search for: direct_media

While trying to use direct_media I'm seeing RTP payload mismatches after succesful reinvites. Initial INVITE from endpoint A to asterisk has rfc4733 DMTF m=audio 35648 RTP/AVP 9 8 111 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 >From asterisk to upstream U: m=audio 14338 RTP/AVP 9 8 111 18 0 101 a=rtpmap:101 teleph...

Hey guys, have issues with reinvite, no matter what endpoint is calling asterisk always tries switch simple_bridge to native_rtp Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge technology to native_rtp in endpoints table ?direct_media? sets to ?no? on all endpoints but it doesn?t help. if native_rtp not work for some reason I have oneway audio. how can I fix this? if I add mix_monitor it works, but it?s not a right way to fix this issues. Asterisk 13.2.0 -------------- next part -------------- An HTML attachment was scrubbed.....

I'm trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures. For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesn't SIP: enduser <-> SBC <-> asterisk 13 <-> uplink RTP: enduse...

...rote: > Hey guys, > > have issues with reinvite, no matter what endpoint is calling asterisk > always tries switch simple_bridge to native_rtp > > Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge > technology to native_rtp > > in endpoints table ?direct_media? sets to ?no? on all endpoints but it > doesn?t help. > > if native_rtp not work for some reason I have oneway audio. how can I fix > this? if I add mix_monitor it works, but it?s not a right way to fix this > issues. > A native_rtp bridge is used for more than direct media. It i...

On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome <jleed at me.com> wrote: > NAT endpoint calling local endpount - switching to native_rtp then no audio, > both of them have direct_media=no, Verbose log: > > -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in > new stack > -- Launched AGI Script /pbx/agi.php > -- AGI Script Executing Application: (Dial) Options: > (PJSIP/99/sip:99 at 192.168.1.73:5060,20) &gt...

...;> have issues with reinvite, no matter what endpoint is calling asterisk >> always tries switch simple_bridge to native_rtp >> >> Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge >> technology to native_rtp >> >> in endpoints table ?direct_media? sets to ?no? on all endpoints but it >> doesn?t help. >> >> if native_rtp not work for some reason I have oneway audio. how can I fix >> this? if I add mix_monitor it works, but it?s not a right way to fix this >> issues. >> > > A native_rtp bridge is u...

NAT endpoint calling local endpount - switching to native_rtp then no audio, both of them have direct_media=no, Verbose log: -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in new stack -- Launched AGI Script /pbx/agi.php -- AGI Script Executing Application: (Dial) Options: (PJSIP/99/sip:99 at 192.168.1.73:5060,20) -- Called PJSIP/99/sip:99 at...

On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome <jleed at me.com> wrote: > Well, it breaks audio for all NAT endpoints, how can I fix this? > Local (packet to packet) bridging should not do that. Remote (direct media) can do that. Can you confirm - by looking at a verbose level 4 log - how Asterisk is bridging the two channels? -- Matthew Jordan Digium, Inc. | Director of Technology

...ng two lines, replace "<publicIP>" with the output of ; curl -s http://169.254.169.254/latest/meta-data/public-ipv4 external_media_address=<publicIP> external_signaling_address=<publicIP> [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=ulaw direct_media=no [auth_userpass](!) type=auth auth_type=userpass [aor_dynamic](!) type=aor max_contacts=1 remove_existing=yes ;Definitions for our phones, using the templates above ;; usernames and passwords etc. below My security group configuration allows TCP, UDP posrt 5060 inbound, outbound from/to 0.0....

...jsip.conf and is identical on both machines: [transport-lan] type=transport protocol=udp bind=0.0.0.0 tos=af31 [pbxbeta] type=endpoint disallow=all allow=g722 allow=ulaw transport=transport-lan context=phone-level3 aors=pbxbeta send_rpid=no send_pai=yes trust_id_inbound=yes trust_id_outbound=yes direct_media=yes direct_media_glare_mitigation=outgoing ;direct_media_method=update tos_audio=46 tos_video=34 t38_udptl=no t38_udptl_nat=no [pbxbeta] type=aor contact=sip:{remote IP address}:5060 [pbxbeta] type=identify endpoint=pbxbeta match={remote IP address} The phones have the following set in pjsip.co...

...s=<scrubbed public ip> cert_file=/etc/asterisk/keys/dev1.crt priv_key_file=/etc/asterisk/keys/dev1.key ca_list_file=/etc/asterisk/keys/ca.crt cipher=AES256-SHA method=tlsv1 ;===============EXTENSION 6001 [6000] type=endpoint context=internal disallow=all allow=ulaw auth=auth6000 aors=6000 direct_media=no rewrite_contact=yes ; necessary if endpoint does not know/register public ip:port ice_support=no force_rport=yes rtp_symmetric=no media_encryption=sdes [auth6000] type=auth auth_type=userpass password=6000 username=6000 [6000] type=aor qualify_frequency=30 max_contacts=1 remove_existing=yes...

