[asterisk-users] No Audio with Gtalk

I configured my * with the instructions found here http://www.voip-info.org/wiki/view/Asterisk+Google+Talk to work with gtalk. The Phone rings and connects - but no audio! I am using a self-compiled asterisk 1.4.2 There is a lot of output on the CLI but I can't make sense of it. Perhaps somebod

[asterisk-users] ISDN PRI DTMF problem

Dear all, Can someone help me about DTMF in ISDNPRI. I'm seting up E1 PRI found problem 90% Hangup when put digit. my E1 card work well with other telco. for this telco it's first time to use (telco in Thailand but i don;t know about equipment they use) I try to debug and got message below. P

Re: [asterisk-users] Which IP Phones have buttons can be assigned to functions with Asterisk

http://www.voip-info.org/wiki-Polycom+Phones#Newphones quote: "WARNING: The IP 30x and IP 50x models do not have on-board Power Over Ethernet chips. Although the phone claims to support 802.3af and the Cisco POE standard (note it says "optional"), the an additional cable (see part list above) is r

Re: [asterisk-users] Question on Priorities

From: "Rizwan Hisham" <[EMAIL PROTECTED]> Date: Sat, 31 Mar 2007 17:01:51 +0500 [inbound-sip] exten => uxbod,1,Dial(sip/1001,20,jt) exten => uxbod,n,Hangup exten => uxbod,102,PlayBack(uxbod) exten => uxbod,103,VoiceMail([EMAIL PROTECTED],s) exten => uxbod,104,Hangup() here if dial fails then n+

RE: [asterisk-users] Re: Paging

From: "Forrest Beck" <[EMAIL PROTECTED]> Date: Fri, 30 Mar 2007 16:52:39 -0400 Forgot to mention. We are using Polycom phones on asterisk 1.4.2 I tried the allpage agi, but it checks for all SIP peers connected to the server. On 3/30/07, Forrest Beck <[EMAIL PROTECTED]> wrote: First off, A lo

Re: [asterisk-users] Which IP Phones have buttons can be assigned to functions with Asterisk

bilal ghayyad wrote: I heared that polycom needs adaptor for the power as it does not provide standard PoE, also I do not know this. You need a special cable to use a 501 with a POE switch, that's all. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759

Re: [asterisk-users] Multi-Level Queue

So does the P option in app_dial seems to me the easiest way to implement is just hack app_dial so it wont prompt to record the name. On 3/31/07, BJ Weschke <[EMAIL PROTECTED]> wrote: On 3/31/07, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: > Peder @ NetworkOblivion wrote: > > > I also had a ques

Re: [asterisk-users] Linksys SPA 3102 causing me problems

On 3/30/07, Gergo Csibra <[EMAIL PROTECTED]> wrote: Friday, March 30, 2007, 5:02:08 AM, Matt wrote: Wehh... He activated the DND function of Linksys. It can be activate with *78 and deactivate with *79. No because if that was a case his sip trace would show something along the lines of "4

[asterisk-users] Re: Meetme question

If you know what you want the conf room number to be, then set that up in meetme.conf. You would have to write your own IVR though, and use Authenticate() with the PIN kept in the DB. Its a hack but it would do what you want: exten => _X,1,Playback("conf-getconfno") exten => _9XX,1,Authenticate

RE: [asterisk-users] Setting a call to be recorded before Xfer?

Maybe have 2 extensions for all the sales people. Calls to 7xx ring their desk extension Calls to 8xx ring their desk extension but also send mp3 of the call via email to sales person and sales manager. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED]

[asterisk-users] Setting a call to be recorded before Xfer?

I need to allow the company operator to decide whether to record a call. (Car dealership that needs to coach salespeople). They don't want to record every sales call, just for the purposes of coaching certain employees on an ad hoc basis. The situation is: a. Call comes in on PSTN PRI b. Call is

[asterisk-users] Understanding the dial flags

I am trying to make a system where a conference user can invite others to join. I am running the 1.2 version of asterisk, so can't use the example on voip-info.org. With use of the X flag on a meetme conference to exit with a single digit, I can get people to join me in a conference with ext

Re: [asterisk-users] Off Topic: Open Source USB Softphone

sip would be the required one as iax..well.. also openwengo wont work.. to much overhead .. broswrer needed.. ie component + flash + css+js etc.. not viable.. so im also asking anyone have one ? since ihave a supply of around 2000 of the vonage usb stick OEM.. On 3/30/07, Michael Van Donselaar

Re: [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 129

[EMAIL PROTECTED] wrote: > Je suis absent du 2/04/2007 au 11/04/2007. Do not auto-reply to list messages. :) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://

[asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 129

Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To U

[asterisk-users] Re: Sponsored development - Monodirectional audio handling

Salvatore Giudice ha scritto: You could put a bounty on this. You may find someone who will be willing to write this for money. My Bounty for that feature is 500 USD -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoI

