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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,

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 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, 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 documentation, no mention. > > This option does not exist. > > > > Yet this option did exist at one time: > > http://lists.xiph.o...

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

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]]

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.f...

Bonjours monsieurs Je suis un ?tudient en 4eme ann?e informatique a l?universite djilali liabes SBA ALGERIE. Je suis entrain de pr?parer un expos? sur l?AFD et j?ai besoin d?un exemple sous R pour bien pr?senter mon travail.

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

madame, monsieur, help me please, Je debute avec R. J'ai r??alis?? une analyse factorielle discriminante avec R. J'obtient les resultat graphiques et les tables assici??es ?? l'analyse. Cependant je souhaiterais savoir si les distances separant les groupes (decades) sont statistiquement significati...

A monsieur Andrew Je suis ?tudiant en fin de cycle de formation en infomatique ? l'institut Africain d'informatique de Libreville.J'?tais sur vos sites et je suis persuad?s que vous ?tes la personne la bien plac?e pour m'orienter et m'aider sur un sujet concernant "l'authentificat...

Madame , Monsieur, J'aimerais utiliser la fonction step dans une boucle "for" pour pouvoir prédire des valeurs (à chaque pas) suivant le modèle proposé par step. Est-il possible de le faire ? Merci d'avance, Cordialement Athanasia

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:

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, 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: