Archives : July-2015
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everyone,A few days ago I had a problem with a couple of extensions. I have about 12Aastra 6731i phones, 6 are at our main office and 6 more on remote branches. We use VPN to communicate to our branches so theres no NATinvolved any where.The prob..
I would like to setup a mechanism to trigger an alarm if user is deal too many numbers within a very short period of time. Safeguard against users hacked accounts.can someone he..
AllI have a problem with mixmonitor in 13.4.0 doing the following:1. Caller phones in2. Reception picks up3. Talks to caller4. Does attended transfer, talks to manager to screen the caller wanting to speak to him5.Complete the transfer by putting d..
Hi.I have a beginner conceptual question about Asterisk:Lets suppose that there are 4 softphones registered in my Asterisk and all of them are currently online. In addiction , there is no call.Suddenly, one of these softphonessends a SIP message to ..
can U help mecaller id in USTM if now working– Starting switch on 4211@4211-1 to 4203– Executing [4203@office:1] DumpChan(USTM/4211@4211-0x7f7ba4228fd0,) in new stackDumping Info For Channel: USTM/4211@4211-0x7f7ba4228fd0:================================================================================Info:Na..
list, Im trying to install gmime22 package which is one of the packages reported as required by ./contrib/scripts/install_prereq test. Whatever I do Im getting to a dead end. On the regular yum repositories that I use (CentOS, epel, rpmforge, asteri..
!I noticed that when the phone of my wife calls the gsm codec will be used, but if someone calls the phone, alaw will be used:00493511111111 calls 00493512222222:OpenWrt*CLI> sip show channels Peer User/ANR Call IDFormat Hold Last MessageExpiry Peer192.168.200..
!Yesterday I set up a voicemail on my Asterisk 1.8. It works as expected, but Id like to have the CID without unnecessary prefix…Right now, if I call from my mobile phone I hear the complete prefix for my mobile number, indeed without 00. So I h..
I am migrating a PABX system based in Asterisk 1.4 to Asterisk 13, with success.I have an application that sends an action Originate to AMI for calling, its working well, but when i see to Asterisks CLI, i see 2calls for just one originate:pftestes40copiabh*C..
I am trying to get Asterisk 11 to co-exist with a CentOS 7 box that has pulse audio running as a l..