search for: vosp

..., etc.) working example that I could look at as reference? Thank you. PS: Here's what I'm thinking of using: ;====================== sip.conf [general] ;map this UDP port on NAT router port = 5060 bindaddr = 0.0.0.0 ;just to be safe context = dummy deny=0.0.0.0/0 permit=<IP address of VOSP server> externip=<public IP address of NAT router> localnet=192.168.0.0/24 disallow=all allow=ulaw allow=alaw allow=gsm ;all RTP packets go through Asterisk canreinvite=no ;incoming calls from VOSP register => me:mypasswd at mysipprovider.com ;for outgoing calls to VOSP [vosp] ;friend...

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 t...

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: ============= [general] ... register => me:pass at vosp.com [vosp_outgoing] type=peer host=vosp.com username=me secret=pass fromuser=me fromdomain=vosp.com nat=yes canreinvite=no qualify=...

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 whic...

...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 to connect it to a VOSP trunk? Thank you.

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 happens. Do I need to add something to extensions.conf for magic...

...h puts party #1 on hold and gives a dialtone 3. Call party #2 4. Once both parties are off-hook, hit "R"+ 3 on handset to bridge both calls and have a conference call Is MeetMe the right way to do this in Asterisk, or should I look at some other way? Ideally, I'd rather go through a VOSP to avoid the extra digital/analog conversion added by going through the FXO module, but free calls are only available when using that port :-/ Thank you.

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