Re: [Asterisk-Users] Snap for Asterisk

in this first release. What is this for? I have set it up, trying to dial some number, a balloon tip says it is dialing but nothing happens. What am I doing wrong? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com

Re: [Asterisk-Users] How to transmit Video

RAHEEL HASSAN wrote: please tell me that what sip based softphone will beused with Asterisk so that i can trasmit and receive video on my LAN . I'm using Vizufon CIP-5500. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided

Re: [Asterisk-Users] Preferred editor(s) dialplan coding?

Tzafrir Cohen wrote: Still: no jump to line, Ctrl + _ -- Best regards, Bartosz Piec ___ --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] Calls forwarding to numbers only in user's context

. Is that possible to check if the user can call this number and inform him in the moment of setting forwarding? Some sort of TryDial() and handling the 'i' extension. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com

[Asterisk-Users] Calls forwarding to numbers only in user's context

exten = s,n(nocfbs),Goto(s-${DIALSTATUS},1) ; NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER ... [incoming] ; ; Incoming calls. ; exten = XYY,1,Macro(call-forwarding,YY) -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided

Re: [Asterisk-Users] how to run asterisk?

Bayrouni wrote: Can I run asterisk under normal user? Or any other user than root? Try this: http://www.voip-info.org/wiki/view/Asterisk+non-root -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk

Re: [Asterisk-Users] about operator

[EMAIL PROTECTED] wrote: I would like to know which kind of solutions are available, both software and hardware, for human operator in an asterisk environment. Take a look at Flash Operator Panel. -- Best regards, Bartosz Piec ___ --Bandwidth

Re: [Asterisk-Users] spandsp debug or logging

(/fax/file/path|debug) or the same with txfax. The logs are then written to (default) /var/log/asterisk/full -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update

Re: [Asterisk-Users] How can I debug spandsp?

it. I just don't find anything. By default it is in /var/log/asterisk/full file. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Problem with receiving faxes with spandsp - full log included (long)

Congestion(SIP/my.cisco.router.ip-08314d98, ) in new stack Feb 22 12:39:26 DEBUG[22100] channel.c: Driver for channel 'SIP/my.cisco.router.ip-08314d98' does not support indication 8, emulating it -- Best regards, Bartosz Piec ___ --Bandwidth

Re: [Asterisk-Users] voicemail recording format

: This is an uncompressed sound format, so the file size is very large. Not recommended. * g723sf: The sample voicemail.conf file has this option commented out. If you try to activate it, you will find that it doesn't work. So I don't think you can use mp3 or au format here... -- Best regards, Bartosz Piec

[Asterisk-Users] Broken faxes when other call disconnects

? -- Best regards, Bartosz Piec ___ --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] Receiving faxes with spandsp - strange problem

Bartosz Jozwiak wrote: Check if rxfax actually receives anything... How? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

Re: [Asterisk-Users] Receiving faxes with spandsp - strange problem

in this configuration? -- Best regards, Bartosz Piec ___ --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] Receiving faxes with spandsp - strange problem

Matt Riddell (IT) wrote: I'm trying to receive faxes with asterisk. My configuration is like this: Codec? In Asterisk or in Cisco? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing

Re: [Asterisk-Users] Receiving faxes with spandsp - strange problem

Bartosz Jozwiak wrote: Check if rxfax actually receives anything... How? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

Re: [Asterisk-Users] Receiving faxes with spandsp - strange problem

in this configuration? -- Best regards, Bartosz Piec ___ --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

[Asterisk-Users] Receiving faxes with spandsp - strange problem

it is an issue with Cisco router configuration? Normal calls (not faxes) from PSTN lines work great... I have installed asterisk 1.2.4 and spandsp 0.0.2pre23. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk

Re: [Asterisk-Users]how to send fax using Spandsp

or something else? When I'm dialing my number handled by Asterisk, there is a fax signal but the fax machine doesn't start sending. What can be wrong? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

[Asterisk-Users] Preventing Asterisk from transfering the call

A user has set in his phone to transfer each call to another number. Is it possible to configure Asterisk not to transfer the calls? Or is it only phone setting? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com

Re: [Asterisk-Users] transfer, recording ...

