similar to: SIP Listen to multiple ports

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?

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!

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

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

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

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

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.

Can anyone point me to a reference or sample config for bypassing a nailed up (point to point) t1 between two PBXs with asterisk and a pair of t1 cards? Right now I have 2 Nortel norstars connected to each other via a leased line t1. I also have a solid 10mbps low latency microwave link between the 2 sites. My goal is to run an asterisk box at each end with a t1 card and Ethernet card to

This has been an interesting discussion for me (except for the sniping). The last post led me, out of curiosity, to this wiki entry: http://www.voip-info.org/wiki-Asterisk+TDMoE I was unaware of this feature, and it looks pretty good. I've been pondering replacing some T1's by leveraging IP capacity but of course have run up against the QoS issue. My idea was different... I

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,

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:

Is there a clean way to disable music on hold for a specific user sip user? I have seen one example that creates a class called [none] that points to an empty directory, which creates log errors that are annoying (but not harmful?) -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Can someone shed some light on why the 'hint' feature was implemented in the 'priority' field that is purely an integer in the rest of the dialplan? There seems to be a conflict with realtime and the hint priority, in order to put in the hints you would have to change the priority column in the database from int to char and give up some performance (since int indexes better and

I set up hints and presence monitoring on some Polycom phones connected to an asterisk server with the expectation that the phones that are "watching" other extensions would be notified when the other extension sis ringing, in addition to the other statuses (on the phone, statuses set by the user on the phone, not registered, etc). I can see when the line is in use, and when it is

Qualify=yes will send a SIP OPTIONS periodically and keep the NAT open, if you use 1 to 1 NAT (versus PAT where it is "many to one NAT") it will work because port 5060 on the private address will still be port 5060 on the public address. With PAT the port could be anything over 1024, but usually much higher, and the originator will send to port 5060, which your NAT router will drop.

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