Re: [asterisk-users] asterisk performance

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Re: [asterisk-users] Registration problem: UA - SER - Asterisk

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Re: [asterisk-users] Registration problem: UA - SER - Asterisk

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Re: [asterisk-users] DS3 Interface

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[asterisk-users] VoIP, Asterisk, What Do you really need ?

Hello folks, I was wondering, talking about VoIP, Asterisk or whatever related to it What is the Function or Service you really need to create your own business, simplify service issue, increase your market-cap ? Is it there but is it not open-source or free ? I would like collect informations

[asterisk-users] VoIP, What Do you really need ?

Hello folks, I was wondering, talking about VoIP, Asterisk or whatever related to it What is the Function or Service you really need to create your own business, simplify service issue, increase your market-cap ? Is it there but is it not open-source or free ? I would like collect informations

Re: [asterisk-users] SIP users with Database

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Re: [asterisk-users] say Asterisk to answer

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Re: [asterisk-users] accountcode and amaflags?

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Re: [Asterisk-Users] Config File Management

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Re: [Asterisk-Users] Authorization by ip

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Re: [Asterisk-Users] Call Simulator

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Re: [Asterisk-Users] echo problem + choppy sound

Of course,Echo is 2 types: electric and ambiental.If u gain rx o tx more than you need, its return in recive and gen echoTry to decrase value, try to set 0 or .. in samecase -1 -2... 2006/3/13, sdgesa gaeharth [EMAIL PROTECTED]: Can you explain why?Giovanni Miano [EMAIL PROTECTED] wrote

Re: [Asterisk-Users] echo problem + choppy sound

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Re: [Asterisk-Users] CallerID popup

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Re: [Asterisk-Users] Events when the target of the call answer

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Re: [Asterisk-Users] (newbie) using dtmf during a call

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Re: [Asterisk-Users] Problem with ISDN HFC-S card

thankful, if you could give me some advice.CheersAndrew ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano

Re: [Asterisk-Users] Reducing echo on FXS port

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Re: [Asterisk-Users] Call forwarding for particular extension

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Re: [Asterisk-Users] CD (call deflection) on Bristuff/zaphfc?

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Re: [Asterisk-Users] ChanSpy via external application

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Re: [Asterisk-Users] callback on busy

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Re: [Asterisk-Users] Incoming PSTN Calls

rs mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visi

Re: [Asterisk-Users] remote sip client fail to register

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Re: [Asterisk-Users] Is it possible to get caller and called numberwith Asterisk Manager

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Re: [Asterisk-Users] CrystalFontz LCD display

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Re: [Asterisk-Users] Three Way Calling with HFC PCI Card

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Re: [Asterisk-Users] Queue features

You can check status of Peer with Asterisk Management Interface (AMI) www.voip-info.org/wiki-Asterisk+manager+API Cheers,Giovanni Miano 2005/12/30, Dov Bigio [EMAIL PROTECTED]: Hi, I am using the Queue application for 5 queues I have in my Call Center, and will by the end of January

Re: [Asterisk-Users] zaptel TDM21B 4-5 second pause

Try to append # or * to numberDial(ZAP/g0/0199255#) or Dial(ZAP/g0/0199255*)Cheers,Giovanni Miano 2005/12/29, Eck [EMAIL PROTECTED]:Hi, Sorry if this is a little off topic as its really more zaptel related, but hopefully someone will have come across this.,I am noticing a 4-5 second pause

Re: [Asterisk-Users] Asterisk Server Hangs

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Re: [Asterisk-Users] Easiest way to use HFC-S?

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Re: [Asterisk-Users] ISDN

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Re: [Asterisk-Users] AGI Problem

, I've changed that and it's the same problem. I've this problem with all applications. Results from agi are not correct. Regards, Cyrille -Message d'origine- De: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] De la part de Giovanni Miano Envoyé: vendredi 2 décembre 2005 12:52 À: Asterisk Users

Re: [Asterisk-Users] Sound problem

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Re: [Asterisk-Users] Include a variable from another file in config files

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Re: [Asterisk-Users] IAX Configuration

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Re: [Asterisk-Users] (no subject)

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Re: [Asterisk-Users] (no subject)

