similar to: E1 PRI to and from SIP screeching

I have an asterisk system where the asterisk daemon dies typically at least once per day. It is running in the wrapper safe_asterisk, which automatically starts the daemon back up. But we find this unacceptable because when the daemon dies, we usually have active calls drop, and sometimes we have to run asterisk -r -x "module reload" after the daemon starts back up before everything is

I have been looking at the rhino r1t1, and digium single t1, and the sangoma, but from what i read they all sound like good products. Anyone have anything bad about any one of them? I am leaning towards the sangoma as it seems to have a better following here on the lists. Any advice is appreciated.Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello! I have been struggling with Asterisk 1.6 and DAHDI for the past few weeks. We are an outgoing call center with 30 internal analog phones hooked up to 2 Rhino CB24 channel banks. The banks are connected to a Rhino R4T1 card in a Dell 2950 server with 8 gigs of RAM. The 2 other ports on the R4T1 go to our 2 PRIs. In this configuration, we have trouble maintaining stability. It may be fine

We are a voip terminating company, we're using Channelbank with FXS modules, Rhino, CAC, etc.. What we're wondering is, is how to would you echo cancel a channelbank. Of course we're realizing that cancel'ing on the T1 (on Ast) does no good (we think?) because the analog conversion is at the channelbank. Suggestions? Lowering the gain helps but we're looking for a real solution

Hey guys! I'm the past week and a half, I have really learned a lot from the mailing list and the wiki's posted online. Now I have a question regarding different ways I can setup my asterisk server for a small business with 12 extensions in the office. Cost is a great concern, so I know cheap analog phones at the desks is what we are looking at. My question is, should I go do a fractional

Hi, A most peculiar and vexing problem for you all. I hope I have been verbose enough without being a firehose ;) The set up: I have a channel bank, using the r1t1 rhino driver with a rhino T1 card (the channel bank itself is a very legacy piece of equipment)- this supplies FXS for all the house phones. Also, a Wildcard TDM400P, using the wctdm module with 1 FXO module, this supplies FXO to the

Hi I've had an asterisk setup for the past 15 months, based on the debian asterisk packaging. Until late August of this year, I had no problems once initial setup was complete- the system worked essentially flawlessly. Since August I have been having exceedingly infuriating intermittent problems that are causing me occasional periods of nasty trouble: 1. No Dial Tone. Every Sunday night at

Hello, My first attempt to get dahdi running on 1.4.28... with a Rhino 8 port modular card and a single FXS module. Got the Rhino card installed and the machine sees it: root at pbx:/etc/dahdi# dmesg | grep rcbfx [ 71.985309] rcbfx 0000:04:00.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21 [ 71.985440] rcbfx 1: Rhino PCI BAR0 50100000 IOMem mapped at ffffc90008d7c000 [ 71.985504]

Hi, Is there any native software for OS X to convert an audio file to speex? Even a command line would be great. I looked for hours and couldn't find a program. I tried VLC but it just outputs screeching noises. Sincerely, Patrick Krol patrick9@net-link.net

Hello all, First, the vital stats: FreeBSD 6.1-STABLE, dovecot 1.0.rc7 from ports. I have run into a vexing little problem that has brought my deployment of a shiny new dovecot server to a screeching halt. Using 'periodic' I've configured regular backups to occur using 'dump'. I have a monthly level 0, a weekly level 4, and a daily level 8. These dumps take advantage of

Is there any possibility of DAHDI supporting Automatic gain control on TDM ports? I'm having issues at a couple of offices where calls made to local numbers are fine but a when a calls from or goes to a large percentage of long-distance or 1-800 numbers the person at the remote end cannot hear the person in my office. Boosting the gains in zapata.conf (I'm still using 1.4.21) to 8

Hello All, I'm looking to get some feedback on which solution of providing FXS is going to have the best results with faxing. I'm only looking to see what method is going to provide the best digitization into Asterisk, not for transmission from Asterisk to else where. Any recommendations of specific channel banks are welcome. I will need to provide approximatly 216 FXS Ports and need

Hi All, I know that this may not strictly be an asterisk issue, but if anyone has any ideas on this prob it would be appreciated. I have a working asterisk setup using Suse 9.1, a BT ISDN BRI card (Fritz?), sipura 3000 and a few cisco 7940's. I wanted to do add faxing into the equation, so I thought I would try and setup the following: PSTN->asterisk->hylafax->pc At the moment

Hi Folks, Struggling with a new * installation with 2 Rhino R2T1 cards. For some reason, the system is locking up tight when you run ztcfg to configure the card(s). Configuration is asterisk 1.2.14, zaptel 1.2.12, and rhino's 1.05rxt1 drivers. The cards seem to load fine with a "modprobe rxt1", but once you run "ztcfg -vvv", the system will lock up within a few seconds, no

I received an additional module for my TDM400P yesterday. I installed it last night. Before I installed it, I had a working 1x1 PBX. After installing the physical hardware, I spent a couple of hours unsuccessfully attempting to get dial-tone on the new FXS part. I gave up, made sure the old FXS still rang and went to bed. I should have tested further. This afternoon, I got a call at work

hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. it will be a slow migration, the asterisk server will be inserted between the telco E1 and the hicom. new phones will be sip ones. the customer has several fax machines and analog phones (some of them have to be explosion-proof). around 50 analog ports in total are needed. as we are in

I have set up a couple of test Asterisk servers and have never had a problem with sound. I've just done a fresh install on a dual 1GHZ PIII Asus box running Fedora Core3 with the Digium PCI Dev kit and following all the various Core 3 How-To's. I can make calls ok but when any sound is sent from the Asterisk box such as voice prompts and music on hold the sound is completely chopped up in

Hi all, I've setup a PBX in production environment last week but we have immediatly find out drop conversations and lot of errors (in asterisk logs) like: Apr 10 17:30:48 NOTICE[25154] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 My configuration is: Motherboard: SuperMicro P8SCT (Intel E7221 chipset) Hypertrading and APIC enabled CPU: P4 3GHz Digium TE205P

For 2.6 to really be a useful, we need this bug fixed: http://llvm.org/bugs/show_bug.cgi?id=4963 Would it be possible to get this merged into 2.6? -Tomas

I'm running the samba-client-3.0.20-0.1 SUSE RPM. I was using the version that came with 9.3 but upgraded to see if this specific problem would go away. Guest access does not appear to be working correctly, and it looks like the problem is due to guest not getting mapped into the LDAP query correctly. Specifically, I can login with local account, join workstation to the domain, browse