search for: sip01

Any got experience with these? I couldn't fint anything in any postings... it seems they have a h.323 on voip01.go2call.com and a sip on sip01.go2call.com I have tried to register with some of the same as I use for nikotel, but Asterisk does not want to register. I've tried to use both the user name (ingvald) and the PIN code 440.... as authentication. ---from sip.conf---- register => 440686267684:xxxxx@sip01.go2call.com [g...

Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at

...40 for this. Session target x.x.x.x But calls will always get routed to the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried. My situation: PSTN -> CISCO -> ASTERISK OK ASTERISK -> CISCO -> PSTN OK ASTERISK -> CISCO -> VOIP NOT OK (only needs outbound calls) SIP01#sh dial-peer voice summary dial-peer hunt 0 TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT 10 pots up up 0 down 1/0/0 20 pots up up 0 down 1/0/1 30 voip up up 2012345.....

...isk CVS-08/02/04 my intention is, that if the extension is not available in Sip channel, I will send the call somewhere else my extensions file contains the following: exten => _[123]XX,1,ChanIsAvail(sip/${EXTEN}) exten => _[123]XX,2,dial(sip/${EXTEN},30) exten => _[123]XX,102,Dial(IAX2/sip01-xxxxxx:xxxxxxx@xxxxxxxxxxxx/${EXTEN}) if I understand ChanIsAvail correctly this should give med following: if i dial extension 111, and that is a local extension, it dials the sip channel on the other hand, if extension 111 is not avaliable in the local sip channel, it dials on IAX2 But it doe...

...t: Asterisk SIPv2 (http://www.asterisk.org CVS-HEAD-11/27/04-15:07:27 ) ..Content-Length: 0.... # U myhost:5060 -> mysipprovider:5060 INVITE sip:callednumber@mysipprovider SIP/2.0..Via: SIP/2.0/UDP 212.33.133.1 95:5060;branch=z9hG4bK251f9e48..From: "4721677891" <sip:4721677891@sip01.ph onect.no>;tag=as11f66db2..To: <sip:callednumber@mysipprovider>..Contact: <si p:4721677891@myhost>..Call-ID: 5ae4b68378e5edf8081a79253ddf8aef@sip 01.phonect.no..CSeq: 103 INVITE..User-Agent: Asterisk SIPv2 (http://www.ast erisk.org CVS-HEAD-11/27/04-15:07:27 ) ..Authorizati...

I have several 7960 phones with SIP image (7.3) and Asterisk 1.0.1 on FreeBSD. When I have 2 active SIP calls on the 7960 phone there are no available lines for additional calls. I tried to configure 2 lines to the same SIP server but it's still limited to 2 calls. How to utilize all lines? -- Called user -- SIP/user-acc6 is ringing -- SIP/user-acc6 answered SIP/x.x.x.x-09a9a000 --

...using Mis-typed, yes. The SIP server is the Asterisk server and is 192.168.1.10 > (hardware/software, brand, version)? Grandstream BudgeTone-100 - can dial 1-800-CALL-ATT and talk with an operator through the sipphone.com SIP proxy, quality is adequate (changed the SIP server to sip01.sipphone.com of course) - when the SIP server is Asterisk, can be dialed from ISDN without any problem (maybe a slight delay), quality is good both directions. - can dial to Asterisk, in that case Asterisk's debug shows the call, but fails. Nothing is hearable on the Budget...

Hello good ppl, Any way of stopping asterisk writing into syslogs or any other file, if I set verbose 6 on the console? All I want is the verbose output only on the console, nowhere else. My logger.conf says : console=> notice,error ;messages => notice,warning,error Thanks in advance. - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential