto reverse will do, since when the power is out it will still play it. Connect that to an ATA (radio shack sells a plug that will connect to a phone line and take the input of a regular headphone jack to the phone line).

Re: [Asterisk-Users] Recording a call

> I'm trying to figure out how to setup "live" recording of a phone call. > I've read all the docs at the wiki, but can't seem to figure out how > to implement it. > > I'm running asterisk 1.2 > I have the Polycom IP500 SIP phones. > > In a perfect world, I would dial something to start recording,

[Asterisk-Users] Lucent TNT / Asterisk help

Title: Message Anyone out there have any experience with setting up asterisk and a Lucent TNT...I need some help with setup and configuration...I have no experience with Lucent...I have referenced the wiki on voip-info.org but does not do much good since i also need help configuring the E1's

[Asterisk-Users] A company that sells Toll Free Number in USA

Hi any one can recommend me a company in the USA that can sell me a Toll Free Number and send me the call via IP. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] IAX2: Don't know any of 0xf800 formats

I seem to recall a similar issue where the guest section HAD to be the last section of iax.conf. It's been that way for years. Ryan Courtnage wrote: On Wed, 2005-07-12 at 16:41 -0600, Eric "ManxPower" Wieling wrote: Sounds like you have an allow=all somewhere. Thanks for the response. No "

Re: [Asterisk-Users] IAX2: Don't know any of 0xf800 formats

On Wed, 2005-07-12 at 16:41 -0600, Eric "ManxPower" Wieling wrote: > Sounds like you have an allow=all somewhere. Thanks for the response. No "allow=all" anywhere. You got me thinking though, and it appears that * has a nasty little parsing problem or something: Here's my iax.conf ---begin---

[Asterisk-Users] Announcement only Devices that work with Asterisk for Dedicated 24/7 Conferencing

Does anyone know an IP Phone or Device  that works with Asterisk as an announcement only device with a loud speaker and that is online forever and will not hungup for any reason and even if it hangsup, it will reboot itself to connect to a VOIP Server with a given configuration to receive

RE: [Asterisk-Users] Help iaxmodem

Hi: I want to use the same phone number for the fax and voice conversations. How do I redirect a call to the iaxmodem extension? Should my VOIP provider support the slinear codec? Thanks Miguel -Original Message- From: Miguel Soto [mailto:[EMAIL PROTECTED] Sent: Thursday, December 01,

Re: [Asterisk-Users] IAX2: Don't know any of 0xf800 formats

Sounds like you have an allow=all somewhere. Ryan Courtnage wrote: Hi all, I'm finding with Asterisk 1.2.1 (and 1.2.0) that when connecting over an unauthenticated IAX2 connection (ie: as [guest] in iax.conf), a codec will always fail to be negotiated (see trace snippet below). The problem app

[Asterisk-Users] Asterisk 1.2.1 and queue_log

Upgrading from Asterisk 1.0.x to the new Asterisk 1.2.x branch and I noticed when running Asterisk that it does not like the line in logger.conf that let Asterisk know to rotate the queue_log. It was the simple line: queue_log => Now in 1.2.x, Asterisk issues a warning about the line...is th

[Asterisk-Users] Recording a call

Hello, I'm trying to figure out how to setup "live" recording of a phone call. I've read all the docs at the wiki, but can't seem to figure out how to implement it. I'm running asterisk 1.2 I have the Polycom IP500 SIP phones. In a perfect world, I would dial something to start recording, an

Re: [Asterisk-Users] UK ISDN2e with DDI?

David Cook wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dogers Sent: 07 December 2005 16:24 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK ISDN2e with DDI? Quoting John Darago

Re: [Asterisk-Users] Connecting asterisk over consumer wifi network

I was thinking ethernet-over-power myself, but I haven't tried it yet That plus those 8 port netgear POE switches might work well. PaulH - Original Message - From: "Chris Bagnall" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Thursday, De

RE: [Asterisk-Users] UK ISDN2e with DDI?

> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > David Cook > Sent: 07 December 2005 21:26 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] UK ISDN2e with DDI? > > Try adding the following to your hands

[Asterisk-Users] AstManProxy Segmentation Faults

List users, I am experiencing segmentation faults in AstManProxy. If anyone could help me identify their source, it would be appreciated. The pertinent information is below. Please let me know if you need any more. Asterisk Version Asterisk ABE-A.2-beta AstManProxy Version

Re: [Asterisk-Users] Unable to compile zaptel / ztdummy

Hi,Have you followed the instructions outlined in README.udev ?HTH,KunalOn 12/8/05, Insider KT < [EMAIL PROTECTED]> wrote: Hi. I just upgraded my asterisk server to a better P4 2.6 Ghz. I thought everything went smooth until someone tried the Meetme. I seems the ztdummy won't compile on the

RE: [Asterisk-Users] UK ISDN2e with DDI?

> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Dogers > Sent: 07 December 2005 16:24 > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - > Non-Commercial Discussion > Subject: Re: [Asterisk-Users] UK ISDN2e with DDI? > > Quoting John Daragon <[

Re: [Asterisk-Users] Door Phones

We have the PanCode Door Phones which works with asterisk via the ata box. If you are interested please contact me off list. I am sure it will work anywhere. C F wrote: I'm not in Mexico, but I'm sure what I use here works in Mexico as well (BTW, it's on the wiki). I have successfuly used:

[Asterisk-Users] IAX2: Don't know any of 0xf800 formats

Hi all, I'm finding with Asterisk 1.2.1 (and 1.2.0) that when connecting over an unauthenticated IAX2 connection (ie: as [guest] in iax.conf), a codec will always fail to be negotiated (see trace snippet below). The problem appears to be specific to only unauthenticated IAX2 connections. Authent

[Asterisk-Users] Re: Unable to compile zaptel / ztdummy

On Wed, Dec 07, 2005 at 09:20:38PM +0100, Insider KT wrote: Hi. I just upgraded my asterisk server to a better P4 2.6 Ghz. I thought everything went smooth until someone tried the Meetme. I seems the ztdummy won't compile on the new server. I am running Mandriva 2006 on the new server. After dow

Re: [Asterisk-Users] Unable to compile zaptel / ztdummy

Insider KT wrote: Hi. I just upgraded my asterisk server to a better P4 2.6 Ghz. I thought everything went smooth until someone tried the Meetme. I seems the ztdummy won't compile on the new server. I am running Mandriva 2006 on the new server. After downloading in /usr/src/ I uncommented the

[Asterisk-Users] Sip behind the NAT

Hi list: i have an asterisk box behind the NAT ,when i try to send calls through Sip to the voip provider server the call is answered but in a one way calling,I hear the voice of the other side just for 4 seconds and then stop but the call do not hangup. my sip.conf is: [voip provider] type=peer

Re: [Asterisk-Users] Unable to compile zaptel / ztdummy

On Wed, Dec 07, 2005 at 09:20:38PM +0100, Insider KT wrote: > Hi. I just upgraded my asterisk server to a better P4 2.6 Ghz. > I thought everything went smooth until someone tried the Meetme. > I seems the ztdummy won't compile on the new server. > > I am running Mandriva 2006 on the new server. >

[Asterisk-Users] app_queue on 1.2 ?

Hi We're trying to migrate our platform from 1.0 to 1.2 and we're seeing some oddness in app_queue. We use local_channels a lot for things like persistent agents, call-forwarding on agents and such. Now on our 1.2 server we notice that the queue is listing all members as 'Invalid' (thus any

[Asterisk-Users] Unable to compile zaptel / ztdummy

Hi. I just upgraded my asterisk server to a better P4 2.6 Ghz. I thought everything went smooth until someone tried the Meetme. I seems the ztdummy won't compile on the new server.   I am running Mandriva 2006 on the new server. After downloading in /usr/src/ I uncommented the #ztdummy to z