Ronald Wiplinger wrote: does still not do the trick! Show your Dial command from extensions.conf file. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update

Re: [Asterisk-Users] transfer, recording ...

. -- Best regards, Bartosz Piec ___ --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] transfer, recording ...

Ronald Wiplinger wrote: I tried to transfer a call, pickupcall and onetouch recording, but have not got it to work! You must uncomment the lines in feature.conf (remove the ; character from the beggining). -- Best regards, Bartosz Piec

Re: [Asterisk-Users] no nat, but one way only audio (more info)

Simone Cittadini wrote: I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Turn off any firewall both on the caller and the called side. -- Best regards, Bartosz Piec

Re: [Asterisk-Users] AEL2 -- The Future --

extensions.ael? From the same site: You can include other files with the #include filepath construct. 1. include /etc/asterisk/testfor.ael2 Read the site for more details. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided

Re: [Asterisk-Users] VizuFon CIP-4500 with Asterisk through SIP

Ian White wrote: Make sure you have a recent copy of the firmware. There was a bug preventing registrations from succeeding until Nov 08 2005 and newer firmwares. Where can I find the firmware? -- Best regards, Bartosz Piec ___ --Bandwidth

[Asterisk-Users] Wrong caller id num on Swissvoice IP10S

: callerid=Someone 11 I've even tried this in extensions.conf: exten = 22,1,Set(CALLERID(num)=11) exten = 22,2,Dial(SIP/22) But it doesn't work. NoOp says the correct CALLERIDNUM. What's wrong? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation

Re: [Asterisk-Users] FAX

Russ Price wrote: So, are there any IP faxes? Sort of. But I'm talking about hardware IP faxes. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options

Re: [Asterisk-Users] Change Inbound CALL ID Asterisk

Oliver Vermeulen wrote: Is here anyway to change the name asterisk on the caller id inbound to the client/sip app? In sip.conf type in the section of desired user: callerid=Caller Name 11 Where 11 is your caller number. -- Best regards, Bartosz Piec

Re: [Asterisk-Users] FAX

C F wrote: Yeah, it shoud NOT work 100% of the time (maybe not even 50%) So, are there any IP faxes? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update

Re: [Asterisk-Users] activex

Vladimir Montealegre wrote: thanks kerry but you have the link? http://www.voip-info.org/wiki/view/Asterisk+.NET -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE

Re: [Asterisk-Users] Simple Dial for If Busy Send to Voicemail

,${EXTEN}) And it works. Try to fit it for your config. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk

Re: [Asterisk-Users] SIP Configuration

, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-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

Re: [Asterisk-Users] ast_streamfile failed

bails napisał(a): Nov 9 10:37:29 WARNING[2083]: Unable to open crim/main-menu.mp3 (format ulaw): Permission denied ^ Any Ideas? Maybe this is a problem with permisions to this file? -- Best regards, Bartosz Piec

Re: [Asterisk-Users] [Asterisk-User] Festival help

asterisk183 napisał(a): therefore don't show error. Test the Festival server console (festival --server). I had permision denied for localhost.localdomain. You must change it in festival.smd file (maybe the name is a bit different). -- Best regards, Bartosz Piec

[Asterisk-Users] Sensing fax with txfax

version. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit

Re: [Asterisk-Users] Sensing fax with txfax

: Call completed to SIP/[EMAIL PROTECTED] Is there a way to debug this somehow? Maybe there is a problem with libtiff? I have the latest 3.7.4 version. And the second question. Do you know some software fax that will work under Windows and send and receive faxes over IP? -- Best regards, Bartosz

Re: [Asterisk-Users] Sensing fax with txfax

g723 to slin Nov 8 15:52:50 WARNING[6429]: app_txfax.c:167 txfax_exec: Unable to set to linear read mode, giving up What is the last warning? It comes from txfax but I don't know how to correct this. -- Best regards, Bartosz Piec ___ --Bandwidth

Re: [Asterisk-Users] Sensing fax with txfax

Matt Riddell napisał(a): You cant do fax in g723. So what to do? Change the fax machine? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

Re: [Asterisk-Users] GSM sound player for windows?

