search for: avpf

...ts are successfully connected. But when I made a call from one of the users, the other user doen'st have call notification and for a while the calling process ended. I check the /var/log/asterisk/messages got the following log: [Dec 17 14:54:11] WARNING[11471][C-00000000] chan_sip.c: Received SAVPF profle in audio offer but AVPF is not enabled: audio 52760 RTP/SAVPF 103 104 0 8 107 106 105 13 126 [Dec 17 14:54:11] WARNING[11471][C-00000000] chan_sip.c: Received SAVPF profle in video offer but AVPF is not enabled: video 52760 RTP/SAVPF 100 101 102 [Dec 17 14:54:11] WARNING[11471][C-00000000] c...

...SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF for this peer icesupport=yes ; Tell Asterisk to use ICE for this peer ignorecryptolifetime=yes context=sameer ; Tell Asterisk which context to use when this peer is dialing ;directmedia=yes ; Asterisk will relay media for this peer transport=udp,ws ;Asterisk will all...

...: OK (using alaw codec) 3) Call from phone2 to phone1: OK (both using alaw) 4) Call from phone1 to phone2: immediate disconnect after answering (might not be related) console says: [Aug 20 09:19:57] WARNING[7080][C-0000011f]: chan_sip.c:10434 process_sdp: Received AVP profile in audio answer but AVPF is enabled: audio 7200 RTP/AVP 8 101 [Aug 20 09:19:57] WARNING[7080][C-0000011f]: chan_sip.c:10819 process_sdp: Failing due to no acceptable offer found I enabled debug on the IP of the dect-phone (full log attached), but it does not make me any wiser... set_destination: Parsing <sip:dect...

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as Kamailio. The version is 11.10.2. With Kamailio I use rtpengine, which affects SDP descriptions...

...route the messages between the clients. Asterisk 11.11.0 Here is my client sip config: [1060] type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=fee50 ; The SIP Password for SIP.js ;encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF for this peer icesupport=yes ; Tell Asterisk to use ICE for this peer context=default ; Tell Asterisk which context to use when this peer is dialing directmedia=no ; Asterisk will relay media for this peer transport=ws ; Asterisk will allow this peer to register on U...

...of the 3 I mentioned). Previously I had problems like 'rejecting secure audio stream without encryption details', no audio or BYE messages sent immediately after call has begun etc, but according to sip.js documentation ( http://sipjs.com/guides/server-configuration/asterisk/) the settings avpf and force_avp affect the way Asterisk handles the rtp profiles and now my calls do work ok but I'd need to move the rtp profile handling to rtpengine. Here's my sip.conf: bindport = 5070 ;Kamailio is at port 5060, and it's always used as outbound proxy bindaddr = PU.BL.IC.IP tcpenable...

....conf: [00493511111111] fullname = luca secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = myproxy host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgroup= pickupgroup= dial=SIP/00493511111111 [00493512222222] fullname = fax secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = myproxy host = dynamic dtmfmode=rfc2833 canreinvite=no sendrp...

...ssip = yes > hasiax = no > hash323 = no > hasmanager = no > callwaiting = no > context = myproxy > host = dynamic > dtmfmode=rfc2833 > canreinvite=no > sendrpid=pai > type=friend > nat=force_rport,comedia > qualify=yes > qualifyfreq=60 > transport=Auto > avpf=no > force_avp=no > icesupport=no > encryption=no > callgroup= > pickupgroup= > dial=SIP/00493511111111 > > [00493512222222] > fullname = fax > secret = MYSECRET > dahdichan = 1 > hassip = yes > hasiax = no > hash323 = no > hasmanager = no > callwaiti...

...0493511111111] fullname = 00493511111111 secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = default host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgroup= pickupgroup= dial=SIP/00493511111111 [00493512222222] fullname = 00493512222222 secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = default host = dynamic dtmfmode=rfc2833 canreinvit...

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

...alue -------------------- -------------------- id 4 type friend name 660 host dynamic secret encryption yes avpf yes icesupport yes <---- ICE is enabled ipaddr PU.BL.IC.IP port 5060 regseconds 1410185500 defaultuser 660 fullcontact sip:660 at PU.BL.IC.IP:5060...

Am 23.06.2020 16:22, schrieb Marek Greško: > It seems your problems lie in something other. Most probably it is not > mtu problem. All my suspections are contradicted. If it is true you > have inter vlan voice quality problems, it is definitely something > different. Formerly I assumed you were trying only LTE vs LAN using > internet. I'm not sure what you mean with the last

Dne 11.8.2015 v 12:18 Joshua Colp napsal(a): > Vinicius Fontes wrote: >> I'm having the same issue! The difference in my case is Asterisk server >> has a public IPv4 and the browser is behind a single NAT. >> >> I'm forwarding my configuration below (which I posted previously on >> asterisk-users). >> >> How can we debug ICE negotiation? >

...a known issue" or "no, and this should be reported to Mozilla", that would be very helpful for me as well. Here is the error I see in the Asterisk console after it successfully parses the SDP a lines: Rejecting secure audio stream without encryption details: audio 62583 UDP/TLS/RTP/SAVPF 109 0 8 101 Trying to put 'SIP/2.0 488' onto WS socket destined for www.xxx.yyy.zzz:5060 No compatible codecs for this SIP call. Here is the sip.conf info. I have tried various permutations of the dtls and encryption parameters with no luck. I do have openssl and srtp built into Asterisk...

...SCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 801 v=0 o=root 969416519 969416519 IN IP4 1.1.1.1 s=Asterisk PBX 11.11.0 c=IN IP4 1.1.1.1 t=0 0 m=audio 18740 RTP/SAVPF 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=ice-ufrag:50d777041673316422560b90281fcd2e a=ice-pwd:0093fdde724f8a411742661c31c90f21 a=cand...

...tpkeepalive=0 checkmwi=10 notifyringing=yes notifyhold=yes nat=yes [1000] deny=0.0.0.0/0.0.0.0 secret=6ff108122cce3b0b45e0abf374c14ef4 dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic trustrpid=yes sendrpid=no type=friend nat=yes port=5060 qualify=yes qualifyfreq=60 transport=udp avpf=no icesupport=no dtlsenable=no dtlsverify=no dtlssetup=actpass encryption=no callgroup= pickupgroup= dial=SIP/1000 mailbox=1000 at device permit=0.0.0.0/0.0.0.0 callerid=Usuario 1 elx4 <1000> callcounter=yes faxdetect=no [1001] deny=0.0.0.0/0.0.0.0 secret=ce93963b0751ed9a88ec1badbc073fce dtm...

...e remove member" queue_log entry. (Closes issue ASTERISK-21826. Reported by Oscar Esteve) * --- chan_sip: Do not increment the SDP version between 183 and 200 responses. (Closes issue ASTERISK-21204. Reported by NITESH BANSAL) * --- chan_sip: Allow a sip peer to accept both AVP and AVPF calls (Closes issue ASTERISK-22005. Reported by Torrey Searle) * --- chan_sip: Fix Realtime Peer Update Problem When Un-registering And Expires Header In 200ok (Closes issue ASTERISK-22428. Reported by Ben Smithurst) For a full list of changes in this release, please see the ChangeLog:...

...e remove member" queue_log entry. (Closes issue ASTERISK-21826. Reported by Oscar Esteve) * --- chan_sip: Do not increment the SDP version between 183 and 200 responses. (Closes issue ASTERISK-21204. Reported by NITESH BANSAL) * --- chan_sip: Allow a sip peer to accept both AVP and AVPF calls (Closes issue ASTERISK-22005. Reported by Torrey Searle) * --- chan_sip: Fix Realtime Peer Update Problem When Un-registering And Expires Header In 200ok (Closes issue ASTERISK-22428. Reported by Ben Smithurst) For a full list of changes in this release, please see the ChangeLog:...

...g set for webrtc :   Realtime peer: Yes, cached   Prim.Transp. : WS   Allowed.Trsp : WSS   Codecs       : (alaw|g729|gsm)   Useragent    : SIP.js/0.10.0   Reg. Contact : sip:u79mer6v at 1u7hp86jdg67.invalid;transport=ws   RTP Engine   : asterisk   Encryption   : Yes   RTCP Mux     : Yes avpf = yes force_avp =yes icesupport = yes dtlsenable = yes dtlsverify = fingerprint dtlssetup = actpass dtlsfingerprint = sha-256 Why is there "UNSUPPORTED OR FAILED" in the log when processing "a=ice-ufrag" and "ice-pwd" ?? Asterisk gives no "a=ice-ufrag&quo...

...y 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.crt dtlsprivatekey=/etc/pki/tls/private/rapidssl.key dtlssetup=actpass sip dump <--- SIP read from WS:2.2.2.2:8558 ---> INVITE sip:887 at ipbx SIP/2.0 Via: S...