search for: hariga

Hi, VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll Take a loock on http://www.voiceglo.com/ The softphone is IAX :) Best regards, Chris HARIGA Techselesta Inc. http://www.techselesta.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/307c10e9/attachment.htm

...ind to context = internal ; Default for incoming calls tos=lowdelay disallow=all allow=ulaw allow=gsm allow=alaw register => 12047440600:1234@213.137.73.178/toti [iconnect] type=friend port=5060 username=12345678 secret=1234 host=213.137.73.178 dtmfmode=inband callerid="Chris Hariga"<2407440600> - extensions.conf - [general] static=yes writeprotect=yes ignorepat => 9 [globals] MYPHONENUMBER=12407440600 MYNAME=Chris HARIGA [incoming] exten => s,1,Answer() exten => s,1,Wait(0) exten => s,2,Dial(SIP/jim&SIP/jimoffice&SIP/sean&SIP/seanhome&am...

Hi, Can U tell me the Vonage ATA 186 settings? I would like to try to have a web interface on my adapter :-)) Best regards, Chris Hariga

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 2280 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20031012/70396f74/smime.bin

Hi, si posible on SIP phones to have the dial tone after 9 like on the FXS card? I set ignorepat => 9 on my extensions.conf... Best regards, Chris HARIGA

...dy can access your DB as user `localhost' from host `localhost' : WITHOUT supplying a password. : Be very careful about it!! BEWARE: Accessing the db as an anonymous user. : Your username has no relevance Any suggestions are welcome. Best regards, Chris HARIGA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040820/c5d7b8df/attachment.htm

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3179 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041007/4b01c29c/smime.bin

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3179 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041007/98aec720/smime.bin

Hello, I have several * servers behind a SER server (in a local ip range). The SER server is also publicy reachable. On the other site, I have SIP clients that are behind another NAT or in the same NAT range as the * server. Can someone give me some directions/hints etc. on how to make this work. I think I should be using MediaProxy with SER. But do the SIP clients need to register at the SER

Hi all I have been googling and wiki-ing and have found a number of potential solutions to my questions, but I don't want to have to play about for too long and risk messing up my * box now I've just got it working, if one of you kind folk could offer your 2 penneth, (being a Brit I'll have none of this cents business ;] ). I want to send an SMS message whenever I get a voicemail

No, you don't need a sound card. Do you have ztdummy loaded or zaptel device in your system? Regards, Gus ----- Original Message ----- From: "Chris Hariga" <contact@techselesta.com> To: <asterisk-users@lists.digium.com> Sent: Sunday, October 19, 2003 8:19 PM Subject: [Asterisk-Users] Music on hold... > Hi, > > I need a sound card and mpg123 for music on hold??? When I call Digium > the guys toll me "is not necessa...

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

Help: Checkout as of 17:00 UCT Does anyone know if: chan_zap.c: In function `zt_train_ec': chan_zap.c:1081: `ZT_ECHOTRAIN' undeclared (first use in this function) is expected at the moment? Dave Kitchen

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3097 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040712/2a1bda93/smime.bin

Aloha, Did anyone ever get the formating manual for the XML brwoser on the Polycom IP600? Does anyone have a sample? Aloha, Matt

Haven't seen email since the 29th.. just testing. -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 288 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050801/96555713/mhess.vcf

Skipped content of type multipart/alternative-------------- next part -------------- _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

...192.168.11.10> Call-ID: 299a873b30ad20f90bbcb66e3d505e68@192.168.10.199 CSeq: 102 OPTIONS Content-Length: 0" Question: How I can setup asterisk to get the sip call without authentication? I check on voip-info.org but I didn't find a sip.conf sample :-( Best regards, Chris HARIGA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/d9719f07/attachment.htm

Hello there, I am trying to get Asterisk set up by using the book Asterisk: The Future of Telephony. I am on Chapter 4. I have have set up Zaptel and zapata.conf and also set up extensions.conf and when I run "asterisk -r" at the Gnome-terminal to connect with Asterisk I get the following message: Unable to connect with remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It

Hi! I am doing a research about the prices of SIP telephones. If someone can tell me which one are the cheapest and have an acceptable quality... it will be very kind. Best Regards, Mireia