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  • Asterisk SIP digest authentication username mismatch

    - by Matt
    I have an asterisk system that I'm attempting to get to work as a backup for our 3com system. We already use it for a conference bridge. Our phones are the 3com 3C10402B, so I don't have the issue of older 3com phones that come without a SIP image. The 3com phones are communicating SIP with the Asterisk, but are unable to register because they

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  • Asterisk SIP digest authentication username mismatch

    - by Matt
    I have an asterisk system that I'm attempting to get to work as a backup for our 3com system. We already use it for a conference bridge. Our phones are the 3com 3C10402B, so I don't have the issue of older 3com phones that come without a SIP image. The 3com phones are communicating SIP with the Asterisk, but are unable to register because they

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  • asterisk queues priority and penalty

    - by MealstroM
    queues.conf shared_lascall=yes strategy=rrmemory wrapuptime=15 A1,A2,A3 are members of 2 queues: queue1(Q1) and queue2(Q2) A3 has penalty 3 in Q1 where min/max penalty are 0/3 and A3 has penalty 0 at Q2 where min/max penalty are 0/3. A3 has just ended a call and is on wrapuptime pause. User1 (U1) enters Q1 with priority 10, and user2 (U2)

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  • Asterisk not playing custom sounds on Ubuntu Server 11.04

    - by jochy2525
    I've installed Asterisk on my Ubuntu Server, all works fine, excepts playing the custom sounds. Asterisk sounds work, but this file I've uploaded does not play (on other servers it works, it is a .WAV PCM 16bit 8000). Here is some log output: [Feb 6 22:55:45] WARNING[11045] file.c: File custom/sohoitsoluciones does not exist in any format

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  • Skype Connect as SIP/Trunk for Asterisk

    - by Kaurin
    First off: I'm not sure if this should be on superuser or here. I have recently built a few Asterisk boxes with OpenVOX FXO/FXS ports little or no trouble. My current project is building an Asterisk box with SIP trunks. My current employer insisted on getting Skype Business/Skype connect for that purpose. After reviewing the Skype

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  • cli commands not working asterisk on ubuntu

    - by Mian Khurram Ijaz
    hi guys today my first day on asterisk on ubuntu. I installed vmware and then ubuntu and then on ubuntu i am running asterisk. i started the asterisk server successfully by following a tutorial. After making few changes into the sip.conf i want to reload the sip.conf i issue the command sip reload and nothing happens neither

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  • Asterisk failing at startup after upgrading to asterisk18

    - by Supratik
    I was using asterisk16 and asterisk16-skypeforasterisk, which was working fine. I have recently upgraded to asterisk18 and asterisk18-skypeforasterisk, after that I am receiving the following error message. Asterisk ended with exit status 1 Asterisk died with code 1. Asterisk could not start! Use 'tail /var/log/asterisk/full'

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  • Can we manage incoming call number landing on asterisk?

    - by user194469
    I am using Asterisk 1.4.2 in two different machine. I have configured some extensions in asterisk. When any caller, dial my extension number with local number then if I see asterisk console (asterisk -r) then incoming number is starting with 0, but if caller dial same extension number using STD number then in asterisk console

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  • Suggestions for a SOHO/Home Asterisk system

    - by James
    I would like to buy a small embedded system that runs Linux and Asterisk. I would like two FXS ports ( to plug analogue phones into ) and one FXO port ( to plug in to my real line to allow access to the POTS ). I would like it to have a USB port to hold storage for voicemail. I really want it for home use so I would like it

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  • Parse and validate Asterisk dialplan before commiting

    - by adaptive
    I recently made a number of changes to my Asterisk dialplan and would like to validate these changes before I commit. I am thinking more from a "write code" - "compile" - "debug" approach. I am very new to Asterisk and am trying to build my dialplan slowly but the server is already in use (by the spouse) so I'd like to

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  • PSTN Trunk TDM400P Install on Asterisk / Trixbox

    - by Jona
    Hey All, I'm trying to get a TDM400P card with FXO module to connect to our PSTN line. The card is correctly detected by Linux: [trixbox1.localdomain asterisk]# lspci 00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface I've run setup-pstn which produces the following

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  • Interpreting and using the Asterisk "timing test" command

    - by zigg
    Timing is very important for certain kinds of applications in Asterisk. If DAHDI is the timing source, the dahdi_test command can be used to check the timing provided by the DAHDI kernel module. If dahdi_test returns exclusively measurements above 99.975%, the DAHDI timing source is generally considered good. Since

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  • Asterisk, IAXModem & Hylafax how-tos?

    - by Brian Postow
    I'm trying to set up Asterisk and IAXModem to send faxes via T38 (Yes, I know I'm swatting a fly with a Buick...) However, since I'm trying to do something so small with a product so large, I'm having trouble finding samples or how-tos that show me how to set this up. I've got all three installed, and I THINK I have

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  • Cant connect to asterisk internal database [on hold]

    - by Bilbo
    Im trying to get a PHP script to connect to Asterisks internal mysql database. I tried the to use the standard method for example $con = mysqli_connect("192.168.1.126","root","mysql","asterisk"); However when I log into the asterisk server to access the mysql database all i need it to type "mysql" and im logged

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  • Unable to call through asterisk

    - by sk
    I want to create a voip service. I have installed asterisk-1.4 on a dedicated remotely hosted debian lenny distro. I made a sip.conf and extensions.conf so as to place a call between two sip phones(i am using xlite 3.0) installed in some other Windows PC. Whenever i switch this phones the asterisk console shows

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