search for: comedia

Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten => _1X.,1,Dial(SIP/usman02/${EXTEN}) exten => _1X.,n,Hangup Server2 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.10.10.81 context=us02-trunk-inbound port=5060 qualify=yes disallo...

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > What settings have you got for directmedia? > > Could you try > > nat=force_rport,comedia > directmedia=no Tried. Peer always unreachable, call not possible... :( Other idea? Thanks Luca Bertoncello (lucabert at lucabert.de)

...=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 allow this peer to register on UDP or WebSockets ;disallow=allow ;allow=vp8 canreinvite=yes ;directrtpsetup=yes nat=force_rtp,comedia dtmfmode=rfc2833 qualify=yes [1061] ; This will be the legacy SIP client type=friend username=1061 host=dynamic secret=sameer context=sameer ignorecryptolifetime=yes nat=force_rtp,comedia encryption=yes avpf=yes ; Tell Asterisk to use AVPF for this peer icesupport=yes ; Tell Asterisk to use ICE fo...

...logging in AsterikNOW **AND** Messagenet - 2 VoIP phones, logged into Ubuntu-PBX (my phone, my wife's phone) - A Twinkle instance on my PC, logged into AsteriskNOW On AsteriskNOW: localhost*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description 00493513333333/0049351333 172.16.34.133 D Yes Yes A 5060 OK (2 ms) 00493511111111/0049351111 172.16.34.133 D Yes Yes A 5060 OK (1 ms) 0049351...

Hello, I have an asterisk box with a peer configured with nat=force_rport,comedia, but asterisk keeps sending the audio to the private IP address and ignoring the client peer nat settings. If I check the "sip show peer extension", I see both symmetric RTP and Force Rport are set to yes, but asterisk seems ignoring them. Force rport : Yes Symmetric RTP: Yes Aste...

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes

...,30,r) exten => _X.,n,Hangup And here my users.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 =...

...k and am completely stumped. I have created two accounts as instructed. Please see below for the config of the user accounts. [Peter] type=friend host=IP address disallow=all allow=ulaw allow=alaw callerid=Peter <6004> secret=XXXXXXX context=default port=9060 nat=force_rport,comedia deny=0.0.0.0 permit=IP/255.255.0.0,IP/255.255.0.0,PrvIP/255.255.0.0, PrvIP/255.255.0.0 When attempting to register there appears to be something not allowing the authentication of the client against Asterisk. I am getting a 401 Unauthorized on first attempt and then 403 (Bad auth) on second....

...> 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 &gt...

...ons happened within users. I am using sip.conf [1091] type=friend context=sip-phone call-limit=2 trustrpid=no callerid="dev1" <1091> disallow=all allow=ulaw allow=alaw username=1091 secret=XXXXX dtmfmode=rfc2833 host=dynamic mailbox=10091 at default nat=force_rport,comedia canreinvite=no extensions.conf exten => 1091,hint,SIP/${EXTEN} exten => 1091,1,Dial(SIP/${EXTEN},15,t) exten => 1091,2,Voicemail(${EXTEN}@default,u) exten => 1091,102,Voicemail(${EXTEN}@default,b) exten => 1091,103,Hangup [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:...

On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote: > How is your endpoint currently configured in asterisk? It's configured as a chan_sip peer. > Have you tried > rtp_symmetric to see if the endpoint sends audio to asterisk if > asterisk > can send audio back to the client? That would require using chan_pjsip wouldn't it? Not that I am opposed to trying that. I

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > Have you tried NAT=force_rport ? OK, tried... I can transmit from my phone (aka: I hear my voice on another phone), but I'm not able to receive data (aka: I cannot hear what I say on the other phone). Other suggestion? Thanks Luca Bertoncello (lucabert at lucabert.de)

I recently received a GoIP-32 for a client project -- primarily outbound calling. How should a GoIP be configured for Asterisk? No fancy shmancy Elastix or FPBX GUI -- just using the configuration files. Single Server Mode, Config By Line, and Trunk Gateway Mode all seem likely suspects. How did you configure your GoIP and why? What do your relevant sip.conf section(s) look like? What does