Chuck Bunn napisał(a): Is there a way to play .gsm sound files on Windows. Is there an extension for Windows Media Player or Real Player to allow playing of these files? http://www.voip-info.org/wiki/view/Asterisk+sound+files Section Playing GSM files on Windows. -- Best regards, Bartosz

Re: [Asterisk-Users] Meetme Conference-reg

I don't have app_meetme.so file neither in /usr/lib/asterisk/modules, nor /usr/src/asterisk/apps. How to get it? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users

Re: [Asterisk-Users] Re: Meetme Conference-reg

Tony Mountifield napisał(a): You need to get, build and install zaptel on your system, and then rebuild Asterisk. ztdummy is enough? Will building Asterisk break something in my working installation? :) -- Best regards, Bartosz Piec ___ --Bandwidth

[Asterisk-Users] Dropping last digit from phone number

Hello, {$EXTEN:1} is used for dropping the first digit. But hot to get rid of the last digit? Is it possible? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users

Re: [Asterisk-Users] Re: Meetme Conference-reg

it properly and with understanding. Backuping /etc/asterisk is enough? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman

Re: [Asterisk-Users] Dropping last digit from phone number

Matt Riddell napisał(a): -1 I've tried it. It just leaves the last 1 digit and drops the rest. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] Voicemail

: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com

Re: [Asterisk-Users] Dropping last digit from phone number

Matt Riddell napisał(a): You could try ${EXTEN:-LEN(${EXTEN:1})} When I have 61* number, it isn't the same as ${EXTEN:-2}? But it isn't work anyway... -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com

Re: [Asterisk-Users] Dropping last digit from phone number

Matt Riddell napisał(a): What does that result in? I have this in extensions.conf: exten = _XX*,1,NoOp(${EXTEN:-LEN(${EXTEN:1})}) And when I dial 61*, the result in Asterisk console is: -- Executing NoOp(SIP/65-aad1, 61*) in new stack just like with ${EXTEN}. -- Best regards, Bartosz

Re: [Asterisk-Users] Dropping last digit from phone number

Matt Riddell napisał(a): What do you want 61* to become? Checking the voicemail of 61 number. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

Re: [Asterisk-Users] Dropping last digit from phone number

Erik napisał(a): exten = _XX*,1,NoOp(${EXTEN:0:-1}) exten = _XX*,1,NoOp(${EXTEN:0:2}) :) It works, thanks. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users

Re: [Asterisk-Users] HELP!

case, when the authentication on a phone is enabled, the calls cannot be permorfed and this message is displayed in Asterisk CLI (I wrote about it yesterday). So it should be an error I think... -- Best regards, Bartosz Piec ___ --Bandwidth

[Asterisk-Users] Swissvoice Vizufon firmware

is: [xx] type=friend host=dynamic qualify=yes callerid=asd fromuser=xx username=xx secret=pass context=somecontext In phone I have user name and auth. id set to xx and password to pass. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation

Re: [Asterisk-Users] Problem with Swissvoice IP10S and Asterisk

[EMAIL PROTECTED] napisał(a): At this moment we work with build 12 (Vers. 1.0.0 - Build 12). We have other version (build 15), but this firmware don't work fine. So when I do firmware update, the phone should work? -- Best regards, Bartosz Piec

Re: [Asterisk-Users] Siwssvoice IP10S telnet password

%20guide%20Valid_28-02-05_IP10S.doc -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE

[Asterisk-Users] Problem with Swissvoice IP10S and Asterisk

] section: exten = xxx,1,Dial(SIP/xx) (xxx is the number I calling from) -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com

[Asterisk-Users] Siwssvoice IP10S telnet password

Hello, Does anyone know what is the default password for telnet in Swissvoice IP10S phone? I didn't find any in documentation... -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list

Re: [Asterisk-Users] Problem with Swissvoice IP10S and Asterisk

compliant to be used with Asterisk, right? i can send you offlist a 1.0.0 build Version Would be very appreciated... -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users

Re: [Asterisk-Users] Help on Asterisk and Client SIP setup

lite xlite will require me to specify username, password, and domain, and yet I am not clear where to define the user and the domain on the Asterisk PBX. Read about sip.conf and extensions.conf. This is good place to start: http://www.voip-info.org/wiki-Asterisk -- Best regards, Bartosz Piec