similar to: Incoming voice "disappears"

I've searched the Wiki and this forum with little success. I have a TDM400P in my server which functions fine. Except it will continue ringing about 3 times after hangup. I.e. it's failing to detect the hangup tone. I was previously running a Sipura 3000 and had the same issue. After researching and some timely assistance I was able to determine the hangup tones applicable to

Hi, Not sure if this is OT or not - apologies if it is. I'm looking for an asterisk consultant in the Sydney area to help with the rollout of a couple of Asterisk setups for clients of ours. If you're interested please reply to me /*off-list*/ at tonyd -at- zeroeffortnetworking -dot- com -dot- au (fixing the address of course :) thanks, tony Zero Effort Networking Pty Ltd ABN 38

Is there a a consensus on which of these is the better phone.I've personally been using an 841 and have learned to live with its shortcomings. I now need to recommend some phones for some sites we're installing. I'm looking at the BT102 for desktops that don't want/need a headset but need a phone for the higher end users (without costing the earth). TIA, tony Zero Effort

Le 19/11/2018 à 15:00, Julien TEHERY via samba a écrit : > Le 19/11/2018 à 12:33, Julien TEHERY via samba a écrit : >> Le 19/11/2018 à 11:14, Marco Gaiarin via samba a écrit : >>> Mandi! Julien TEHERY via samba >>>    In chel di` si favelave... >>> >>>> Is there a good pratice when adding new remote DCs in terms of >>>> replication

Cordialement, Doe Corp <https://www.openevents.fr/> <https://www.facebook.com/OPENevents-172305449504004/> <https://twitter.com/SocOPENevents> <https://www.linkedin.com/company/openevents/> Julien Téhéry Ingénieur Systèmes & Réseaux | OPENevents 15 avenue de l'Europe 86170 Neuville de Poitou phone : +33 5 49 62 26 03 <tel:+33549622603> mail :

Hello all, I was wondering if anyone know where i can find a book on Asterisk, i have been told about VoIP With Asterisk but i am unsure where to find it, any ideas plase?

Hi all, I have some problem to send a FAX to a given number. I use asterisk 1.2.18, on a openSUSE 10.2, i586 host. The FAX is sent out via an ISDN PRI interface, I'm in Germany, and the destination FAX devices are in Germany too, but in different areas, so I have to use a city prefix. I did set the pri device in debug mode, below are two calls, to two different FAX numbers, the first is

I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip extensions and a regular phone connected to the box. All routing works fine from the regular phone connected to the box, whether its going to FWD, broadvoice or the PSTN. The problem I am experiencing comes from making calls from the sip phones. They get routed correctly to the sip and iax trunks but when making calls

Hi Guys, I'm having sound problems when diverting a call using asterisk@home 1.5. I am using the following configuration in extensions_custom.conf, extensions_additional.conf and extensions.conf [custom-Sales] exten => s,1,SetVar(DivertNumber=02XXXXXXXX) exten => s,2,Dial(SIP/116, 15) exten => s,3,Goto(outrt-010-outside3,9${DivertNumber},1) (i have replaced the diverted phone

Dear All, I have an asterisk server version 1.2.12.1 along with trixbox and I am having this nasty problem, I have a TE200P and have an E1 pri attached to it and zttool says it's OK, I have configured the whole 31 channels into one group as follow: Zapata-auto.conf: callerid=asreceived signalling=pri_cpe switchtype=euroisdn context=from-zaptel group=0 channel=>1-15,17-31

Hi! I have configured a SIP trunk with a dialing rule. The trunk behaves normally for incoming calls but when in used for outgoing call a strange thing happens. When I place a call I cannot hear the tone confirming that the remote phone is ringing. I simply hear the voice as soon as the party picks up. When the remote phone start ringing Asterisk receives a SIP packet stating that the call is

Hello, I have *@Home 1.5 installed and all is working fine for incoming calls and sometimes outgoing calls. Installed in the box is a digium TDM04B (4xFXO Ports) setup as ZAP1 to ZAP4. I have incoming calls coming in on lines 1-4 in that order and outgoing calls prefering ZAP4 then ZAP3 then ZAP2. When i try to dial out to the PSTN from a SIP phone it sometimes works (normally after a reboot)

I am posting this in case someone need help.. ========================================================= YOU THA MAN!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! No sure how I will repay you, but anything you need, just let me know!!!!!!!!!!!!!!!!!!!!!!!!!!!!! Thank you, thank you, thank you!!!!!!!! -- Executing GotoIf("SIP/2007-12c7", "0?4") in new stack

Does anyone have a solution for the problem of file and directory names containing characters which are OK in DOS/Windows but illegal (or at least unwise) in Unix? I have Windows clients who have written directory names containing "$" characters to Samba shares on a Solaris box. This is fine for the clients, but plays havoc with Solaris programs needing to access the files, eg. backups

I've a call center using IAX softphones provided by a third party. We've observed problems where the IAX phones seem unable to use our PRI trunks. A sample anonymized call is provided below with the PRI debug calls embedded. Any thoughts, comments or suggestions would be welcome. In anonymizing it, I preseved the format and number of digits sent. -- Accepting AUTHENTICATED

I just grabbed 1.9.18p2 and tried compiling with gcc 2.7.2.1 for a Solaris 2.5 box, but it didn't get too far - after a bunch of macro re-defines (eg. "/usr/include/sys/termios.h", line 32: macro CTRL redefines previous macro at "/usr/ucbinclude/sys/ttychars.h", line 65 "/usr/include/sys/termios.h", line 150: macro CINTR redefines previous macro at

Hello, I have a TDM400 and currently have 2 of the ZAP Trunks configured on it. Zap/1-1 and Zap/2-1. I am Running Asterisk@home Version 2.4 with AMP version 1.10.010 In my Outbound Routing I have the Trunk Sequence set up so that 0 is Zap/1-1 and 1 is ZAP/2-1 What I see is that when Trunk Sequence 0 is full, it does not open Trunk Sequence 1. I have found that this is true even if I

Hi all, I've recentrly starting to play around with *, when all I wanted is to configure an fritz ISDN card with A@H. Currently I'm stuck at the phase of what do I do with capi after everything is installed. I'm trying to understand how to setup incoming and outgoing calls at A@H since I'm getting a bit lost with the default dial plan. It seems that * answers but disconnect

Hi, We are using VOIP-SIP gateway to route outbound PSTN calls. Recently, I am getting == No one is available to answer at this time message, after making 5 SIP attempts (Retransmitting #5 (no NAT):), and the calls are going out through alternate Zap-trunk. I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls. Strange thing is that this is happening randomly,

Hello everybody, and first of all sorry for my poor English. I'm having trouble with Quadbri installed on Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e. Everything is working fine, except calling to switched off or "out of coverage" cell phones. In this case I have to wait 40 seconds until Asterisk tell me that "all circuits are busy now" instead of receive cell phone