search for: sdp

Hello list when trying to set up webRTC communications with sipjs client package (tried 0.7.0, 0.10.0 and 0.19.0), I see in the asterisk debug log-file the following : DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 99.88.77.66... OK. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:9 IN IP4 0.0.0.0... UNSUPPORTED OR FAILED. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=candidate:4275385644 1 udp 2122260223 192.168.0.18 57987 typ host gener...

...tput of "tcpdump ip port 5060" on the Asterisk server. Asterisk is 10.0.1.39 and Pingtel is 172.16.9.39. User ID on the Pingtel is 2219. My sip.conf looks like this: [foo4] host=172.16.9.39 type=friend context=foo canreinvite=no insecure=1 1046.210483 172.16.9.39 -> 10.0.1.39 SIP/SDP Request: INVITE sip:17035551212@10.0.1.39:5060, with session description 1046.220483 10.0.1.39 -> 172.16.9.39 SIP Status: 100 Trying 1046.230483 10.0.1.39 -> 172.16.9.39 SIP/SDP Status: 183 Session Progress, with session description 1047.810453 10.0.1.39 -> 172.16.9.39 SIP/SDP Status: 200...

Hello, I think I met a case similar to the one solved by [1] . Quoting this case : * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk sends 200 OK with SD...

...not the res_fax and res_fax_digium that comes with FFA. What happens is sometimes the T.38 negotiation goes well and others it fails completely. That's what I got from the debug info on two different calls, without changing any configs: [Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP o=PVG 1265107050040 1265107050040 IN IP4 10.152.0.164... UNSUPPORTED. [Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP...

Hello! An extension registered at asterisk 13.23 initiates an external call (pjsip). After the Invite, the callee (-> ISP) sends a 100 Trying 183 Session Progress (*without* SDP) Asterisk now sends to the extension: 183 Session Progress (*with* SDP) 183 Session Progress (*with* SDP) (really two times) The callee meanwhile sends 180 Ringing (*without* SDP) which is "forwarded" by Asterisk to the extension with 180 Ringing (*with* SDP) The problem: T...

Hi everyone, I am sending out a multicast page using the following in my dialplan: Page(MulticastRTP/linksys/${MULTICAST_IP_ADDR}:6061/${MULTICAST_IP_ADDR}:6061,q) Everything works great, but I had a question about SIP and SDP: Should I be seeing a SIP/SDP message from the asterisk server containing media information and the multicast IP address? On wireshark, I see SIP/SDP from the admin phone I am using to dial the extension and initiate the page. But I never see a SIP/SDP message with the multicast address sent from...

...50.38 SIP Request: OPTIONS sip: 69.67.250.38 10.818903 69.67.250.38 -> 66.218.1.47 SIP Status: 200 OK 10.820676 192.168.0.102 -> 69.67.250.38 SIP Request: OPTIONS sip: 69.67.250.38 10.821626 69.67.250.38 -> 192.168.0.102 SIP Status: 200 OK 10.829019 66.218.1.47 -> 69.67.250.38 SIP/SDP Request: INVITE sip:95694033@69.67.250.38, with session description 10.830792 69.67.250.38 -> 66.218.1.47 SIP Status: 407 Proxy Authentication Required 10.835473 66.218.1.47 -> 69.67.250.38 SIP Request: ACK sip:95694033@69.67.250.38 10.841651 66.218.1.47 -> 69.67.250.38 SIP/SDP Reque...

Hello, I have a issue between asterisk and windows based VoIP system (Client). Vendor SIP Server --> My asterisk --> Client Here is ethereal trace between asterisk and client. 1 0.000000 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE sip:1978525648 at 192.168.4.23 <sip%3A1978525648 at 192.168.4.23>, with session description 2 0.042380 192.168.4.23 -> 192.168.3.222 SIP Status: 100 Trying 3 0.044235 192.168.4.23 -> 192.168.3.222 SIP Status: 183 Session Progress 4 0.046546 192.168.4.23 ->...

I need to get hooked up with this class, I could have students doing projects for homework :) Interested in RTCP? j On 6/26/23 7:45 PM, TTT wrote: > > I’m in training, so I have to demonstrate something SIP related.  I > figure it would be cool to hack a call, hanging it up while in > progress from outside Asterisk.  Doing so will demonstrate > use/knowledge of ARI, AMI, SIP,

Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through which I connected an external PSTN line. I use it as carrier for VoIP calls. I can make successfully calls, but there's one problem, I receive 200 OK with SDP with delay (sometimes more than 30 seconds). So when I make a call through asterisk I receive intially: - 100 Trying - 183 Session Progress, with SDP when the called number respond, I start receiving RTP with voice, also the called receives voice from me, but only after a while asterisk sends 200...

