similar to: Asterisk + VOSP account working configuration?

Hello I don't have a landine and use a VOSP to provide access to the telephone network. In case a call comes in and I'm not home, I'd like Asterisk to log the call, and then send an SIP message to my VOSP so the call is forwarded to my cellphone and is thus charged to the caller, without Asterisk having to dial out to my cellphone through my VOSP at my expense and bridge the two

Hello I've done some googling, but still puzzled at my working configuration. Apparently, a "user" can only receive calls through Asterisk, a "peer" can only make calls, and a "friend" can do both. If that's correct, I don't understand why my VOSP requires the following settings in sip.conf to let my Asterisk server make/receive calls to/from the PSTN:

Hello At this point, I have an Asterisk 1.4 + PC running XLite behind a NAT set up with a VOSP trunk that I can use to make/receive calls to/from the PSTN. Now, I'd like to be able to call any number on the Net that is advertised as "sip:user at domain.com", such as those: www.voip-info.org/wiki/view/Phone+Numbers Do I need to register a second trunk (FWD, etc.) through which

I am using the latest CVS version of * (just upgraded incase it was a bug) and I am having problems with the directory. Whatever I enter when dialing it the only result I get back is '6000' which is not even in the voicemail.conf extensions.conf contains :- ... [voip] exten => 6010,1,Macro(cst) exten => 6011,1,Macro(std) exten => 6012,1,Macro(std) ... exten => 6503,1,Answer

Hello I was wondering if someone knew of an application that could check that the user has a firewall and a broadband connection that will work OK with Asterisk and VoIP. The app would first perform some bandwith + jitter tests, and will then call a STUN server to check that the firewall isn't symetric. BTW, is Asterisk now STUN-capable, or is it still to map ports manually on the firewall

Hello This is a newbie question : With a simple Asterisk server on a private LAN, an FXO port to handle the PSTN, and an ADSL connection to the Net, ie. with no VOSP in the mix... how should I configure Asterisk so that SIP clients can dial SIP numbers on the Net, such as those below to perform an echo test? www.voip-info.org/wiki/view/Phone+Numbers I tried pasting numbers in XLite, but nothing

Hello I've never used Asterisk for a three-person call, and would like to check that MeetMe is the way to do this. The ADSL modem provided by my ISP offers free calls to landlines/cellphones when using a handset connected to an RJ11 port on the modem. A three-person call can be set up by using the standard PBX sequence: 1. Using the handset, call party #1 2. Hit "R" key on

I'm having trouble connecting a asterisk server to a SIP Express router. Inbound calls to my asterisk server works just fine, but when i try to make outbound calls I get the following error message: Nov 27 22:40:48 NOTICE[4687]: chan_sip2.c:7967 handle_response: Failed to WWW-authenticate on INVITE to '"username" <sip:username@mysipprovider>;tag=as5399a078' I'm

If you update your source from the CVS, you'll get a new SIP channel that supports a new syntax for SIP calls in extensions.conf If you define a SIP peer in sip conf, like [mysipprovider] ... You can now use dial(SIP/mysipprovider/extension) Where the part "mysipprovider" is related to the sip.conf section. Also, you can dial any SIP URL by

Hello all: For fun, I am learning about Asterisk, and trying to get Asterisk working at my house. I installed Asterisk@Home. It seems to be functioning fine. I installed a couple of softphones, and have them registered with Asterisk. I actually work for a CLEC, and I have registered my Asterisk box with SER (which I don't begin to understand yet) at the office. In order to try to

Can asterisk act as a SIP registrar or location server? I would like to be able for a user agent(client) to register with whatever client they are using as "username@domain-name.com". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages

Hi folks! I bought a X100P a while ago and know I've tried to get it working here at home again ... but I can't manage to get my X-Lite client working with Asterisk (CVS from a day ago) ... I've downloaded the latest version of X-Lite and I believe that I've set it up correctly ;-) But I cant get it to register with my Asterisk - I only get "Login timed out, contact your

Hi all, I have a small system of two hardware boxes (residential gateways) running Linux with Asterisk on them. Each RG has some FXS ports to which analog telephones can be connected. I already had a working system including an external SIP provider, where both RGs would register to that provider with a telephone number and they could call each other via that telephone number. Each RG had a line

Dear all, I would appreciate it if you could teach me "Asterisk SIP NAT Config". I'm trying to capture SIP Register with externip that should set in contact header at External SIP Server as shown below, but I haven't seen it. I need your help. My experiment environment is as follows.

Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine;

Hi, Could you please help me!! I am trying to configure the Asterisk server. I have a analog phone connected to a FXS port of a Cisco 3745 router. This router is connected to a Asterisk server via Fast Ethernet interface. I am trying to make a call from the analog phone to a SIP client. This SIP client is registered to the Asterisk server. Analog phone number: 999 SIP client : 202 Sip client IP

Hi, the documentation of sip.conf is telling me this: ;register => 1234:password@mysipprovider.com ; ; This will pass incoming calls to the 's' extension In reality it jumps to the extension 1234 in the context and not to s So it is much more complicate to write an proper dialplan. Is this an bug or is the documentation not up to date? best regards Thomas

Full_Name: Paul Vos Version: 2.0.1 OS: windows XP Submission from: (NULL) (150.216.148.20) > zapsmall(.3-.2-.1,digits=7) [1] -2.775558e-17 This should be zero. By changing the condition if (mx > 0) in zapsmall to if (mx > 1) we get > zapsmall(.3-.2-.1,digits=7) [1] 0

Rowland, I've done almost all permissions change, I forgot bind-dns directory. Now, the named service still doesn't start and journalctl -xe showed me that this occurs because permission denied to run dlz_bind9_9.so. I've checked out and the lib and directory /usr/local/samba/lib/bind9/ have execute permission to named group. The output of ls command, journalctl -xe and

hi there, I have just downloaded and installed asterisk a couple of days ago, it compiled correctly and starts up and runs, on a Redhat 9 system freshly installed for testing. I don't have any extra hardware installed so far, was attempting to just try out connectivity. I am having some probs with the configuration, maybe someone out there can give me some tips : firstly on modifying the