similar to: DTMF Debug

Hi, Want to do some experiments with the G.723 codecs - where can I download the 723 source code for Asterisk? I know there are some ongoing discussion regarding patents and license fees for the g.723 but I have some hardware on which I only have the 723 and need to test it privately. Thanks! Dan _________________________________________________________________ Use MSN Messenger to send

Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get handed off to the Asterisk server via SIP, at least not in any format that Asterisk

hi, i have strange problem with no rtp packets from asterisk after dns query. see pcap below centos6/asterisk 13.9 + chan_sip 172.23.0.3 - asterisk 172.23.5.1/2 - voip phones any ideas/hints? 1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49318, Time=1442112256 1171 25.045556000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711

Hi list, I know * generates it's outgoing RTP stream based on the incomming RTP stream, i'm having some audio problems after i recieve an rtp reinvite from my carrier. Situation: Phone -- Asterisk A --- Asterisk B --- Carrier --- PSTN Asterisk A: reinvite = no Asterisk B: reinvite = no If i dial out on phone via asterisk A, Asterisk B relay's the INVITE to the carrier, after the

Hello list, We're currently facing some issues concerning T.38 gateway faxing. This is a device used almost exclusively for receiving faxes. Calls are incoming to asterisk on a SIP trunk (sangoma netborder) using G711A. Gateway mode is activated in the asterisk dialplan towards a Cisco SPA 112 running firmware 1.3.5. We are using asterisk 1.8.13.0 with the T.38 gateway patch applied (I know I

Audio packets are running... 961 16.150421076 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28402, Time=73280 962 16.170411284 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28403, Time=73440 963 16.190381989 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28404, Time=73600 964 16.210387990

Hi LLVMers, there were a gcc ICE problem discussed in current mail list. Chris was right here: http://mail.cs.uiuc.edu/pipermail/llvmdev/2003-December/000693.html saying that the PR 12544 is not really the corresponding issue :) The correct one is PR 13392: http://gcc.gnu.org/bugzilla/show_bug.cgi?id=13392 Interesting fact is that -O2 (or -O3) goes somehow around this

Hello, I was testing with sdp and something came up worth asking: While calling from a webrtc client to another (chrome, sip.js) Asterisk receives the following sdp and rejects it with 488 Not Acceptable. Why does this happen, what's wrong with the sdp? The second sdp body below is accepted instead. Both have rtp profile RTP/SAVPF, difference is that the second one was produced by rtpengine,

Hi All, Could somebody explain me how the timestamps are computed in asterisk while bridging two sip channels ? I've got situation with my provider, who changed some things in config and added some codecs (that much i know) and after that we got one way audio issues. It seems that the problem is with RTP timestamps. Within one outgoing stream the RTP timestamps are growing, as it should

Hi All. I'm running some tests with the latest Asterisk SVN-branch-12-r410493M compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS machine (2.6.32-358.18.1.el6.i686). As a client I'm using the sipMLP WebRTC javascript softphone running on Chrome 33.0.1750.146 m. I have the softphone correctly registered on the Asterisk machine but as soon as I try to start a new call

Hi Everyone. I am getting this error WARNING[31977][C-00000009]: chan_sip.c:10657 process_sdp: Can't provide secure audio requested in SDP offer >From the sdp can anyone suggest why secure audio cannot be provided ????v=0 ????o=- 6611325078116277019 2 IN IP4 127.0.0.1 ????s=- ????t=0 0 ????a=group:BUNDLE audio ????a=msid-semantic: WMS YxFi1hLhslP6PiA3D1xi2RxV5i1iATmDOz4l ????m=audio

We are having a weird problem where calls get cut off in the middle. I'm not a SIP expert but could the INVITE with an empty SDP be the problem? |Time | 209.220.119.18 | | | | 208.88.61.150 | |9687.369 | INVITE SDP (g729 telephone-eventRTPType-101) |SIP From: <sip:vmax at 209.220.119.18

Hello, I've been trying to send traffic to a Cisco Call Manager 3.2, but with no luck. Here's whats happening: * Call gets to CCM * Call gets to the gateway * Rings a couple times on destiny * Call gets hungup. On the CCM I get the following error: MediaManager - ERROR wait_AuConnectErrorInd On the Gw (Cisco AS5300) I get a disconnect cause of 2F (Resource not available) On asterisk:

facing problem with originating webrtc calls 1-when iam doing call from webrtc iget ice working <--- SIP read from WS:91.196.158.205:1466 ---> INVITE sip:0669197533 at 77.91.132.9 SIP/2.0 Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315 Max-Forwards: 69 To: <sip:0669197533 at 77.91.132.9> From: "Anton" <sip:1065 at 77.91.132.9>;tag=5i21qaop43 Call-ID:

I have a problem that has developed within about the past 3 months with my backup outgoing SIP provider (I am not sure when this problem started since it involves only my backup provider which is used rarely). The problem is that most (not all) outgoing calls fail during the earliest stages of call setup, specifically after the provider sends back a "407 Proxy Auth Required" response.

v2: Check for PFIFO, don't pause if it's not yet running. This should fix reclocking on boot Signed-off-by: Roy Spliet <rspliet at eclipso.eu> --- drivers/gpu/drm/nouveau/Makefile | 1 + drivers/gpu/drm/nouveau/core/engine/device/nv50.c | 4 +- .../gpu/drm/nouveau/core/include/subdev/clock.h | 4 + drivers/gpu/drm/nouveau/core/subdev/clock/nvaa.c | 439

v2: Check for PFIFO, don't pause if it's not yet running. This should fix reclocking on boot v3: Tiny clean up Signed-off-by: Roy Spliet <rspliet at eclipso.eu> --- drivers/gpu/drm/nouveau/Makefile | 1 + drivers/gpu/drm/nouveau/core/engine/device/nv50.c | 4 +- .../gpu/drm/nouveau/core/include/subdev/clock.h | 4 +

Signed-off-by: Roy Spliet <nouveau at spliet.org> --- drivers/gpu/drm/nouveau/nvkm/subdev/clk/gf100.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) diff --git a/drivers/gpu/drm/nouveau/nvkm/subdev/clk/gf100.c b/drivers/gpu/drm/nouveau/nvkm/subdev/clk/gf100.c index f9a4918..80c6dd6 100644 --- a/drivers/gpu/drm/nouveau/nvkm/subdev/clk/gf100.c +++

v2: fix typo it's -> its Signed-off-by: Roy Spliet <nouveau at spliet.org> --- drivers/gpu/drm/nouveau/nvkm/subdev/clk/gf100.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) diff --git a/drivers/gpu/drm/nouveau/nvkm/subdev/clk/gf100.c b/drivers/gpu/drm/nouveau/nvkm/subdev/clk/gf100.c index 026baff..89d5543 100644 ---

Hello. Is someone here using VoIPmonitor? I am using just the sniffer and I found some pcap files that contain some odd streams. For example, I have a file with 3 streams, but the weird stuff is that 2 streams are the same (e.g., have the same source address and port and same destination address and port). Example: "Source Address","Source Port","Destination