Hi virendra, Dialed same command.. I got below output
ast18*CLI> originate sip/test02 application dial == Using SIP RTP CoS mark 5 [Jan 4 14:13:07] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to '"Anonymous" <sip:[email protected]:192>;tag=as417a5527' Best Regards, *Jayesh Labade* e-mail: [email protected] On Wed, Jan 4, 2012 at 4:35 PM, virendra bhati <[email protected]> wrote: > Hi checked your debug like. > > Did you check that your SIP device ir registered with server ? > if yes then dial below command from CLI > > *originate sip/test02 application dial* > > > > > On Wed, Jan 4, 2012 at 3:33 PM, Jayesh Labade <[email protected]>wrote: > >> 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 >> > > > > -- > > 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 >
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