similar to: contacting

Hello, I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work! I placed this in my peer: (sip.conf) sendrpid=yes trustrpid=yes or sendrpid=yes trustrpid=no (and restarted Asterisk) and the line "Remote-Party-ID" does not appear in my sip debug! Please help me, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am running an AGI script in a command dial, or call a SIP trunk. I want to execute after 10 minutes a voice message (stream file) on the channel to warn the person that the call is about to end. How to do that? Thank you, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, When I get a SIP INVITE as follows: INVITE sip:s at 10.1.0.191:5060 SIP/2.0 Max-Forwards: 69 From: "0475XXXXXX" <sip:1053212 at sip.domain.com>;tag=as7df9ab18 To: <sip:02XXXXXX at IP:5060> Contact: <sip:1053212 at IP:5060> Call-ID: 344d42bd16975a54141d11f635bdfc71 at sip.domain.com CSeq: 102 INVITE Date: Wed, 26 Mar 2014 15:06:01 GMT Allow: INVITE, ACK, CANCEL,

What is the interest to supply binary of Asterisk, under debian for example, while to use MeetMe it is necessary to COMPILE Asterisk ??? :-)) Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100611/e4d749f6/attachment.htm

Good morning, or Good afternoon! It depends :-) I have a standard Asterisk configuration: SIP friends (phones) <-----> Asterisk <-----> SIP gateway to PSTN converter 80.236.215.61 109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0) When analyzing traffic on a SIP friend/phone I see this: INVITE sip:xxxx at 80.236.215.61:64946;ob

Hello I just coding a AGI script for billing. - For external calls, I pass the call directly on a trunk. I do : Dial(trunk1/extension) -> OK ! - For internal calls (shortcode, others users ...) I am Dial(Local/extension at context/n) The problem is that through chan_local.so, I sound as it cut! Example if I call the voicemail ... "You have No messa ..." or "You have

Hello, I look ARI (Asterisk Recording Interface) the publisher site is closed... http://www.littlejohnconsulting.com/ari Thank you, Mickael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100623/a8d923ae/attachment.htm

Hello, I have a Cisco AS5300 connected to Asterisk (1.6.2.9) Between 15-16 minutes, the call is disconnected without reason. Here is what is displayed in the debug: Received an SDES from 10.4.0.10:17399 -- Got SIP response 420 "Bad Extension" back from 10.4.0.10 -- Stopped music on hold on SIP/as5300-1-0000004d == Spawn extension (dialin, 065939191, 2) exited non-zero on

Hi, Where is the mount option 'limit-rate' in the current version? I checked in cfgfile.c and in the documentation, no mention. Yet this option did exist at one time: http://lists.xiph.org/pipermail/icecast/2010-October/011703.html http://lists.xiph.org/pipermail/icecast/2009-January/011391.html I try to limit the bitrate of a mount-point, is there another solution? Do you know why this

Hello, Thank you for your response. It is on the kh version.. https://github.com/karlheyes/icecast-kh Le sam. 3 nov. 2018 à 21:47, Thomas B. Rücker <thomas at ruecker.fi> a écrit : > Hi, > > On 11/03/2018 07:33 PM, Mickael MONSIEUR wrote: > > Hi, > > Where is the mount option 'limit-rate' in the current version? > > I checked in cfgfile.c and in the

Hello ! I want to call a line and play a sound from the callee before putting it in connection with the caller. Is this possible? Example: Dial(SIP/111111, m) // Ring or Music... if(call==ANSWERED) Play(announce) // Play 'announce' to the called // To connect caller and called ? Best regards, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, The MeetMe application refuses MusicOnHold personalized and skip always in the default! Have you any idea how to fix this? -- Executing [028883899 at default:1] Set("SIP/109.10.214.1-00000002", "CHANNEL(language)=fr") in new stack -- Executing [028883899 at default:2] Answer("SIP/109.10.214.1-00000002", "") in new stack -- Executing

All, I want to know if it's possible to create a log file per context? and each time a context is restarted a ne x log file is created. regards Mickael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091020/673c0eb4/attachment.htm

On 11/03/2018 09:53 PM, Mickael MONSIEUR wrote: > Hello, > Thank you for your response. > It is on the kh version.. That's not a version. That's completely different software at this point. It's also not Xiph.org, but published by Karl. TBR > Le sam. 3 nov. 2018 à 21:47, Thomas B. Rücker <thomas at ruecker.fi > <mailto:thomas at ruecker.fi>> a écrit :

All, According to my readings CDRs are stored at the end of the call. My concerns is when asterisk goes down (I know that it's never happen but it's just in case) or when the is a power shutdown of the server. then CDRs are not stored in mysql. is there a way to store periodially CDR during a call, and set the periodical timer regarding the context. if no is there a way to retreive CDR,

Hi, I am working on Trixbox. I want to create my own dial() function (named specificdial()) and I want to know how I can create a module and integrate the module in the trixbox plateform. thanks a lot Mickael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091006/b69d4103/attachment.htm

Le 19/06/2015 09:04, Steffen Kaiser a ?crit : > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On Fri, 19 Jun 2015, Mickael Choisnard wrote: > >> Date: Fri, 19 Jun 2015 08:29:08 +0200 >> From: Mickael Choisnard <mickael.choisnard at free.fr> >> To: dovecot at dovecot.org >> Subject: [Dovecot] question about /var/mail/xxx >> >> Hi,

Hello Les, Thanks for that info. I'm playing with this now and although the 'failover' process seems rather slow, it does seem to be doing what I need. I setup a subdomain entry to point to 4 IP's, only one if which is actually working, and indeed, when IE get's a non-active IP, it eventually goes to the next one until it finally finds the actual live IP. Once it gets the

Hello CentOS, I've just taken 2 old servers running CentOS 3.x, wiped it out, and installed CentOS 4.1 The systems seemed to work fine with CentOS 4.1, but when I proceeded to run yum -y update, all the RPM's (101) download and seem to install fine - however, upon reboot, partitions such as /usr/ and /var will not mount (ext3). These machines both use the Intel S875WP1 P4 Mainboards.

Hi, I am using Matt McCray''s theme generator. http://rubyforge.org/projects/theme-generator/ Everything works fine, but rendering an action in a controller using the AppMailer raises this: undefined method `current_theme'' for #<AppMailer:0x39df950> Is there someone who could help me ? Thanks ! Mickael. Traces: