Hello everyone! On IAX topic. Is there a way to know from which * box the call has originated and onto which box the call is terminating before call terminates? Can the call be trapped efficiently (from dial-plan or such) before leaving network of * servers to PSTN (e.g. voice prompt "Your calling party is outside of free-phone domain. Do you want to proceed?") Suggestions, thoughts - appreciated. Vel
hallo, does anybody successfully managed to get swissvoice ip10s with h323 firmware work with asterisk ? mgcp firmware works fine, but with h323 i'm still getting one way audio. regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------------------------------------------------ A mind is like a parachute... it only works when it's open.
Hi!> does anybody successfully managed to get swissvoice ip10s with h323 > firmware work with asterisk ? mgcp firmware works fine, but with h323 > i'm still getting one way audio.Never tried, no clue. But I can tell you that newer ip10 firmware and latest head CVS (yesterday) don't play together at all - see bug 881. Appli version IP10 M v1.0.0 (Build3) Boot version IP10 Boot v0.3.6 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) Protocol MGCP 1.0 Cheers, Philipp
On Tue, 2004-04-06 at 07:44, Velimir Novkovic wrote:> Hello everyone! > > On IAX topic. > > Is there a way to know from which * box the call has originated and onto > which box the call is terminating before call terminates? Can the call > be trapped efficiently (from dial-plan or such) before leaving network > of * servers to PSTN (e.g. voice prompt "Your calling party is outside > of free-phone domain. Do you want to proceed?") > > Suggestions, thoughts - appreciated.I think that depends on how you implement your network. If you blindly dial from one side to the other, then no because you are transfering control to the remote machine to take control. If you use switch statements, you effectively share your dialplan with the remote machines and they then can query if you have matches. You could implement a new application that looked for a dialplan match without doing a real dial. Then you could then implement your dialplans with a free context and a non-free context which are linked to your local dialplans via switch statements. You can then testdial a extention in your free context to see if it comes back with a match, then test dial using the non-free context. I think you will want to look at pbx_find_extension in pbx.c -- Steven Critchfield <critch@basesys.com>
hallo, i would like to ask if somebody have sip firmware for this ip phone from swissvoice. they announced sip firmware in April 2004 but so far i'm unable to contact product manager and get the sip firmware. best regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ majo at sunteq dot sk ------------------------------------------------------------ A mind is like a parachute... it only works when it's open.
I've noticed quite a few posts on the list about the swiss voice ip10s phone. We recently purchased a few of these phones and have had no luck getting the services button to work any ideas? are the example .cfg files for this phone? any idea when sip firmware is coming? Any help/info would be great. Thanks, Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 ?Fax: 360.694.0219 Email: mhohman@newheights.org -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 526 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040623/7ec66dd3/attachment.bin