RE: [Asterisk-Users] Static on inside end of conversation

Correct. The issue is that most of the echo is between internal stations. SIP -> SIP. The users with the system using the sipura's don't report any echo when calling outside the office or receiving a call. The users with the system using the audiocodes report an echo for the first 1 - 2 sec

Re: [Asterisk-Users] Door Phones

I'm not in Mexico, but I'm sure what I use here works in Mexico as well (BTW, it's on the wiki). I have successfuly used: Valcom VikingElectronics doorfonebell I might have spelling wrong on the last one. Too lazy to look it up on the Wiki :) On 12/7/05, Anton Krall <[EMAIL PROTECTED]> wrote: > G

Re: [Asterisk-Users] Can Asterisk act as a media gateway?

Ken D'Ambrosio wrote: I've got an account that's looking at doing some cable/VoIP integration. They were wondering if it were possible to set up something like this: PSTN (T1) -> Asterisk -> (some VoIP protocol, probably SIP) -> Siemens soft switch -> their product It sure sounds nice in t

Re: [Asterisk-Users] Static on inside end of conversation

Jeff: I'm not an Asterisk person but I play one on TV :-) Actually I just use it for voicemail and ancillary services from my SER proxy so take this message with the appropriate caution. I'd look at and possibly "tweak" parameters in the phone.conf file. echocancel, txgain and rxgain look in

[Asterisk-Users] Door Phones

Guys, Im wondering, is anybody in Mexico using any kind of door phone with asterisk? Please drop me a note. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.d

Re: [Asterisk-Users] Static on inside end of conversation

The device that interfaces to the PSTN is the interface that must cancel echo. If I read your post correctly, that is the SAP-3000 and the Audiocodes boxes in your case. Jeff Busch wrote: Update on this... And it is still not solved. This is actually fairly interesting. I have two installat

Re: [Asterisk-Users] IConnecthere dial out problems

On Wednesday 07 December 2005 11:10, John Voss wrote: > I can't seem to get my outgoing connections to work with IConnecthere. At > one time it did with v1.0 > > I can register and receive calls just fine. But can't make them. > > Ultimately, the trace ends with a "400 Bad Request" error when you d

RE: [Asterisk-Users] Static on inside end of conversation

Update on this... And it is still not solved. This is actually fairly interesting. I have two installations at a construction company. They are both running similar class machines (I was wrong in my initial post) they are: System "A" 2.4 ghz Celeron 1 gb RAM IDE Drives [EMAIL PROTECTED] 1.13 (A

[Asterisk-Users] SIP Video Recording

Is is possible to record video and audio when using SIP with video? Regards, Chris___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/list

[Asterisk-Users] TNT / Asterisk Help

Anyone out there have any experience with setting up asterisk and a Lucent TNT...I need some help with setup and configuration...I have no experience with Lucent...I have referenced the wiki on voip-info.org but does not do much good since i also need help configuring the E1's... any help would be

[Asterisk-Users] Can Asterisk act as a media gateway?

I've got an account that's looking at doing some cable/VoIP integration. They were wondering if it were possible to set up something like this: PSTN (T1) -> Asterisk -> (some VoIP protocol, probably SIP) -> Siemens soft switch -> their product It sure sounds nice in theory, but I've never t

RE: [Asterisk-Users] Nortel Meridian Option81C to TE405P

Yes, according to the document link that you provided and most other sources, a T1 crossover cable is required to connect the Nortel Meridian to an Asterisk server. Here is a summary of the important settings that I have: Nortel Meridian Asterisk Digium TE405P

[Asterisk-Users] Recording a call

Hello, I'm trying to figure out how to setup "live" recording of a phone call. I've read all the docs at the wiki, but can't seem to figure out how to implement it. I'm running asterisk 1.2 I have the Polycom IP500 SIP phones. In a perfect world, I would dial something to start recording, an

