similar to: What would make Asterisk Ignore INVITES?

Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX -

[macro-hangupcall] exten => s,1,ResetCDR(w) exten => s,2,NoCDR() exten => s,3,Wait(5) exten => s,4,Hangup Hi all, currently i've been getting troubles with SIpphone Sjphone and Xlite seems also to get delay but no crash on hanging. I found that Ast@home is executing this Wait(5) and it seems to me that Sjphone is giving timeout error because of it... Why is this 5 seconnds? any

---------- Forwarded message ---------- From: Marco Mouta <marco.mouta@gmail.com> Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: asterisk-users-request@lists.digium.com Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me

Hi all, I've around 10 people on my network getting this error, Critical transaction failed: Client non-INVITE transaction[trying]: Time out I'm using Asterisk 1.2.5 and Sjphone. Any tips??? Best regards, Marco Mouta

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Whey you 'destroy' a Sip extension you are only removing the entrys that allow you to make and receive the auth needed to do so. Your voicemail files are not tied to an extension but are independent and are only 'married' when you specify it in your sip.conf or other channel configs. Removing a confi from a channel will not touch voicemail. You need to go into the voicemail.conf

Boa tarde, Ap?s alguma experi?ncia com o Asterisk, e com muito ainda para aprender, gostaria de saber se h? algu?m nesta mailing list que pretenda criar um Asterisk Users Group para Portugal. Visto que acaba sempre por ser uma enorme aprendizagem ( valor acrescentado) a partilha de experi?ncias/problemas e solu??es nas implementa??es Asterisk. H? spre detalhes que variam entre os Telco's de

Hi all, I've the current scenario: User "A" - Zaptel call incoming in my Asterisk to my SIP user "B". "B" gets the Call. "A" says : "B" i would like to call PSTN user "C" "B" places a call to user "C" and asks if "C" wants the call from "A". "C" says yes i want, then B needs to

Hi guys, I've setup on box with a TE110P and time to time I need to access remote equipment outside of our office and use a data channel. I'm wondering if do I need to buy a POTS line only for this time to time acess or what's the easiest way to do that via my TE110P on asterisk box. I know that is possible data transmission with this Digium Card, I'm wondering how... Any tip any

Hi all, I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When I'm in a call and i press Hold button, the other party starts listening Music on Hold but then when i press the button again to get the call back it doesn't work! I've checked asterisk CLI: -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 --

Hi guys, I've been looking on wiki, but i could find it only for chan_capi: http://www.voip-info.org/wiki/view/Asterisk+PBX+functions In the CAPI channel See Asterisk CAPI channels * Call Deflection (CD) (redirect without answering): Implemented by chan_capi How can i do it with my Softphone Xlite? Any one can help me? I want to redirect a call without answering it. Best regards,

HI all, My asterisk for all my users, everything was fine for 3 days, but now i can't access it. But it is running... Could any one help me on this? Best regards, Marco Mouta

Hi all, I've my Server running well, then sometimes Sjphones looses registry and it only works well again if i restart the pc running sjphone. Has any one experience this? Best regards, Marco Mouta