Instead of Transfer to <a href="http://sphone.vopr.vonage.net">sphone.vopr.vonage.net</a>, try Transfer to one of your Vonage peers ie.<br><br>exten => 5,1,Transfer(SIP/18005551212@vonageTrunk)<br><br><br><div><span class="gmail_quote">
On 1/10/06, <b class="gmail_sendername">Lea</b> <<a href="mailto:lea123@wp.pl">lea123@wp.pl</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Olle,<br><br>Thank you very much for replying !<br><br>The info that the transfer() application works differently from<br>the Dial( ,t) application is not anywhere in the documentation.<br>Please consider adding it for the poor unaware users.
<br><br>Anyway, I tried to transfer an incoming SIP call (<br>after it has been answered) using the following dialplan:<br>exten => 5,1,Transfer(<a href="mailto:SIP/18005551212@sphone.vopr.vonage.net">SIP/18005551212@sphone.vopr.vonage.net
</a>)<br>using the same channel as for the incoming call.<br><br>...and the Asterisk SIP debug output showed a REFER request<br>being generated (see its listing below). Great !<br><br>Unfortunately this REFER request was not accepted by the Vonage
<br>server and later Asterisk retransmitted it several times after<br>which Asterisk gave up. Thus the TRANSFER FAILED :(<br><br>IMPORTANT: When I tried to do an identical transfer of incoming<br>call using an X-Pro Softphone, I immediately received a "202
<br>Accepted" response, thus the TRANSFER SUCCEEDED :)<br><br>The REFER request generated by the X-Pro Sofphone is very<br>similar to Asterisk's (see the listings below).<br>The differences are:<br>- the host address after the REFER keyword is different
<br>- the Refer-To header has an additional ;method=INVITE<br>- the Proxy Authorization header has a different uri=<br><br>Can you shed some light on the logic used by Asterisk when<br>transfering calls using the REFER method ?
<br>Do you understand why X-Pro Sofphone works, but Asterisk does<br>not work ?<br><br>Regards,<br>Daniel Leeds<br><br><br>Debug output of Asterisk:<br>-------------------------------------------------<br>Reliably Transmitting (no NAT) to
<a href="http://sphone.vopr.vonage.net:5061">sphone.vopr.vonage.net:5061</a>:<br>REFER sip:<a href="http://transit.vonage.net:5060">transit.vonage.net:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP asterisk.aa.eu:5060;branch=z9hG4bK35d81ab7;rport
<br>Route: <sip:pstn_7748712@<br><a href="http://inbound4.vonage.net:5060">inbound4.vonage.net:5060</a>>,<sip:<a href="http://transit.vonage.net:5060">transit.vonage.net:5060</a>><br>From: <<a href="mailto:sip:pstn_7748712@inbound4.vonage.net">
sip:pstn_7748712@inbound4.vonage.net</a>>;tag=as2dd427cd<br>To: "us_pstn_caller" <sip:<a href="http://transit.vonage.net">transit.vonage.net</a>>;tag=1180845188<br>Contact: <<a href="mailto:sip:pstn_7748712@asterisk.aa.eu">
sip:pstn_7748712@asterisk.aa.eu</a>><br>Call-ID: <a href="mailto:33D60779-816711DA-BA1CDF96-A870513F@transit.vonage.net">33D60779-816711DA-BA1CDF96-A870513F@transit.vonage.net</a><br>CSeq: 103 REFER<br>User-Agent: Asterisk
<br>Max-Forwards: 70<br>Proxy-Authorization: Digest username="pstn_7748712",<br>realm="<a href="http://sphone.vopr.vonage.net">sphone.vopr.vonage.net</a>", algorithm=MD5,<br>uri="sip:<a href="http://sphone.vopr.vonage.net">
