similar to: First second choppy

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' ==

Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users

I've been looking for a good iax or sip <==> ptsn provider. Someone with very low cost usa calling and can offer incoming ptsn connections in most markets. The only decent providers I could find were iconnecthere and nufone. Has anyone found someone that really stood out? Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office:

Hi Folks, I'm trying to test out Asterisk overall. I'm having some problems with DTMF. Currently I'm playing with DISA, but I'm worried this will happen when I get to implementing AAs etc. I have a free SIP trunk from IPKall that I'm trying to make work. I'm able to receive calls, and I've now setup and extension with DISA and a password. I connect ok from the

Hy all, Recently a have a little problem with a Cisco device, SPA3102. I use this device with asterisk to dial out with outbound trunk. (SPA3102 has 1 FXO port) It working ok , but the device SPA3102 do this : when a call is placed for outgoing in asterisk and send to SPA3102 , this device "answer and dial the number in the same time" , in my CLI I see the channel is open , but on

Hi All, I Have a Digium Card TDM40P, the specification say: OEM TDM40B: TDM400P + 4 PORT FXS Bundle, my question is: Can I to install analog lines of PTSN?, other detail is: this card have 4 card green. I need to know what is the best card for the following scenario: I need a IVR for my comapny and a PBX, but i want that my extension not use FXS I want IP phone . Thanks ins advanced,

Hello Asterisk List, My company is running a bunch of Asterisk servers behind a Kamailio (openSER) SIP proxy gateway. Calls come in from our PTSN to VOIP service to Kamailio, which then randomly chooses an Asterisk server to handle the call. All Asterisk servers are 1.6.0.9, but the issue I'm about to describe exists in 1.6.1.5-rc1 as well. Ultimately what I want to do is cap each

Hi all. I`ve installed a X100P on my box and is working well with incoming and outgoing calls as a trunk with one PTSN line. I want to connect the X100P to my Panasonic 616 EASA-PHONE as an internal extension to permit to users to make calls to SIP devices from analog phones, the problem is when I dial the ext number where the X100P is connected I get busy tone. What config I need to change to

Hi everybody, Well, I've finally got asterisk to to talk nicely with my Intertel pbx. Currently there is a outside T1 line (e&m wink start, esf, b8zs) connected to asterisk, and then asterisk connected similarly to my Intertel pbx. For right now all asterisk is doing is passing calls between the two. When I call out from the pbx, I can connect perfectly to the outside world. When I

Noob here, my apologies in advance. Recently my employer decide to stop paying for my home POTS line, so I ordered packet8 for a home line instead of another POTS line, since I really dislike my local phone company, and the POTS line without any long distance would cost more than the $20 to packet8 with unlimited US calling. Anyway, this whole thing got me started thinking about VOIP a lot

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

This indicate that the connection with the local provider PTSN it is ok? : -- Attempting call on Zap/10 for s@inbound:1 (Retry 2) -- Making new call for cr 32781 > Protocol Discriminator: Q.931 (8) len=28 > Call Ref: len= 2 (reference 13/0xD) (Originator) > Message type: SETUP (5) > Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) >

Hi, I am new in asterisk. Is it possible to use it with common dialup modem to connect ptsn to the server? Thanks Regards, Soragan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040123/8e77b5cf/attachment.htm

I have a Dell SC440 with Centos and Asterisk 1.4.21 and a card openvox D110PG, T1, when a person calling from the PTSN will listen to them but then begins to distort the voice I heard that name. I probe the card in another computer and it works perfectly. Anyone has any idea or help. I'm going crazy with this problem. Install Debian on this server and the same thing happened to me. I bought

Dear list, I am playing with Sipura 3000 since last week. Through the wiki pages I could get running it reasonably well. My setup is that of a Sipura, linked with a local analog cordless phone, a local PSTN line and the setup to link to an asterisk server located at a remote static ip address. I can dial the cordless phone from other extensions located at the asterisk server; I can dial out

Hi all, I have setup a Dell PowerEdge 2550 with a Sangoma A200 card with 2xFSO and 1XFS modules. The PowerEdge specs are 1 x P3 1133MHz, 512MB RAM. Sangoma A200 has 3 analogue PSTN lines connected. This server is based in Office 1, with 5 users all with a Linksys SPA942 VoIP Handset. There is another Office (Office 2) connected to here using VPN. There are two users in Office 2 with the

I am looking for an ATA that has had very reliable results when passing FAX over IP. I was thinking of testing the Cisco (not Linksys) ATA 186 I1, ATA 186 I2, ATA 188 I1. This is what I'm looking for: FAX -> PTSN -> through Asterisk -> ATA -> Fax Machine. I have QoS from PSTN entry to ATA on the network so I can assure precedence. What has everyone out there been using

Hi ! I ma having trouble with my PTSN line. When I call to my asterisk I get this.. -- Executing [s at from-pstn:3] Hangup("Zap/5-1", "") in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' -- Starting simple switch on 'Zap/5-1'[Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17

We have an asterisk frontend terminating all our SIP phones to, and an asterisk backend with a wildcard PRI card in it connecting to the PTSN. The frontend handles 99% of dialplan logic and just hands off anything outgoing to the backend via IAX2, which dials out on one of the open channels. Lately we've been getting a disconnected calls. Keeping the consoles running it doesn't seem to be

Hi, I'm a student and my thesis work consist in testing Asterisk with Swyx(SwyxWare). My approach is to declare asterisk as h323 gateway for the Swyxserver using oh323 Plugin. Is there any possibility to connect Asterisk with Swyx? how? the outgoing call must pass from Swyxit->to Swyxserver-> Asterisk->to PTSN Thanks