search for: eutelia

...to call the external world. on fissun, it runs asterisk. On kphone I set as 192.168.1.4 as host and <username> as autentication username. when I run asterisk on fissun, I see only thos warning messages: Oct 25 01:31:55 WARNING[5549]: pbx.c:6438 ast_context_verify_includes: Context ' eutelia' tries includes nonexistent context 'out_eutelia' [...] Oct 25 01:31:55 WARNING[5549]: pbx.c:6438 ast_context_verify_includes: Context ' eutelia' tries includes nonexistent context 'out_eutelia' [...] Oct 25 01:31:55 WARNING[5549]: chan_iax2.c:9582 load_module: Unable t...

...erisk server, but I can't understand how and what should I configure in asterisk in order to accept the redirected call. In asterisk console I can't see nothing when ekiga passes the call. If I turn asterisk's sip debug, I can see that the call arrives to asterisk from 0108392222@voip.eutelia.it (skypho provider) via something containing my external IP address, and asterisk tries to communicate with a host on my external IP address, obviously unsuccessfully, and in ekiga I get a occupied tone. Note that in ekiga I have an account which is in sip.conf, and ekiga registers without proble...

...isk, from which i can see incoming calls and also make calls, mainly only sip, i don't mean trixbox or freepbx. anyhow my main problem is the sipgate.de thing, below are my sip.conf and extension.conf thx for any help sip.conf: [general] language=en disable=all register => user:pass at voip.eutelia.it/number register => user:pass at budgetphone.nl/number register => user:pass at sipgate.de/number register => user:pass at sip.voiparound.com/number register => user:pass at sip.webcalldirect.com/ register => user:pass at ixcall.net/number register => user:pass at freedigits.net...

Hi, I have asterisk 1.6.2.6 on a Debian Lenny system. When I try to send a fax in T.38 mode I receive this error ERROR[15035]: res_fax.c:795 set_fax_t38_caps: channel 'SIP/eutelia-sirio-out-00000000' is in an unsupported T.38 negotiation state, cannot continue. In my sip.config general section I have added this lines t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no If I comment this lines, the fax is sended by G711 but this modality is too slow and instable. How can I do t...

Hi, I continue the discussion from https://issues.asterisk.org/view.php?id=19103 If T.38 reinvite detection should still work, why it doesn't? If I use faxdetect = t38 it does never detect the fax, even using alaw. Cheers, Darkbasic

...9;true'') (''pae'' ''true'') (''viridian'' ''true''))) (''PCI_bus'' '''') (''other_config'' (''struct'' (''install-repository'' ''ftp://ftp.eutelia.it/CentOS/5.4/os/x86_64/'') (''mac_seed'' ''69060b5c-02cf-d8a7-fed2-691f55d1b777'') (''linux_template'' ''true'') (''machine-address-size'' ''36'') (''install-methods'' ''...

Hi! from 2 days I'm trying to run hylafax server and iaxmodem with Asterisk 1.8.5. I have 2 computers in the lan, one is the Asterisk PBX and the other is the server with hylafax and iaxmodem installed. In Asterisk I set up an IAX trunk in this way: ___________________________ iax.conf [iaxmodem] type=friend context=outgoing-fax disallow=all allow=ulaw username=iaxmodem secret=password

Is there anyone with any experience of using the AMD app and the settings that worked for them in the UK ? Any help would be appreciated. Julian

...ot 200 OK on REGISTER that isn't a register I guess it is related to my problem i have with one of my voip providers: i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when picked up, but after a while it stops working, incoming calls for this provider are not shown...

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when picked up, but after a while it stops working, incoming calls for this provider are not shown...