similar to: [MSG]TDM Error on ASUS Pundit-R

Just to bug you all (feel free to rant at me), a client wants to set his caller*ID number for outbound calls though us to PSTN. the client is using SIP to us, he can set the caller*ID name fine. if he sets his caller*ID number to anything other than his account number (8440101), the call is dropped into the default context (and then hung up by our dial plan). To get around this i

Can someone help me, im using latest CVS, asterisk and cdr_mysql, when I make a bridge call (using .call files in outgoing/) I always get 'billsec=12' in the cdr, both mysql and Master file even if the call lasted longer, watching the Master file while making a call I see it updated at 12 seconds even while im still 'in' the DIAL app and the call continues on just fine. Iv looked

Hello all, Recently I bought a TDM02B digium card to conect to the PSTN. We pluged it on a Pentium IV 2,8 Ghz, Asus Motherboard, but when we try to start asterisk, the box hangs. Someone have the same card running with asterisk in a similar machine? Could you tell me your box hardware details? Thanks for your time, Ismael Gil.

I keep getting this error randomly. I have a TDM400P, I've tried disabling everything in the BIOS. I'm running Fedora core 2.6.11 and asterisk 1.0. Any pointers would be greatly appreciated. Thanks! -Paul

Hi, I'm specifically have a problem with idmap entries not being created in my LDAP backend for trusted domain logons - Local accounts appear to be fine. I have installed the Sernet enterprise packages from: http://ftp.sernet.de/pub/samba/experimental/rhel/5/i386/ I'm preparing the server as follows: 1. smbpasswd -w '<password>' 2. net rpc trustdom establish SANDBOX

We are seeing this weird problem, it seems to happen at random periods throughout the day from a few minuets to a up to an hour. [Phone A] >--SIP--> [Asterisk] >--SIP--> [Phone B] Both phones are snom 360's. Asterisk is Stable 1.0.9 Pretty simple config, just a dial direct to each other like Dial(SIP/phoneA,30,t) Running Gentoo linux When we make a call during one of the problem

Oh. So how can I do this? If I write something in PHP, how do I make it output to an Asterisk variabel? I need to set a variable in asterisk to TRUE or FALSE based on the result of the PHP-script. ________________________________ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Morgan Gilroy Sendt: 8. februar 2006 15:28 Til:

Hello all, recently i stumbled upon the I-TDM standard, e.g. see here http://www.picmg.org/v2internal/news2005.htm "SFP.1, also known as I-TDM (Internal TDM), is a companion protocol specification to SFP.0 that is optimized for TDM traffic over high-speed fabrics such as 1 and 10 Gigabit Ethernet (PICMG 3.1), Advanced Switching (PICMG 3.4), Infiniband (PICMG 3.2), etc. SFP.1 and SFP.0

Hi: I'm using a TDM-400 to terminate PSTN lines at my Asterisk server with kewlstart signalling. When an outside caller calls the server, the TDM-400 goes off-hook and provides a ringing tone to the caller. If the caller hangs up before the receiving party answers the phone, Asterisk fails to detect the hang-up. The TDM-400 stays off-hook, hogging the line, while Asterisk rings the

I have 2 TDM04b cards currently running in an asterisk at home box that I am ready to replace with the CVS version of asterisk. What I am looking for is thoughts / recommendations. I want to move this to a small form factor ( shuttle ) machine and was wandering what expeience / advice there was for this? I have seen the incompatible motherboard list at digium ( and in fact I think my current

For those using the new tdm fxo modules, please review and add your comments to: Bug Description 1898 tdm fxo does not sense broken pstn line 1899 tdm fxo senses pstn line disturbances as ringing 1902 tdm fxo echo when calling from sip phone There is at least a suspicion on my part that bug 1902 might be related to specific "systems" that have not yet been diagnosed or

I sent a query about this before, but have some further information and am hoping somebody has a suggestion as to what to try next to debug this. I'm using an Asterisk box primarily for MeetMe conferencing. There are two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works fine between TDM channels. But when a SIP phone calls the conference, there's no voice path *to*

Massey, Ricky would like to recall the message, "EXTERNAL: [CentOS] A bridge problem".

Im having a problem getting the 2 cards working together. I added the TDM with 3 FXO and 1 FXS board. The TE405p is working fine. THe TDM registers and identifies the FXO's and FXS but cant get them to go off hook. I hope someone can help. Asterisk seems to THINK its dialing it sends manager status as dialing, and it just sits there with no tones, on the phone dialing out. If I dial,

Hi, I'm experiencing an issue with my x86_64 machine containing a Hauppauge PVR-500 (ivtv) and a Digium TDM400p (wctdm, part of zaptel) PCI cards. Independently of each other both cards work fine, but once the wctdm driver is loaded and mythtv tries to record something on the PVR-500 the wctdm driver freaks out. The error message is see is "TDM PCI Master abort" printed over and

A bit about my experience with the TDM-04 FXO. Only saw a few post on this subject, thought I would contribute a little about my experience to save others the hassle. a. As an earlier poster noted, the driver for the FXO is in the wcfxs module. Perhaps it should be renamed to something less confusing. b. You need the zaptel,zapata libraries from the cvs, the ones with Asterisk 0.7.2 won't

Hi, sorry for my insistence but this is a big problem for me..my steps for remove card are ok ? Thanks. ------ Salvatore. ----- Original Message ----- From: "Sasa" <sasa at shoponweb.it> To: <asterisk-users at lists.digium.com> Sent: Monday, November 26, 2007 4:25 PM Subject: [asterisk-users] Remove a TDM Card > Hi, I would like remove a Digium TDM2400P from

I haven't found anything about this in the FAQ's... Despite having a translation from an all uppercase (or all lowercase) to the mixed case Unix username in the username map, the samba server (1.9.18p1) does not allow browsing from a Win95 client. If I change the username to all lowercase on the Unix server, then browsing works fine. Have I missed something? -- Phil Etheridge

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Eric "ManxPower" Wieling > Sent: Thursday, December 14, 2006 4:30 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Hardware TDM Switching > > asterisk@nicox.org wrote: >

Dear fellows, how are you? I?m offering TDM routes for Brazil (landline and mobile destinations) with low prices, TDM ccts (no GSM), ASR and ACD great. Pre paid, by paypal. If you have interest, please just let me know. With Best Regards Josue -------------- next part -------------- An HTML attachment was scrubbed... URL: