Hi, I am using asterisk ver 1.8.8.1.
My SIP trunk conf details are below.. [general] context=default ; Default context for incoming calls realm=192.168.1.55 allowguest=yes realmauth=yes send_rpid=pai register => test02:[email protected] [test02] type=peer nat=no canreinvite=no host=192.168.1.55 ;[email protected] context=incoming secret=test02 permit=192.168.1.0/255.255.255.0 username=test02 fromuser=test02 fromdomain=192.168.1.55 defaultuser=test02 insecure=invite,port outboundproxy=192.168.1.55 promiscredir=yes userphone=yes For more details you can find my paste in pastebin.. Links given below. While Dialing call fro Xlite send following Sip header F= sip:[email protected]. And if tried to register same account in asterisk trunk i got F=sip:[email protected] in sip header. I dont know why asterisk sends anonymous.invalid instead of domain name..Help me Best Regards, *Jayesh Labade* e-mail: [email protected] On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati <[email protected]> wrote: > Hi, > > Give the complete details about the asterisk version, and SIP trunk conf > details > > > On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade <[email protected]>wrote: > >> Please help me.. >> >> Best Regards, >> *Jayesh Labade* >> e-mail: [email protected] >> >> >> >> On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade >> <[email protected]>wrote: >> >>> Hello Experts, >>> >>> I have pasted my issue in http://pastebin.com/zBGVmdcY >>> >>> I Cant able to Originate call from SIp trunk..I got this [Jan 3 >>> 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to >>> authenticate on INVITE to '"Anonymous" <sip:[email protected] >>> >;tag=as57d3a806' >>> i am unable to make outbound call from this trunk. while if i registered >>> this trunk in softphone like Xlite, there is no problem with outbound >>> calls. Help me. >>> >>> please find sip.conf file in http://pastebin.com/zBGVmdcY >>> >>> I have pasted sip debug with verbosity of failed call >>> http://pastebin.com/jL2ki0s8 >>> >>> >>> Best Regards, >>> *Jayesh Labade* >>> e-mail: [email protected] >>> >>> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > > Thanks and regards > > Virendra Bhati > +91-8885268942 > Software Engineer > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
