similar to: Help Solving Asterisk Lockups

Displaying 20 results from an estimated 4000 matches similar to: "Help Solving Asterisk Lockups"

2011 May 20
2
Faxing with Asterisk 1.8.4 & T.38
Hi - I am looking for suggestions for ITSPs for faxing with asterisk 1.8. We are based in the US, so would need an ITSP with US DIDs. #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to receive faxes via T.38. Sending faxes is not a requirement. Does anyone have a working asterisk 1.8.4 configuration and ITSP provider that they can recommend? We have been trying T.38
2005 Jun 24
4
Tellabs Echo Canceller
I am getting ready to experiment with the Tellabs 2752 echo canceller. I have a 255D shelf (and power supply), but am struggling a little on connecting the echo canceller to a PRI. The shelf has 4 25-pair amphenol connectors. The two on the line side are marked "Receive In" and "Send Out". The 2 connectors on the drop side are marked "Send In" and "Receive
2006 May 25
2
Volume configuration on Polycom Soundpoint 501phone
Could not find your post for 4 months ago. -------------- Original message -------------- From: "Anton Krall" <akrall-lists@intruder.com.mx> > Yes, check a post that I made about 4 months ago, I posted the cofig for > setting the speaker, handset and ring volumes .. > > |-----Original Message----- > |From: asterisk-users-bounces@lists.digium.com >
2005 Feb 09
6
Cisco 7960 Beating a Dead Horse
Hi all, So I have been reading through the docs available online and the different threads on this list, but I cannot seem to get this phone to work. I have configured the OS79XX.TXT and SIP/SEP*.cnf files (see attached), when I configure the phone to point to my tftp server and reboot it I get this message: Connection received from 10.6.0.224 on port 50608 [09/02 12:16:11.750] Read request
2005 Aug 26
1
Is LDAPget module stable enough for enterprise usage?
Hi, all. I am building a SER+asterisk PBX airming at around 10k persons' usage. For authentication purpose I am in favor of ldap storage, while I am not sure the current ldap module for asterisk(0.9.9.2) is stable enough? sorry I do not master the proper testing mechanisms to find out myself. Thanks in advance.
2004 Aug 24
2
SIP Provider in India/Pakistan/Bengladesh
Hello All, We are looking for a SIP provider teminating calls in India, Pakistan and Bengladesh. Any one knows a good one? Regards, Cesar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040824/4d5dd4b4/attachment.htm
2004 Jan 16
2
ISDN30 - HW ?
Hi, Are there any hardware for ISDN30 ? if yes any problem with this ? is i out-of-box like ISDN2 but with 30 linies ? Do I need more than the cable from my teleprowider and a PCI-card ? /HHA _________________________________________________________________ Find high-speed ‘net deals — comparison-shop your local providers here. https://broadband.msn.com
2007 Dec 16
0
LDAPget question, usage
Hi, I've recently come across LDAPget (version 2.0rc1) and I've been trying to get it functional in my test environment (Asterisk 1.4.15 and MS Active Directory 2003) but I can't seem to get it working. I put together a test extension to try to change the CALLERID(name) by way of a LDAP query to AD: extensions.conf exten => 100,1,Answer() exten =>
2005 Sep 13
1
SetCIDName question
Hi all, I tried to set the calleridname of an incoming call to get different incoming labels displayed for different incoming numbers. This does work for hidden number-calls so I can set the displayed CIDName on my cisco7960 from "CID withheld" to "abc CID withheld" If the incoming CID isn't hidden it works to use SetCallerID but not to change only the CIDName with
2005 Jul 07
3
isdn30 / pri lines in the UK
anybody recommend a supplier in the UK for a pri/isdn30 line (other than BT) thanx very much __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
2009 Jul 28
3
CIsco 7960 + asterisk: hepl needed
