similar to: SBC ADTSe - Sending DP digits

I am trying to make an in/out trunk group comprised of 4 DS0's using E&M Wink signalling. The first four channels of a DS1 on a T100P are being used for the group. Outbound calls work fine, but inbound calls fail. The other 20 DS0 channels are used for a PRI. Does the configuration shown below look okay? I've tried setting 'immediate => yes' without success, but it

Hi All, I'm trying to configure a Digium T100P to talk to a legacy voicemail system. I have the signaling specs verbatim from the original manufacturer documentation as follows: [T1 Signaling] Service Type: T1,D4 format, AMI(Super Fram) Signaling: Four wire, terminated, E&M (Robbed bit) Start Protocol: Wink start; 250msec duration Dial Tone: Enabled Digits: DTMF, 4-digits DTMF: 50msec

Hi all, i use asterisk 1.2.7 and i have a problem with callerid. i use sangoma a200 cards. one fxo one fxs module i have these scenario - bob calls adam, where bob calls into my asterisk and adam picks up "from" my asterisk - bob and adam are speaking to each other - meanwhile eve calls adam, adam hears a beep, and knows there is an other caller on line. - bob and adam stop seaking

Hi list, I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS 5. I am having trouble to make my simple IVR extension work, here is relevant config: zapata.conf ---- context=incoming signalling=fxs_ks channel => 4 context=internal signalling=fxo_ks channel => 1 ----- extensions.conf: ---- [office] exten => s,1,Dial(Zap/1,30) [home] exten =>

Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 0412222222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a "failed checksum" like

Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 0412222222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a "failed checksum" like

hi all, We have a TDM400 card with 4 wfo modules. now the modules load fine and when i start asterisk with on phone line connected it just starts spewing these messages: -- Starting simple switch on 'Zap/4-1' Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! Making some changes in extensions.conf to test the incoming calls so that these are derived to a SIP extension, I found something that draws attention to me: if I test calling to my PSTN line from a mobile phone, when take the call from the SIP extension (softphone), if the mobile phone releases the call, sofphone do it too without problems,

I've updated to the latest CVS as of today (and rebuilt RedHat 9). My setup is as follows: Wildcard X101P - channel 1 TDM400P - channels 2-3 - fxs cards with fxo signalling, channels 4-5 - fxo cards with fxs signalling I got CallerID to work on channel 4 with an old CVS, despite the usual "Didn't finish CallerID spill" message. However, as soon as I insert the following

Aloha, I'm finishing up the final touches on this install, and have run into an odd problem. I can't seem to get Caller ID on the analog phone lines working. It's a Digium AEX 410 card. I have Verbose set and a line to print the CID: I have usecallerid=yes and callerid=asreceived set in both chan_dahdi.conf and users.conf [analog] include=>default exten =>

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 have an FXO module in my TDM400P configured to receive caller*id (see zapata.conf below). I get a curious behavior: When I call this line with my cell phone, I see caller ID received just fine, with no warnings or errors.. When I call from another landline, I get different results: calling from external line, caller ID off: WARNING[1233021872]: chan_zap.c:4980 ss_thread: CallerID

Can anyone tell me what this message means? Got event 17 (Polarity Reversal)... I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0. It appears that I get this Polarity Reversal each time an inbound call hangs up. This results in another ring, but no one is there. It appears as an unknown caller, but I believe its a phantom. Thanks, Jim [Oct 10 12:47:54] NOTICE[6669]:

Using the latest SVN of zaptel and asterisk, I can no longer receive incoming analog calls. The caller just hears it ringing but Asterisk doesn't pick up. I am seeing these error messages: [Apr 9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 9 09:38:50] WARNING[16570]: chan_zap.c:6470 ss_thread: CallerID

A very good day to you all, We can't get the phones to pick up on an incoming call on analog trunks. We're using the digium products in the box, with snom phones internally. This is the output from the asterisk console: linux*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo pstn-incoming en default 1 pstn-incoming

I have a Toshiba DK424 connected via T1 E&M to a TE110P card. Intermittently when a user dials a number I am getting 'getdtmf' errors on the Ast server and the calls do not go through. If they dial the number once or twice more, it works fine and I receive no DTMF problems. On another note, end users are complaining about intermittent disconnects. T1 is ESF/B8ZS - 24 chan. Other

Yo! I changed callprogress to no, and in wcfxo.c source around line 334 i changed the value 32000 and -32000 to 10000 and -10000 because it had something to do with the DC voltage when it was ringing. I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an interesting diagram of wiring that was incorrect for sending voltage to a phone or something like that. So put it

pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4

Sorry if this is a daft question but when a PSTN call comes in on my X100P the console shows the following; NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2 (Ring/Answered)... NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2 (Ring/Answered)... NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2 (Ring/Answered)...

I am a little green when it comes to all this but I am trying to connect our PBX to an asterisk server using a TDM400 with 4 FXO modules. I am able to dial an extension on my PBX handset and I get a dialtone from the PBX. After 2 rings I then hear the asterisk server connect and I get a dialtone from asterisk. I am then able to dial an extension on another asterisk server. My question is: How do