similar to: Verizon, SBC local company?

Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a "basic" Linux/Asterisk system? Just re-boot and config. -- James Taylor jltaylor@metrotel.net 903-793-1953 --

Greetings, I'm ready to start and setup Asterisk. Any preference on which Linux to use? Windows & FreeBSD in use here. I'd like to get up and running as quickly and easily as possible. Thanks -- James Taylor jltaylor@metrotel.net 903-793-1953 --

I've seen the embedded posts. Is anyone running Asterisk on the MINI ITX? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

I am looking for help in writing the scripts for numeric paging via Asterisk. Here's the general flow: answer lookup D-I-D if "on" prompt with "please enter your telephone number" store in variable prompt "please hangup now" Shell to SNPP executable and send message to server. I have the code in "C" for the command line SNPP client. Three parameters

Any means of killing a .call file that is in progress? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

I've got echo problems. *** I'm looking for paid support. *** I'll accept free support, but don't mind paying if someone really knows what they are doing. I've read the wiki, etc. Played with the settings in zapata.conf Using V400P PSTN->_T1->_ASTERISK->_BROADVOICE->_PSTN ECHO ON CALLED PHONE PSTN->_T1->_ASTERISK->_T1->_PSTN NO ECHO

*Hi All, I have starting to reading About SBC and found one artical reagding SBC and they gives a solutions like this. i want to know is this true in realtime sceanario while we think of an big implementation and is it possible with cloud computing. i have found from http://www.smartvox.co.uk/products_gateways_explained.htm Asterisk as a Session Border Controller* Equip the Asterisk server

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress on we are having a few disconnects while calls are in session. I have talked both to some local phone contractors and SBC directly and

Hello, I would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to

Hi, I am new to the world of Asterisk PBX. I have been given the task of coming up with a solution to our office phone situation. From what I have been reading, Asterisk sounds like it could be ideal. However, most of the information I am finding is focused on the VoIP aspects, which is something we want to use for our remote employees, but the first hurdle is integrating Asterisk with the

Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM

Hi, I'm trying to get my * server connected to a softswitch through an SBC. I get the following error when * trys to register. Got SIP response 423 "Interval Too Brief" back from xxx.xxx.xxx.xxx Jan 20 12:43:54 NOTICE[2138]: chan_sip.c:5473 sip_reg_timeout: -- Registration for '1234561234@xxx.xxx.xxx.xxx ' timed out, trying again (Attempt #9) Is there something I can

I have what my SBC/ATT representative calls a "Supertrunk," but he can't tell me the specifics I need to know to configure Asterisk to work with it. By fiddling about, I've come up with the "almost working" configuration below. This works except that it takes about 4 seconds from when the console says the line is answered until it plays my prompt. Watching the

Hi, This is a bit OT. If I have a phone behind NAT and the phone registers via an SBC can I set NAT=NO and canreinvite=yes in asterisk or will I still have issues ? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080313/41d639ee/attachment.htm

This is a really strange one, IMO. I just bought an SBC 900Mhz two-line portable phone. My intent was to use it with one of my TDM20 lines with asterisk, and have the other line be a PSTN line. But it appears as if for some reason I can't intermingle the two types of lines. If I hook *both* lines from the TDM20 to the unit it works just fine. If I hook *both* of my PSTN lines to the

SBC installed a T1 ADTSe (Digital Trunking Service Enhanced) e&m wink start with 24 1 way trunks. The CO says they dial pulse DP the seven digit dnis number. The channels work now but take long time to answer and get these messages repeating until I guess the CO stops Pulse dialing the number. Nov 23 19:08:58 WARNING[1827865]: chan_zap.c:4718 ss_thread: getdtmf on channel 8: Operation now

Hi.. I am about to replace my aging Nortel Venture system with an Asterisk system and 6 Polycom IP 501 phones, and a couple sipura 841's for less used areas. We have 3 phone lines here. One is SBC, one Vonage, and one Voipjet... One hangup is that I can't figure out how to pass through a voicemail waiting indication from SBC. This is important because my wife and her family all

What started out as a good thing for the community has veared it ugly head and will come back to bite us in the ass. I give my respect to the two companies that decided to put themselves 'out there' and attempted to bring 'real world' certifications of knowledge in an area that is unregulated, open, and currently has no measurement of how much one person has over another. What

Hey all, quick question: does asterisk work okay in a Cygwin environment? I want to install it on my cygwin setup for local testing/demoing and save me the hassle of using a pure linux machine........ As long as it doesn't take a huge huge performance hit from running out of Cygwin, then I'll have a go there for a start confirmation appreciated! thanks -- C h r i s E a r