[asterisk-dev] Voicemail + IMAP Problems again (svn trunk)

Hello, can anyone tell me which version is suggested for testing voicemail with imap support? Ive installed a svn trunk version about 10 days ago which compiled well but had problems with sip+rtp(+nat). Today i downloaded svn trunk again and now it doesnt even compile anymore (see errors below).

Re: [asterisk-dev] PRI: help to understand why

Giorgio Incantalupo wrote: Hi Doug, thanks for answering. I do not understand what could change if I used G instead of gg makes asterisk call starting from channel 1, G from the last but wouldn't be the same?? Asterisk would complain about last channel instead of the first. Don't know u

: Re: [asterisk-dev] Forwarding sip requests from none local domains

OK, Are you able to help me ? Anyway communigate pro server will be my choice to do business ? Regards Harry --- Tzafrir Cohen <[EMAIL PROTECTED]> a écrit : > On Tue, Sep 12, 2006 at 04:33:21PM +0200, > [EMAIL PROTECTED] wrote: > > I've ever post this question many times on > asterisk > > us

Re: [asterisk-dev] PRI: help to understand why

Hi Doug, thanks for answering. I do not understand what could change if I used G instead of gg makes asterisk call starting from channel 1, G from the last but wouldn't be the same?? Asterisk would complain about last channel instead of the first. I'd like to know why this happens...is it a

RE: [asterisk-dev] ast_channel_masquerade in a queue consultation

Hello, In http://ftp.digium.com/pub/asterisk/releases/ChangeLog-1.2.10 i see:  “2006-07-10 21:01 + [r37361]  Kevin P. Fleming <[EMAIL PROTECTED]> * channel.c: do masquerade-behind-proxy checking with better      control over locks”. I suppose that the code for that correcti

Re: [asterisk-dev] internal_timing on 1.2? clock drift?

Kevin P. Fleming wrote: - Daniel Pocock <[EMAIL PROTECTED]> wrote: Is the internal_timing patch likely to be backported into a future 1.2.x release, or do people just have to wait for 1.4? We do not add new functionality to release branches, except in cases where it is necessa

Re: [asterisk-dev] PRI: help to understand why

Giorgio Incantalupo wrote: I read on internet that it is due to the telco which send request to establish a call on a channel but Asterisk has not freed it yetis it true? Is it a bug or can be avoided setting some parameter inside zapata.conf? This may be glare. If your incoming calls

RE : Re: [asterisk-dev] Forwarding sip requests from none local domains

I've ever post this question many times on asterisk users without success ? My config : SER => outbound proxy presence/im server ASTERISK || || proxy/SER ===sip agents + rtpproxy If a sip agents dial local uri

[asterisk-dev] mute calls through a sip proxy

Hi guys, Does anyone know why when I copy this file in /var/spool/asterisk/outgoing/  the call is mute.  ? Any help will be appreciated. ===  sip.call Channel: SIP/[EMAIL PROTECTED] MaxRetries: 2 RetryTime: 60 WaitTime: 60 callerid: '"magik" <299>' extension: 299 This is my confi

Re: [asterisk-dev] Forwarding sip requests from none local domains

12 sep 2006 kl. 15.23 skrev <[EMAIL PROTECTED]> [EMAIL PROTECTED]>: Hello, I use asterisk svn-trunk . I wish asterisk to forward all sip requests from non local domains to a proxy . For example asterisk handle domainA a sip agent send a invite to a domainB . Is asterisk able to check the d

Re: [asterisk-dev] PRI: help to understand why Asterisk drops calls(Got restart ack)

Hi Paul, you were right...the message disappeared. I'd like to have no warnings in my log...it seems that the message: WARNING[21314] chan_zap.c: Ring requested on channel 0/1 already in use on span 1. Hanging up owner appears too many times (and randomly..). I read on internet that it is du

Re: [asterisk-dev] rewriting standard stuff?

then perhaps there should be some sort of autosensing if it needs to be overridden. AFAIK very few OSes doesn't support stuff like getloadavg() unless you're talking about windozes, and afaik there's no windoze support back in asterisk. roy On 11. sep. 2006, at 19.48, Jason Parker wrote:

[asterisk-dev] Forwarding sip requests from none local domains

Hello, I use asterisk svn-trunk . I wish asterisk to forward all sip requests from non local domains to a proxy . For example asterisk handle domainA a sip agent send a invite to a domainB . Is asterisk able to check the domain and so forward the request to a context (with outboubounproxy) Har

[asterisk-dev] ast_channel_masquerade in a queue consultation

Hello, With asterisk 1.2.12 i need restart asterisk to free channels. This isn’t happen with 1.2.7.1. The scenario is: -  Sip1 has a call with Sip2. -  Sip1 make a consultation call to queue1 with originate action. -  The call is delivered to sip3 (agent3 logged

[asterisk-dev] asterisk-1.2.12 iLBC decode cause core dump

I just installed asterisk-1.2.12, zaptel-1.2.8(ztdummy). I use this server talk with another * server through IAX2 trunk. and codec is iLBC between these two servers. I run new asterisk about 2 hours, and calls reach 40-50 all the times. Then it crash with core dump. I use "gdb, bt" got the dump

[asterisk-dev] Warning and/or question (Zaptel MMX)

Hi, Firstly, a note of caution for those compiling MMX into Zaptel... I use an EPIA processor, which may be part of the problem, but enabling the Zaptel MMX code caused sufficient corruption that it affected other userspace processes! I suddenly started getting page-errors in a web application, a