similar to: Asterisk 1.8 Dimensioning.

Hello, does anybody know about an existing skill-based routing solution for asterisk? I found only some theoretical documents on voip-info.org. I would like to have finer control over who can get which call in which order. Example: Several operators with several topics. Each operator may have a given knowledge-base for given topic. Topics may be weighted in question of complexity as well. Some

Using pjsip 2.7.2 on Asterisk 15.5 Really struggling to make sense of translating these old 1.8 SIP instructions into a neat pjsip_wizard conf suitable for 2018 http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 In pjsip_wizard.conf, I have the following, which seems to get me registered, and it responds to an incoming call, but I always get this: [Jul 28 18:32:29]

Dear all, I have enabled stun module and configured it in asterisk , but asterisk not using stun returned public ip address for any of the sip requests going out of my network. i have done settings as below res_stun_monitor.conf settings: [general] stunaddr = stun.ideasip.com stunrefresh = 30 stun show status Hostname Port Period Retries Status ExternAddr

I'm trying to configure sip2sip, which says: http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk "Asterisk, is currently unable to handle more that one result for a DNS SRV lookup, and the Asterisk configuration needed for getting it work with the SIP2SIP service is not trivial" It then gives a complex multi-section workaround in SIP. I remember reading there'd be

Hi, Some help with dimensioning the server will be gladly accepted. -50 sip phones (g729) or g711(to avoid transcoding) in LAN -an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN -Some sporadic conferencing with no more than 2 sip phones and maybe 2 or 3 calls coming from the E1 for a total of 5 people in a conference. The asterisk server will get an E1(pri) via one

Hi, I have to create scenario like following, I have 2 sip soft phone.I configured Asterisk server on local network, on Linux.With two soft-phone , local asterisk sever, i able to communicate.Now i have communicate with other network SIP client.For that i have opened account at @sip2sip.info, they provided me credentials.Then i registered one SIP phone to local Asterisk sever and another to

Hi, I have some customers complaining about moh quality, but never about voice quality. The moh files themselves sound great when listen to from a media player, and given that I am streaming the files through DAHDI (while voice goes from DAHDI to SIP, so more chances of problems) I would expect moh to be of more consistent quality than voice itself, but it turns out to be the opposite.

Has anyone noticed a problem with runaway mpg123 processes for music-on-hold eating up ~100% CPU and driving the load on the machine way up? I've seen this problem consistently with multiple Asterisk installs, 1.2.x and 1.4.x, although admittedly it was more common with 1.2.x as far as I can tell. There is no clearly identifiable sequence of events that causes this to occur, although it

I'm having trouble getting the quota plugin to work with deliver. The quota in the database is not getting set. I am using dovecot-1.0-rc15. This server is not yet in production. Some values below are because I'm trying to test with the simplest possible case, and some because it's reading from a different database. Also, I noticed that "dovecot -n" doesn't display any

Hi, is there any good calculator/table/reference about proper dimensioning? I read the wiki and they basically say "xx users run fine in yy hardware" http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning. SO far I read that: -Run up to 4 E1s per CPU (which one? an i386 or a dual core? -it is very CPU intensive to do transcoding. Try to minimize it. -you can help the CPU

I have been trying to get a feel for scaling or dimensioning using asterisk 11. if I desire to use something like a dell r320, hardware RAID, 2G E5-2420, 4G RAM and only SIP trunking using gsm (least bandwidth and no transcoding) how many calls "out" can I expect to make at one time and asterisk still be OK and responsive? Thanks, Jerry -------------- next part -------------- An HTML

Where can I find some numbers of Asterisk Dimensioning on newer processor like i7 like number of parallel encoding/decoding sessions. I can see some data on wiki but all are for dated systems. Jim

Would greatly appreciate any input into this currently-unanswered question on the forum: http://forums.asterisk.org/viewtopic.php?f=1&t=96496 I posted it on Jan 6th, have tried so many things, so much forum/list searching and late nights since, but have had to admit defeat. Rather than duplicate it all here, I've posted my logs and conf files on that thread, too. Problem is that while

See comments below. Henry Devito wrote: > Here's the problem. When I call 555 to login, it asks for the agent ID > which I enter as 501, it asks for the password which I enter as 1234, > then it asks for the extension I dial 501 It then says that extension is > not valid. What am I missing? Of course 501 is valid I can make and > take calls from it now. > > >

I'm considering using the Agent "login/logoff" function to add to a queue that will be our "main number during the day" to answer. Periodically our receptionist is not at her desk and would be useful for her to login elsewhere and get the main number calls to transfer as she sees fit. If the agent's don't pick up in a specific amount of time, it's

Has anybody got the polycom acd function to work? I have the following setup: Debian 3.1 - 2.6.8 linux zlib-1.1.4 libpri-1.2.3 zaptel- 1.2.6 Asterisk - the bweschke/polycom_acd_funtions branch version - I get one error when doing a make install about needing a newer version of libpri and zaptel, I got the above versions from asterisk.org, are there newer version anywhere else? In the sip.conf

Hi, I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk 1.4 + CDRTool with freeradius telephony system. Asterisk is used only for voice mail and redirectioning calls. Every calls should pass through mediaproxy so that i can account them. The goal was to create a simple prototype of what could be a VoIP provider. Now i need to dimensioning this system to work

Good day, Our installation needs a robust ACD application (as I'm sure others do) that can be dynamically reconfigured (if possible) maybe by a MySQL database. I have looked at Bill Heckel's ACD work and Andreas Otto's DynExtendb as well as James Sharp's ACD. None of these seem to be quite finished. Is there a chance a unified (* blessed) ACD application can be put

Hello Sir, I have few clarifications as we are planning to work with asterisk. If you don't mind, please clarify the following:- Q1. Do Asterisk support ACD functionality? If Yes, can you give information on how to configure or work with ACD (and it's usage). Q2. From the list of features listed in www.asterisk.org , I see "Predictive dialler" is listed

Hello, I cannot call my server over the internet with SIP anymore. Even when I do a maximum logging on my firewall, I don't see packets coming from outside. I've tried it from an ekiga.net account and an sip2sip.info account. What could be wrong? I would expect incoming traffic on port 5060 UDP... The account is "paul at vandervlis.nl". This should connect trought DNS to the