similar to: Asterisk on laptop connected to POTS line

Hi I looking a good IAX service for a *emerging * voip provider. Better with a test account to try. Thanks in advance. roberto -- Ing. Roberto Pereyra ContenidosOnline Servidores BSD, Solaris y Linux Soporte t?cnico ISPs Jabber ID: rpereyra@lugmen.org.ar For reliable and professional DNS, use DNS Made Easy! http://www.dnsmadeeasy.com/u/14989 -------------- next part

Hi everyone ! So here's my question of the day ! I need to make a decision on whether or not to go to a voip solution or configure an existing pbx (norstar) that my company has available. We are a small startup. I'm wanting a solution that will support up to about 200 people, with direct dial-in capability, up to about 30 concurrent phone calls and good voice quality. Right now I have

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!

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

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

Hi, It's been forever since I dealt with POTS lines. We have a client that needs FXS and FXO support. If memory serves correct we used the TDM400P with fxs_gs/fxo_gs. What's the equivalent of that card today? TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello List, I know this has been brought up many times but I wanted to know if anyone had any expirience in the following. I setting up several voice mail systems. Each one is going to have a TDM400P. Two FXO for people to leave messages and two FXS for POTS phones so people can listen. Anyone know if there are any simple specific dell models that will handle this without a problem ? 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

I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad.

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!

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

Hello, I'm currently looking at FXO options to provide a POTS line to Asterisk to trunk calls with. Does anyone have any experience using the Linksys Sipura 3201 as an FXO device for Asterisk? Thanks

-----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: jueves, 02 de febrero de 2006 10:15 Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 19, Issue 15 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To

Sure would like to hear experiences using various FXO to VoIP gateways with *. It seems that any thread that has anything to do with problematic FXO interfaces goes on forever with speculation about everything under the sun. Unless there is someone out there with the engineering experience to build a better one it is a waste of time, let Digium deal with it. If the TDM400P can ever be made 99.99%

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

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

Hi all, I was wondering whether anybody here would help me clarify this minor issue please. If I have the following setup; Asterisk ------ Sipura SPA 3000 (fxo) --------- Pstn Line Would a call coming in on the pstn line be answered by the ATA or just get passed through to the * server (depending on dialplan) to handle? So basically, the caller does

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