similar to: Psgw

I'm sitting in the Emerging Telephony Conference, so this seems a particularly apt place to pre-announce this... I've wanted to be able to gateway calls between Skype and Asterisk for a while, which of course would require some type of protocol converter (IAX or SIP to Skype, probably.) This of course is directly not in Skype's interest, since they would like to keep the network

Hello, I bought a copy of this software in the hope of bridging Skype into my * box. It sort of works but the audio is all distorted and there's huge latency. Does someone have this working well with their * server? According to the wiki someone does, but I don't know who. I'm needing advise on what to fiddle with to get it working correctly. Thanks Michael -- Michael Graves

Sorry, i also tried this: exten => 6000,1,Answer exten => 6000,2,MusicOnHold(default) and i got this result: *CLI> -- Executing Answer("SIP/2391-8cdd", "") in new stack -- Executing MusicOnHold("SIP/2391-8cdd", "default") in new stack Jun 29 19:33:47 WARNING[1616]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class

Hi all, i'm tryinf to install chan_capi but i get this error root@obelix:/usr/src/chan_capi-0.3.5# make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DNEVER_EVER_EARLY_B3_CONNECTS -DCAPI_ES -DCAPI_GAIN -DDEFLECT_ON_CIRCUITBUSY -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO

I cannot use CVS, is there anoyher way to use direct pickup ? Thanks again Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Alexander Lopez Inviato: venerd? 16 settembre 2005 17.53 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: RE: [Asterisk-Users] direct sip call

This is my musiconhold.conf and my folder: root@voip:/etc/asterisk# less musiconhold.conf [classes] default => quietmp3:/var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random => mp3:/var/lib/asterisk/mohmp3,-z ;unbuffered => mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters

Ok, but is there a way to check if echo cancellation is active on a call in progress ? Thanks Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Steve Davies Inviato: marted? 28 marzo 2006 16.43 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Echo cancellation

Hi, I just installed spandsp 0.0.3pre6 with libtiff 3.7.1 and evrythinghs is ok. I recompiled asterisk and run ldocnfig, but whrn I start asterisk I get this error: [app_rxfax.so]Jan 12 16:40:42 WARNING[1569]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Jan 12 16:40:42 WARNING[1569]: loader.c:440 load_modules: Loading

Hi, just a question: anyone has never installed g729 codec on VIA motherboard with C3 processor ? I'm having problem with IPP libraries, and Intel said that it works only on Inter processor. Any suggestion? Thanks Giordano -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm looking for a PCI card which i could install on asterisk box, with purpose to use 15-20 cordless dect phone in a very "dect cell". Is there anyone that could help me pls ? Thanks Giordano -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have to re-compile also app_rxfax.so and app_txfax.so or just spandsp ? Thanks Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Colin Anderson Inviato: gioved? 12 gennaio 2006 17.20 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE: [Asterisk-Users]

Hi group, anyone can explain me the exact difference between pri value in zapata.conf ? ; PRI Dialplan: Only RARELY used for PRI. ; ; unknown: Unknown ; private: Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN If I use it, I also must use prilocaldialplan = local ? Thanks Giordano -------------- next

I known. I'm using the 1.44 firmware version relesed on 26 may. I worked for italian IVR an HTTP pgaes. So i can only update asterisk with CVS and try atxfer. Thanks for all -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill Inviato: luned? 30 maggio 2005 18.40 A:

Good...it almost works fine! I just have 't' in command Dial, but i also have 'T'. Is it a problem ? This is my Dial() exten => 605,1,Dial(${GIORDANO NAT},60,Ttr) I have only a problem: A and B are speaking, B calls C and ask it if wants speak with A, C accept but if B hang up A is waiting and C get busy tone. To make it works B don't have to hangup but habe to press

I'm going to try, Thanks very much Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Remco Barende Inviato: luned? 30 gennaio 2006 20.04 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Kirk IP600 Hi! Yes, it works (sort of) but I still have some issues.

Which models of Diva could work with CAPI and asterisk? Thanks Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di gw@adcomcorp.com Inviato: sabato 1 ottobre 2005 23.46 A: asterisk-users@lists.digium.com Oggetto: RE: [Asterisk-Users] Diva Nope. At least I tried and never could get it

Thanks...which version of IPP did u use ? I do not have Makefile file....there is only a .sh script Thanks Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Juan Salas Inviato: luned? 3 ottobre 2005 15.41 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE:

Hi, Thanks for yuor answer. The boot time of the phone is very very fast, 10 sec to startup and 2 or 3 second to login to asterisk. I set the NTP server to 255.255.255.255 so it don't try to get time. I thinked carefully to your scenario and i am going to try it, but i don't known if it could like to my customer I will try also to use CVS, but i am skeptic to utilize asterisk to

Hi all, i'm going to install asterisk with a 4 BRI (HFC chipset) on a Celeron at 2.6 GHz I don't known Celeron performance, but i listen that is not very good. Could I have some performance isuue with this kind of processor ? Thanks for all Giordano -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have to bought a PCI with 4 PRI but on digium site I saw that there a re two different kind (3,3V and 5v). What's the difference? Which one I have to buy for do not have any problem with this motherboard? (Gygabyte GA-8S661FXM-775). I checked on Gigabyte website but I don't find any kind of this value :-| Thanks all Giordano -------------- next part --------------