search for: force_avp

I'm using v11.11 I tried setting: force_avp=yes in a SIP peer in sip.conf and it seems to be ignored. The WebRTC client sends an INVITE with "RTP/SAVPF" and Asterisk is still sending back 183 and 200 responses with the UDP/TLS/RTP/SAVPF string Are there some limitations with this option or does it depend on any other settings?...

...m MagicJack would go silent after about 10 seconds.  This happened while in the automated attendant area.  This problem did not occur with Asterisk 13 LTS.  I reverted PJSIP back to SIP and the problem still occurred, so that was not it. We connect to Flowroute for our SIP provider.  I added "force_avp = yes" to the Flowroute endpoint section in the pjsip.conf and the problem appeared to be solved after I tested it a dozen times.  However, this morning this issue has reappeared.  Any thoughts on what might be causing this? My Flowroute pjsip.conf config: [transport-udp] type = transport pro...

...[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 sendrpid=pai type=f...

...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 = yes limiton...

...t; 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 > conte...

...11111] 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 canreinvite=no sendrpid...

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,

...isk 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 UDP or WebSockets force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11 nat=force_rport,comedia accept_outofcall_message=yes outofcall_message_context=messages ;dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer ;dtlsverify=no ; Tell Asterisk to not verify your DTLS certs ;dtlscertfile=/etc/asterisk...

...efault directmedia no deny 0.0.0.0/0.0.0.0 permit PU.BL.IC.IP nat force_rport,comedia language disallow allow force_avp yes callerid amaflags mailbox regexten regserver fromdomain testers.com videosupport no contactpermit contactden...

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

...efaultuser: 660 fullcontact: sip:660 at 1.1.1.1:5060 regserver: useragent: lastms: 0 host: dynamic type: friend context: default deny: 0.0.0.0/0.0.0.0 permit: 1.1.1.1 secret: NULL md5secret: NULL avpf: yes force_avp: yes icesupport: yes directmedia: no encryption: yes nat: force_rport,comedia callgroup: NULL pickupgroup: NULL language: NULL disallow: NULL allow: NULL setvar: NULL callerid: NULL amaflags: NULL videosupport: no maxcallbitra...

...c for debugging support p.s.2 relevant part of pjsip.conf [global] [transport-wss] type=transport protocol=wss ;udp,tcp,tls,ws,wss bind=0.0.0.0 ;===============ENDPOINT TEMPLATES [endpoint-basic](!) type=endpoint transport=transport-wss context=route_phones disallow=all allow=alaw allow=ulaw force_avp=yes use_avpf=yes ; Determines whether res_pjsip will use and enforce usage of media_encryption=dtls ; Determines whether res_pjsip will use and enforce dtls_verify=no ; Verify that the provided peer certificate is valid (default: dtls_rekey=0 ; Interval at which to renegotiate the TLS sess...

...brtc :   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" and "ice...

...erisk 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: SIP/2.0/WSS df...

...deny=0.0.0.0/0.0.0.0 > secret=NearlyANastyThat > dtmfmode=rfc2833 > canreinvite=no > context=from-internal > host=dynamic > trustrpid=yes > sendrpid=no > type=friend > nat=no > port=5060 > qualify=yes > qualifyfreq=60 > transport=tls,udp,tcp > avpf=no > force_avp=no > icesupport=no > encryption=yes > callgroup= > pickupgroup= > dial=SIP/41712 > mailbox=41712 at device > permit=192.168.6.0/255.255.255.0 > callerid=James B Byrne <41712> > callcounter=yes > faxdetect=no > cc_monitor_policy=generic > > If I change th...

Asterisk 13.8 Is CALLERID(all) supposed to wok for pjsip? When I do this: exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) same => n,Dial(PJSIP/phone123, 30) I expect the callerid to be as set, but is always seems to be "phone123", the name of the endpoint. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL:

...=10100 users.conf: [00491773333333] fullname = 00491773333333 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=yes qualify=yes qualifyfreq=60 avpf=no force_avp=no icesupport=no encryption=no callgroup= pickupgroup= dial=SIP/00491773333333 And on my Firewall: /sbin/iptables -t nat -A PREROUTING -i ppp0 -p udp -m udp --dport 10000:10100 -j DNAT --to-destination 192.168.20.120 /sbin/iptables -t nat -A PREROUTING -i ppp0 -p udp -m udp --dport 5060 -j DN...

...isk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1 [6000] host=dynamic secret=mysecret context=default type=friend icesupport=yes directmedia=no disallow=all allow=ulaw qualify=yes [6001] host=dynamic secret=mysecret context=default type=friend encryption=yes avpf=yes force_avp=yes icesupport=yes directmedia=no disallow=all allow=ulaw dtlsenable=yes dtlsverify=fingerprint dtlscertfile=/etc/asterisk/keys/asterisk.pem dtlscafile=/etc/asterisk/keys/ca.crt dtlssetup=actpass *extensions.conf:* [default] exten => _6XXX,1,Dial(SIP/${EXTEN}) *rtp.conf:* [general] rtpstart=...

hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side

...lsclientmethod=tlsv1 And I have this for the test device context: [41712] deny=0.0.0.0/0.0.0.0 secret=NearlyANastyThat dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic trustrpid=yes sendrpid=no type=friend nat=no port=5060 qualify=yes qualifyfreq=60 transport=tls,udp,tcp avpf=no force_avp=no icesupport=no encryption=yes callgroup= pickupgroup= dial=SIP/41712 mailbox=41712 at device permit=192.168.6.0/255.255.255.0 callerid=James B Byrne <41712> callcounter=yes faxdetect=no cc_monitor_policy=generic If I change the transport setting to TLS then I get this reported: [2015-03-0...