similar to: PRI crashing Asterisk

Displaying 20 results from an estimated 1000 matches similar to: "PRI crashing Asterisk"

2008 Feb 20
2
Skype Users
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 found this today, I am not a skype user but have read on chan_skype and don't like aspects of how it is implemented. My thoughts on it are only theoretical as I haven't used it I just cringe at adding X to a server. Anyhow there is a new project called sippyskype that appears to do a similar sort of thing with a couple differences. 1. Its
2008 Mar 06
14
FXS channel banks
Greetings list, I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. If anyone's had experience using channel
2008 Feb 15
7
Digium stopped TDM400P production: alternatives??
Hi, Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a "fully-compatible" Openvox product...but is it really "fully-compatible"? Any experience about Openvox products (card and zaptel versions, etc...)? Thank you! Giorgio.
2008 Feb 08
3
Question about Asterisk versions (newbie)
Hello, I would like to consulate with you guys. I'm setting up an Asterisk server on Debian. The problem is that Rhino drivers are only compatible with Zaptel 1.2. By default debian stable offers asterisk 1.2 and zaptel 1.2, and that suits our needs. Is there a bleeding need to use latest version of asterisk? I have managed to install Asterisk 1.4 and Zaptel 1.2 but then i got the
2008 Feb 13
3
Analog DID
Does anyone have any suggestions for connecting analog DID trunks? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4
2008 Feb 18
1
Zaptel 1.4.8 breaks tor2 support on CentOS 5.1? (kernel panic)
Hi all... I did some Google searches and didn't find any info on this so I'm posting it here... if this was recently discussed, I apologize for the duplication -- please point me to the appropriate thread. System Description: Supermicro SuperServer 5015M-MF w/ PDSMi Motherboard Intel Pentium 4 2.8 Ghz CPU 2 GB DDR2 Memory Digium T400P 4 Port T1 Card CentOS 5.1 (Final) Kernel:
2008 Feb 27
3
Simultaneous Inbound and Outbound calls on analog lines...
Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this
2005 Jan 09
5
Help in E1-T1 encoding
I have an asterisk with a TE110P configured as T1 which is behind a PSTN gateway. This gateway has an E1 to PSTN and a T1 to asterisk. This T1 is configured as Network and * as CPE. Every call I receive in E1 gateway is directly switched to asterisk using T1. Remember E1 is alaw. Both E1 and T1 have Natural Microsystems boards with a very simple software. When I call to E1 asterisk signalling
2009 May 22
2
BT ISDN-30 Pri getting 'stuck' on outgoing calls.
I've having problems with a BT 2 span ISDN-30/Digium TE205P asterisk setup with outgoing calls not completing and requiring an Asterisk reset to 'unstick' span 1. Sorry this is a bit long but I'm completely out of my depth :-( This system has been in use for some while and I recently upgraded it to asterisk 1.4.24, zaptel 1.4.11 and libpri 1.4.9. I didn't change
2006 Mar 11
1
FW: I need to set NO CRC4 on zaptel.conf?
Hi all, Can somebody help me out, to get the call going through to my provider? I connected my A104D Sangoma card to E1/isdn and each I tried to make call I get the errors below. My protocol analyzer can see only setup info and release complete. WIRELESS2*CLI> set debug 9 Core debug was 0 and is now 9 WIRELESS2*CLI> pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 T203
2004 Jan 12
1
E100P - connected to Cisco
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="Content-Type" content="text/html;charset=ISO-8859-1"> <title></title> </head> <body text="#000000" bgcolor="#ffffff"> Hi,<br> <br> have any one sucessfully connected E100P to Cisco? I am
2005 May 19
2
MusicOnHold Loudness/Distortion
For whatever reason, the music on hold is extremely distorted and loud. It didn't used to be this way and I haven't changed anything, yet it persists. This is on all the channels we use (SIP, IAX2, Zap, ALSA). Can anyone help with this, or has anyone seen this? The mp3s play fine on any computer and haven't changed since they did work. Those wishing to hear for themselves, feel
2004 Mar 18
4
zaphfc problem
Hi, I have a partial working installation with zaphfc. Incoming call : For incoming call, seems work fine. But the sound is very bad with bounce short crashing sound. Same sound with echo cancel off or on. SDA work fine. Another problem, it's seems that's zaphfc don't reset correctly the line. I have one of my D channel how was busy even after stop communication. Outgoing call :
2004 Oct 07
1
T100P Pri Audio
I've been working on an asterisk box at work for a few weeks now, things were finally starting to sail smoothly until I hit this head scratcher this morning. It's a rather intricate problem, so bear with me. Heres the scenario. What works: If I call from my sip phone -> sip phone everythings ok If I call from sip phone -> external pots number ok as well If I map one of our
2007 Feb 28
2
No Caller ID Name PRI NI2
I there, I have some trouble to do working caller id name for outgoing calls on the PRI we just installed. Caller id name work on incoming calls only. Caller id number work on incoming and outgoing calls. The provider, Goup Telecom, said that's in what i'm sending. They said that I send the cid name in ascii code and to do it working, I need to send it in hex. So I take some traces
2007 Oct 31
4
PRI over T1 calls dropping, cause 100
I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a Meridian Option 61C. Calls either way drop with error "Channel 0/23, span 1 got hangup, cause 100". Can anyone offer insight into the cause and solution/workaround? (I tried upgrading to Ast 1.4.13, and upgrading matching zaptel & libpri, put the problem is identical). For testing, I tried a call from the
2005 May 19
1
New IAXy from Digium
I was just browsing Digium's web site and noticed they are taking orders for the new IAXy. Has anyone purchased and tested one of these yet?? I have thought about buying one for testing, but want to make sure it isn't going to be a flop like the last one. Robert
2005 Jun 14
3
How to setup a test number to know my extension number
I would like to setup a test number, that speaks back my phone number. How can I set this up? bye Ronald
2005 Jun 15
1
Changing caller ID on a Zap channel
I have asterisk with two zap channels which are analog ports off a T1. They each have a inward DID number If they are used for outgoing they show the T1 main number not the DID's number. Is there any way to send caller ID of the inward DID number not the main number Jeff
2006 Feb 16
1
Problem making outbound calls on TE210P using NFAS
Hello, I'm running Asterisk@home 2.5 asterisk 1.2.4 zapatel 1.2.2 libpri 1.2.2 on a Dell Poweredge 2850 (1 CPU) with a TE210P I have 2 t1 circuits using NFAS with dchan on 24 and no backup dchan. I am able to receive inbound calls on all channels and can only make outbound calls on channels 25-48. Attempting to make an outbound call on channels 1-23 results in congestion.