search for: directrtpsetup

Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool> or can u just sniff? regards, ron

Hi, I want to be able to bridge two sip channels using direct RTP between my endpoints (Audio IP : not local) but without using reinvites. So I set up my asterisk sip endpoints as follows: [test1] type=friend host=dynamic username=test1 dtmfmode=info context=test_rtp allow=all canreinvite=no directrtpsetup=yes [test2] type=friend host=dynamic username=test2 dtmfmode=info context=test_rtp allow=all canreinvite=no directrtpsetup=yes ... but it doesn't work. How can I ensure that the RTP is not going through my asterisk box and that the re-invite method is not used? P.S. Both endpoints are using...

is there a trick to get asterisk (1.6.2.13) to propagate codec-changing sip reinvites when directrtpsetup=yes? i'm trying to route calls to a gateway without keeping asterisk in the rtp stream. the gateway is first routing the call to a media server. when connecting the call to the downstream carrier a different codec is selected. the reinvite makes it to asterisk but asterisk isn't sending...

...1: Sip.conf [VoipDirect777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret=xxxxxxxxxxxx accountcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12 File: sip_custom.conf [VoipDirect777821] type=friend host=141.122.139 username=VoipDirect777821 secret=wsPiOov8830 accountcode=5260477782 amaflags=billing context=Incomming disallow=all allow=g729 ;all...

...d response from the downstream server it accessed in attempting to fulfill the request" from the rfc. Below is the current configuration for the peers. If anyone knows anything about why this could be happening your help is very much appreciated. [DID_Provider] type=peer canreinvite=no directrtpsetup=no context=incoming host=DID_PROVIDER_IP insecure=invite,port qualify=no [outbound_provider] type=peer disallow=all allow=G729 context=incoming host=outbound_provider_ip insecure=invite,port canreinvite=no directrtpsetup=no qualify=no nat=no

...ossible. Sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 192.168.1.243 ; x = Asterisk server IP address disallow=all ;allow = ulaw ; Allow all codecs ;allow = alaw context = bogon-calls ; Send SIP callers that we don't know about here canreinvite=no directrtpsetup=yes nat=no ;subscribecontext= localextensions ;default allowsubscribe=yes ; Disable support for subscriptions. (Default is yes) [App] type=friend username=App ;regexten=1234 ; When they register, create extension 1234 ;secret=password host=dynamic...

...ou in advance. Sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 192.168.1.243 ; x = Asterisk server IP address disallow=all ;allow = ulaw ; Allow all codecs ;allow = alaw context = from-sip ; Send SIP callers that we don't know about here canreinvite=no directrtpsetup=yes nat=no ;subscribecontext= localextensions ;default allowsubscribe=yes ; Disable support for subscriptions. (Default is yes) [App] type=friend username=App ;regexten=1234 ; When they register, create extension 1234 ;secret=password host=dynamic...

...neral] port=5060 bindaddr=0.0.0.0 language=de allowguest=no ;echocancel=yes ;echotraining=yes alwaysauthreject=yes disallow=all allow=alaw deny=0.0.0.0/0.0.0.0 permit=XXX.XXX.X.X/29 permit=192.168.1.0/24 ;jbenable=yes ;jbforce=yes ;jbmaxsize=20 ;jbresyncthreshold=1000 tos=0x10 directmedia=no nat=no directrtpsetup=no [provider] type=peer host=XXX.XXX.X.X insecure=port,invite context=XXXXXXXXX dtmfmode=rfc2833 directmedia=no nat=no directrtpsetup=no ;qualify=300 [one-phone] [10] type=peer context=XXXXXXXXX secret=XXXXXX host=dynamic ;qualify=300 directmedia=no nat=no directrtpsetup=no dtmfmode=inband Any h...

...the phone doesn't support "bypass" or re-INVITEs? > Is there a way to disable this functionality and instruct asterisk to just > stay in the middle of the conversation (bridging or native-bridging) for the > duration of the call? I thought that setting directmedia=no and > directrtpsetup=no would disable re-INVITEs and force asterisk to use bridging > mode, but perhaps something else is required? That should be all that is required. If that were broken I'd expect issue reports to implode - what's the configuration? -- Joshua Colp Digium, Inc. | Senior Software Develo...

...or 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 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 ; T...

Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username

...s, does anyone have any ideas how to get this working correctly? Asterisk version is 13.25.0 The settings I think are relevant (I'm using chan_sip): (sip.conf) ignoresdpversion=yes internal_timing=yes progressinband=never silencesuppression=no prematuremedia=no (Per peer) progressinband=yes directrtpsetup=no dtmfmode=rfc2833 directmedia=no silencesuppression=no prematuremedia=no TIA Mark. -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190614/4d36bd8a/attachment....

...ss" or > > re-INVITEs? > > Is there a way to disable this functionality and instruct asterisk to just > > stay in the middle of the conversation (bridging or native-bridging) for > > the > > duration of the call? I thought that setting directmedia=no and > > directrtpsetup=no would disable re-INVITEs and force asterisk to use > > bridging > > mode, but perhaps something else is required? > > That should be all that is required. If that were broken I'd expect > issue reports to implode - what's the configuration? > Here's the sip...

...> Bridge 475050e7-9d99-43f0-a9bf-7aa581a97fd9: switching from simple_bridge technology to native_rtp > Remotely bridged 'SIP/63000-00000012' and 'SIP/63009-00000013' - media will flow directly between them I added in general section of sip.conf (chan_sip in use) directrtpsetup=no directmedia=no but yet I still see "media will flow directly between them". HOW do I turn this off - RTP has to go through the server. Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/a...

...N The SIP settings are: [1532497439] type=friend host=dynamic username=1532497439 secret=wspiov8729 accountcode=1532497439 callerid=90002 regexten=90002 amaflags=billing context=OutboundWS disallow=all allow=g729 trunk=yes qualify=6000 qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 directrtpsetup=no Thanks Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081029/8e0c38e8/attachment.htm

...5.50, a soft phone on IP 10.100.210.51 and a gateway at 10.100.210.254 I have set both gateway and peer to "directmedia=yes" but still on gateway I see RTP from asterisk's IP, have tried setting nat=yes/no and also specifying localnet values but not sure where I am doing wrong. Also directrtpsetup is set to yes A sip debug and sip show peer output is here http://pastebin.com/5PwqJ1KW Please assist Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130309/57d558fb/attachment.htm>

...ast where it's configurable), Asterisk and the firewall. All ports that we're using have been opened in the firewall and incoming UDP traffic is routed to Asterisk. In Asterisk settings, localnet is defined as the LAN that both endpoints are on, externip is the public address of the server. Directrtpsetup and directmedia are both set to "no" and nat is set to "yes". So, what could be causing this issue? Best wishes, Norman -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141113/b02...

...gt; That should be all that is required. If that were broken I'd expect > > issue reports to implode - what's the configuration? > > > > Here's the sip.conf (only showing a single extension since they're all the > same): > [general] > directmedia=no > directrtpsetup=no > dtmfmode=rfc2833 > context=asterisk-internal > allowsubscribe=no > qualify=no > disallow=all > allow=ulaw > allow=alaw > allow=gsm > localnet=10.10.32.0/255.255.248.0 > localnet=192.168.32.0/255.255.255.0 > > [146] > secret= > host=dynamic > type=f...

...on bindaddr=192.168.xx.xx ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls limitonpeers=yes notifyringing=yes rtupdate=yes[authentication] [104] type=peer context=phones host=dynamic fromuser=104 secret=xxxxxx canreinvite=update directrtpsetup=no call-limit=3 nat=yes qualify=yes register=no session-timers=accept session-expires=90 session-minse=120 session-refresher=uac register => 104:xxxxx at xxxxxx.com/104 defaultip=192.168.xx.xxx mailbox=104 disallow=all allow=ulaw,alaw artcachefriends=yes notifyhold=yes incomingli...

...is that new in 1.4.22.1? Or does it default to 'no' now? (Didn't have it in either 1.4.18 or 1.4.22) Thanks. -- James > > This setup works well on my * > [104] > type=peer > context=phones > host=dynamic > fromuser=104 > secret=xxxxxx > canreinvite=yes > directrtpsetup=no > call-limit=3 > nat=yes > qualify=yes > register=no > session-timers=accept > session-expires=60 > session-minse=120 > session-refresher=uac > register => 104:xxxxx at yyyyyy.com/104 > defaultip=192.168.23.114 > mailbox=104 > > -----Original Message----...