Dear All, I'm trying to configure my new phone Cisco 7960 to work with asterisk. I followed http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html and I got into the point where I can see on the the display line indication showing "55 <phone icon with x>" so it looks like the phone is not registered. The phone and the asterisk are in the same local
2006 Mar 15
3
Double-ring tone
I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works fine. Except that when I make an outbound call, I get a double-ring sound. I also found that if the target number is engaged, I get a ring sound and at the same time get a busy sound. If I revert back to 7-4, there is no problem. Anyone else had this, or any clues on how to fix it ? All of our other phones are still on
2007 Aug 17
2
No audio on ISDN PRI calls
Hello, I have a Sangoma A101 connected to an ISDN30 (E1 in the UK) with some Snom 300 and Idefisk softphones. I can do SIP and IAX2 calls just fine, however I cant get any audio in either direction on the Zap channels. When I call in or dial out over the ISDN30 (UK E1) I can see the call answered/placed on the CLI and then silence follows. I've been provisioned 25 out of the 31 channels only
2009 Nov 27
1
ISDN30 Timing Sources (Jon Morgan)
Quoth Jon Morgan <jon.morgan at motors.co.uk> > >We have a 2 port Digium TE220P card, one span is configured to connect to our ISDN30 provider (British Telecom), the other span connects to our internal PBX. Here's the zaptel.conf snip: > >span=1,1,0,ccs,hdb3,crc4 >bchan=1-15 >dchan=16 >bchan=17-31 > >span=2,0,0,ccs,hdb3,crc4 >bchan=32-46 >dchan=47
2008 Dec 16
4
RDNIS and asterisk
I have a couple of numbers that are diverted to a number that is conected to an isdn30 card, running asterisk 1.4. eg. 123456 => 22334455 654321 => 22334455 What I would like to know is the number of the orginal number dialled (123456 or 654321). I thought that RDNIS was the answer, but it is always coming up blank. When I did a debug on the pri span, I saw the following message
2003 Jul 18
5
Again Asterisk and VMWare - it works now!
Hi, I have succeed using Asterisk on VMWare on an Athlon@1GB with 128 MB allocated for the Linux virtual machine. I have connected this PBX with another one using IAX/GSM. I can call the other part and the sound is great, without any interruption. The phone used is a Cisco7960 with G.711, so still a codec conversion is in place (GSM/G.711) and Asterisk/VMWare Wkst performs very well. The problem
2004 Nov 24
3
Haven't got a clue ...
On how to even search for this "feature" as I have no idea on what it can be. I've got a meridian linked to * (by EuroISDN) which is linked to a ISDN30. I can make calls from the meridian, and receive calls into the meridian. Great stuff. However, if someone dials an invalid number, then instead of hearing a "three tone", the line just drops and goes dead. The console
2010 Mar 15
2
Problem with an older VB program
I'm fairly new to linux and completely new to Wine, so please bear with me. I am trying to run a game, Extreme Warfare, an old wrestling simulator, from a thumb drive. When I browse to it through "browse C: drive" and double click, nothing happens. No error messages or anything. Doing it through the command line, I get the message > kirk at kirk-laptop:~/.wine/drive_c/EWR$
2004 Sep 21
3
chan_sccp/SEP<mac>.cnf.xml
HI all: I have spent a large amount of time configuring/installing phones connected to Asterisk. Halfway through the process I discovered that my Cisco7960 with 2 7914 expansions was not supported in the SIP protocol. After reverting to SCCP 6.0(4.0) I am now perplexed with the hassle of configuring SCCP to properly work with Asterisk. So far I have gotten the phone to dial and receive calls
2003 Oct 30
6
Info on UK ISDN30e?
Hi :) My employer is looking to move a call centre to a new office, and has been increasingly frustrated with their legacy PBX (call-logging licensing and hardware upgrade costs). So I've stepped forth as the Open Source Pedant and suggested Asterisk so we can do all our own CallerID / call logging / analyses, and make use of IP Phones / teleworking, etc. The problem begins in that I only