similar to: DID Terminations

Displaying 20 results from an estimated 3000 matches similar to: "DID Terminations"

2004 Jul 02
2
Zaptel, Line Impedence and Echo
Hi, I'm not sure if I just missed something somewhere along the way, but I noticed while I was going through the CVS logs that there is an option in the wcfxs module to set an "opermode" - which apparently might help with echo issues around the globe (like the ones I'm seeing - some times). So I'd love to use it and give it a go. Apparently, there are some options that
2004 Jul 15
4
ZyXEL 2000W
Hi, I know we've talked about this phone to death. I have pretty good voice quality, with and without WEP enabled, using the G729a codec and DLink & Netgear access points. I am facing one obstacle that is driving me insane. Does anyone have the call hold feature working? If you do... how did you make it work? The instructions say to press the left button to place the call on hold,
2004 May 23
3
NetJet and RAS
Hi, Last weekend I was planning to buy a physical PBX system, but instead I have been blown away by the fact that VoIP really works, that Asterisk is so easy to set up and use... and free! We're in Australia, so as I understand it, we aren't allowed to use the Zaptel cards. We need to set up our system to route incoming 56K data calls to the PPP daemon on our Linux box or to our
2004 Jun 25
2
Latest CVS fax detection & grandstream bug
As a follow up to my previous post, I have now identified what is causing the bug with the grandstream phones. When the line faxdetection=incoming is in the zapata.conf file, the grandstream phones will not ring, nor connect a call to the zaptel interface. Can anyone else confirm this bug? I'm going to play with the different options (incoming/outgoing/both) to see if it makes a
2004 Jun 05
1
FXO answering quicker
Hi, I don't know if this is possible - but can I set up asterisk to answer the FSO line after one or two rings rather than four? I haven't (yet) found a configuration variable to let me do this... Thanks in advance, Andrew _________________________ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568
2020 Jul 24
4
Remove ANSI colour trings from log files only
Hi, Is there a way to drop the ANSI colour strings from log files? In particular, I've got JSON logging throwing logs over to ES, but they have the ANSI colour escape sequences. Ideally I don't want to lose coloured logs from the console though, and I can't "see" a way to do this. Ast 16 at the moment… Andrew -------------- next part -------------- An HTML attachment was
2020 Jul 18
2
PJSIP AoR vs Endpoint
Hi, I realise this is an old question, but I’m struggling to get my head around it. The ERD suggests that endpoints can link to multiple AoRs In what situation would you actually use this? Given that mapping of inbound calls is primary done to the endpoint, it looks to me like most of the scenarios where this might be beneficial are actually not possible? One example I had envisaged was being
2004 Jun 24
1
Latest CVS, Grandstream and Zaptel bug?
Hi, I'm confused as anything by this bug. I'm hoping that it is just something screwy in my config. I have 1 Cisco 7960 and several Grandstream BT101 & 102's, and a Digium TDM31B. I'm running the latest CVS (CVS-HEAD-05/27/04-17:22:40) of both Asterisk and the Zaptel driver on Fedora Cora 1. When I make an outgoing call on the Cisco phone, everything works fine. I'm
2013 Mar 11
2
Serviced Office operator panel
Hi, I'm trying to find (with some desperation now) a decent web based or application based UI that integrates with an Asterisk based PBX and is designed for a Serviced Office environment. Key features we're looking for: Concept of a client with multiple staff/associates Recognition of inbound DIDs with display of company details and greeting Support for multiple receptionists Some
2004 Jul 11
1
Echo issues (again...)
OK... so I'm not sure what I'm looking at. I've had the good old echo problems on my Rev C FXO again this morning, so I thought I'd attempt some debugging, though I'm not sure what I'm looking at. This call has echo. Channel: 2 File Descriptor: 20 Span: 1I> Extension: Dialing: no Context: incoming Caller ID string: "External Call" <99999999> Destroy:
2004 Jul 07
2
Parking call problem
I been having a issue with call parking. I can park calls from internal extensions. But call from the outside can not be parked. When I recieve call from the outside I press the # key and nothing happens. Does any one have any thoughts? P.S. I am allowing the to be transferable. James --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com).
