search for: 8000

...- 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 15000 adpcm 15000 15000 15000 15000 15000 - 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 15000 slin8 6000 6000 6000 6000 6000 6000 - 8000 8000 8000 8000 8000 8000 8000 8000 6000 6000 8250 6000 slin12 14500 14500 14500 14500 14500 14500 8500 - 8000 8000 8000 8000 8000 8000 8000 14500 14500 14000 14500 slin16 14500 14500 14500 14500 14500 14500 8500 8500 - 8000 8000...

Hi all - I am using asterisk 13.27.0 with Linphone. I turned off all codes on linphone except the one I want to try. For example: opus and speex (so only one enabled at a time). Then did this same on asterisk for the linphone extension. disallow=all allow=speex (for example). Then I place my call and the call fails. if I enable something like gsm, ulaw, alaw the call works fine. Why does the

...translate.c:490 ast_translator_build_path: No translator path: (starting codec is not valid) [Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw frame when native formats are (siren7) (rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000)) [Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw frame when native formats are (siren7) (rd:ala...

Hi All, I am trying to get the opus codec working with linphone. I followed the instructions... This shows me its loaded core show translation paths opus --- Translation paths SRC Codec "opus" sample rate 48000 --- opus:48000 To g723:8000 : No Translation Path opus:48000 To ulaw:8000 : (opus at 48000)->(slin at 48000 )->(slin at 8000)->(ulaw at 8000) opus:48000 To alaw:8000 : (opus at 48000)->(slin at 48000 )->(slin at 8000)->(a...

...17000 17000 17000 17000 17000 *adpcm *15000 15000 15000 15000 - 9000 15000 15000 15000 23000 15000 15000 17250 17000 15000 23000 17000 17000 17000 17000 17000 17000 17000 *slin *6000 6000 6000 6000 6000 - 6000 6000 6000 14000 6000 6000 8250 8000 6000 14000 8000 8000 8000 8000 8000 8000 8000 *lpc10 *15000 15000 15000 15000 15000 9000 - 15000 15000 23000 15000 15000 17250 17000 15000 23000 17000 17000 17000 17000 17000 17000 17000 *g729 *15000 15000 15000 15000 15000 9000 15000 - 15000...

...tween theses two pins" And here is the issue : My guess is there should be a common media type. When I inspect the different media type the matroska splitter can return (using the EnumMediaTypes method of the pins) I can see : f72a76a0-eb0a-11d0-ace4-0000c0cc16ba 73646976-0000-0010-8000-00aa00389b71 32595559-0000-0010-8000-00aa00389b71 VIDEOINFO2 Video YUY2 05589f80-c356-11ce-bf01-00aa0055595a 73646976-0000-0010-8000-00aa00389b71 32595559-0000-0010-8000-00aa00389b71 VIDEOINFO...

...f: Code: Load Charts Error Code: 429 Description: ActiveX component can't create object Source: Boutique40 And in the terminal, I get the following error log: Code: vchilds at Penfold:~$ wine .wine/drive_c/PMBoutique4/PMSystem40.exe err:ole:CoGetClassObject class {00000514-0000-0010-8000-00aa006d2ea4} not registered err:ole:create_server class {00000514-0000-0010-8000-00aa006d2ea4} not registered err:ole:CoGetClassObject no class object {00000514-0000-0010-8000-00aa006d2ea4} could be created for context 0x5 err:ole:CoGetClassObject class {00000535-0000-0010-8000-00aa006d2ea4} not r...

...ng an integerized version for Windows CE. But that will take longer. An ACM for Vorbis on FLAC may follow if this project works out. Cheers. Rate Channels Quality Bitrate BlockAligned Padding Loss ------------------------------------------------------------------ Narrowband Mono 8000, 1, 0, 2150, 2400, 10.4% 8000, 1, 1, 3950, 4000, 1.2% 8000, 1, 2, 5950, 6000, 0.8% 8000, 1, 3, 8000, 8000, 0.0% 8000, 1, 4, 8000,...

...00, 40, 400, 4000... How do I plot so that Cn is plotted on the x-axis in an ascending order: 40, 60, 80, .......10000? Thanks for your help. Anjan ID Cn read_count 1 MJ-2000-79 10,000 6876 2 MJ-2000-80 10,000 23440 3 MJ-2000-87 10,000 18787 4 MJ-2000-100 8000 4775 5 MJ-2000-81 8000 1542 6 MJ-2000-82 8000 1550 7 MJ-2000-101 6000 15322 8 MJ-2000-83 6000 7023 9 MJ-2000-84 6000 834 10 MJ-2000-102 4000 4216 11 MJ-2000-85 4000 1174 12 MJ-2000-86 4000 404 13 MJ-4000-131 1000...

...n 2006 11:21:10 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Content-Type: application/sdp..Content-Length: 463....v=0..o=root 16102 16102 IN IP4 10.2.11.35..s=s ession..c=IN IP4 10.2.11.35..t=0 0..m=audio 14640 RTP/AVP 0 8 4 111 18 3 97 7 110 5 101..a=rtpmap:0 PCMU/8000..a=rtpmap :8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:111 G726-32/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:3 G SM/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:7 LPC/8000..a=rtpmap:110 speex/8000..a=rtpmap:5 DVI4/8000..a=rtpmap:101 teleph one-event/8000..a=fmtp:101 0-16..a=silenceSu...

