Re: [OpenSIPS-Users] Asterisk Unrecognized sip header

2016-04-01 Thread Schneur Rosenberg
shov > Sent: Thursday, March 31, 2016 11:51 AM > To: users@lists.opensips.org > Subject: Re: [OpenSIPS-Users] Asterisk Unrecognized sip header > > On 03/31/2016 02:49 PM, Travis Manson-Drake wrote: > > > I would be more than happy to send it to you privately if that's ok? > >

Re: [OpenSIPS-Users] Asterisk Unrecognized sip header

2016-03-31 Thread Travis Manson-Drake
, 2016 11:51 AM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Asterisk Unrecognized sip header On 03/31/2016 02:49 PM, Travis Manson-Drake wrote: > I would be more than happy to send it to you privately if that's ok? Of course! -- Alex Balashov | Principal | Evariste Systems

Re: [OpenSIPS-Users] Asterisk Unrecognized sip header

2016-03-31 Thread Alex Balashov
On 03/31/2016 02:49 PM, Travis Manson-Drake wrote: I would be more than happy to send it to you privately if that's ok? Of course! -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) /

Re: [OpenSIPS-Users] Asterisk Unrecognized sip header

2016-03-31 Thread Travis Manson-Drake
, 2016 10:50 AM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Asterisk Unrecognized sip header Well, can we see the requests being sent to Asterisk? :) On 03/31/2016 01:48 PM, Travis Manson-Drake wrote: > Hello everyone. > > Hope your all doing well! > > I seem to be

Re: [OpenSIPS-Users] Asterisk Unrecognized sip header

2016-03-31 Thread Alex Balashov
Well, can we see the requests being sent to Asterisk? :) On 03/31/2016 01:48 PM, Travis Manson-Drake wrote: Hello everyone. Hope your all doing well! I seem to be having an issue in which when a call is sent through OpenSIPS to my Asterisk PBX asterisk with eventually send a BYE with a hang

[OpenSIPS-Users] Asterisk Unrecognized sip header

2016-03-31 Thread Travis Manson-Drake
Hello everyone. Hope your all doing well! I seem to be having an issue in which when a call is sent through OpenSIPS to my Asterisk PBX asterisk with eventually send a BYE with a hang up Cause of 111/unrecognized sip header. I looked at the headers of all my packets but can't find anything