...server with 11.2 and 11.2 it works fine. I tried both IAX and SIP. the trunk in sip.conf what i have is, [serverb] type=friend username=serverb secret=serverb host=10.10.10.5 port=5060 context=default insecure=port,invite dtmfmode=rfc2833 relaxdtmf=yes directmedia=no qualify=3000 nat=force_rport,comedia disallow=all allow=g729 allow=ulaw allow=alaw deny=0.0.0.0/0.0.0.0 permit=10.10.10.5/255.255.255.0 Is there any issue with 11.1? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130703/f748a537/attach...

...ood, the following configuration allows a sip client only to receive calls (type=peer) but not to place calls (http://www.voip-info.org/wiki/view/Asterisk+sip+type). Why can I place calls though with this config? sip.conf ... [thorsten] type=peer host=dynamic context=my_context nat=force_rport,comedia secret=... dtmfmode=rfc2833 disallow=all allow=g722 allow=g729 allow=g729 ... extensions.conf ... [my_context] exten => _X.,1,Dial(DAHDI/g1/${EXTEN},60) ... Of course: when removing a valid context the client can not place the call. But I thought this behaviour can be controlled via "typ...

I'm asking about this scenario: Asterisk(public IP) <--> Internet <--> Router (public IP) <--> SIP client (private IP and NAT) What settings in sip.conf will give this the best fighting chance of working? We already have nat=force_rport,comedia

...so i might be doing something fundamentally wrong... That being said I got a FreePBX 32bit stable 6.12.65. I am having some issue with the NAT and sound, both phones are ringing but there is sound, I had some talk on IRC: <[TK]D-Fender> Note for elfranne's situation, : nat=force_rport,comedia" should have returned the public IP the call arrived on, but it is not. Can anyone comment on why it wouldn't have pulled it? A call sample 202 calling 203 (ignore 403): http://pastebin.com/sPB6FJEu

...daddr = PU.BL.IC.IP tcpenable = yes limitonpeers = yes rtcachefriends = yes rtupdate=yes tos_sip=cs3 tos_audio=ef realm = testers.com autodomain=no domain=testers.com allowexternaldomains=no allowguest=no ;avpf=yes ; encryption=yes transport=ws,wss,udp icesupport=yes srvlookup=yes nat=force_rport,comedia videosupport=yes directmedia=no And here's the way I've defined my websocket peer to my sippeers table: id: 4 name: 660 ipaddr: PU.BL.IC.IP port: 5060 regseconds: 1407744248 defaultuser: 660 fullcontact: sip:660 at PU.BL.IC.IP:5060 r...

...5) exten => _X.,n,Dial(SIP/123&SIP/456,30,oxX) ... exten => fax,1,NoOp(**** FAX DETECTED ****) exten => fax,n,Goto(fax-rx,receive,1) in the sip.conf i specified [general] sendrpid=rpid trustrpid=yes language=de videosupport=yes callevents=yes caninvite=yes qualify=yes nat=force_rport,comedia faxdetect=yes t38pt_udptl=yes ... [abcde] type=peer insecure=invite defaultuser=12345678912 fromuser=12345678912 fromdomain=abcde.ab secret=guess-what host=abcde.ab qualify=yes context=from-sip dtmfmode=rfc2833 callbackextension=12345678912 but all i can see if i try to send a testfax is == Us...

.... (my definition follows): [526] type=friend defaultname=526 defaultuser=526 secret=XXXXXXXXX dtmfmode=RFC2833 host=dynamic description=Polycom context=sip qualify=yes rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 callerid="Polycom " qualify=no canreinvite=yes timezone=1 nat=force_rport,comedia disallow=all allow=ulaw allow=alaw allow=gsm Thoughts on what is happening here or what to try? Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200806/77068536/attachment.html>