similar to: Big problems with TDM2400 :(

Hello All I have a TDM2400 card with 4 FXO, and the the following problem: This card answered all the calls, but for the caller, the call is ringing and I don't hear nothing when it has picked up. Here is piece of my log: Thanks for any help. == Starting post polarity CID detection on channel 21 -- Starting simple switch on 'Zap/21-1' -- Executing [s at entrada:1]

Hi all... I have a problem with caller id on my asterisk server. here is my configuration : centos-5, asterisk 1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2 ibm X-3200 series, xorcom astribank (16fxo, 8fxs), 16 line telco (hunting) everything fine until I try to feed my app with caller id. My extensions.conf : [incoming1] exten =>

Hi, I'm trying to connect an HDL F10 device for a friend living in Brazil to the TDM2400 on his Asterisk server. That device should behave like a normal doorbell and it is if connected to an analog PBX. I connected to the TDM2400 and everything works fine except for one thing: when the called party hangs up his phone, the F10 HDL device does not hang up. I'm not brazilian and not

Hi List... I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's, and it also has the echo canceller... I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel 2.6.9-34.0.2.EL I'm using Polycom's 501 with the SIP 1.6.2.0041 The problem is when someone dials to or from the PSTN through the TDM2400, the voice quality is crappy...Instead of hearing:

Hello list, I am encountering a bit of a problem in working with incoming calls with a TDM2400 and * 1.2.4; when a call comes in, * will correctly detect the ringing, but will sometimes report multiple "Attempting native bridge". What I do is basically that when a call comes in, I dial a different box through the same Zaptel interface and I get logs like this: [call comes in]

Hi, >From a chat with Kevin Fleming during last Astricon Europe in Paris, I understood that Digium was about to support TDM bridging with TDM2400 board. This feature had to be introduced with 1.4 release. Is my understanding correct ? If correct, would it be reliable enough to bridge 14400b/s fax calls though I know, fax were not officially supported or shall I use Sangoma boards for that ?

Hi Adam, I have the same problem. Are you sure is an echo canceller problem? Following advices from this list I discovered that I had an IRQ shared. Untill now I didn't try the new setup but I really hope that this was the problem. If you manage to solve the problem in any way please give me advise. Ciao You can make sure the card is sitting on

Hello, I've installed two Digium TDM2400 cards on my server. One has 24FXS and the other has 16 FXS and 4 FXO. They are both connected to power. Unfortunately some of the FXS module fail to initialize and I find following messages in the logs (the rest of the FXS modules work well). Could someone give me some advice? !!!!!!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 Feb 9 18:09:45

i have a full fxo TDM24 and i have problem with installing it. when i run modprobe wctdm24xxp dmesg shows the following messages. and it waits for ever and nothing will happen. i'm sure that: - the power is plugged into tdm24 board - udev is configured and is working with other tdm cards. - zaptel.conf is configured. Zapata Telephony Interface Registered on major 196 Zaptel Version:

does TDM2400 work on 3.3v pci slot? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070211/ac7d0c79/attachment.htm

Hello everyone, I have noticed thar our dahdi-channels.conf has some repeating directives, for instance for channel 2 (FXO) we have these settings: ;;; line="2 WCTDM/0/1 FXSKS" signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel => 2 callerid= group= context=default As you can see, a few directives are repeated (callerid, group, context). This was generated by

Does anyone know if a TDM2400 will fit into a Dell 2850? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com -------------- next part -------------- An HTML attachment was scrubbed...

Good Day List, I'm having some issues with echo cancel on my Asterisk box, and have done extensive reading and have gained some useful pointers from this list but have a couple of hopefully fairly simple questions. The Asterisk box is connected via 20 FXO ports on a TDM2400 with the Hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a

We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. So far, this has happened on both times the

I have one TDM2404E digium card on asterisk box, after configuring the zaptel and zapata configuration files, I am getting these errors when reloading asterisk: ast_unregister_indication_country: Removed default indication country 'us' setup_zap: Ignoring signalling setup_zap: Ignoring answeronpolarityswitch unable to recognize channel 13-5 what is the reason for that? Thanks,

Good morning, I have a digium wctdm24xxp in my asterisk box, i am not able to see the callerid when the call is incoming from the fxo line, i live in Brazil, how can i change the signaling from fsk to dtmf? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Let us know more information about your setup. Hardware/software details details such as. server configuration PSTN cards you are using?? ( E1 or FXO card) sip.conf, zapata.cons, zaptel.conf config details?? Thanks & Regards, Vidura Senadeera, Sri Lanka. Tel - +94114520001 Mobile - +94777766596 yahoo/skype Ids - vidurased ================== Message: 5 Date: Mon, 10 Dec 2007

Hello, Since last few days I have noticed some people complaining that their call gets disconnected while they are in the middle of the conversations. Looking in the log I found these error messages, Dec 10 11:18:56 DEBUG[8833] channel.c: Bridge stops bridging channels SIP/5060-b7a03560 and SIP/219.206.2.291-089d8768 Dec 10 11:26:41 DEBUG[10410] channel.c: Didn't get a frame from channel:

Hello, I've come across the following situation in R-1.8.1 (compile + running under RedHat 7.1): > phyper(24, 514, 5961-514, 53, lower.tail=T) [1] 1 > phyper(24, 514, 5961-514, 53, lower.tail=F) [1] -1.037310e-11 I'd expect the later to be 0 or some very small positive number. Is this a number under-flow of the calculation? Do you think I'm safe if I just set the result to 0

Hi I'm building new samba server (on Debian 6.0, software RAID10 2TB, Xeon CPU). Generally everything is working fine, so I have decided to run some stress tests. My choice was dbench. Old server is Debian 4.0 (samba 3.0.24, Athlon 3000+, one ATA 160GB disk). So run dbench 16 on both old and new server The results are strange old serwer: about 300MB/sek (dbench 3.0) and below are first