similar to: Outbound dialing issue with FXO

I can call my Broadvoice DID from a outbound caller-id blocked phone, and BV happily delivers the CID to Asterisk (and then on to my IP phone display.) I've tested with the *67 prefix from a PSTN phone to make sure it was supposed to be blocked. The number is always correct, but sometimes the the caller ID name is set to something funky (like a CO or switch center name.) I *think* this

I started out happy as a clam with my new Broadvoice account and asterisk machine. About 10 days ago things began to change. Inbound calling has been down for 2 days. Beyond the "We are currently experiencing in-bound call issues with a carrier partner in some areas. We are aware of the issue and our engineers are working to have it resolved as soon as possible" mantra their

Hello all, We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond....) We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line

I would like to install a TDM04B with Asterisk@Home 2.4 and Asterisk@Home 1.5 but I didn't find documentation about this installation. Thanks in advance, Nelson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060206/85ff5252/attachment.htm

I don't know what's changed, but four SPA841s and a SPA3000 are no longer answering when they get an inbound call from *. This has been a working configuration for weeks. I *have* been fiddling with the server config; however, the configuration is under version control and I've reverted everything to exactly how it was when the server was working. Doesn't fix it. I reset one of

I would like to know if it is possible to slow down the dialing process in asterisk. I have 4 of my 8 phone lines that are VoDSL. When we try and dial out these 4 VoDSL Lines, the number is often miss dialed, or incomplete. I added a wait before Asterisk tries to dial the whole number, but that has not solved my problem. If I use a regular phone and dial out these lines, they work fine. My

hi, To post, and to reply to a post, i have to goto my email. Now, if there is a web interface to these mailing list, things would be easier.

OK, here's the setup, AAH 0.8, Grandview 2000 phone, Digium TDM04B interface to POTS lines. Everything seems to be working just fine, but I have some questions on how to access voicemail options. I can leave a message for an extension, but when I try to retrieve it by using *97 it asks for the password and even though I type in the same password I gave the extension for vmail, it tells

Hello. I have a MinITX motherboard with only one pci slot and one onboard ethernet interface, I have a TDM04B card plugged into that motherboard and the /proc/interrupts: CPU0 0: 169626332 XT-PIC timer 1: 1270 XT-PIC i8042 2: 0 XT-PIC cascade 8: 4 XT-PIC rtc 12: 170166219 XT-PIC eth0, wctdm 14:

I have server with a total of 6 Analog ports...using TDM04B and TDM02B. I have 3 Lines that are DIDs and 3 are Main/Roll Over lines and I have worked through getting the DIDs to work and route to the extensions...now what I need to do is when Extension 1111 picks up the phone to dial, I would like them to use their DID analog line first, unless someone has called in on it and they are trying to

Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount,

Hey all - we have two TDM400P cards in an SMP Redhat9 box, with 4 FXO ports each running thel latest Asterisk CVS. Users connect to this to share POTS lines using analog phones connected to Sipura SPA-2000 boxes. All works reasonably well, except every day (or more) a line will get "locked up". No other way to describe it really -- the line just stops working, so if someone tries

Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco

I currently have a Toshiba Strata DK424 with a Stratagy voicemail system (4 ports). I am looking to go from having a receptionist answering the phone to an automated attendant. It appears that Asterisk can be the solution, but I have some questions. Do I just replace the Stratagy with the Asterisk or do I need to reconfigure the routing of the phone lines? Currently, the POTS lines come in to

I have a client who wants a Meetme box with 12 FXO ports, to connect to Analogue lines coming from an Ericsson PBX. It looks like I could do this with four different hardware configurations: a) three TDM04B cards (based on TDM400P) b) one TDM04B and one TDM808B c) one TDM804B (or TDM854B?) and one TDP808B d) one TDM2403B (half filled TDM2400P) Apart from considerations of cost and PCI slot

I just install a Digium TDM04B card. I created 4 separate Zap channels and one outbound routing containing zap channels from 1 to 4. If a phone line is plug in Zap/1 then works fine, but if I unplug phone line from Zap/1 (simulating fail) the system keep dialing out on Zap/1, even with no dial tone; Only if Zap/1 is busy it will use Zap/2. There is any work around or different setting to avoid

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

Hi all, I've been trying to wrap my mind around this one for several days now. How can I 'debug' the CallerID reception on a Zap/POTS channel? I have a POTS line with CallerID and a Digium TDM11B card right now. I have my signalling set to ks for both sides, can make and receive calls just fine. But I never get CallerID on incoming calls. I get the following messages: Aug 11

Hello, I am looking for 4 port FXO low profile PCI card that could be used with Asterisk. Digium TDM04B sound like a good choice but it is half high PCI card and I can not plug it in my Dell box (small box). I am looking for adequate low profile PCI card (55mm high or similar but definitely smaller than TDM04B so I can plug it in). Does anyone know where to search for it? Thank you in advance,

Hi, I just wonder, if it is possible, to suppress my own number on outbound dials with chan_capi. I took a look into the sources and think it might work with toggeling the "@" in front of the outbound msn in the dialstring. (Dial(CAPI/@msn... vs. Dial(CAPI/msn... But it doesn't work. Maybee I'm wrong and misunderstood the code. Thanks for any answers! Karsten