Re: [Asterisk-Users] FAX

Yes there is, using TDM, but not VoIP. On 12/7/05, Alejandro Vargas <[EMAIL PROTECTED]> wrote: > 2005/12/6, C F <[EMAIL PROTECTED]>: > > Yeah, it shoud NOT work 100% of the time (maybe not even 50%) > > Then... ¿there is not any way to connect a hardware fax to an asterisk pbx? > > -- > Alejandro

Re: [Asterisk-Users] FAX

On 12/7/05, Bartosz Piec <[EMAIL PROTECTED]> wrote: > C F wrote: > > Yeah, it shoud NOT work 100% of the time (maybe not even 50%) > > So, are there any IP faxes? Sure, it's called email. ___ --Bandwidth and Colocation provided by Easynews.com -- Asteri

[Asterisk-Users] chan_bluetooth Audio Sensitivity

Hello all: I'm currently running the latest version of asterisk, and using chan_bluetooth I am using the usb/bluetooth dongle from Compusa: http://www.compusa.com/products/product_info.asp?product_code=312330&pfp=SEARCH I am also using the M2500 Plantronics headset - 29.95 from Frys things are

Re: [Asterisk-Users] Asterisk 1.2.1 released

http://ftp.digium.com/pub/telephony/asterisk/ChangeLog-1.2.1 On 12/7/05, Jared Armstrong <[EMAIL PROTECTED]> wrote: > Its also on www.asterisk.org. > > > Jared Armstrong > > -Original Message- > From: Remco Barende [mailto:[EMAIL PROTECTED] > Sent: Wednesday, December 07, 2005 12:39 PM > T

RE: [Asterisk-Users] Asterisk 1.2.1 released

Its also on www.asterisk.org. Jared Armstrong -Original Message- From: Remco Barende [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 07, 2005 12:39 PM To: Asterisk Users List Subject: [Asterisk-Users] Asterisk 1.2.1 released It seems that Asterisk 1.2.1 is on the Digium FTP, but n

[Asterisk-Users] Zaphfc as a timing source?

Hello all, I know the TDM cards (and I assume the TE cards) provide a timing source to be used for IAX trunking etc., but is it possible to use a BRI card running under zaphfc as a timing source, or should one run ztdummy as well? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Mi

[Asterisk-Users] Asterisk 1.2.1 Released

We are proud to announce that Asterisk 1.2.1 has been released! This release of Asterisk contains a number of bug fixes over version 1.2.0. See the ChangeLog at http://ftp.digium.com/pub/telephony/asterisk/ChangeLog-1.2.1 for more details. It is available from the ftp.digium.com FTP servers, as

[Asterisk-Users] Lucent TNT / Asterisk Help

Anyone out there have any experience with setting up asterisk and a Lucent TNT...I need some help with setup and configuration...I have no experience with Lucent...I have referenced the wiki on voip-info.org but does not do much good since i also need help configuring the E1's... any help would be

[Asterisk-Users] VoIP US - Toll Free Origination / Termination Providers

I'm looking for a provider that can offer VoIP origination and termination in the domestic US and Puerto Rico. To be more exact, Toll Free numbers origination is a must. I'm looking for a block of 100 domestic toll free numbers and 100 local DIDs. Estimated traffic is about 100K origination

[Asterisk-Users] Asterisk 1.2.1 released

It seems that Asterisk 1.2.1 is on the Digium FTP, but no posts to the users lists, nothing in the wiki? Everybody still asleep? Looking forward to the changelogs :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing li

Re: [Asterisk-Users] Error when compiling asterisk

jourdan lemieux wrote: I am compiling a app file writting in C for asterisk and I am getting the following errors: ../include/asterisk/file.h:27:2: #error You must include stdio.h before file.h! Any help please on this!! How much clearer can that be? Your source file is out of date and

Re: [Asterisk-Users] Error when compiling asterisk

I am compiling a app file writting in C for asterisk and I am getting the following errors: ../include/asterisk/file.h:27:2: #error You must include stdio.h before file.h!In file included from app_akEventsProxy.c:17:../include/asterisk/file.h:56: error: syntax error before '*' token../include/aste

Re: [Asterisk-Users] UK ISDN2e with DDI?

