search for: srsergio

I want use iLBC and have following in mind, please help me is it possible ? ISDN <-----(ALAW)-----> * <-----(ALAW)-----> SNOM SIP??<-----(iLBC)----->?*?<-----(ALAW)----->?SNOM 1. ISDN incoming codec is ALAW SNOM codec should be ALAW (can't be iLBC because a lack of codec). 2. SIP incoming codec should be iLBC (snom is ALAW). 3. SIP outgoing codec should be iLBC /snom

Hi: Did anyone use apt-get install zaptel successfully? After apt-get instal zaptel, use "modprobe zaptel", get a "FATAL modul zaptel not found". Thanks. Yan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040722/4608e1e5/attachment.htm

Hi, Which is the correct syntax to call using IAX? I have two Asterisk boxes behind a NAT and one of them use the default port 5036 for IAX, the second one use 5038. To call an extension of the first one, the line in extensions.conf is: exten => _9XXX,1,Dial(IAX/user:pass@195.3.32.191/${EXTEN:1}) and for the second one: exten => _8XXX,1,Dial(IAX/user:pass@195.3.32.191:5038/${EXTEN:1})

Can someone tell me how to do this...Given the following line: exten => *97,3,VoicemailMain(${CALLERIDNUM}@default) Is it possible to add some logic to manipulate the CALLERIDNUM to send back 801 even if the extension is 601 and 901 even if the extension is 701? I have 2 branch offices where users have both Office and Home SIP phones. I want them to share a VM box. Branch1 = 8XX , Home =

Hello- What is the .gsm format? Ie: what's the encoding method and sample rate please? Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott@evtmedia.com URL: www.evtmedia.com

Hi, I would like to know how suppress number for outside dialling in CDR table. For example, if I need press 9 key to make an outside call, I would like that the number in dst field in cdr table was the outside number without 9 key. It's possible? Thanks in advance, srsergio

Hello, Is there a asterisk developer guide/source code doc or something like that? I want to see if I can implement the admin menu function for meetme. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030922/3ff8a388/attachment.htm

Anyone have any suggestions on free sip phone software for windows?? Only have one IP phone and want to have one other computer hooked up to my Asterisk box for testing.

...1846141,971846142,97 1846143,971846146,971846015,971846148 incomingmsn=* controller=1,2,3,4 context=default softdtmf=0 echosquelch=1 echocancel=0 echotail=64 callgroup=1 devices=8 and the script I use to load diva Cars is diva_cfg.rc generated by Diva Server for Linux tool. Any idea? Regards, srsergio

Perhaps stupid question but, is Asterisk a statefull or stateless proxy? Ignace

I cannot get Festival to work with asterisk. I have the following: exten => 555,1,Answer exten => 555,2,Festival(mary has a little lamb) exten => 555,3,Hangup I get the following from asterisk: "Festival returned ER" and the festival logs shows the following: client(1) Fri Jul 16 15:35:54 2004 : disconnected client(2) Fri Jul 16 15:40:26 2004 : accepted from localhost

Hi, I have to link an Asterisk Box with a PBX Matra 6501. System look like this : E1------Te110P Asterisk Te110P-----E1----Matra 6501-----Phones | | Ip Phones Incoming call from E1 will enter on asterisk, if incoming number is _800n then go to IP phones. In this case no problem. But if it's an another call i want to return call to my

I'm sorry for the several messages with holidays message. Regards, srsergio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050714/28ba6f00/attachment.htm

Hi. Today I was asked about a function of asterisk. That's what it should to: a call arrives -> put it in a queue -> remain here ;) Then, when someone wants to answer, just dial an extension and the older call that's in the queue is picked up. A sort of app_queueonly + app_pickolderqueuecall . As far as I know asterisk doesn't support that, so was wondering if someone

Hi all, again I try configure Hylafax with asterisk. I would like configure Asterisk in the next way: 1)An incoming fax go into through X100P 2)Asterisk detects Fax and redirect fax to Hylafax Is it possible? Any idea woluld be great idea? regards, srsergio -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.5.4 - Release Date: 15/12/2004 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041217/b4fb...

Hi All, Has anyone had success with using the AVM Fritz PCI ISDN Card (v2) CAPI drivers with a distro other than suse? The AVM driver is geared towards suse, and after trying them with RedHat 9 I've not got very far. The module compiles but dies when I attempt to modprobe it. I've googled until I almost lost the will to live! What little information I can find seems to refer back to

Hi, all I am trying to install E100P card, the 'modprobe zaptel' is ok, but when I did 'modprobe wct1xxp', I got such error, so can not load the driver for the card. /lib/modules/2.4.20-8/misc/wct1xxp.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog

Hi list, I have a "Wildcard TDM400P REV I Board 1" with 4 FXO modules and * 1.0.9 up-and-running. Only 2 FXO ports are used for 2 analog phones and are doing fine. I now wanted to use the 3rd and 4th port, but when I insert an analog phone, take it off hook, I do not get a dial tone. With my 1st and 2nd port, I get messages like: -- Starting simple switch on 'Zap/13-1'

...Bus 2, device 5, function 0: Network controller: Tiger Jet Network Inc. Model 300 128k (rev 0). IRQ 12. Master Capable. Latency=32. Min Gnt=1.Max Lat=128. I/O at 0xa400 [0xa4ff]. Non-prefetchable 32 bit memory at 0xe3000000 [0xe3000fff]. Any idea? regards, srsergio -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041229/0df...

Hi, I am using Asterisk on a DSS1 ISDN-BRI with ISDN4Linux (and a Fritz Card PnP). The ISDN-BRI is in PTP-Mode (Point to Point "german: Anlagenanschluss") which is enabled within I4L with "hisaxctrl fcpcipnp0 7 1". Everything works fine except that I can not see the called number/MSN of incoming calls within Asterisk and because of this I can not route incoming calls