..._frequency = 60 contact = sip:64.2.142.93 [outbound.vitelity.net] type = endpoint context = TestApp transport = transport1 aors = outbound.vitelity.net dtmf_mode = rfc4733 force_rport = yes rtp_symmetric = yes rewrite_contact = yes send_rpid = yes trust_id_inbound = yes disallow = all allow = ulaw direct_media = no *from_user=<your main vitelity account name> ; Not subaccount* [outbound.vitelity.net] type = identify endpoint = outbound.vitelity.net match = 64.2.142.93 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-use...

Hi All, We are using asterisk 16.5 and having an issue with the first re-invite after the call has been established. We can see the call gets up and you see in the logs the bridge type has changed and after that a re-invite is triggered. Is there any possibility to deactivate this kind of reInvite? We have some race conditions while have multiple asterisk in the call flow and the different

...has had good results with tls my config [transport-tls] type=transport protocol=tls bind=0.0.0.0:5061 cert_file=/etc/asterisk/keys/asterisk.crt priv_key_file=/etc/asterisk/keys/asterisk.key method=tlsv1 [XXXX] type=endpoint context=XX-Xip disallow=all allow=ulaw allow=alaw transport=transport-tls direct_media=no force_rport=yes rtp_symmetric=yes mailboxes=XXXX at default auth=XXXX aors=XXXX media_encryption=sdes dtmfmode=rfc4733 regardss -- rickygm http://gnuforever.homelinux.com

...seems to confuse the RTP engine in Asterisk (see the console output at the end of this mail)    which also seems to break transcoding, because audio doesn't work in either direction. My questions are: - Is there a way to make Asterisk always try to get out of the media path?   I thought direct_media=yes and an overlapping set of codecs would suffice. - Even if direct_media is disabled: Is there a way to make Asterisk always use a common codec between SIP endpoints,   so it doesn't need to transcode? - When Asterisk receives unexpected RTP packets or RTP packets of an unexpected type...

...ting up the Asterisk, I had to deal with NAT. The Asterisk config object of type=transport knows about essential entries: local_net= 192.168.254.1/24 bind= 192.168.254.1:5060 external_media_address= dyndns FQDN external_signaling_address= dyndns FQDN direct_media= no rtp_symmetric= yes force_rport= yes dyndns FQDN is the FQDN of my broadband access point provided by some dynamical DNS provider. This setup is not working properly with when external_media_address= and external_signaling_address= are set that way, but commenting out both makes all of t...

...ootai](!) ; type = wizard sends_auth = yes sends_registrations = yes accepts_auth = no accepts_registrations = no endpoint/call_group = 1 endpoint/pickup_group = 1 endpoint/accountcode = TOOTAi endpoint/language = fr endpoint/allow = !all,ulaw,alaw,g729 endpoint/context = incoming-Provider endpoint/direct_media = no endpoint/dtmf_mode = inband registration/retry_interval = 20 registration/max_retries = 0 registration/expiration = 3600 registration/transport = transport-udp aor/max_contacts = 2 aor/qualify_frequency = 2000 [Provider](Provider-tootai) ; remote_hosts = sips.provider.eu endpoint/callerid = &...

...ton using again T38 protocol? Below my pjsip.conf? file: ;===============TRANSPORT========================= [simpletrans] type=transport protocol=udp bind=0.0.0.0:5060 ;===============TRUNK============================== [trunk-provider] type=endpoint transport=simpletrans context=in_provider direct_media=no disallow=all allow=alaw allow=g729 aors=trunk-provider t38_udptl=yes t38_udptl_ec=redundancy t38_udptl_maxdatagram=400 [trunk-provider] type=aor contact=sip:X.X.X.X:5060 [trunk-provider] type=identify endpoint=trunk-provider match=X.X.X.X match=X.X.X.X [trunk-patton] type=aor max_contacts=5...

...working on TCP. My transport looks like this. My box is not behind NAT. [transport-tcp] type=transport protocol=tcp bind=0.0.0.0:5061 My endpoint looks like this: [user1] type=endpoint transport=transport-tcp context=local_out disallow=all allow=alaw allow=ulaw allow=g722 auth=user1 aors=user1 direct_media=no rtp_symmetric=yes force_rport=yes rewrite_contact=yes [user1] type=auth auth_type=userpass password=123456 username=user1 [user1] type=aor remove_existing=yes max_contacts=1 I have two endpoints user1 and user 2. Both are able to register fine. With both endpoints I can call into asterisk an...

...oftphones within EC2 local_net=192.168.1.0/24 external_media_address=<publicIPOfEC2Instance> external_signaling_address=<publicIPOfEC2Instance> ;Templates for the necessary config sections [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=!all,ulaw direct_media=no rtp_symmetric=yes On Thu, Mar 5, 2015 at 5:52 PM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > Hello All, > > I have an Asterisk server v13.1.0 running on EC2 and I am able to connect > and register SIP devices and "see" them on the asterisk CLI....