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...How can I make Asterisk do the SIP-SIP transfer with the REFER / NOTIFY methods according to RFC-3515 ? Below is X-Pro Softphone's debug output of a successful transfer according to RFC-3515 : -------------------------------------------------------- SEND TIME: 26251796 SEND >> sphone.vopr.vonage.net:5061 REGISTER sip:sphone.vopr.vonage.net SIP/2.0 Via: SIP/2.0/UDP aster.autoattendant.eu:5060;rport;branch=z9hG4bKAD14EC1C2A2542DF877A3DF9D1358091 From: Vonage User <sip:pstn_6523567@sphone.vopr.vonage.net>;tag=1997595095 To: Vonage User <sip:pstn_6523567@sphone.vopr.vonage.net&...

...I know that Vonage does a lot of nasty stuff which impacts UA's but Xten Eyebeam handles it correctly at least. I have tried pedantic=yes but no difference. Here is the sip.conf and the BYE dialog with numbers replaced: [vonage] type=friend secret=<pass> username=<no> host=sphone.vopr.vonage.net <http://sphone.vopr.vonage.net> dtmfmode=rfc2833 port=5061 fromuser=<no> fromdomain=sphone.vopr.vonage.net <http://sphone.vopr.vonage.net> canreinvite=no context=context insecure=very BYE sip:(PSTN Number)@216.115.20.171:5060 <http://216.115.20.171:5060>SIP/2.0...

...; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw allow=gsm allow=g729 allow=g723 context=from-sip-external ; Send unknown SIP callers to this context callerid=Unknown register=1XXXXXXXXXX:PASSWORD@sphone.vopr.vonage.net:5061 [vonage] username=1XXXXXXXXXX type=friend secret=PASSWORD port=5061 nat=yes host=sphone.vopr.vonage.net fromuser=1XXXXXXXXXX fromdomain=sphone.vopr.vonage.net dtmfmode=rfc2833 auth=md5 [vonage_inbound] username=1XXXXXXXXXX type=friend secret=PASSWORD port=5061 nat...

Hi Nitesh, check out my config that I have for the Faktortel config in the asterisk@home sourceforge forum, you'll probably be able to work out how to set it up from there. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Wednesday, February 23, 2005 4:12 PM To:

...p.conf; your mileage may vary: [general] dtmfmode=inband port=5060 bindaddr=<enter yours> context=incoming disallow=all allow=ulaw externip = <enter yours> localnet = <enter yours> localmask = 255.255.255.0 nat=yes register => .. ... register => 16125551212:password@sphone.vopr.vonage.net:5061/99612 [sip99612] secret=password username=16125551212 host=sphone.vopr.vonage.net port=5061 type=peer nat=yes canreinvite=no dtmfmode=rfc2833 fromuser=16125551212 context=incoming fromdomain=sphone.vopr.vonage.net For what it's worth, my initial "host=sphone.vopr.vonage....

I've yet to successfully register and receive calls from my Vonage softphone. I've tried what few examples are given in former post this list and some other forums, but nobody seems to be stepping forward saying it works recently. Either they broke something to where you simply cannot use *, or the config examples need updated. If anyone can show a still working config,

...;diax' (AUTHENTICATED) at 59.93.73.0:4569 -- Registered 'diax' (AUTHENTICATED) at 59.93.73.0:4569 -- Accepting AUTHENTICATED call from 59.93.73.0, requested format = 2, actual format = 2 -- Executing Dial("IAX2/diax@manoj/5", "IAX2/561514445:_54rge86541@sphone.vopr.vonage.net/") in new stack -- Called 56165651:_4516rgf@sphone.vopr.vonage.net/ Jan 18 04:37:43 WARNING[30192]: chan_iax2.c:1477 attempt_transmit: Max retries exceeded to host 216.115.20.41 on IAX2/216.115.20.41:4569/7 (type = 6, subclass = 1, ts=7, seqno=0) -- Hungup 'IAX2/216.115....

...fixing my problem and configure Asterisk. The error I get after I make a call to outside number like 18007633555 is -- Accepting AUTHENTICATED call from 59.93.69.218, requested format = 1024, actual format = 1024 -- Executing Dial("IAX2/diax@diax/2", "SIP/18006733555@sphone.vopr.vonage.net:5061|30|r") in new stack -- Called 18006733555@sphone.vopr.vonage.net:5061 == No one is available to answer at this time Jan 19 06:26:36 NOTICE[29115]: chan_sip.c:4045 sip_reg_timeout: -- Jan 19 06:26:38 WARNING[29115]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 't...

...but I have not been able to get them to work. My Asterisk server is behind a NAT firewall. The files below are the closest I have gotten to complete connectivity. Any feedback is appreciated! One note: I was able to receive calls only if I used atlas-east as the Vonage server. If I used sphone.vopr.vonage.net, no calls came in. sip.conf: [general] externip=X.X.X.X port=5060 bindaddr=X.X.X.X context=vonage-out disallow=all allow=ulaw allow=alaw nat=yes register=####:PASSWORD@atlas-east.vonage.net:5060/201 [vonage] username=#### type=peer secret=PASSWORD port=5060 nat=yes host=atlas-east.vo...

I know Vonage doesn't officially have a "bring your own device" type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060329/5bc9f644/attachment.htm

atlast after working of 7 hours i got voange soft account working on asterisk.

I am new to the mailing list, but I am very interested in running my small home business office phone system using Asterisk. However, Broadvoice, a VoIP provider of choice based on my research, is not available in my area. I currently use Vonage VoIP. Their website mentions nothing about being able to link to Asterisk. I was wondering if any US subscribers have been able to configure Vonage

>When I call a SIP user, the phone should ring in more than one >extentions. Also more than one phone should be able to register with >asterisk. Right now it is not the case. There is no issue here. You seem to be confused, that's all. A SIP account is a SIP account and an extension is an extension. You can assign an extension to an account (or to multiple accounts) and the tool for