similar to: How to reply with 480 Call-limit to incoming SIP call ?

Hi, For a call center, I'm studying how I can offer agents the ability to reject an incoming call using a custom application. As you can guess, in this case, rejecting a call means "let another agent answer this call" (it doesn't mean "end this call"). The only way I could imagine for this to happen, would be to redirect the caller to a conference room, then hangup

Hello, When sending this AMI request ... 192.168.64.5 -> Action: Originate 192.168.64.5 -> Channel: SIP/9122 192.168.64.5 -> Async: True 192.168.64.5 -> Callerid: 9122 Guest2 <9122> 192.168.64.5 -> Exten: 9123 192.168.64.5 -> Context: local 192.168.64.5 -> Priority: 1 ... I've got this INVITE from Asterisk INVITE sip:9121 at 192.168.100.198:5060;user=phone SIP/2.0

Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David

Hi, What are the main differences between Aastra SIP phones 6730i and 6750i series ? Aastra corporate web site mentions : "The Aastra 6730i Series offers exceptional features and flexibility in an open-standard enterprise grade IP telephone" for one "The Aastra 6750i Series offers features and flexibility in an open-standards based, carrier grade IP telephone." for the

Hi , I had some problems in the past with sip trunks, asterisk-users Digest, Vol 68, Issue 4, message 6, and had a reply (message 9) saying that It could be a dns issue. Well today I had a problem again with sip module and it really seams a dns issue. I have an asterisk, version 1.4.26.1, that has 4 bri access and two sip trunks. I'm having internet access problems and when this happens

Hi , I am trying to get some estimator based on lognormal distribution when we have left,interval, and right censored data. Since, there is now avalible pakage in R can help me in this, I had to write my own code using Newton Raphson method which requires first and second derivative of log likelihood but my problem after runing the code is the estimators were too high. with this email ,I provide

Hi, Many SIP phones implement list-based Notify-Subscribe mechanism with the phone may request to be notified of status changes from a whole list of resources. Thanks to PJSIP inclusion in Asterisk 12, I'm wondering how a Resource List Server could be implemented with Asterisk 12. 1. I couldn't see RFC4662 itself is implemented in PJSIP. Is this correct ? 2. Which architecture

https://bugzilla.samba.org/show_bug.cgi?id=5407 Summary: hlink.c:480: finish_hard_link: Assertion `flist != ((void *)0)' failed. Product: rsync Version: 3.0.2 Platform: x86 OS/Version: Linux Status: NEW Severity: major Priority: P3 Component: core AssignedTo:

Hi,

Hi, I am trying to get the maximum likelihood estimator for lognormal distribution with censored data;when we have left, interval and right censord. I built my code in R, by writing the deriving of log likelihood function and using newton raphson method but my estimators were too high " overestimation", where the values exceed the 1000 in some runing of my code. is there any one can

Can you tell more about your setup? Is zarafa and samba on the same server for example. Which MTA are you using postfix/exim?   My top was about 150 users, and all my printers are connected also so about 200 devices do ldap searches. but my setup is split over 10+ servers ( 2 are AD DC )   So best is to tell what you can about your setup, anonimize if needed.   Greetz,   Louis  

hi@all i have a problem when hangup an incoming call, i receive this error: Incoming call: Got SIP response 503 "Server error" back from xxx.xxx.xxx.xxx. and the caller stay connected and don't receive hangup any idea? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, when I still had ISDN, I was using Hangup(causecode) to send e.g. "Wrong number" to unwelcome callers. Meanwhile, I am only using SIP providers (no PSTN lines any more) and I would like to do similar, i.e. send specific SIP headers. Besides "wrong number", I would especially like to send 302 temp moved with a specified address to deflect certain calls. Is there any way to

Hi list! I'm trying to configure Asterisk to record incoming calls, if the called press *3. I added in features.conf: automixmon => *3 then, in my dialplan: exten => 1,n,Dial(SIP/00493511111111,20,RcxX) Well, if I **CALL** a number I'm able to record the call, but if I'll be called, and press *3 nothing happens... In the console I can't see anything, too. Could you

Is there a way to divert incoming calls on DAHDI T1 channels so telco gets the diversion and send the call to new number and releasing the channel? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140425/d50c1f18/attachment.html>

Hi All, Can someone please tell me how to limit incoming calls to SIP channels using the SetGroup & Checkgroup command. I don't want any call waiting on SIP channels and you are somehow meant to be able to do it with these commands. Many Thanks Daniel Niasoff

Hi all, I've been using the following method to limit calls to sip clients to 1: exten => 200,1,SetGroup(200) exten => 200,2,CheckGroup(1) exten => 200,3,Dial(SIP/200) exten => 200,103,Busy This works fine for a single extension. However, I also need to dial groups of sip clients. It appears that SetGroup can only be used once per channel. This (useless) example would not

hi i am creating some variables from same data, but somewhere is different rouding. look: P = abs(fft(d.zlato)/480)^2 hladane= sort(P,decreasing=T)[1:10]/480 pozicia=c(0,0,0,0,0) for (j in 1:5){ for (i in 2:239){ if (P[i]/480==hladane[2*j-1]){pozicia[j]=i-1}}} period=479/pozicia > P[2]/334 [1] 0.0001279107 > hladane[1] [1] 0.0001279107 > P[2]/334==hladane[1] [1]

On Mon, 2017-03-27 at 10:43 +0200, Gaetan SLONGO via samba wrote: > Zarafa is not on the same server as Samba  > > We only have 2 AD/DC Samba 4.5 (CentOS 7) and we put required indexes > on LDAP .  > > Arround 1000 mailboxes but not all are simultaneously in use (approx > 1/3 in use).  > MTA is postfix (and is still connected to Samba AD, this one is not > causing the

I'm looking for a way to delay the disconnection of a call to a SIP extension (or pad it with silence) for a few seconds, after an incoming call to that extension hangs up. Rationale: I have an Asterisk PBX (current 1.4.20 codebase), with a Leadtek BVP8051S ATA hooked to an analog phone which has a built-in answering machine. Incoming SIP connections to the appropriate extension are dialed