[Asterisk-Users] "Non-blocking" Dial (and other commands): is there a way?
In order to use a with GrandStream BT-488 as "pass-through" gateway, I need a way of sending the FXO port off hook when I'm using the FXS port for VoIP communications, because I want to use the "hunting line" feature to let incoming call skip that FXO port and move on to the next free line. The only way I have found to engage a device without getting blocked until the call ends passes through an AGI script that drops a callfile into the /var/spool/asterisk/outgoing directory, telling Asterisk to dial the FXO port and then connect the channel to, say, the MusicOnHold() application. When I'm done, I can then issue a SoftHanghup() to the FXO device. This method strikes me as pretty clumsy: aren't there better ways of issuing commands from the dialplan in "detached mode", perhaps getting a handle useful to regain control later, and proceed to do other things? Enzo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: g729 to asterisk to g729 voip provider
- Original Message - From: "Christian B" <[EMAIL PROTECTED]> To: Sent: Saturday, September 17, 2005 5:44 AM Subject: Re: [Asterisk-Users] Re: g729 to asterisk to g729 voip provider > On Fri, 16 Sep 2005 16:09:37 -0500 > Erick Perez <[EMAIL PROTECTED]> wrote: > > > Hi, your project is indeed interesting, however for learning purposes > > i do need to know the answer of at least: > > it is not my project. > > > 1- Using sipura sip/g729 to connect to an asterisk server that will > > server as a gateway to a VOIP provider(g729), all in g729 will require > > to purchase codecs from Digium? > > read the page, it provides you with a free version of the g729 Erick's question is better answered here: http://www.voip-info.org/wiki-Asterisk+G.729+pass-thru Enzo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 ATA
- Original Message - From: "Rosario Pingaro" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, September 14, 2005 9:07 PM Subject: Re: [Asterisk-Users] T.38 ATA > I can confirm that sipura spa-2100 has t.38 suppurt from firmware 3.2.1 > > and it seems to work fine in our test with some t.38 providers. Are they pay-as-you-go providers? If so, do you mind sharing their names with us? Enzo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone knows how to receive a SIP call withoutregistering gateway?
Well, a SIP authorization does not require a registration (in fact, registration should be primarily used to inform a registrar about the whereabouts of a UA with dynamic IP address in order to handle incoming calls _for_ that UA). CS can just create for his Asterisk a "type=user" entry in sip.conf containing "username" (equal to the section's title) and "secret" both matching the remote peer's own: his Asterisk will then react to an INVITE from that peer with a "401" reply containing a nonce as challenge; the peer will then retry the INVITE with valid credentials based on the shared secret and the nonce. Enzo - Original Message - From: BJ Weschke To: C. Savinovich ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, September 14, 2005 2:49 PM Subject: Re: [Asterisk-Users] Anyone knows how to receive a SIP call withoutregistering gateway? What they're asking you to do is quite insecure to be doing over public IP. At the very least, you should confirm that there is a static IP that these calls will be coming from and only accept calls from that IP, but that's still not quite as secure as digest authentication that would be available via registration. If you know what IP the calls are coming from, you simply insert a host=XX.XX.XX.XX instead of host=dynamic in your sip.conf for that peer and calls should then come in as they did before without them having to register. If they are pre-pending digits on to the front of what you're interpreting as the dialed number/extension, you may choose to lop them off in extensions.conf, but aside from that this is fairly straight forward. On 9/14/05, C. Savinovich <[EMAIL PROTECTED]> wrote: Hello everyone, I am pulling my hair here because a carrier threw me curve early today. They want to send calls to my asterisk server using SIP. Then they said that their gateways don't have to register with my server, that all they have to do is send a prefix for validation. Whereas I can think of several ways to authenticate their incoming number string, I am only used to the orthodox SIP way which is: client registers to my proxy. Guess what, I can't find any samples on this!!, Can anyone please help?, I will probably need a sample sip.conf. and then, to make a test call, I can use another asterisk box and try asterisk to asterisk sip calls (without register) via the cli prompt. But I have no idea and I am intrigued. Thanks CS___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitoring status of ISDN lines
When Asterisk uses an ISDN interface, it periodically sends to CLI messages such as: == Primary D-Channel on span 1 down [...] == Primary D-Channel on span 1 up Is there a simple programmatic way of capturing them for monitoring purposes? Enzo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
