similar to: Early media support for Asterisk behind NAT

Hi, I have a SIP client which is registered to asterisk. Asterisk is registered to a SIP trunk and also handles the media. Now since my client has some issues in its RTP Tx, which seems to have some amount of jitter (mean jitter as per ethereal trace is about 17ms, max jitter is 20 ms and max delta is 85 ms), to over come that I have enabled jitter buffer in the SIP channel by setting sip.conf

Hi, I have asterisk register two users (client-1, client-2) with a SIP proxy. I have the same two SIP client registered with asterisk. Now my dial plan setup is such that any call from client-1/client-2 is forwarded to the SIP proxy and the SIP proxy then takes the routing decision. Calls coming from SIP proxy will dial out the respective user. Asterisk is required to stay in the signaling as

Hello, I have an Asterisk 15 with PJSIP behind NAT (Amazon EC2). Now I would like to get Early Media Video working between clients in different NATed networks. The 183 signalling goes trough perfectly, but asterisk doesn't forward the Early Media RTP stream from the caller to the recipent. I have the following configuration: [6001] type = endpoint context = internal rewrite_contact = yes

Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. It looks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126

Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: ---- sip.conf: ---- [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk

wohoo, so if I unterstand it correctly with that patch early media video works over the Asterisk server? In other words the Asterisk server get's able to (process/)forward the early media video stream with that patch? 2018-04-09 17:57 GMT+02:00 Joshua Colp <jcolp at digium.com>: > On Mon, Apr 9, 2018, at 12:04 PM, Benjamin Marty wrote: > > My understanding based on Wireshark

Hi Mayur, > On 24 Nov 2014, at 07:00, MAYUR PANDEY <mayur.p at samsung.com> wrote: > In the assembly generated with O0, we are getting the "blx" instruction whereas with O1 we get "bx" (in 3.4.2 we used to get "blx" for both O0 and O1). > > Is this because of this patch: [llvm] r214959 - ARM: do not generate BLX instructions on Cortex-M CPUs

Hi Benjamin! You're obviously using a similar scenario that I have in place for testing. I initially had issues with early media (not only video also audio) as well in that scenario. What I had to do was to additionally set external_media_address=<your external IP> in pjsip.conf Also, as I wrote the patch for early-media video I'd be interested in any feedback from it. ? ? With

I have a couple asterisk boxes, running sip between both boxes. 1.4.43 on both. both are installed from source, both have default settings, My config for one box is: [devgeis] type=friend defaultname=devgeis username=devgeis secret=yes disallow=all allow=ulaw allow=alaw allow=gsm rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 host=192.168.1.8 context=panel The other box is the same. There

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

I am using asterisk 13.33.0 and POlycom phone with the latest firmware. The polycom phone is behind a firewall, the server is in the cloud. If the polycom has just booted - it receives a call, after some time (couple minutes) it no longer receives a ring. I see no errors in the CLI - looks just like the previous call as far as I can tell. Then reboot the phone and as soon as its ready call it

My understanding based on Wireshark analysis is that the signaling works (also the recipent phone is displaying the video frame before accepting the call), also the calling phone send video (i see that also via Wireshark) but the recipent phone doesn't get any video from the Asterisk before the call. 2018-04-09 17:02 GMT+02:00 Joshua Colp <jcolp at digium.com>: > On Mon, Apr 9,

Yes, media is flowing through Asterisk because both client's are behind different NAT's. Do I need to do something special in the Call Flow? Or anything additional to the pjsip.conf? 2018-04-09 16:50 GMT+02:00 Joshua Colp <jcolp at digium.com>: > On Mon, Apr 9, 2018, at 11:42 AM, Benjamin Marty wrote: > > Hello, > > > > I have an Asterisk 15 with PJSIP behind

Hi; How can I know the duration that the DAHDI channel is still used? I need to know its status and since when it is in this status, how? Also, is it possible to hangup the channel if it has been openned more than 90 minute? Other than using the timeout in the Dial command (because this I know it). What is happening with me that from time to time, I find some DAHDI channels are stayed connected

On Wed, Apr 11, 2018, at 4:33 AM, Benjamin Marty wrote: > I added the bind_rtp_to_media_address=yes on all endpoints but still the > same behaviour. The funny thing is that the G711 audio early media works > and doesn't have that Private IP issue. I was also able to cross check with > chan_sip on Asterisk 15, exactly the same wrong behaviour. See following > capture (PJSIP): As

Hi all, Are there any scale limitations in terms of how many nodes can be in a single Gluster Cluster or how much storage capacity can be managed in a single cluster? What are some of the large deployments out there that you know of? Thanks, Mayur ***************************Legal Disclaimer*************************** "This communication may contain confidential and privileged material for

Hi, This is regarding a TODO mentioned in getIdentityValue function in InstructionCombining.cpp file. //TODO: We can handle other cases e.g. Instruction::And, Instruction::Or etc. I wanted to know what could be the use cases of implementing these. When I tried implementing these and wrote test cases for the same, the test cases would be optimized in InstructionSimplify before hitting the code

Actually it should have not thrown error at all. it works fine with gcc. And the part of code which you mentioned is not getting hit at all. Maybe some difference in parsing is there. On Wed, Sep 25, 2013 at 5:29 AM, Eli Friedman <eli.friedman at gmail.com>wrote: > On Mon, Sep 23, 2013 at 11:43 PM, Mayur Pandey <mayurthebond at gmail.com>wrote: > >> for the following

Hello! Could you help me with Asterisk 11.21.2 and AsteriskNow platform. The problem is: My Asterisk PBX has SIP (chan_sip) trunk to provider. Asterisk periodically loses trunk registratrion: *sip show registry:* /Host??????????????????????????????????? dnsmgr Username?????? Refresh State??????????????? Reg.Time???????????????? // //X.X.X.X:5060??????????????????? N????? <LOGIN>

For the given test: class A1 { int x; int y; public: A1(int a, int b) { x=a; y=b; } A1 operator+(const A1&); }; A1 A1::operator+(const A1& second) { A1 sum(0,0); sum.x = x + second.x; sum.y = y + second.y; return (sum); } int main (void) { A1 one(2,3); A1 two(4,5); return 0; } when the exectable of this code is debugged in gdb for i386, we dont get the