My name Sohaib. I am new in this forum.
I am having a issue with DID forwarding, when i am forwarding the DID to gmail and call the DID, the receiver answers the call but at other end it keeps ringing, and on asterisk CLI it does not show that the call has been answered, it just shows the called XXXXX@gtalk2voip.com
But when i forward the same DID directly from the providers server to my gmail it works fine.
Please help me out if there is something wrong with my asterisk.
Thanks in Advance.
Please provide your dial plan and a verbose console trace of a failing call.
Hello Thank you For you prompt reply.
exten => 1253xxxxxxx,1,Dial(SIPfirstname.lastname@example.org)
exten => 1253xxxxxxx,2,Hangup
Executing [1253xxxxxxx@a2billing-did:1] Dial(“SIP/13602776421-00000153”, “SIPemail@example.com”) in new stack
– ast_get_srv: SRV lookup for ‘_sip._udp.gtalk2voip.com’ mapped to host core2-ba.gtalk2voip.com, port 5060
– Called firstname.lastname@example.org
the call has been received at the receiving end, but at the dialing end i keep listing the dial tone
% is special in SIP URLs. You need to provide the sip set debug trace to make sure it is getting URL encoded, otherwise you may need to pre-encode it.
Failing to break dial tone is because either you haven’t had a response from the remote end or possibly because nothing is getting back to the phone. That might hint at a routing or NAT problem.
It may vary between phones as to whether or not 100 Trying breaks dial tone. Remember that dial tone is faked by the phone, for SIP phones.
Deleted. I misread another thread as suggesting Asterisk does URL encode, but it actually looks like it provides evidence that it doesn’t URL decode.
Sorry for the delay reply, actually i didn’t have internet access for last couple of days.
I thought when forwarding to gtalk we should use % sign, but after removing the that sign and i forwarded the DID directly like email@example.com, it still has same problem…
The URI encoding of % is %25 I would try that.
you mean i should forward it in this way
Yes. That is what should go over the wire. Asterisk may not be doing the encoding.
thanks for your quick response.
Now after doing this exten => XXXXXXXXX,1,Dial(SIPfirstname.lastname@example.org) call is even not coming to my gmail
how can i set sip debugging on incoming calls or on any DID?