...ere a config option to force * to just passthrough the codec list sent by the 7960 in the invite? Question 3: What are SDP codec matching rules for SIP endpoints? How do they decide on common codec. Comparing the SDP sent and receive all systems claim support for 3 common codecs: a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 Now of course each device specified these 3 codecs in a different order. Under normal circumstances I feel this call should complete why is * claiming a codec mismatch? - Dustin - From phone v=0 o=Cisco-SIPUA 5892 12461 IN IP4 192.168.68.12 s=SIP Ca...

...efault nat=no LOG X-LITE (10:16:24) : ? 2004 Xten Networks, Inc. All rights reserved. X-Lite release 1105d build stamp 99999 License key: 31AC0B511918201B7ED760CE6BC073B6 Established SIP protocol listen on: 10.44.1.20:5060 Firewall Discovery Skipped SIP: 10.44.1.20:5060 RTP: 10.44.1.20:8000 NAT: 10.44.1.20 SEND TIME: 3079422292 SEND >> 0.0.0.100:5060 INVITE sip:100 SIP/2.0 Via: SIP/2.0/UDP 10.44.1.20:5060;rport;branch=z9hG4bK42C9872FBBB23E82A267FF9CA39FD454 From: informatica <sip:102 at 10.44.1.20>;tag=93961341 To: <sip:100> Contact: <sip:102 at 10.44.1...

I thought - what about the software - maybe it needs updated. After doing so I get a list: Audio codecs PCMU (8000 Hz) PCMA (8000 Hz) opus (48000 Hz) L16/16000 (16000 Hz) G.726-32 (8000 Hz) L16/8000 (8000 Hz) speex/16000 (16000 Hz) speex/8000 (8000 Hz) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200617/cb8d295...

...192.0.0.0> Call-ID: 265fdf1e0f61bd2a3422940c4ec90131@192.0.0.0 CSeq: 102 INVITE User-Agent: Asterisk PBX Content-Type: application/sdp Content-Length: 384 v=0 o=root 21604 21604 IN IP4 192.0.0.0 s=session c=IN IP4 192.0.0.0 t=0 0 m=audio 27916 RTP/AVP 4 3 18 0 8 14 5 10 7 110 97 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:14 MPA/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:97 iLBC/8000 (no NAT) to 66.33.146.12:5060 -- Called 1800XXXXXXX@net2phone Sip read: SIP/2.0 4...

...aces Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Content-Type: application/sdp Content-Length: 394 v=0 o=root 15363811 15363812 IN IP4 192.168.2.1 s=sipgate VoIP GW c=IN IP4 192.168.2.1 t=0 0 m=audio 7070 RTP/AVP 8 0 3 97 18 112 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:112 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <--- Transmitting SIP response (733 bytes) to UDP:217.10.79.9:5060 --->...

...option for a full log. requests are coming in from SipGates server etc but my asterisk box is not transfering the calls to the phones. I have the register string in my sip.conf as so: register=6698221:(MYSECRET)@sipgate.co.uk/6698221 Port on my IPCOP box as follows: UDP/5060 UDP/10000:20000 UDP/8000:8012 UDP-TCP/3478 Thanks for your time. Paul. -------------- next part -------------- Sip read: INVITE sip:6698221@MY_ISP_IP:5060 SIP/2.0 Record-Route: <sip:6698221@217.10.79.219;ftag=as6a04ebdf;lr=on> Max-Forwards: 9 Record-Route: <sip:6698221@217.10.79.8;ftag=as6a04ebdf;lr=on> Via...

...4DA4630E2@192.168.11.9. CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9. s=X-LITE. c=IN IP4 my.openser.ip.addr. t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 2007/05/17 13:31:36.325713 my.openser.ip.addr:5060 -> my.asterisk.ip.addr :5060 INVITE sip:03939749001@my.asterisk.ip.addr:50...

...ugging with tcpdump, I have seen that all the successful calls have SDP negotiation that look like this: (inside INVITE request body from SIP carrier) v=0 o=msw.chance4minutes.net 1234 0 IN IP4 38.126.208.46 s=sip call c=IN IP4 38.126.208.46 t=0 0 m=audio 30552 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 (inside 200 OK response body from asterisk) v=0 o=root 835643920 835643920 IN IP4 201.234.196.171 s=Asterisk PBX 11.10.0 c=IN IP4 201.234.196.171 t=0 0 m=audio 1...

> Try > > ping6 -I eth1 2000:8000:12:6:192:168:11:8 Thanks for the response. I tried. However, It did not resolve.... # ping6 -I eth1 2000:8000:12:6:192:168:11:8 PING 2000:8000:12:6:192:168:11:8(2000:8000:12:6:192:168:11:8) from fe80::a00:27ff:fe03:5b8a eth1: 56 data bytes On 2015/11/13 2:22, Paul Heinlein wrote: > On Thu,...

...Answering with capability 0x100(G729A) Answering with capability 0x200(SPEEX) Answering with capability 0x400(ILBC) Answering with capability 0x800(UNKN) Answering with capability 0x1000(UNKN) Answering with capability 0x2000(UNKN) Answering with capability 0x4000(UNKN) Answering with capability 0x8000(UNKN) Answering with non-codec capability 0x1(G723) 12 headers, 20 lines Reliably Transmitting: INVITE sip:8664113278@munged SIP/2.0 Via: SIP/2.0/UDP munged:0;branch=z9hG4bK304da88f From: munged To: munged Contact: munged Call-ID: 29cc2fe50f4e9c827dcc7e57676564b7@munged CSeq: 102 INVITE User-Agent:...