similar to: Directmedia Question

Hi, I am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF

Andrew Martin wrote: > ----- Original Message ----- >> From: "Joshua Colp"<jcolp at digium.com> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users at lists.digium.com> >> Sent: Monday, May 11, 2015 12:32:06 PM >> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32

Hi all I've got an issue where when I call a number that just plays early media back to me. Instead of hearing the full sequence of tones I hear a short ringing then part of the sequence. What seems odd is that I can see the telephone-event/8000 being passed up the chain but when it gets to Asterisk, it is never sent back to the phone. Instead I just see the usual RTP flows. I've been

I have a hosted server. I have TWO different locations what have phones. Chicago and Indiana If I send audio direct from server to Chicago I hear it - same with indiana. But if indiana calls chicago - NO AUDIO. I see this in the CLI -- Channel SIP/63009-00000013 joined 'simple_bridge' basic-bridge <475050e7-9d99-43f0-a9bf-7aa581a97fd9> -- Channel SIP/63000-00000012 joined

----- Original Message ----- > From: "Andrew Martin" <amartin at xes-inc.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Monday, May 11, 2015 1:35:07 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > > That should

Morning, We recently pushed our Asterisk video bridge into a DMZ and since then, local calls have been unreliable to say the least. While offsite calls work nicely, calls on our internal server usually fail to ring the far end. Two test calls that were made 4 minutes apart yielded different results: one rang the far end, the other kept trying to transmit the Invite. The configuration didn't

Hello, I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All the SIP clients are on a LAN, so no NAT is involved. I have been experiencing an intermittent problem where a call will be successfully answered, but then dropped by Asterisk 32 seconds after it is answered (with a "Retransmission timeout reached on transmission" error). Here is an example of this

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

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

Please help. Asterisk 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

Hi all my asterisk server, 2 sip client softphones are the same LAN asterisk ip address : 192.168.1.5 sip client 1 : 192.168.1.4 sip client 2 : 192.168.1.2 asterisk starts ok with sip setup the sip.conf [test] type=friend username=test secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes [1000] type=friend username=1000 secret=1000 host=dynamic context=cucku

Hey guys, I've been having a very odd problem that happens intermittently. I've had this happen with only a couple of providers and somewhat rarely but its to the point now that we need to fix it to be able to do business. The scenario is as follows: We have a DID provider that routes calls to our asterisk boxes and we have an outbound provider to whom we send the calls of the person

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

Hi all again, I've got a problem with sporadic one way audio calls, which means sometimes I can't hear the calling party (call is established, but audio is missing). Today I received ~90 calls, one of them got this problem. I've got two networks involved, without NAT: - 192.168.1.X, in there one nic of my server and all the phones - a private net to my provider, in there a nic of my

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

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]

Andrew Martin wrote: > ----- Original Message ----- <snip> > > By doing a number of test calls today, I have managed to reproduce this while > sip debugging was on, so I have that information available now as well: > http://pastebin.com/ZJqzdvY3 > > This was a call from 113 to 146 via a queue. Note that the asterisk server is > at 10.10.32.251. I see the following:

Hello all, I'm setting up a couple of test boxes and I'm running into a problem. What I need help with is determining whether I'm going something wrong or if I need to post a bug report. I have two asterisk 13.0-beta 3 machines set up with extensions connected to each as such: 3700 ----> AST-A <------> AST-B <---- 3800 & 3801 When I place a call from 3800 to

----- Original Message ----- > From: "Joshua Colp" <jcolp at digium.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Monday, May 11, 2015 1:24:53 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > > Could this

Hey guys, I'm using asterisk 1.6.2.13 and have an endpoint which uses silence suppression which I can't turn off. I've set rtpkeepalive=10 in sip.conf [general], as well as under the peer details for our sip provider but it doesn't seem to do anything. Rtp debug shows that we are receiving RTP from the SIP provider, and forwarding it to the end point, but no RTP packets are sent