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Re: [Asterisk-Users] Hint: how to include dialplan files from remotesystems

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Re: [Asterisk-Users] AGI Problem

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Re: [Asterisk-Users] Limiting DID calls

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Re: [Asterisk-Users] voice problems under 8 concurent calles

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Re: [Asterisk-Users] AGI Problem

-Message d'origine- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Giovanni Miano Envoyé: vendredi 2 décembre 2005 12:52 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: Re: [Asterisk-Users] AGI Problem Try print EXEC ChanIsAvail IAX2/24\n; Channel type

Re: [Asterisk-Users] Call transfer with voicemail password

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Re: [Asterisk-Users] Motherboard choice for asterisk?

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Re: [Asterisk-Users] Voice Mail

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Re: [Asterisk-Users] unable to make calls out using [EMAIL PROTECTED]

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Re: [Asterisk-Users] unable to make calls out using [EMAIL PROTECTED]

type in console: sip show registry and verify status of your trunk 2005/12/1, ram [EMAIL PROTECTED]: yes i got new account from provider and i have registered no one using, iam using that account for testing ram On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote: Have u

Re: [Asterisk-Users] unable to make calls out using [EMAIL PROTECTED]

, Giovanni Miano [EMAIL PROTECTED] wrote: type in console: sip show registry and verify status of your trunk 2005/12/1, ram [EMAIL PROTECTED]: yes i got new account from provider and i have registered no one using, iam using that account for testing ram

Re: [Asterisk-Users] unable to make calls out using [EMAIL PROTECTED]

On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote: pastme context for outgoing 2005/12/1, ram [EMAIL PROTECTED] : Hi here is the results asterisk1*CLI sip show registry HostUsername Refresh State x.x.x..2:5060 xx

Re: [Asterisk-Users] WG: App_rxfax problem

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Re: [Asterisk-Users] Help transfer call

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Re: [Asterisk-Users] Newbie question

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Re: [Asterisk-Users] route call based on codec? (g723 gets message, g729 goes to conf connection)

is appreciated. Thanks - Jon Carnes ___ --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 -- Giovanni Miano

Re: [Asterisk-Users] Newbie question on 1.2 extension configs

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Re: [Asterisk-Users] Tone busy in zaptel

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Re: [Asterisk-Users] [EMAIL PROTECTED] isdn

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Re: [Asterisk-Users] [EMAIL PROTECTED] isdn

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[Asterisk-Users] Asterisk + FXO + FAX SIP

I've 1 tdm400 with 2 fxo modules. FXO-ASTERISK-SIP Adapt-Fax-Machine I've more problems where receive or send fax because poor quality of signal i'm in italy and use fxo with adsl + spliter ? is this problem ? How can i resolve ? -- Giovanni Miano

[Asterisk-Users] Bristuff 0.2.0-RC8o or 0.2.0-RC8n (* 1.0.9)

, thirdcall = -1 Nov 5 11:58:04 DEBUG[8299]: Not yet hungup... Calling hangup once with icause, and clearing call Why ? Please help me. -- Giovanni Miano ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk

Re: [Asterisk-Users] Zaptel + No Hangup

Any problems with bristuff ? 2005/10/26, Julian J. M. [EMAIL PROTECTED]: You can try this patch (www.maxosystem.net/asterisk/asterisk-stable-polarity.html), if your telco sends your polarity reversals on answer and hangup. Julian J. M. On 10/26/05, Giovanni Miano [EMAIL PROTECTED] wrote

[Asterisk-Users] Zaptel + No Hangup

I've TDM400P with 2 cards fxo and asterisk 1.0.9 + zaptel 1.0.9.2 All works perfectly but command Hangup or Hangup() in dialplan dont hangup call (zapata.conf within busycount=4 and busydetect=yes) Why ? Country is ITALY -- Giovanni Miano

Re: [Asterisk-Users] 0.2.0-RC8o (* 1.0.9) + No Caller ID

/10/24, Massimo De Nadal [EMAIL PROTECTED]: Giovanni Miano wrote: I've 2 hfc billion and one TDM400P 1fxs/1fxo with bristuff 0.2.0-RC8o and * 1.0.9 I dont recive callerid from TDM400P fxo port but isdn hasnt problems If i try to use only TDM400P 1fxs/1fxo without bristuff.. all work ok

