similar to: Default extension

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 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

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 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

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

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

Hello, Is it possible to connect two *callers* without going through a conference (meetme) ? Example: 06:50pm - User 1 call extension 600 and musiconhold / parked call .. 06:51pm - User 2 call extension 600 and connect to User 1. Thank you in advance, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Say I have a list of variables,  listVar <- list(age,sex) I am looking for a way to either 1- create a vector c("age","sex") from it, or 2- get the names one by one in a for loop such as these     a)  for (i in 1:length(listVar)) rownames(result)[i] <- ???     b)  for(i in listVar) print (variable's name) Any help much appreciated. [[alternative HTML version

I've seen a number of issues with the loading of workspaces discussed previously, but here's another one... I simply can't load any saved workspace at all... Here's an example, starting with an empty workspace and creating a single variable "a". > a<-1:5 > save.image("a.Rdata") > rm(a) > load("a.Rdata") Error in function ()  : unused

Hello ! 1) I have this error logged by bdrb: undefined method `send_request'' for nil:NilClass framework/packet_master.rb:58:in `ask_worker'' backgroundrb/server/master_worker.rb:59:in `process_work'' backgroundrb/server/master_worker.rb:16:in `receive_data'' backgroundrb/framework/bin_parser.rb:29:in `call'' backgroundrb/framework/bin_parser.rb:29:in

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,

I have a dataset and a list of labels. I simply want to apply the labels to the variables, all at once. The only way I was able to do it was using a loop:     for (i in 1:length(data))  label(data[,i]) <- data.labels[i] I'd like to find the non-loop way to do it, using apply or the like... Any help appreciated. [[alternative HTML version deleted]]

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

Hi, I need your help to use dovecot with sendmail I use a fresh debian 8 jessie and i installed sendmail and dovecot 2. Sendmail works fine. All mails go to /var/mail/xxx this is problem for dovecot. dovecot cannot get it. # 2.2.13: /etc/dovecot/dovecot.conf # OS: Linux 3.16.0-4-amd64 x86_64 Debian 8.1 auth_mechanisms = plain login auth_verbose = yes auth_verbose_passwords = yes

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.