[asterisk-users] NEED ASTERISK DEVELPER : OH323-asterisk driver and openh323
Dear List, I'm looking for a coder/developer that can modify oh323 return codes on asterisk Example on based on SIP and h323. Right now we are receiving : Call Rejected (code 21) Network Out of Order (code 38) Need to able to replace dose codes with - No Circuit/Channel Available (code 34) Please contact me on [EMAIL PROTECTED] Thanks, Oliver Vermeulen ___ --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] JAMAICA DID'S - 1-876
JAMAICA DID'S - 1-876 NOW ACTIVE ON www.didx.org Oliver Vermeulen World Venture Group Telecom Corporate Address: 147 New Haven Point Lane West Palm Beach , FL , Miami USA DID: +1 (305)722-1457 BE DID: +(32)9-395-5620 UK DID: +(44)870-478-8896 SIP : [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] msn: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.wvg-tele.com ___ --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] JAMAICA DID'S - 1-876
JAMAICA DID'S - 1-876 NOW ACTIVE ON www.didx.org Cheers, Oliver VermeulenWorld Venture Group Telecom Office: +(40)21-569-4700Office2: +(40)31-860-0030Fax: +(40)31-860-0031USA DID: +1 (305)722-1457BE DID: +(32)9-395-5620UK DID:+(44)870-478-8896msn: [EMAIL PROTECTED]http://www.wvg-tele.com ___ --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] new DID's
Hi List, We have new DID's aviable for the following countrys : - Romania Bucharest 40-21+ - Jamaica 1-876+ Go to www.didx.org Or contact me of the list. Cheers, Oliver VermeulenWorld Venture Group Telecom Corporate Address:Str Avionului Nr 35/bl16J/3Bucharest, 014333 RomaniaOffice: +(40)21-569-4700Office2: +(40)31-860-0030Fax: +(40)31-860-0031USA DID: +1 (305)722-1457BE DID: +(32)9-395-5620UK DID:+(44)870-478-8896SIP : [EMAIL PROTECTED]msn: [EMAIL PROTECTED]http://www.wvg-tele.com ___ --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] SkypeOUT proxy
Dose anybody know how to put skype behind a usa proxy ? Thanks, O ___ --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 Radius Module
Hi List, I'm looking for a Asterisk radius module ... Anybody has one ? Thanks, Oliver Oliver VermeulenWorld Venture Group Telecom Corporate Address:Str Avionului Nr 35/bl16J/3Bucharest, 014333 RomaniaOffice: +(40)21-569-4700Office2: +(40)31-860-0030Fax: +(40)31-860-0031USA DID: +1 (305)722-1457BE DID: +(32)9-395-5620UK DID:+(44)870-478-8896SIP : [EMAIL PROTECTED]msn: [EMAIL PROTECTED]http://www.wvg-tele.com ___ --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] Codec problem from SIP to H323
Try to upgrade asterisk to version 1.2.4 Are you using OH323 or H323 ? I had same problem with 1.2.1 using H323(addon) , Installed 1.2.4 and OH323 and everything worked fine. Cheers, Oliver -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Mejía Evertsz Sent: Thursday, April 20, 2006 12:44 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Codec problem from SIP to H323 Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to transcode (I don't have licences for g729) - sip.conf has disallow=all allow=g729 - h323.conf has disallow=all allow=g729 The problem: [SIPphone] [sip.conf] [h323.conf] [H323gw] g729---allow=g729 ---allow=g729 ---g729 When I dial to the gateway from the SIPphone using g729 as my sip phone's default codec I get: -- Executing Dial(SIP/amejia-8be1, H323/[EMAIL PROTECTED]) in new stack Apr 19 15:02:14 WARNING[68595]: channel.c:2504 ast_request: No translator path exists for channel type H323 (native 4) to 256 Apr 19 15:02:14 NOTICE[68595]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'H323' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) I don't get it why is it trying to translate anything. There's nothing to translate, cause I'm using g729 in both ends. Well, to make it more interesting, I tried this way: [SIPphone] [sip.conf] [h323.conf] [H323gw] g711---allow=all ---allow=all ---g729 This way, it passes the call to the gateway just giving a waring that it can't find a codec to translate. But at least it passes the call. It rings on the other side, and of course as I don't have any g729 licenses installed it drops the call when answered. -- Executing Dial(SIP/amejia-1fc8, H323/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw -- H323/H323gw-2 is making progress passing it to SIP/amejia-1fc8 -- H323/H323gw-2 is making progress passing it to SIP/amejia-1fc8 -- H323/H323gw-2 is ringing -- H323/H323gw-2 answered SIP/amejia-1fc8 Apr 19 15:23:45 WARNING[75484]: channel.c:2685 ast_channel_make_compatible: No path to translate from SIP/amejia-1fc8(4) to H323/H323gw-2(256) Apr 19 15:23:45 WARNING[75484]: app_dial.c:1553 dial_exec_full: Had to drop call because I couldn't make SIP/amejia-1fc8 compatible with H323/H323gw-2 == Spawn extension (test, 444, 1) exited non-zero on 'SIP/amejia-1fc8' Does anybody know how can I get rid of the problem I get on the first scenario? Why does it try to use codec 4 (g711u) if both ends are configured with g729? Please give me some light. I don't know what else to try. Thank you all. Alejandro Mejia ___ --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 ___ --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] DID'S Romania - Bucharest
Dear List, We have Romania Bucharest DIDs available with area code 4021 and 4031 For more information go to www.didx.org Best Regards, Oliver Vermeulen World Venture Group Telecom Tech / Admin Corporate Address: Str Avionului Nr 35/bl16J/3 Bucharest, 014333 Romania Office : +(40) 21-569-4700 Office2 : +(40) 31-860-0030 Fax: +(40) 31-860-0031 USA DID: + 1 (305) 722-1457 BELGIUM DID: +(32) 9 395-5620 UK DID: +(44) 870-478-8896 SIP : [EMAIL PROTECTED] website : http://www.wvg-tele.com ___ --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] DID's Now Offering Romania Bucharest 4021+ and 4031+
Hi All, We are offering Romania Bucharest DIDs NXX : 4021+ and 4031+ We have plenty available on http://www.didx.org/ Regards, Oliver Vermeulen World Venture Group Telecom Tech / Admin Corporate Address: Str Avionului Nr 35/bl16J/3 Bucharest, 014333 Romania Office : +(40) 21-569-4700 Office2 : +(40) 31-860-0030 Fax: +(40) 31-860-0031 USA DID: + 1 (305) 722-1457 BELGIUM DID: +(32) 9 395-5620 UK DID: +(44) 870-478-8896 SIP : [EMAIL PROTECTED] website : http://www.wvg-tele.com ___ --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] Grandstream unit HT-488
Hi All, Anybody knows how to terminated calls using Grandstream Ht488 and the FXO port ? I can ring the FXO port fine , rings 1once then give me dial tone. Thanks, Oliver Vermeulen World Venture Group Telecom Tech / Admin Corporate Address: Str Avionului Nr 35/bl16J/3 Bucharest, 014333 Romania Tel Romania: +(40) 31-860-0030 Fax: +(40) 31-860-0031 USA DID: + 1 (305) 722-1457 DR DID: +1(809) 202-6932 BELGIUM DID: +(32) 9 395-5620 UK DID: +(44) 870-478-8896 SIP : [EMAIL PROTECTED] website : http://www.wvg-tele.com ___ --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] DTMF relay problem
hey all, I'm trying to double forward a DID , from PSTN to SIP DID then to my *. Double forwarding my DID , but the Digits are not getting relayed correctly ... sending double digits and sometimes missing digits? Is here anywhere I can put a delay on DTMF? Or a different solution for this? I use asterisk 1.2.1 , centos 4.2 , codec ulaw Let me know [EMAIL PROTECTED] Oliver World Venture Group Telecom Tech / Admin Oliver Vermeulen USA: tel: +1 (360) 469-0481 DR tel: +1 (809) 412-3457 DR cel: +1 (809) 443-3588 SIP/IAX: [EMAIL PROTECTED] World Venture Telecom Your Connection to VoIP ___ --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] Change Inbound CALL ID Asterisk
Hey everybody, Is here anyway to change the name asterisk on the caller id inbound to the client/sip app? Thanks, Oliver ___ --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] Multi process of *
Hi , Do anybody know how you can run multi proccess of * on a server ? Thanks, O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - Zaptel - DIGIUM x 4 T1
Hi all, Anybody try to configure Asterisk and Digium card on linux-2.6.5-1.358 FEDORA CORE2 ? Making the zaptel getting error: storage size. Thanks, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicepulse error
Hi Guys, I anybody having problems with voicepulse out/in bound call ? On the outbound calls im getting this error : (removed the username) -- Executing Dial(SIP/103-296e, IAX2/[EMAIL PROTECTED]/917707840009) in new stack -- Called [EMAIL PROTECTED]/917707840009 Mar 4 12:51:31 WARNING[131081]: chan_iax2.c:4515 socket_read: Call rejected by 66.234.228.132: No such context/extension -- Hungup 'IAX2[voicepulse-out]/3' And on the inbound call im getting fast busy? Thanks, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
