similar to: Reconfiguring a Zap Channel on the fly

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

We are using Bellsouth 8867 phones on our TDM400B FXS lines (asterisk-1.0.3). It has a "Voicemail" light, which appears to be MWI (according to the manual it works with voicemail from the telco that sends a FSK signal). The dialtone stutters when a line has voicemail, so I know that I have the mailbox setting right in zapata.conf, but the light doesn't go on. I am also getting

Hi all, I've got one of those cool free incoming IPKall phone numbers from www.ipkall.com. These numbers just connect to the SIP proxy of your choice, they default to Frreworld Dialup. You can use them with your own sip proxy on asterisk. My config for this is below. The trouble I'm having is the incoming calls do not seem to hit the section in sip.conf for the call. With sip

Hello, I have following problem with Sangoma A104 card: CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 NOTICE[16509]: chan_zap.c:7494 pri_dchannel: PRI

There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;

Jay, I have had a lot of trouble with the FXO ports on Adtran TA750. Unless the incoming POTS lines have a balance impedance, they will buzz very bad. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jay Milk Sent: Sunday, April

Sirs, I can't compile the source spandsp-0.0.2pre10; when i try to do the "make" sentence the following errors appear: # make Making all in src make[1]: Entering directory `/export/usr/src/spandsp-0.0.2/src' make all-am make[2]: Entering directory `/export/usr/src/spandsp-0.0.2/src' if /bin/sh ../libtool --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -MT

Greetings, I have just purchased several Astra 390 phones ready for asterisk. I have placed a line with adsi=yes in the Zapata.conf file just before channel => 13 I have also added an extension exten => 6199,1,ADSIProg(asterisk.adsi) exten => 6199,2,Hangup in the extensions.conf file. When I try to program the phone I get the following: Asterisk CVS-03/28/04-12:02:10,

I just need to buy 5-6 ADSI Phones. I wondering which of these models to choose - aastra 390 - I don't know is this an ADSI phone at all. Is there versions with and without ADSI - aastra 350 - I'm sure that it have ADSI. If there is some other good working model, it will be great, if someone points me. Thanks

Hi Zide, I think I found the right way to reach my goal. I used the following command: clang -O3 -Xclang -disable-llvm-optzns main.c -S -emit-llvm -o main.ll to generate an IR file enriched by all the metadata that otherwise wouldn't be generated with -O0. Moreover, -disable-llvm-optzns flag ensures that none of the optimization passes has been applied yet to the IR. In this way, I can

Ok, so I figured out my problem with my pt480s. But, now I have a few more. 1. When I dial into the voicemailmain or voicemailmain2 application, the phone and * start talking adsi, but then the phone tells me "programming download canceled, services is full.", but my services list isn't full, only "Asterisk PBX" occupies slot 2, slots 1, 3 and 4 are available. Any ideas?

Hi, Would anyone know of a way to set the time automatically on an ADSI capable phone from *? The phone in question is a Aastra 480e. While I am at it, does anyone have any helpful docs on the ADSI script programming? I have managed to do basic functions by modifying the asterisk.adsi file using stuff gleaned from the app_adsiprog.c file, but docs would be really helpful at this point. Tia,

Hey Everybody, You may remember be as the one that was generally running in circles trying to get ADSI phones to work? Well, five phones later, I've got a phone and a code that works to program it. I have been discouraged from just posting it to an e-mail forum, but I have been encouraged to get it embedded in the program. Is this possible? How should it be done? Next, after getting it

I'm trying to work out whether I want an ADSI phone or not, and in fact, whether it is ... useful/works with Asterisk. I have decided that for the cost/performance of the various IP based phones, I am not interested, I seem to get significantly better quality from a plain analogue phone using the TDM40B card. Is it beneficial to get an ADSI phone as opposed to a plain old analogue phone from

I was wondering if anyone has had any experience with PowerTouch 350 & ADSI. I am able to configure the phone with ADSI.. Which is pretty cool in it self. But I have a question regarding the Service feature. I have to choose the service button and choose the Asterisk PBX ADSI. But let's say I make a call it does not default back to this. I have to go back into Services and choose it

I just received an unlocked ADSI phone and I am playing with the ADSI script. I was wondering how I can include Voicemail functions (Check new messages, Delete message) into the soft buttons. I checked in app_voicemail.c and it looks like these functions have already been programmed. Is there a voicemail.adsi script somewhere? If not, then how do I get the functions I want onto my phone? Thank

Hello Everyone, About a month ago, someone put a question to the list about which ADSI spec to purchase from Telcordia. I looked in the archives, and it appears that this question was never answered, so I'll put it to the list in a slightly different manner: Do I need to purchase the Telcordia specs in order to learn how to write my own ADSI scripts? If so, which one? I found the Black

I know I heard people looking for inexpensive ADSI phones a while back. Can't vouch for these guys, but this looks like a reasonable deal for what perhaps are new phones. No experience with ADSI, myself, but thought I'd pass it along. These folks also carry the Nortel analog units that do intercom and 2-line, as has been discussed in prior threads. JT ---------------------

Hi, Stefano I also have the problem as described by Emanuele. You say that clang schedules target-independent and target-dependent passes. However, when I use lli to execute bitcode generated by opt with -O3 or with the same optimization passes as -O3, the performance are still different. So, are there some special operations by -O3 option? I read the source code of opt, but I cannot find the

Hi all, I have a problem with the cvs head zaptel library: I cannot update my zapata.conf into asterisk when I issue the reload command from the CLI prompt; only when I stop and restart the asterisk service. The system sends the following message: Jun 20 09:49:14 WARNING[3754]: chan_zap.c:10364 setup_zap: Ignoring switchtype Jun 20 09:49:14 WARNING[3754]: chan_zap.c:10364