...with: t38_udptl=yes ;fax_detect=yes ;fax_detect_timeout=30 rtp_ipv6=yes Both sides are T.38 capable and detect fax tone so no need for fax detection on asterisk. Voice calls between the two work fine. But on a Fax call, I see this situation: A <=> Asterisk <=> B A: INVITE + Audio SDP => Asterisk => (same SDP) => B B: 200 OK + Audio SDP => Asterisk => (same SDP) => A * B Detects Fax-Tone! B: Re-Invite + UDPTL => Asterisk => (same SDP) => A A: 200 OK + UDPTL => Asterisk => 488 => B I tweakted the udptl setting in various ways, but I am una...

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia canreinvite=no encryption=yes avpf=yes force_avp=yes icesupport=yes directmedia=yes transport=wss,ws dtlsrekey=60 dtlsverify=no dtlscertfile=/etc/pki/tls/certs/rapidssl.c...

...> > <snip> > >> >> The problem: The extension doesn't create a ringback locally, because >> it most probably expects it to >> be sent by the callee - but the callee doesn't send anything (not >> surprising, because there has been >> no SDP). >> >> Or should Asterisk create the ringback (Asterisk doesn't send any RTP >> package)? Or should the phone >> create the ringback itself because there is a 180 Ringing (even if it >> contains SDP)? >> >> I'm wondering: Why does Asterisk create...

...a NetVanta 7100 PBX system that has a SIP trunk connection to my Asterisk box. The NV 7100 has a public IP on it that doesn't have any NAT between it and my Asterisk system. When the customer transfers a call from one handset to a voicemail box, the NV 7100 sends a RE-INVITE to Asterisk with SDP information for a different RTP port number. Asterisk is ACKing the RE-INVITE, but never changes media over to the new port number. AdTran is saying it's Asterisk's problem, since the Wireshark trace shows Asterisk is ACKing the re-invite but not changing ports. I do see that the Session...

...-in?<phone number>:<secret>@sip.internode.on.net/<phone number> externip = <my static ip> localnet = <my local subnet> canreinvite = no hasvoicemail = no qualify = yes nat = no ;rtptimeout = 120 rtpkeepalive = 5 ;ignoresdpversion = yes ;directmediapermit = <my local subnet> [sip-in] type = peer host = sip.internode.on.net context = internode-incoming ;externip = <my static ip> ;domain = internode.on.net,internode-incoming ;fromdomain = sip.interno...

...9; The 'Flow' diagram from Wireshark from a tcpdump on the OpenSIPS host follows, hopefully the email clients will not mung it too much. |Time | Client | Asterisk | | | | OpenSIPS | |7.158764 | INVITE SDP (g711U g7 | |SIP From: "760xxxxxxx" <sip:760xxxxxxx at client To:<sip:317xxxxxxx at OpenSIPS | |(5060) ------------------> (5060) | | |7.159003 | | INVITE SDP (g711U g7 |SIP Request |...

We're dialing a disconnected number via Level 3's vector network, and are receiving this. The response has SDP in it. Apparently, Level 3 is playing early media. Asterisk doesn't seem to know what to do with SDP in a 180 RINGING, and just plays ringing. What am I missing here? How can Asterisk see there's SDP, early media, in the response and act accordingly? SIP/2.0 180 Ringing. Via: SIP/2.0/U...

Asterisk receives a 180 Ringing with SDP from the called side, then it sends 180 without SDP to the calling side. We would like asterisk to sends to the calling side the same response that was received from the called side. This is Asterisk cert 13.1, is that a new behavior, is there a setting to change this ? ?

...pt all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see A1 is offering codecs: [May 6 16:43:19] WARNING[25404]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at <IP HIDDEN> port 14958 Adding codec 0x2000 (amr) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP But when A2 makes the same call to B, it only offers amr: [May 6 16:38:44] WARNING[20408]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at <IP H...

...--------------+ > | Product | Asterisk | > |--------------------+---------------------------------------------------| > | Summary | Two stack buffer overflows in SIP channel's T.38 | > | | SDP parsing code | > |--------------------+---------------------------------------------------| > | Nature of Advisory | Exploitable Stack Buffer Overflow | > |--------------------+---------------------------------------------------| &g...