Jayesh Labade
2012-Jan-04 09:37 UTC
[asterisk-users] Failed to authenticate on INVITE to Anonymous
Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.labade at gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade <jayesh.labade at gmail.com>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:test02 at anonymous.invalid > >;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: jayesh.labade at gmail.com > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120104/c2097393/attachment.htm>
virendra bhati
2012-Jan-04 09:46 UTC
[asterisk-users] Failed to authenticate on INVITE to Anonymous
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 <jayesh.labade at gmail.com>wrote:> Please help me.. > > Best Regards, > *Jayesh Labade* > e-mail: jayesh.labade at gmail.com > > > > On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade <jayesh.labade at gmail.com>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:test02 at anonymous.invalid >> >;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: jayesh.labade at gmail.com >> >> > > -- > _____________________________________________________________________ > -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120104/2c65db90/attachment.htm>
sean darcy
2012-Jan-04 19:50 UTC
[asterisk-users] Failed to authenticate on INVITE to Anonymous
On 1/4/2012 4:37 AM, Jayesh Labade wrote:> Please help me.. > > Best Regards, > *Jayesh Labade* > e-mail: jayesh.labade at gmail.com <mailto:jayesh.labade at gmail.com> > > > > On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade <jayesh.labade at gmail.com > <mailto:jayesh.labade at gmail.com>> 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:test02 at anonymous.invalid>;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: jayesh.labade at gmail.com <mailto:jayesh.labade at gmail.com> > > > > > -- > _____________________________________________________________________ > -- 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-usersTry: register => test02:test02 at 192.168.1.55/s sean