sphone.vopr.vonage.net</a>", nonce="1384655670",<br>response="b29c87eeb7ec31847c4348b2e0b3bcbf", opaque=""<br>Date: Tue, 10 Jan 2006 23:25:30 GMT<br>Refer-To: <<a href="mailto:sip:18005551212@sphone.vopr.vonage.net">
sip:18005551212@sphone.vopr.vonage.net</a>><br>Referred-By: <<a href="mailto:sip:pstn_7748712@asterisk.aa.eu">sip:pstn_7748712@asterisk.aa.eu</a>><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,<br>
NOTIFY<br>Content-Length: 0<br><br>Debug output of X-Pro Softphone:<br>-------------------------------------------------<br>Transmitting to <a href="http://sphone.vopr.vonage.net:5061">sphone.vopr.vonage.net:5061</a><br>REFER
<a href="http://sip:pstn_7748712@sphone.vopr.vonage.net:5061">sip:pstn_7748712@sphone.vopr.vonage.net:5061</a> SIP/2.0<br>Via: SIP/2.0/UDP<br>asterisk.aa.eu:5060;rport;branch=z9hG4bK1CFB3639BC0147BE98519AA14<br>B1FE8A1<br>
Route:<br><<a href="http://sip:pstn_7748712@inbound4.vonage.net:5060">sip:pstn_7748712@inbound4.vonage.net:5060</a>>,<sip:transit.vonage.c<br>om:5060><br>From: <<a href="mailto:sip:pstn_7748712@inbound4.vonage.net">
sip:pstn_7748712@inbound4.vonage.net</a>>;tag=3655015093<br>To: "us_pstn_caller" <sip:<a href="http://transit.vonage.com">transit.vonage.com</a>>;tag=2112301987<br>Contact: <sip:pstn_7748712@asterisk.aa.eu:5060>
<br>Call-ID: <a href="mailto:CF6FC3F9-800111DA-999DDF96-A870513F@transit.vonage.com">CF6FC3F9-800111DA-999DDF96-A870513F@transit.vonage.com</a><br>CSeq: 7237 REFER<br>User-Agent: X-PRO release 1105x<br>Max-Forwards: 70<br>
Proxy-Authorization: Digest<br>username="pstn_7748712",realm="<a href="http://sphone.vopr.vonage.net">sphone.vopr.vonage.net</a>",nonce="529<br>605995",response="ff4eadf3e625217233fdbff90fd67ec5",uri="sip:pstn
<br>_7748712@<a href="http://sphone.vopr.vonage.net:5061">sphone.vopr.vonage.net:5061</a>",algorithm=MD5<br>Refer-To: <<a href="mailto:sip:18005551212@sphone.vopr.vonage.net">sip:18005551212@sphone.vopr.vonage.net
</a>;method=INVITE><br>Referred-By: Vonage User<br><<a href="mailto:sip:pstn_7748712@sphone.vopr.vonage.net">sip:pstn_7748712@sphone.vopr.vonage.net</a>><br>Content-Length: 0<br><br><br>On 10-01-2006,. 16:36 Olle E Johansson wrote:
<br>> We do support REFER/NOTIFY.<br>> I am working (and have been for a long time) on improving the<br>support, but the basic support for transfers with REFER is there.<br><br>In another topic Olle E Johansson wrote:
<br>>We only issue an outbound REFER by you using the transfer()<br>application in the dial plan.<br><br>> Daniel Leeds wrote:<br>> > I noticed that in the file "chan_sip.c" there is a constant<br>> > SIP_REFER.
<br>> ><br>> > Does this mean that Asterisk can initiate SIP-SIP transfers<br>with<br>> > the REFER / NOTIFY methods according to RFC-3515 ?<br>> ><br>> > If not, why not ? ...after all RFC-3515 is almost 3 years
<br>old<br>> > and I thought Asterisk is on the cutting edge of VoIP...<br>><br>> ><br>> > Anyway, if somebody has some code to make Asterisk compliant<br>> > with this RFC-3515 or some hidden #defines or maybe compiler
<br>> > options to enable this crucial transfer functionality,<br>please<br>> > scribble something back.<br>><br>> /O<br>> _______________________________________________<br>> --Bandwidth and Colocation provided by
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