2020 Jul 11
1
16.11.1 release removed from current
Hi, This is not a big issue… I just noticed a build script that was pulling the 16.11.1 release from https://downloads.asterisk.org/pub/telephony/asterisk has started to fail and went to investigate (I need to test a patch for a bugifx). It looks like when the 16.12-rc1 RC was released the 16.11.1 was pulled from the current release directory. I realise it's probably better practice to pull
2003 Nov 24
3
Cisco to asterisk termination with h323 and g729 finally works.
Hello, I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323. I tested both oh323 from inaccessnetwork and JerJers chan_h323. I used 1.12.2 version of oh323 and 1.5.2 version of pwlib. After latest changes from JerJer chan_h323.c works ok when receiving traffic from ciscos. I havnt found any audio problems although I didnt send much traffic. Latest oh323 has some
2004 May 25
0
Compiling on OSX 10.3.3
Hi, Just thought I'd let you guys know about my time in compiling on Panther. Not being the world's greatest Makefile expert, and after applying the patch suggested on this list (and modifying it so that the 7.2.0 was replaced by a 7.3.0 - for the latest Darwin kernel) I still had issues with my system requiring endian.h and byteswap.h . Being ignorant of most things, after much
2004 Jun 08
1
Outgoing call via Fritz!
Hi, I'm trying to get a fritz working in our asterisk box, however I'm getting the following error and was wondering if anyone could be of any help: -- Executing Macro("SIP/2002-5501", "enum-call|call|0405152568") in new stack -- Executing Goto("SIP/2002-5501", "call|0405152568|1") in new stack -- Goto (call,0405152568,1) ==
2020 Jul 22
4
Failed to authenticate device message
I am getting this message: Failed to authenticate device <sip:2010 at X.X.X.X>;tag=149853321 for INVITE, code = -1 but it does not report the "connecting" address. Who is failing connecting ? I either need to block someone or fix something - I'm thinking block - but I dont know who. How do I found out the connecting IP? Jerry -------------- next part -------------- An HTML
2011 Aug 30
1
Running Modern Warfare 2 on Ubuntu 11.04 Natty.
My hardware is Intel Core i7 4GB ram, Nvidea GeForce 310m graphics card. I installed my graphics drivers in Ubuntu 11.04 using the recommended techniques. My video drivers are noted as experimental however all features and 3D support is working in Ubuntu 11.04 gnome environment. I did not use the .run file downloadable from the Nvidea website. I have read the sticky about no sound. I have no sound
2018 Nov 01
1
Intro
- your *FirstnameLastname* username JohnBoero - the proposed subject of your Wiki contribution(s) To seek the holy grail. But mostly to fix the rampant 404 download links for CentOS Atomic media here: https://wiki.centos.org/SpecialInterestGroup/Atomic/Download - the proposed location of your Wiki contribution(s) https://wiki.centos.org/SpecialInterestGroup/Atomic/Download Ex
2017 Mar 21
1
rename Administrator account
>Sure you can rename it. Being a member of the right groups decite what >an account can do. >However, I don't understand how renaming the admin account improves the >security. For example, every domain user can easily find out who is a >member of the "Domain Admins" group: >> dsquery group -name "Domain Admins" | dsget group -members
2004 Jun 20
1
Data over Voice through Asterisk
Hi, I'm trying to make a dialup internet connection through my asterisk PBX. When I bipass the Asterisk box, I can get 51600bps. When I run through the asterisk box, I'm limited to about 21600bps. I have a TDM31B card. Any help on speeding these connections up would be good - I was on the understanding that if you bridged the channels, then the call should essentially flow straight