search for: directmedia

Hello, I have a question about "directmedia" or "canreinvite", I have experience that whatever I set directmedia=yes or no. After I run sip show settings. all settings looks the same. My question is how I could make sure from "sip show settings" that my "directmedia" configuration is applied. Thanks

...te (not a typo); I have found it in at least one example and one article, but with no documentation, so the difference between canreinvite and reinvite is completely unclear to me. chan_sip seems to be documented very poorly, unlike chan_pjsip / res_pjsip. I'm just going to comment on the 'directmedia'/'canreinvite' points here. 1) There is no 'reinvite' setting in chan_sip. If you patched Asterisk, than your mileage may vary. 2) 'directmedia' is the same thing as 'canreinvite'. They are the same setting. 'directmedia' replaced the nomenclature 'c...

Hi, when using some non dynamic host eg. host=192.168.111.153 in sip.conf asterisk is not considering specific peer options eg. directmedia=off, transport=tcp if I set host=dynamic and register the sip phone it works as expected. Is this a bug or feature - I wanna disable the usage of directmedia for some peers with fixed ip but wanna allow it in general. Same with transport=tcp. [97] type=peer host=192.168.111.153 transport=udp c...

I've got several SIP clients with dynamic IP addresses Asterisk has one public and one private IP address SIP clients might connect to Asterisk from either the internet or the private network (192.168.1.255) - they're portable By default, directmedia/canreinvite is enabled and Asterisk sets up direct media connections between clients. In this case clients on the internet can make calls between each other, and clients on the private network can make calls between each other, but calls between clients on the internet and clients on the private ne...

...----> AST-A <------> AST-B <---- 3800 & 3801 When I place a call from 3800 to 3700 or the other way around , asterisk seg faults on both machines at roughly the same time. All connections are done using PJSIP. The crash occurs when the ringing extension is answered. If I set (directmedia=no) OR (directmedia=yes & t38_udptl=yes) on the trunk then the call completes fine. All phones and servers are on the same LAN with no firewalls active. The trunk between AST-A and AST-B is configured like this in pjsip.conf and is identical on both machines: [transport-lan] type=transport...

Hello List, I have some doubt about direct media settings. I have an asterisk 1.8.14 instance running on 172.20.255.50, 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...

Asterisk 11 CentOS 6.4 Cisco 7971 phones Does chan_skinny support directmedia? Jacob Miles Software Engineer jacob.e.miles at l-3com.com 903.457.4422 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130604/745a7918/attachment-0001.htm> -------------- nex...

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)

Hi, with canreinvite=no and directmedia=no I and getting the message in the logs for all calls "switching from simple_bridge technology to native_rtp" -- Executing [102 at mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/102 -- SIP/102-00000018 is...

...,n,NoOp("Callerid is " ${CALLERID(all)} ) exten => _j.,n,GoTo(from-outside,${3digitexten},1) [from-outside] exten => 123,1,NoOp() exten => 123,n,Answer() exten => 123,n,Dial(SIP/jnctn/1212xxxyyyy) exten => 123,n,HangUp() sip.conf: [general] externaddr=xx.yyy.zz.aa nat=yes directmedia=no ; tried nonat sip show peer jnctn: Insecure : invite Force rport : Yes ......... DirectMedia : No sip show peer teliax: Insecure : port,invite Force rport : Yes ........ DirectMedia : No And the cli doesn't show any problems: NoOp("SIP/teliax-0...

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)

Hello, at first, thanks for helping! In the meantime, I have done a lot of research and trial and error, and I could solve that specific problem. Obviously, the dialplan application "Answer" was playing a key role here. My original dialplan snippet (which produced that problem) was: exten => _00., 1, NoOp() same => n, Set(FAXOPT(gateway)=yes) same => n,

...ng a problem getting >> chan_sip to quit ignoring re-invites from Vitelity. Our side ends up >> sending a reinvite which their side & they do not support us sending a >> reinvite. Ive tried: >> >> canreinvite=no which was supposedly replaced by: >> >> directmedia=no >> >> Can anyone shed any light on this matter? I'd love to get this fixed. >> > > Those options *should* influence chan_sip's reinvite behavior - at > least they have from my experiences working with chan_sip. Do you > know what is triggering the reinvit...

