similar to: asterisk SIP domain (in LAN or DMZ)?

Hello I'm using eyeBeam, and Asterisk keeps sending my clients a second call, when they are still in one call (because eyeBeam has lots of channels). I was using X-Lite (with 3 channels) and Asterisk never sent the client a second call. How can I force Asterisk (or eyeBeam) just to send one call at each time. Is this a configuration I need to do in eyeBeam or Asterisk? Thanks Regards Joao

Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx -> the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr

Hello list, Im writing my dial plan, in witch every SIP phone begins with 74 and has more 3 numbers (like 74XXX). So, I want to route all 74XXX calls to my sip channel. For this I wrote this line: exten => s,1,Dial(SIP/74118@193.136.252.5,30,r) but this way all calls go to 74118@193.136.252.5 ..... Then I tried: exten => s,1,Dial(SIP/${EXTEN}@193.136.252.5,30,r) but this way, the

Good evening! I was wondering one thing, I'm using freepbx to configure my asterisk server and I have a problem with some inbound calls. When I receive a call to an INVITE sip:01xxxxxx at myip.com I an set an inbound route! It matches a DID number. How can I route an INVITE sip:s at myip.com? The number only appear in the To: Section. Thanks! Example: With this one, I cannot route it

I have configured call pickup, and this works fine. Although there are 2 problems, perhaps anyone would know a solution to this; - When I pickup a call from another set, the *8 code keeps being displayed in my screen (Snom 220). I would like it to show the phonenumber of the person calling me. - When a caller that I've answered through Call-Pickup disconnects, my phone does not close

Hello to all Im looking for a billing tool for Asterisk, that works with PostgreSQL. All the tools I found in www.asteriskbilling.com just work with MySQL :( Do you know a nice billing tool for Asterisk with PostgreSQL? Thanks Joao Pereira

Hello I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom and Cisco Phones. Can you gurus, please, give me your impression of these 2 brands? I need to focus more in SIP and Asterisk compatibility and less in pricing (yes, I know the Cisco are more expensive). Are there any features that Snom has, that Cisco doesnt? And are these features important? Thanks Joao

Is it possible to implement a dial-out call queue in Asterisk? My idea is to give Asterisk a list of numbers, and then he makes the calls and delivers the calls to a call queue. Then, the agents will answer the calls. Is this possible? Thanks Regards Joao pereira

Hi, As many of you may know, we are undertaking several tests in order to test the interoperability between several PBX IP from different vendors. Until now, we were trusting that the VoIP IP PBX were good enough to be interconnected directly, however, one of the vendors have presented the "SBC" concept. The "SBC" (Session Border Controller) is not a new concept since we

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Chris Mason (Lists) Sent: Monday, September 17, 2007 12:45 PM To: joao.pereira at fccn.pt; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Center SoftPhone with Auto Answer Joao Pereira wrote: > But still, the

Hi all I have a pocketPC Toshiba e750 and I want to make SIP calls from it, but I didnt found any free softphones for my Toshiba. X lite's versions for pocketPC isnt free :( Did someone used before a free softphone for pocketPC? witch one? Thanks Joao Pereira www.fccn.pt

Hello Our NT workstations always ask for the password on each share of the samba server, although the password is the same as the log on password. Is there any way to avoid this ? The first try always fails on the server, but I don't what user/passwd combination NT uses on that first time... Thanks, Joao Pagaime -- FCCN - Fundacao para a Computacao Cientifica Nacional - Tel: 351-1-8440100

Hello, I would like to get help on the instalation of R. I have too few free space in my pc hard disk. So I wonder if it is possible to install R on an external removable hard drive. Can it be done? How should I proceed? Thank you for your help. best regards, Francisca A. S. dos Santos Bronner --

Hi guys, I've setup on box with a TE110P and time to time I need to access remote equipment outside of our office and use a data channel. I'm wondering if do I need to buy a POTS line only for this time to time acess or what's the easiest way to do that via my TE110P on asterisk box. I know that is possible data transmission with this Digium Card, I'm wondering how... Any tip any

Dear all, I've a live system that needs to be upgraded but, before I proceed to the upgrade I want to assure the rollback process. That's why I'm requesting your feedback, in fact this asterisk in live system isn't going so bad but.... the upgrade is essential NOTICE that the upgrade will keep the same version 1.2 not from 1.2 to 1.4 Requirements: -backup /usr/sbin/asterisk

Hi List; Is Digium the best telephony cards to be used with Asterisk? The prices are some how high, any suggestion? Regards Bilal ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

I know this issue has been covered with at least 2 different patches, and probably a dozen different discussions, however I'm a bit unclear as to what my options are. I have a DSL line coming in with 8 IP addresses going to an OpenBSD firewall doing 1:1 NAT for machines behind the firewall. My asterisk box is one of these machines, and I'd like to allow foreign SIP clients

Hi, I have our software with SIP running on it.I configured asterisk server as proxy. How do I implement the call screening features(incoming and outgoing) using asterisk server.Please suggest me how to proceed on this. Thanks & Regards, Jahnavi. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps

Dear all , Is there any subset regression (subset selection regression) package in R other than "leaps"? Thanks and regards Alex [[alternative HTML version deleted]]