Re: [asterisk-users] LUSYN patches

On Sat, Mar 31, 2007 at 05:17:02PM +0100, Marc McLaughlin wrote: > Hello all, > > The history buffer patches maintained by myself have been updated to > work with Zaptel 1.4.1 and Asterisk 1.4.2. They are available along > with installation instructions at www.lusyn.com. BTW: for zaptel 1.2 >=

[asterisk-users] LUSYN patches

Hello all, The history buffer patches maintained by myself have been updated to work with Zaptel 1.4.1 and Asterisk 1.4.2. They are available along with installation instructions at www.lusyn.com. Rgds, Marc LUSYN Limited is a company registered in England and Wales with company number 04803

RE: [asterisk-users] Re: wireless desktop phones

You can always using a gaming bridge for phones that do not support wireless. I've done this before with this: Linksys / WGA54G / 54Mbps / 802.11g / Wireless Bridge Setup is pretty easy. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Trainin

RE: [asterisk-users] Sponsored development - Monodirectional audio handling

You could put a bounty on this. You may find someone who will be willing to write this for money. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phon

RE : RE : [asterisk-users] wireless desktop phones

Hi Tobias and the list, Yes, I have, I use and sell them to integrators ;-) But only the "600v3 family", not the older ISND or analog versions, and the current DECT handsets 40XX. Any Digium interfaces run well with them as any SIP IP-Phone, of course. The sound quality is GREAT and the infrastru

[asterisk-users] Sponsored development - Monodirectional audio handling

Hi Guys, we're needing a special implementation on Asterisk Our intention is to contribute the development and share back the code to Asterisk community Here is what we need: - An option to Asterisk Dial command which, if used, when calls is answered gives monodirectional audio (Call

Re: [asterisk-users] Problem while using asterisk Realtime

- "Sanjay Rajdev" <[EMAIL PROTECTED]> wrote: > Another thing I found is if i do the following > # make menuselect > go in option 2. Call Detail Recording > here the option 4. cdr_odbc and option 5. cdr_pgsql both have XXX > marked infront of them. And at the bottom of screen it says ODBC CDR >

Re: [asterisk-users] Multi-Level Queue

On 3/31/07, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: Peder @ NetworkOblivion wrote: > I also had a question about "acking" a call. It appears that acking a > call is under agents.conf. I want to specify members as SIP/1234, etc, > rather than having users login all the time. I don't want to

Re: [asterisk-users] Multi-Level Queue

Peder @ NetworkOblivion wrote: > I also had a question about "acking" a call. It appears that acking a > call is under agents.conf. I want to specify members as SIP/1234, etc, > rather than having users login all the time. I don't want to have to > login from my cell, I would prefer it to just k

[asterisk-users] Re: wireless desktop phones

> "JN" == Jordan Novak <[EMAIL PROTECTED]> writes: JN> Okay, I get it. I still have a problem though. I have no way to JN> wire 30% of these end-points. P{hysically impossible. They do have JN> cat3 twisted pair to each phone. But of course they want IP. Are JN> there any adpaters that will gi

Re: [asterisk-users] wireless desktop phones

On 28 Mar 2007, at 21:51, Matt Gorecki wrote: I'm also in the market for a wi-fi phone. My boss currently has a cordless phone and wants to keep the same functionality. We have a robust wireless network in the office and the phone will be staying here, so roaming is not really an issue.

[asterisk-users] Meetme question

Hi, I'm experimenting with the Meetme feature of Asterisk 1.2, exten => 2095,1,MeetMe(|Ds) This almost gives me what I want, where each employee can create their own on-the-fly conferences with a personal Conference Number and PIN. However, as the PIN is actually set by the first callee, the

Re: [asterisk-users] Question on Priorities

also only priorities are added incase of priority jumping, not extensions. On 3/31/07, --[ UxBoD ]-- <[EMAIL PROTECTED]> wrote: Hi, I am attempting to change my dialplan to use 'n' priorities and labels for easier reading, and less re-numbering :) but how do you handle the plus 101 ? In my ext

Re: [asterisk-users] Question on Priorities

[inbound-sip] exten => uxbod,1,Dial(sip/1001,20,jt) exten => uxbod,n,Hangup exten => uxbod,102,PlayBack(uxbod) exten => uxbod,103,VoiceMail([EMAIL PROTECTED],s) exten => uxbod,104,Hangup() here if dial fails then n+101 =102 extension will get executed unless you use j option in dial application

[asterisk-users] Question on Priorities

Hi, I am attempting to change my dialplan to use 'n' priorities and labels for easier reading, and less re-numbering :) but how do you handle the plus 101 ? In my extensions.conf I have a simple plan for testing :- [inbound-sip] exten => uxbod,1,Dial(sip/1001,20,t) exten => uxbod,n,PlayBack(uxbod