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
