search for: pbxinaflash

Hello, is there anyone who can point me to correct information ? Following http://pbxinaflash.com/forum/showthread.php?t=9042 and http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql > Extending CDR does not result in a working environment for me. Any feedback appreciated. Kind regards, Jonas. -------- Original Message -------- Subject: [asterisk-users] Asterisk 1.6.2.10 &...

Any vitelity customers with pbxinaflash boxes? I'm able to call in-house, but failing to make outbound calls. My assigned server at vitelity is not reachable. I can ping to my ISP OK. Any help appreciated. Such as actually how to make email contact with support at vitelity. They're not responding. Thanks, Tom

hi; I am attempting to find a way to install pbxinaflash into hypervm, so to find a templet for hypervm with pbxinaflash. Do you guys know of any i can get or can someone design one for me so i can install that templet in a hypervm virtual machine? I can't appear to figure out where i need to go to do this. please help if you can. Michael Mic...

Hi Guys, I have to upgrade to latest Libpri 1.4.10.2 due to an existing bug in the current 1.4.10 version. I am running Asterisk 1.4.x (in fact it is a PBXinaFLASH system). How can I upgrade to the latest Libpri? Do I need to re-install Asterisk? Won't that break the box? Can I simply do this to upgrade: *rm /usr/src/libpri/*.** *rm -rf /usr/src/libpri/** Download the new libpri and put files in folder /usr/src/libpri/ *cd /usr/src/libpri &&amp...

Hi Everyone, Just noted that PBXinaFLASH failed me again on yet something else. The uniqueid field didn't update on the asteriskcdrdb in the past few months but it is available in the .csv files in /var/log/asterisk/cdr-csv/*.csv I have already re-did the asterisk-addons install with correct parameters to include future calls "u...

Hi List, Another dumb conversion question (I hope). I installed 10.0 and copied my 1.4 configuration files over. With a few tweaks everything works great except for 1 feature that I specifically went to 10.0 for. When I do an attended transfer, I still get the receptionists caller ID on the transferred phone instead of the incoming callerID. My assumption is that there is some

I have been asked by my church to recommend a VoIP system which can do the following. They do internet radio shows which are sometimes broadcast on radio. They are looking for a system which does the following for about 5 agents, exactly as they have described it. 1. Take incoming calls 2. Put them on hold if there is no one to handle the call immediately, or transfer them to an available

Hey; I figured I would ask here as I seem to get better results. I am using the voip.ms <http://voip.ms/> VoIP service. I have no problem configuring my Asterisk server 1.8x to dial out with my Softphone. HOWEVER, for some reason, I cannot get inbound. All that I hear is a busy signal. I know this is not much for you folks to go on, but what would be a good place to start

Our small company is replacing Cisco CallManager with Asterisk (because we are tired of sending them money) and I am documenting the process as I go on my blog. I am trying to make the notes as easy as possible in hopes that I can ease someone else's pain. Here is the link: http://qvlweb.blogspot.com/2009/03/asterisk-pbx-system-install-01-what-i.html Please feel free to comment on the

...logs approximately 50 quantity like this; 2009-01-29 08:17:24 SIP/96.9.1... BenQTelecom "BenQ Telecom" s ANSWERED 00:20 I wonder that have anyone ever faced like this problem ? I think it's a security problem, when I googled I have found one topic explanation in english http://www.pbxinaflash.com/forum/showthread.php?t=3265 but without solution. Has anyone any recommend to solve it ? Thanks in advance Yavuzhan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090129/a57aba8e/attachment.htm

Hi List; I am going to use a post and prepaid billing with my Asterisk vesion 1.4, can anyone advise me which of the billing software will be more stable and it is opensource to be used: ASTCC or a2billing? Any advise that might help to take the proper decision? Regards Bilal

...trunking and zap lines on this server but sip doesn't work. I register to an extension but even dialing *97 for voicemail wont' give me any audio. Picture posted here shows my DD-WRT NAT setting: *http://tinypic.com/r/21cuqlu/5* Any input will be much appreciated. This is running latest PBXinaFLASH (which has FreePBX) and I tried using externip=x.x.x.x/255.255.248.0 in /etc/asterisk/sip_nat.conf but it was of no use. Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100323/fdbdeeb7/atta...

Hello Everyone, I have a system that was working on Sunday 1 P.M. and then gives Congestion on Monday morning. Sometimes over night it probably stopped working. It's a PBXinaFLASH with Asterisk 1.4 and libPRI with a 23 channel PRI connected and 24th D-Channel. This is all I see in /var/log/asterisk/full: [2010-04-19 08:45:50] WARNING[29707] app_dial.c: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) Where can I look to debug the pro...

Hi Guys, I am running a PBXinaFLASH server. I replaced contents of /usr/src/libpri with the new version of Libpri v1.4.11. The installed one was v1.4.10. System is running Asterisk 1.4.21.2. I did the following after: cd /usr/src/libpri/ make make clean make install Install end with these lines.....: *ln -sf libpri.so.1.4 libpri...

Hi Guys, I am looking to delete some of the CDR logged by Asterisk in asteriskcdrdb in a PbxinaFlash system running Asterisk 1.4.x The CDR records to deleted are probably a big chunk and spread out all through the database but I basically want to delete all calls that came in through a specific DID. I think they all show as SIP/did_number I don't want to break the system or break the reporti...

...a value to this column by doing the following in the dialplan : exten => 600,n,Set(CDR(mycolumn)="myvalue") But this value is not written to the column 'mycolumn' together with the other CDR-data. Why is this ?! Do I need further configuration ? (Not according to http://pbxinaflash.com/forum/showthread.php?t=9042) Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110322/ad394233/attachment.htm>

I'm looking to install a basic asterisk system for my church with: 8 inbound sip channels 8 sip handsets basic voicemail room to grow (maybe doubling each of the above) What would be a recomended system as to needed processor and memory? Thanks, Eric

The Asterisk Development Team has announced the release of version 1.4.11 of libpri. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ This release contains many fixes and new features, among them being: 1.) Support for NT-PTMP BRI links, including support for multiple TEIs and connecting of BRI phones. 2.) Support for allowing persistent

The Asterisk Development Team has announced the release of version 1.4.11 of libpri. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ This release contains many fixes and new features, among them being: 1.) Support for NT-PTMP BRI links, including support for multiple TEIs and connecting of BRI phones. 2.) Support for allowing persistent

Hi all, I took my decision to use Asterisk server for handling my VOIP calls...My next step is to choose the best hardware that I should use i order to have the best performance...Here I faced 2 choices for my hardware (CPU)... 1- Using Intel CPU or AMD 2- Use 32 or 64 bits Can you help me please to choose between the above choices and what is the advantage and disadvantage of each of choices