...fax from SPA112 to PSTN fax doesn't work. Using udptl debug, I can see packets between Asterisk and both sides (SPA112 and PSTN fax) but it seems that faxes can't agree how to send image. == sip.conf: [general] tcpenable=yes videosupport=yes transport=udp,tcp dtmfmode=rfc2833 qualify=yes directmedia=no allowguest=no alwaysauthreject=yes rtcachefriends=yes rtupdate=no callcounter=yes t38pt_udptl=yes,redundancy,maxdatagram=200 t38pt_rtp=no t38pt_tcp=no ignoresdpversion=yes disallow=all allow=alaw allow=ulaw externip=82.200.7.184 localnet=192.168.0.0/255.255.0.0 [mtt] type=peer host=80.75.130.13...

On Sat, 15 Aug 2015 16:30:39 +0800 Michael Dupree <michael at easybitllc.com> wrote: > Not 100% ure, but maybe play with the canreinvite or directmedia > settings. Yes! That was it. Just for future searches here is what I did. I added "directmedia = no" in sip.conf. This fixed the issue. I believe that Asterisk was getting confused when one leg was inside NAT and the other was outside. Perhaps there was an "OR" where...

Hello Mynum: 6001 , Othernum: 6002. I can record as follows. But I do not enter individual records for each internal required. I want to do it more smoothly with a Macro. Thanks. exten => _6001,1,NoOp() exten => _6001,n,MixMonitor(${UNIQUEID}.wav,ab) exten => _6001,n,Dial(SIP/6001,20) exten => _6001,n,StopMixMonitor() exten => _6001,n,Hangup() On Sat, Mar 9, 2019 at 6:50 PM

...unk-r247652M built by root @ sip on a i686 running Linux on 2010-01-25 11:10:15 UTC [0197673581] secret=xyz callerid=Input Interior Orebro (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=no context=inputinterior.se directmedia=no dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=yes qualify=yes sendrpid=pai t38pt_udptl=no transport=udp trustrpid=yes type=friend videosupport=no [0851711201] secret=xyz callerid=Input Interior Stockholm (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcoun...

On 2/23/19 2:39 PM, Social Boh wrote: > *DIrect media with SRTP is not supported. All media when SRTP goes > through Asterisk.* > > So you have to open ports on your firewall and disable directmedia=yes > on your configuration. directmedia is not explicitly enabled; I guess it's the default. Joshua basically says there is no way to control which ports are being used for SRTP because that it is "up the endpoint". Such endpoints, in this case, are mobile phones with softwa...

...:34.765: RTP(50493): ps rx s=0.0.0.0(0), d=192.168.0.139(17112), pt=8, ts=BF40, ssrc=2C1690C9 any body knows how should i fix it? this is my ooh323.conf file: [general] port=1720 context=from-trunk gatekeeper=DISABLE bindaddr=192.X.X.X disallow=all allow=all AcceptAnonymous=yes directrtpsetup=yes directmedia=yes faststart=yes h245tunneling=yes mediawaitforconnect=yes tos=lowdelay [sam] type=user host=192.X.X.X directmedia=yes [sam-1] type=peer host=192.X.X.X directmedia=yes any comments or hints are really appreciated. SAM -------------- next part -------------- An HTML attachment was scrubbed... UR...

...e IAX trunks split off to where they need to go. We are having a problem getting chan_sip to quit ignoring re-invites from Vitelity. Our side ends up sending a reinvite which their side & they do not support us sending a reinvite. Ive tried: canreinvite=no which was supposedly replaced by: directmedia=no Can anyone shed any light on this matter? I'd love to get this fixed. There is no firewall on this machine at all. Thanks --Tammy