Im having a look at section 13.1 from SIP Connect v2 doc (see ). It refers to RFC6442 which gives the following example (sorry for its length):INVITE sips:firstname.lastname@example.org SIP/2.0 Via: SIPS/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bK74..
all, I have a strange issue, with a some kind complicate architecture… A router of our internet provider is in front of another bintec rs353j router, at which my freepbx installation is located. However, NAT etc. seems to work fine. BUT: Someth..
Hello. Continuea months-longstrugglethat is associatedwith the transfer from chan_sip to res_pjsip. Where are many gates (GSM gate) that do not supportauthentication whensendingMESSAGE. For example, 4goip when relay incoming SMS. Using chan_sip it ..
, 2 weks ago I asked questions about PJSIP and T.38 but got no replies. I upgraded Asterisk to git as of yesterday (309dd2a), and Im still having the same issues. In the trace below, Im sending a fax from Hylafax server through iaxmodem on Asterisk..
Hello Always returns 401 Unauthorized, because of [Oct 25 11:59:48] NOTICE chan_sip.c: Correct auth, but based on stale nonce received from L6 ;tag=31b9dc9e-684902 L6 is realtime device of type FRIEND (DLINK DVG7022S) Reviewed SIP conversat..
Hallo, any idea whats wrong with that invite?? help would be greatly appreciated! thanks Markus U XX.199.123.185:5060 -> XX.189.169.66:5060 INVITE sip:07111234567@XX.189.169.66 SIP/2.0..Via: SIP/2.0/UDP 192.168.178.26:5060;rport;branch=z9hG4bK98099..Max-Forwar..
Via: SIP/2.0/UDP 10.11.22.161:10000;branch=z9hG4bK-a860600ex0dx0a From: Jian Gao ;tag=7e9c4091bfc704bco0x0dx0a To: Jian Gao x0dx0a Call-ID: email@example.com CSeq: 48998 REGISTERx0dx0a Max-Forwards: 70x0dx0a Contact: Jian Gao ;expires=60x0d..