[Asterisk-Users] Asterisk wengophone
Hi I use Asterisk with some SIP phone (grandstrea), while with my notebook when I'm out of home/office I use X-lite and all work. Some days ago I try to install wengophone and I decided that I want replace X-lite for use wengophone as client for my Asterisk. One of the reasons is that wengophone support g729 codec. I make some test and I see that is possible to configure other sip server (es. Asterisk) but every login wengo download from his site the conf. Now I want that wengo download the conf from my http server with my conf.:) Now I work on this using patient and ethereal, is anyone make wengo and Asterisk work or make this test? -- Pasqualotto Enrico email: pasqu AT linux.it || enrico AT pasqualotto.org web: http://www.pasqualotto.org skype: epasqualotto smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] # and call speed
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, if I append a # after the number asterisk call more fastly, but which step I bypass? Can I append this in all call automaticaly? If yes, how can I do this? - -- Pasqualotto Enrico email: [EMAIL PROTECTED] web: http://www.pasqualotto.org - -BEGIN GEEK CODE BLOCK- Version: 3.12 GIT d? s: a-- C+++ UL P L++ E--- W++ N++ o K- w--- O-- M V-- PS+ PE+ Y PGP- t--- 5 X R tv-- b+ DI- D+ G e h++ r+ y+ - --END GEEK CODE BLOCK-- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFETyY7c25ND+sg2LkRAmSjAJ9TZPAk51OL2u7nwhQHfrtCRYt3sQCgi2KF 2wdSh8JLkyLgKgf53T1m+S0= =Qk2a -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with HT 488 FXO
Macro(SIP/300-3bb9, hangupcall) in new stack -- Executing ResetCDR(SIP/300-3bb9, w) in new stack -- Executing NoCDR(SIP/300-3bb9, ) in new stack -- Executing Wait(SIP/300-3bb9, 5) in new stack -- Executing Hangup(SIP/300-3bb9, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/300-3bb9' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/300-3bb9' asterisk1*CLI but the call is send only to extension 204. Is possible that the Inbound routing routed only from-pstn? My FXO (300) is in a from-internal! Where is the problem? Thanks -- Pasqualotto Enrico email: [EMAIL PROTECTED] web: http://www.pasqualotto.org -BEGIN GEEK CODE BLOCK- Version: 3.12 GIT d? s: a-- C+++ UL P L++ E--- W++ N++ o K- w--- O-- M V-- PS+ PE+ Y PGP- t--- 5 X R tv-- b+ DI- D+ G e h++ r+ y+ --END GEEK CODE BLOCK-- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with HT 488 FXO
Pasqualotto Enrico wrote: Is possible that the Inbound routing routed only from-pstn? My FXO (300) is in a from-internal! Yes, is possible! -- Pasqualotto Enrico email: [EMAIL PROTECTED] web: http://www.pasqualotto.org -BEGIN GEEK CODE BLOCK- Version: 3.12 GIT d? s: a-- C+++ UL P L++ E--- W++ N++ o K- w--- O-- M V-- PS+ PE+ Y PGP- t--- 5 X R tv-- b+ DI- D+ G e h++ r+ y+ --END GEEK CODE BLOCK-- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with HT 488 FXO
Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug: -- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK669516e2;rport From: Unknown sip:[EMAIL PROTECTED];tag=as073738f8 To: sip:192.168.1.157:5062;tag=ebc4a8e2 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Grandstream HT488 1.0.2.16 Contact: sip:[EMAIL PROTECTED]:5062;user=phone Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' asterisk1*CLI -- SIP read from 192.168.1.157:5062: SIP/2.0 481 No Such Call Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK669516e2;rport From: Unknown sip:[EMAIL PROTECTED];tag=as073738f8 To: sip:192.168.1.157:5062;tag=52242a6b Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Grandstream HT488 1.0.2.16 Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.1.157:5060: REGISTER sip:192.168.1.157 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK42700737;rport From: sip:[EMAIL PROTECTED];tag=as558874a4 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 120 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Destroying call '[EMAIL PROTECTED]' asterisk1*CLI -- SIP read from 192.168.1.157:5060: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK42700737;rport From: sip:[EMAIL PROTECTED];tag=as558874a4 To: sip:[EMAIL PROTECTED];tag=3a733fa7 Call-ID: [EMAIL PROTECTED] CSeq: 120 REGISTER User-Agent: Grandstream HT488 1.0.2.16 Content-Length: 0 --- The register string ?? Can anyone help me?? Thanks -- Pasqualotto Enrico email: [EMAIL PROTECTED] web: http://www.pasqualotto.org -BEGIN GEEK CODE BLOCK- Version: 3.12 GIT d? s: a-- C+++ UL P L++ E--- W++ N++ o K- w--- O-- M V-- PS+ PE+ Y PGP- t--- 5 X R tv-- b+ DI- D+ G e h++ r+ y+ --END GEEK CODE BLOCK-- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