[Asterisk-Users] 0.2.0-RC8o (* 1.0.9) + No Caller ID

I've 2 hfc billion and one TDM400P 1fxs/1fxo with bristuff 0.2.0-RC8o and * 1.0.9 I dont recive callerid from TDM400P fxo port but isdn hasnt problems If i try to use only TDM400P 1fxs/1fxo without bristuff.. all work ok is it bug of bristuff ? -- Giovanni Miano

Re: [Asterisk-Users] AstriCon 2006 Location

-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth

Re: [Asterisk-Users] CTI and Asterisk

-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] I need a Asterisk tech

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Re: [Asterisk-Users] Dialing With Backgound Music

Dial(SIP/100,30,tm) On Apr 9, 2005 5:50 PM, Ugur GUNCER [EMAIL PROTECTED] wrote: How can play music when is clients phone ringing in dial progress. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] I4l + HiSax

I need HELP pls! BRISTUFF: Bad Sound quality CAPI: PTP Mode dont supported mISDN : kernel is 2.4.x and not 2.6.x HISAX : PTMP ok, PTP incoming ok but in outgoing asterisk dont compose number(i listen dial tone and than i can compose number via dtmf) Asterisk CLI (g3 is group of

[Asterisk-Users] Asterisk + Call hangup

Scenario PSTN - ZAP CHANNEL - ASTERISK - SIP When i recive call i fwd it to SIP Phone - SIP PHONE ringing If From External Line PSTN hungup call SIP Phone Ringing too, why ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Asterisk@home 0.6 + Modem.conf

How i use Modem.conf with AMP? It allow only ZAP IAX or SIP Trunk is there a patch to manage it ? And how configure extension.conf to use Modem/g1 channels when Zap/g0 channels are busy ? ___ Asterisk-Users mailing list

[Asterisk-Users] Asterisk@home 0.6 + bristuff

How install bristuff in [EMAIL PROTECTED] ? i tried version 0.2.Rca to last RC7k and when try to compile zaptel (after patched it) i've this error: make: *** [zaptel.o] Error 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Asterisk@home 0.6 + mISDN

Hi, I've a billion isdn0 card but i suppose than i cant patch kernel to use mISDN support because i've kernel 2.4 and patch on CVS i4l is for = 2.6.8 How do it ? help me please Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Chan_Capi + HFC Card

I installed capi support, For HFC PCI card i must use Capi4Linux o Isdn4Linunx (my kernel is 2.4, i dont support mISDN) How install HFC Card ? How see HFC Card PCI ? (my card is chip billion s0 isdn card) Help me ___ Asterisk-Users mailing list

[Asterisk-Users] Astrerisk + Conversation OneWay

Redhat 9 Asterisk 1.x isd4linux Teles ISA Card 16.3c PlugPlay I load hisax module: modprobe hisax type=14 protocol=2 irq=5 PSTN - Asterisk - SIP SoftPhone (XTen) Comunication between asterisk and sip client is ok Comunication berween clients sip is ok Comuncation between PSTN and SIP SoftPhone

[Asterisk-Users] SIP with Delay

I use codec g711u or g711a but comuncation between two sip client (XTen lite) have bastard dalay of 0,5 - 1 second Is it normal ? Are there any configuration to solve problem ? Thanks all ___ Asterisk-Users mailing list

[Asterisk-Users] Asterisk with SourdCard

My system is: Redhat 9.0 + Asterisk + ISDN4Linux + Teles 16.3 ISA Passive card I haven't sound card. Comunication between two SIP Clients is OK Comunication between PSTN and SIP Client is OneWay (i cant recive dtmf and voice from pstn) is it needed sound card ?

[Asterisk-Users] i4l + SIP: Audio One-way

Hi all, I use isdn4linux with sip terminal. Audio between two client sip is good but audio between PSTN and client sip is oneway, why ? Help me pls ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] i4l: Quality of Voice

God save this mailling list :) Which is the best settings for the best quality of Audio ? I'use isdn4Linux driver and SIP client but bad quality PSTN - Asterisk and SIP - PSTN ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com