similar to: 3 companies 1 card

Displaying 20 results from an estimated 1000 matches similar to: "3 companies 1 card"

2004 Aug 13
2
not hangup
Good day all I'm using sip protocol and a openline4 card.If I dial out of the pstn and hangup a answered call it does not disconnect the connection.It shows there is still a call on the external phone I called but on my side its says i'm not connected.We are using x-ten soft phones Can someone please help me Thanks Altus
2004 Apr 30
2
South-Africa
Good day all I'm in South-Africa,currently we are using openline4 cards for our pbx systems.Now we first need approval on the cards form icasa(a government standards) before we can use the card.The market here is very big for a system like asterisk.The only problem is to get a card approved(for a small company like us) its just about impossible. Now what I'm looking for is a company that
2015 Apr 07
5
Asterisk Inbound calls, multiple SIP accounts, calledID
Hi Dmitriy and others and thanks for your help so far. The option "match_auth_username=yes" seems to have had no effect. From my reading, this option will try to match the username of the incoming SIP account to a section heading. If that is how it must work then i can see a big problem. I'm trying to present the receptionist with a nice display of which line the call came in on.
2004 Nov 25
1
No hangup(vpb)
Good day all We have a voicetronix openline4 card If someone calls in from the outside the pstn and into the system and hangsup asterisk does not deteck the hangup any Idea why please Help Altus
2004 Apr 05
1
sip no sound?
Good day all So I've installed asterisk with my openline4 card and I've setup sip and I can do sip on the local network,we are using soft clients,x-lite. But... When a call comes in from the outside(PSTN) and the you dial the extension it forwards the call the the client and you see incoming call on x-lite,you accept he call....BUT there is no sound.It shows there is a call and you are
2015 Apr 01
4
Asterisk Inbound calls, multiple SIP accounts, calledID
Hello all, I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts with the same service provides. We have 8 phone numbers in total. Incoming calls from the public are all correctly directed to appropriate office handsets. However, the display on the reception phone (the only one i care about) is always showing the same "SIP/Account1_0843214321" rather than the account
2004 Nov 27
4
very newbie question
Hi everyone! I have very simple question, how to limit SIP phone user making calls to for example longdistant calls? Maybe: Put in his context in sip.conf context which don't provide possibility to make such calls? Is it correct? thanks for any help, regards, Corvin
2004 Apr 03
7
Few question on HTB
Dear All, Sorry to trouble again..... After go through www.lartc.org I have implemented the HTB instead of CBQ for the same scenario. Now following files are under /etc/sysconfig/htb directory. eth0 DEFAULT=30 R2Q=10 eth0-2.root RATE=256kbps BURST=25k eth0-2:10.comp1 RATE=120kbps BURST=12k PRIO=0 LEAF=sfq RULE=192.168.200.0/24 eth0-2:20.comp2
2004 Jun 14
1
Multiple tennants, two DIDs, One IAX provider
I would like to setup a system with two tennants with two seperate DIDs through one IAX provider account. Is it possible to route the calls into different contexts based on the DID dialed? I have searched and found nothing. I do not see anywhere in the console that says what DID was dialed so I am thinking two seperate accounts are needed to make this work. Can anyone confirm? Thanks
2004 Oct 07
1
dial out
Good day all I'm getting this error while trying to dial out on my asterisk server using a openline4 card "exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872" Please Help me
2004 May 26
2
Voicetronix OpenLine4 -- Help Needed
Hi. I need help with my brand new Voicetronix OpenLine4 board that I installed into Asterisk. After building the Linux device driver and inserting the module, I modified the /usr/src/asterisk/channels/chan_vpb.c file to uncomment the US settings and comment out the Austrailian ones. I made the appropriate entries for routing in vpb.conf and extensions.conf.... All appears to be well, except
2004 Sep 06
1
Voicetronix OpenSwitch12
Hi all, I used to have an OpenLine4 card, but decided against using it due to some problems with hangup detect. Does anyone on the list actively use Voicetronix's OpenSwitch12? What are your opinions on the card? Cheers, Flynn
2004 Aug 13
0
incomingcall braking all
Good day all We have a voicetronix openline4 card.Asterisk is configured for sip with all the extensions and all&all. I can call out and internally,to dial out I have to dial 0... My problem is with incoming calls If I call my external PSTN number,asterisk answers with the default message and if I press the extension it goes to the right sip client. BUT As soon as I hangup this call all gets
2004 Sep 13
5
music on hold not strting
Good day all I added the music on hold entry in vpb.conf and commented out default line in musiconhold.conf. Asterisk starts up with the default mp3 but as soon as I remove it and add my mp3 it just doenst start up and gives a broken pipe error? Please Help or advice Thanks ALtus
2004 Sep 08
3
sendmail&hostname
Good day all I'm just wondering for interest sake I have a box,hostname=myname.co.za,running sendmail If I send mail to someuser@myname.co.za it try to deliver to the box,witch does not have the mail box.How do I tell sendmail that it should send mail to myname.co.za's mailserver. I know how easy it is to change the name but there's a lot of reasons why we can.It is not in the
2006 Mar 01
1
SIP contexts being confused
I have an * system running version 1.0.8 and it works mostly fine. I am using it as a virtual PBX and we share the box among companies. I have the calls all staying separate, we well as the companies' extensions, voicemail, etc. The only problem I'm having is with two accounts that use the same SIP termination provider. * seems to be confusing the sip contexts for the incoming calls.
2004 Aug 05
2
personal voicemail
Good day all IS there a way to personalise the voicemail message when you leave a message? Thanks Altus
2005 Sep 15
2
cdr server
Good day all Is it possable to set asterisk up as a cdr server for other voip units We got a quintum dx here and its got a option to log to a cdr server on port 9002 Thanks Altus
2005 Feb 08
2
bri dropping calls
Good day all We have a quad bri card,installed on fedora core1,downloaded the latest bri-stuff that download asterisk 1.0.3 and zaptel 1.0.3 and libpri 1.0.3 All installed and working.BUT after 5min+ of talking it just drops the calls? Any reason why? Please help Thanks Altus
2016 Mar 21
2
transfer FSMO roles from Windows DC
I have the Active Directory domain with Windows 2008 R2 domain controller and Samba domain controller on CentOS 7. Samba is 4.3.5 (self-compiled). Forest and domain levels are Windows 2008 R2. After joining Samba to the domain as the domain controller there were no DC=ForestDnsZones and DC=DomainDnsZones records on "OUTBOUND NEIGHBORS". I fixed it with ntdsutil, as it's written here