Dogers wrote: Quoting John Daragon <[EMAIL PROTECTED]>: snip... When you say "ringtones", do you mean "sounds like a UK phone when it rings", or "sounds like a UK phone when we ring someone else" ? It does actually sound okay when we ring someone else, but when it rings, it has the long

RE: [Asterisk-Users] UK ISDN2e with DDI?

The ringtone on your Grandstreams is indeed set in the phone itself. I think they hold up to 4 ringtones (default, custom 1 2 3) which can be configured either per "line" or different rings on different caller ID. Grandstream have a freely available utility to convert PCM ringtones into the necessa

[Asterisk-Users] IConnecthere dial out problems

I can't seem to get my outgoing connections to work with IConnecthere. At one time it did with v1.0 I can register and receive calls just fine. But can't make them. Ultimately, the trace ends with a "400 Bad Request" error when you do a SIP debug. Has anyone got it to work with v1.2? Don't kno

RE: [Asterisk-Users] Nortel Meridian Option81C to TE405P

Can you do a ISDN message trace in LD 96 on the M1 when you try to bring up the D-Channel? LD 96 enl msgo 10 enl msgi 10 Make sure you later do a dis msgi 10 dis msgo 10 To shut it off. You should see good info there. -Original Message- From: Anthony Rodgers [mailto:[EMAIL PROTECT

[Asterisk-Users] Lucent TNT / Asterisk Help

Title: Message Anyone out there have any experience with setting up asterisk and a Lucent TNT...I need some help with setup and configuration...I have no experience with Lucent...I have referenced the wiki on voip-info.org but does not do much good since i also need help configuring the E1's

RE: [Asterisk-Users] Connecting asterisk over consumer wifi network

> So, I'd like to get some feedback on how it > might work if we >simply put a wireless access point at each workstation, and > used the 4 port switch to connect to the PC + polycom handset. In my experience, wireless signals have a really poor range in elderly buildings - they're usually built

Re: [Asterisk-Users] Nortel Meridian Option81C to TE405P

This might be an obvious question, but should you be using a crossover cable? Information on setting up Nortel to TDM card links can be found at: Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http:

[Asterisk-Users] Polycom 501 remapping keys

I've tried to configure the "services"-key on my Polycom 501 to run a SpeedDial-entry in [MACADRESS]-directory.xml (which would call a asterisk-extension that starts SayUnixTime) but i have not been able to accomplish my goal. Whe configuring the SpeedDial-function in sip.cfg "VolUp" is started

Re: [Asterisk-Users] UK ISDN2e with DDI?

Quoting John Daragon <[EMAIL PROTECTED]>: > Patrick Lidstone (Personal E-mail) wrote: > >>We're about ready to go ahead with a nice 6 line (maybe later > >>8) ISDN setup with [EMAIL PROTECTED] and the quad Junghanns card. > >> > >>Before we do, could anyone confirm for me that BT's ISDN2e > >>line

[Asterisk-Users] Wanted Japan DID

Dear All,   Any one knows where to buy did's from Japan or Exchange them with US and UK did's ?   I need them for re-selling on didx.net-- Rehan Ahmed AllahWalahttp://www.SuperTec.com - Tommrow's Technology, Today. http://www.didx.net - DID Number Exchange and Peering Service. 

[Asterisk-Users] AMP

Anybody have a manual or link for a manual for the AMP? english or spanish thaks in advance __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Co

Re: [Asterisk-Users] UK ISDN2e with DDI?

Patrick Lidstone (Personal E-mail) wrote: We're about ready to go ahead with a nice 6 line (maybe later 8) ISDN setup with [EMAIL PROTECTED] and the quad Junghanns card. Before we do, could anyone confirm for me that BT's ISDN2e lines do actually provide Asterisk with the DDI number? We need

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