similar to: Problem with flashing FXO callwaiting from FXS

Does anybody out there run * on an AT&T broadband phone line? I'm not seeing any callerid and I can't tell if its AT&T doing something funky or if its my setup. I do see CID on my normal phones.... Thanks, Chris -- The face of a child can say it all, especially the mouth part of the face. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!

Hey all...I'm brand new to * and I want to convert my home into a pbx type setup. I've figured out that I want a Wildcard X100P to bring my single POTS CO into my Linux box. My problem is that I'm sure sure what I need to do to get my analog phones connected up into a structured phone system. It *looks* like I can go the route of the Cisco Analog -> VOIP for about $100 on ebay.

Hey all...for whatever reason my caller id doesn't appear to be working. My setup is simple (Wildcard FXO and thats it) and I'm just expecting the Caller ID to show up on the console. I'm seeing this: *CLI> -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line

Hey all...I'm just getting my * setup and right now all I have is an FXO but no FXS. I wan't to get rid of telemarketers by having * pick up the phone if there is no CID present, give the caller the Zapateller tones and then ask the user to input their phone number via Privacy Manager (yes I realize that this won't get us any where given that I can't re-ring the phones without FXS

Hi everybody... I'm having an odd problem with voice mail on a recent CVS of * where it appears not to detect a hangup on FXO and * will keep treating the call as new and continue leaving voicemails until the max has been reached. It will then continue trying to leave voice mails and basically makes the system unavailble to any further incoming or outgoing calls on that FXO..has anybody

Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT.

I'm working on migrating my home POTS phone system into an Asterisk PBX. Currently I have my FXO and FXS setup and working great. Zapateller is nice! I'm working on my voicemail now and currently have my 3 mailbox answering machine plugged into my FXS along with all my other analog phones. I'd like to slowly migrate away from the FXS answering machine in favor of Asterisk Voice Mail.

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

Hey all..in trying to futher troubleshoot my caller id problem I'm looking at some past troubleshooting tips and this struck me as strange: If I cat /dev/zap/1 I *always* see data...no matter if the line is in use or not...is that typical? Just curious... Thanks, Chris -- "People are not homeless if they're sleeping in the streets of their own hometowns." - Dan Quayle

Hey all..I'm looking to start with a single FXS card but with 3 extensions for VM purposes only. I'd like to know if there is a way that you can have different stuttering dialtones depending on which extension has a VM. For example If x103 and x104 have VM can there be a distinctly different stutter for each mail box and have them play back at once or back to back so that when you

I'm having a problem with Asterisk picking up the Zap/1 and thinking its a new call when instead I've already been on the phone talking to a person. This is not my ideal setup and currently I have just an FXO card and Asterisk is in parallel with my phone system instead of being in the front. I'm not sure if the problem would be fixed by adding an FXS card and putting my phones on

Hellow everyone, here's an other newby question. I've got a * configured with the card in the subject line. At times Asterisk fails to notice a disconet from the incoming line going into one of the FXO ports. Consequently it just keeps the line off-hook for ever and that causes my provider to mark the line aut of order. Is there any way to "help" Asterisk notice the disconect?

Hi Gurus. I searched the lists, wiki and the rest of the web but I still do not understand this. My Setup is as follows: [ISDN via chan_capi or IAX2 DiD Provider] => [* PBX] => [IAX2 Clients (Atcom AT-320ED)] I want to get callgroup/pickupgroup and callwaiting working on the IAX phones. Some web sources told me that this was not implemented, others say that the phone has to handle

Hi all, In order to get the CallerID from PSTN (X100P) I have modified callerid.c file like that: callerid.c [line 256] from: case 3: /* Number (for Zebble) */ to /*case 3: Number (for Zebble) */ Without this modification my own number was displayed as the inoming call CallerID. Now I want to go further. I have activated CallWaiting support on the POTS line. When someone calls and I am in

Hi I have been having a rough time getting a Sangoma A200/Remora FXO/ FXS Analog AFT card set up properly. The main issue is that the card has four ports and as far as I can tell Asterisk is only seeing two. On the two that it recognizes the "Green" FXS ports are not green, they just are not lit. The "RED" FXO ports are indeed red, but from what I have read your not

Is it possible to send callwaiting callerid to a channel without actually having a call waiting, I'm thinking if messaging a handset through the manger interface for example? Robb

Hi- I'm totally stumped configuring my TDM400P with one FXS and one FXO module. Before I got the FXO module, I used to have an X101P, and everything was working very well. Now * doesn't seem to recognize the FXO channel. I've searched the wiki and the list archives. Stock Debian 3.0 stable installation. Any advice? Thanks. -- David Here's my configuration: modprobe zaptel

I think the problem it's in your dialplan, extensions.conf: ; voicemail management [voicemail] include => misc exten => 6245,1,VoiceMailMain2() exten => 6245,2,Hangup() Check the last line, I have the same problem and was because I wrote wrong the Hangup instruction... Regards! Date: Thu, 10 Jun 2004 09:54:32 -0600 From: Chris Hirsch <chris@base2technology.com> To:

Hello List, I am looking at implementing auto callwait on my asterisk box ( has single FXS to which my phone is connected). Is there a way to terminate the current active call (soft-hangup) and auto accept the new incoming call ? (I don't care for call indication). Please advice, -Akshay -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, The call waiting tone is very annoying (you hear nothing while it plays the beep). I need callwaiting because of the queues (the phone has to ring as soon as you hangup) but I want to remove the beep on my dahdi channels, how can I do? Thanks, Niccol? -- http://www.linuxsystems.it