search for: suburbanbroadband

What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions?

Hello all I'm trying to get my asterisk config to listen to multiple ports. This is since some clients have port 5060 blocked by their ISP. Does anyone know how to do this in sip.conf or if it is even supported? Thanks!

Has anyone been able to get the IP501 to discover the FTP server IP address (via dhcp or dns) and download 100% of the config from a provisioning server? We are still having to touch each unit to enter the ftp server address and password, as well as set many of the options that will not take from the config file. Have a sample config file you are willing to share? What is required in

Can anyone shed some light on "rules" that might make the task of parsing the country code and city codes from a dialed number in the CDRs? I know that there is almost never a case where a concatenated country and city code could overlap with another country code, but what about city codes and local numbers? Is it possible for a concatenated city code and local number to match another

Hi, I want to know how I can get a value from a table. Say, I have a table sip_buddies for storing sip user account information. There is a field called 'accountcode' that I want to get its value in the dial plan. As I find that there is no direct way to get the value from the table. Does anyone can tell me how can I get its value in the dial plan? Thanks!

Anyone know how to override the 99 message limit in voicemail? (yeah, we have a public VM that gets that many a day). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050812/1bff12e4/attachment.htm

I have added the following to a macro that is used for all extensions so a user can access voicemailmain by pressing * during the voicemail prompt ; check voicemail exten => a,1,voicemailmain(${macro_exten}) exten => a,2,hangup The behavior is a little weird, the * key is not recognized during the portion of the greeting where the extension number is being played back, after it is

Looking for someone with a successful experience similar to this; I have a need to cross connect a 3COM NBX PBX PRI interface to asterisk, but over a long distance. We do not need any IP connectivity and the solution requires G.711u audio so there is no benefit to using IP. Has anyone here successfully cross connected any PBX PRI interface expecting NI2 PRI signaling B8ZS/ESF with an

...insignificant). Is this easier or harder than TDMoE as described? Does the TDMoE shared idle bandwidth? What about stability (I'm thinking of SW releases)? What other drawbacks or advantages are there? >Date: Wed, 25 Jan 2006 23:53:59 -0700 >From: "Damon Estep" <damon@suburbanbroadband.net> >Subject: [Asterisk-Users] * point to point t1 solution? >To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > >Can anyone point me to a reference or sample config for bypassing a >nailed up (point to point) t1...

Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Regards, Dovid __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Just curious... Does anyone know if any companies using Asterisk on embedded hardware (out at the customer premisis), such as the Soekris Net4801, to provide VOIP service? Doug.

Michel Send me the same output for a dial string that only sends the *31* Is this an ISDN line? What type of card/signalling/switchtype are you using? It looks as if the PSTN switch accepts the *31* and then hangs up so you can make the NEXT call with the *31* feature enabled. If so I assume the *31* feature will be enabled for the next call on the ENTIRE SPAN if it is an ISDN trunk group. If

In an earlier thread Andrew Kohlsmith enlightened me on the use of qualify in sip.conf to deal with a peer that is down. Since then I have been searching for information on how the behavior of qualify can be tuned. The wiki is vague on this; " Syntax: qualify=xxx|no|yes where XXX is the number of milliseconds used. If yes the default timeout is used, 2 seconds. If you turn on

Hi, today a customer asked how to use asterisk with 12 to 16 FXO ports. I can use a channel bank with 16 FXO ports and connect the channel bank with a T1 cable to a T1 card in the Asterisk Server. Asterisk will then send the calls to the Voip provider over the internet. However a 16 fxo port channel bank is about USD 1500 + a t1 card USD 500 + a USD 1000 computer = 3 thousand us dollars + my

...done then > in real world? I cannot see any business moving to > voip when you cannot control quality of service. > Can you recomend any free programs which alow you > analyze delay,jitter and packet loss? > Once again thank you. > robert > > > --- Damon Estep <damon@suburbanbroadband.net> wrote: > > > That "program" will be detected by your ISP within a > > day or so, > > determined to be a virus, and your service will get > > disconnected...which > > n turn will not help your latency or jitter at all. > > > > VoIP can...

Are all of the sip phones in the same context? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Robert Jenkins > Sent: Tuesday, January 16, 2007 1:44 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [asterisk-users] Polycom IP601 - some hints working,

Looking for a liitle help if anyone has dealt with this; The options on dial and queue of t (allow called party to transfer call) and T (allow calling aprty to transfer call) seem to work fine (as long as you do not confuse them with the same t and T that indicate timeout!). The problem I am having is the use of the # key to do so. Many times a caller will palce a call to an IVR that requires

Guys'n'Gals vote for bug 3836 - Chanspy is back. Better than ever. Let's get this one into CVS. Julian

...isk-users- > bounces@lists.digium.com] On Behalf Of Michel Koenen > Sent: Tuesday, August 30, 2005 1:46 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] How to use * and # as part of > numberindialcommand > > > From: "Damon Estep" <damon@suburbanbroadband.net> > > Subject: RE: [Asterisk-Users] How to use * and # as part of number > > indialcommand > > > > Good to hear you have found a temporary solution, although I think it is > > the permanent solution. > > > > Keypad protocol is a bandaid to fix...

Does anyone know what date this memory leak was introduced and/or how to check source code for it? I am running a pre-1.2 CVS head version and would like to know if the potential problem exists. > -----Original Message----- > From: asterisk-announce-bounces@lists.digium.com [mailto:asterisk- > announce-bounces@lists.digium.com] On Behalf Of The Asterisk Development > Team > Sent: