similar to: spa-2100 3.2.1 firmware

For redundancy I would like to write the CDRs on tow mysql servers. cdr_mysql.conf accept only one configuration [global], how to add a second host? Thanks Rosario -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050609/9eacbae2/attachment.htm

Sorry about stupid question but I would liek to get help about Zap channel. We would like to get early media on session in progress from zap channel. But using the standard exten => _X.,1,Dial(Zap/g1/${EXTEN}|60|o) I don't hear any audio until someone pickup the phone. Now I can't now if there is a message from a mobile phone comany on session in progress. please help. regards

Hello all ! Can anyone recommend me ATA device that REALLY has T.38 built in. So far I have heard of Telco Systems Access201, which seems to be impossible to bye in Europe (all resselers are droped Telco systems ATAs for some reason (tried in Germany and in UK so far)), and I have heard that SIPURA SPA-2100 should have T.38 built in into newer firmware, but I wasn't able to confirm that

I am experiencing a very wired problem. Some of my cdr are lost. I use logging cdr to csv, mysql and odbc. But some of them are lost. They miss in csv mysql and odbc, so i'm pretty sure it is related to asterisk functioning. I am running asterisk 1.0.7; this is simple configuration file: extensions.conf [general] static=yes writeprotect=no [macro-gw-voipjet] exten =>

We have some speed/performance issues: We have a 100M fullduplex private network setup to handle rsync transfers to our "mirror" server with a command like: time rsync -e ssh -avzl --delete --rsync-path=/usr/local/bin/rsync \ --exclude ".netscape/cache/" --delete-excluded \ bigserver:/staff1 /mirror/bigserver It takes about 20 minutes to check/transfer files from

I have Asterisk configured to be the gateway for sip users. CDR are stored using the mysql module. But in the cdr's source filed is present only the user and not the domain. I'd like to get displayed all the infos in this way: user@doamin.com and not only user. In what way I can add the domain? Thanks Rosario -------------- next part -------------- An HTML attachment was scrubbed...

We need to configure asterisk to authenticate two sip ATAs, but the stream must go directly from one to another ata without tuching asterisk. Is this possible adding canreinvite=yes into sip.conf? is it true laso if asterisk doesn't recognize the spd (t38)? thanks Rosario -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello together, well since I'm close to the despair with my problem I wanna describe it here and maybe anyone has a good idea. As the subject already told you I'm trying to connect from a MS-DOS Client to my Samba Server. I'm running a smbd 2.0.5a on a AMD K6-III machine which has two IBM 16.8 GB IDE drives and is connected to over a 3com 3c905b with 100Mb to a fullduplex hub. So far I

I just bough a Sipura SPA-2100 to use with Asterisk. When I use the analog handset plugged into the SPA-2100, the person on the other end can hardly hear me. I check the SPA-2100 setup and their is no mic/spk gain control. Is this a problem with the SPA-2100 or with Asterisk? Any way for asterisk to compensate for the poor audio level (if the problem is the SPA-2100)? Thanks, Mike

I have been looking into how to setup distinctive ringing on a SPA-2100. So far the only thing i have been able to find is how to define a distinctive ring in the spa config. What i cannot figure out is what SIP message i need to be sending to it in order for it use the ring. I did find out how to add the sip message for distinctive ring i just dont know what variable needs to be passed in order

We've got an SPA-2100 connected to * and then into a paging system on one of the FXS ports. We are having an issue where the paging system doesn't hang up the line, so it stays offhook forever and obviously makes in unusable. The paging company says that the SPA needs to hangup the line once the calling user hangs up the phone. Any idea how to make it do this? It doesn't do it

Hi, I'm looking for a full list of xml provisioning variables of the SPA-2100/3000. Currently the Sipura website has example XMLs only for the SPA-841 [1] and SPA-941 [2]. I'm mostly interested in the CallerID type selector variables and whatever variables control the PSTN<->VoIP settings. Sipura Configuration website form field names are numeral only. :( [1]

I've a problem. I've some spa3000 and spa2100. Asterisk 1.2.4. Prefered codec g711u in both. Calleng from a fxs of spa2100 to the fxo of spa3000, all works ok. Then I call from a sip phone configured for using g729, to the fxo of spa3000, it also works ok. The problem is that after this, when, making again a new call from spa2100 to spa3000, spa2100 receives only white noise. I suspect a

Jon, I need to create a database of basics blocks and their execution time. The only thing I'm concerned is if a block A is more expensive than a block B. Do you think that even with the overhead I would be able to get the A > B information? Like: overhead + time(A) > overhead + time(B) => A > B. If so, I'm not too much concerned about the accuracy. Not sure if I was clear,

Dear Group, I'm running the following setup; Yoper v2, Kernel 2.6.8.1-7, Wine 20040914 on a Toshiba Satellite Pro 4600. The Toshiba has a Yamaha AC-XG. In addition I have a USB Plantronics DSP100. After some tweaking I got wine to install x-ten lite; I have pasted my config file; [WinMM] ; Wine supports the following sound drivers: ; winearts.drv ; for KDE ; winealsa.drv ; for

Hi Friends, I have successfully implemented Intercom, Voicemail and International dialing using Asterisk. Now I want to connect my PSTN Lines to Asterisk server. I have 3 PSTN number (lines) to connect to Asterisk. For this, I want to use Sipura SPA-2100. Is my decession is correct or not? Is there any disadvantages with this Sipura SPA-2100? Please tell me. Thank you. Regards, Chandramouli

Hi all, I'm writing to you becuase I have compiled PAM in SCO (now Caldera) Openserver 5.0.x, and when I tried to use SSH with PAM enabled, y realized that OpenSSH depends on the user to exist en the /etc/passwd, and /etc/shadow databases, or equivalent ones (it uses getpw...() functions to determine validity of the user). In Linux, the simlpe solution is to use nsswitch, but it seems to hard

Dear Stefan, In the paper "Improved Forward-Adaptive Prediction for MPEG-4 Audio Lossless Coding", a non-linear compander is applied to the parcor coefficients prior to quantization. This compander is designed in order to minimize quantization error, especially for magnitudes close to unity. If you determine the typical distribution of magnitudes of the LPC coefficients, you could

Dear list, I am looking for different window functions used in perceptual audio coding. In the mp3 format, these windows overlap with each other, but I haven't found any information about their mathematical expression, nor their numerical values (see attached document). Which temporal windows are used in ogg audio coding? Thank you very much in advance. Kind regards, ------ Fernando A.

Dear list, I wonder if the flac encoded file has a great amount of bits due to the residue. I mean, what percentage of the flac file has information of the residue? Is this sequence represented by more than 80% of the flac file? On the other hand, which is the value typically adopted for the parameter "n" in the Rice coder? I know these results depend on the input wav file, but I