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Hello! Im facing a problem concerning MWI. The problem is: The phone switches off the MWI exactly at the moment the second NOTIFY package for one voice mail arrives. The phone switches off MWI independently if the voice mail has been acknowledged..
I need some help understanding SIP dialog. Some actor is trying to access my server, but I cant figure out what hes trying to do ,or how.Im getting a lot of these warnings.[May 17 10:08:08] WARNING[1532]: chan_sip.c:4068 retrans_pkt: Retransmission time..
–C0QFpPkb1GnU2Jkez22QMuv3Jxua41OsHContent-Type: text/plain; charset=utf-8Content-Transfer-Encoding: quoted-printable Content-Language: en-USall, im getting this error:[Feb 21 09:29:09] ERROR[1250]: pjproject:0 : sip_transport.c Er..
Im having a look at section 13.1 from SIP Connect v2 doc (see [1]). It refers to RFC6442 which gives the following example (sorry for its length):INVITE sips:bob@biloxi.example.com 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..