similar to: PJSIP incompatibility

with wireshark i need decrypt traffic every call which is time consuming. get debug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo m=audio 52421 RTP/SAVPF 8 0 101 c=IN

Asterisk is on public IP (as described in the first email) i have 10 years experience in voip, 4 years webrtc in production. i know about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism but i confess. i dont understand WHY Asterisk SOMETIMES switches destination IP in RTP. this is not only about ICE. its about RTP engine too which is Asterisk specific and Asterisk DEBUG is

On Sun, Dec 21, 2014 at 4:54 AM, Recursive <lists at binarus.de> wrote: > Dear list, > > I am currently trying to send faxes via T.38 using PJSIP (newest version 2.3) with Asterisk 13.0.2. After having configured PJSIP, I have seen several things the cause of which I would like to know. > > 1) Ports and IP addresses which PJSIP bind to > > I have configured one transport

Hello, I have an issue with Asterisk 12 PJSIP. When receving an INVITE with FROM URI that has a port number, the Asterisk removes the port from URI on consecutive Responses / Requests. This causes an issue with one of our SIP servers (it doesn't recognize the response / request). Below you can see an incoming INVITE and the outgoing 200OK response. I have highlighted the issue in Yellow. Does

Trying this again after my first away from work in a couple weeks. Running Asterisk 13.0.0 IP authentication with Vitelity I can Originate with sip, but not pjsip. Here is the sip settings and trace. Action: Originate ActionID: S8 Channel: SIP/8005555555 at outbound.vitelity.net Exten: createcall Context: TestApp Priority: 1 Timeout: 60000 CallerID: John Doe <1234> Variable:

another issues with cisco 7975 I have phone registered on asterisk have 2 different issues on different versions of firmware, on 9-4-2-1S I have not working 3way conference, when I trying to connect second call, phone says ?unable to set up conference? and sending some cisco xml data to asterisk which cannot be handled, thats the problem, I know on firmware 8-5-4 3way conference works just

I am running PJPROJECT 2.3 and Asterisk 13.0.0. I answer the call, about 15 seconds later, vitality hangs up on my cell phone. However, Asterisk is never notified When the OK (for the answer) occurs, the ACK seems to never be accepted. The OK recvd with ACK sent occurs several times. Here are the pjsip.conf settings... [global] type = global debug = yes [transport1] type = transport bind =

I place a call into Asterisk (from SIP phone) and the To header does not have a tag. Asterisk then sends it's Trying response, still no tag in the To header. The phone then replies with OK, this time the To header includes a tag. Is there any way to retrieve this response To header (including the tag field) from the dial plan? I have tried the PJSIP-HEADER read of the To header, but it

Hello, I have installed the latest version 12 that has been released (12.1.0.rc3). I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint that doesn't support it (no telephony event in the rtpmap) the asterisk responds OK in the signalling but DTMF is not working. Is it a known issue? Below you can see the output of the asterisk monitor. <--- Received SIP request

I am not sure what you mean by the ful SIP signaling? Here is the trace for the sip.conf which works successfully. Below that, I will include the trace for the pjsip.conf which it seems Vitelity isn't accepting the ACK in response to the OK ---- SIP --- <--- Transmitting SIP request (1004 bytes) to UDP:64.2.142.93:5060 ---> INVITE sip:8005555555 at 64.2.142.93 SIP/2.0 Via: SIP/2.0/UDP

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, 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 I'm still having the same issues. In the trace below, I'm sending a fax from Hylafax server through iaxmodem on Asterisk-13 (tiare) to a second Asterisk (11.18.0 t0gw) connected to the PSTN via ISDN; the

George, I have the detailed log below. (Resending after trimming the email to 40KB.) The sequence below just repeats ad-nauseam. Is this a SIP trunk issue? Thanks! --------------------- Transmitting SIP request (885 bytes) to UDP:65.254.44.194:5060 ---> INVITE sip:12025551212 at 65.254.44.194:5060 SIP/2.0 Via: SIP/2.0/UDP 18.18.19.123:5060

On 06/14/2017 at 10:17 PM, Joshua Colp wrote: > On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote: > > <snip> > >> >> I can now say, that asterisk / pjsip seams to work *mostly* as expected. >> Just one exception - and that's the package in question, which can't be >> seen in tcpdump. >> >> I extended the above patch by adding

OK, it stopped working. It turns out the transport and endpoints in PJSIP are ok. I can send an invite from my unregistered snom phone and I can see some activity in the CLI. However, when I dial from my snom to Kamailio and have it pass the message to asterisk, PJSIP seems to ignore the sip messages even though they are there. Is there something wrong in the invite that I'm missing? U

I'm trying to set up pjsip to work with an obi202 and google voice. But I can't configure the endpoint. pjsip: [obi202-auth](!) type = auth auth_type = userpass password = <mypass> [obi202-aor](!) type = aor max_contacts = 2 ; ===== endpoints ======== [gv-voice](obi202-endpoint) auth = gv-voice aors = gv-voice identify_by=auth_username ;identify_by=username ; I also tried

Dear list, I am currently trying to send faxes via T.38 using PJSIP (newest version 2.3) with Asterisk 13.0.2. After having configured PJSIP, I have seen several things the cause of which I would like to know. 1) Ports and IP addresses which PJSIP bind to I have configured one transport like that: [tr_wZCMk5MvC2ATNzAr] type = transport protocol = udp bind = 192.168.20.48 Nevertheless, PJSIP

Hello! unchanged asterisk crashes during udptl / t.38 negotiation with telekom - they do not support t.38 / udptl. In detail: fax client -> asterisk -> telekom -> easybell -> asterisk -> fax server Fax server sends t.38 reinvite via asterisk to easybell. Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 2447581897 4 IN IP4 46.17.15.23

Ok after I added tcp transport and disable force_rport phone get registered, but still have issues with calls, when I call from cisco from, it work except hangup. when I call to cisco phone asterisk return congested debug of call <--- Transmitting SIP request (952 bytes) to TCP:192.168.1.61:51179 ---> INVITE sip:111 at 192.168.1.61:51179;transport=tcp SIP/2.0 Via: SIP/2.0/TCP

??? I am having a really bad day trying to get incoming calls to work on Asterisk 13 with PJSIP.? We just migrated from Asterisk 1.8 where everything was working but there seems that something got lost in translation.? No matter what I try I always get a 401 Unauthorized message when receiving a call from the PSTN provider.? I can make calls and the registration is working.? I have tried to

On Sun, Mar 15, 2015 at 5:56 PM, Sonny Rajagopalan <sonny.rajagopalan at gmail.com> wrote: > George, > > I have the detailed log below. (Resending after trimming the email to 40KB.) > > The sequence below just repeats ad-nauseam. Is this a SIP trunk issue? > > Thanks! > I don't see anything obvious. The registration works though, right? You might want to compare