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2004 Mar 29
6
Asterisk + GrandStream SIP phones
...*********** IP Address:192.168.0.105 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1005 Authenticate ID:1005 Authenticate Password:123 Name:1005 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ***************[1004]**************** IP Address:192.168.0.104 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1004 Authenticate ID:1004 Authenticate Password:123 Name:1004 Preferred Vocode...
2003 Sep 27
1
Continuing Budgetone woes
...NFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 257 v=0 o=btel 0 0 IN IP4 192.168.1.21 s=- c=IN IP4 192.168.1.21 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 12 headers, 13 lines Using latest request as basis request Sending to 192.168.1.21 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format ULAW Found audio format UNKN Found audio format GSM Found audio format UNKN Found description format PCMU Found description fo...
2003 Nov 05
0
SIP broken for budgtone.
...UBSCRIBE Content-Type: application/sdp Content-Length: 263 v=0 o=budgtone 0 0 IN IP4 192.168.1.223 s=- c=IN IP4 192.168.1.223 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 12 headers, 13 lines Using latest request as basis request Sending to 192.168.1.223 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format ULAW Found audio format UNKN Found audio format GSM Found audio format UNKN Found description format PCMU Found de...
2006 Jan 19
1
Sound issue with Asterisk
...,SUBSCRIBE Content-Type: application/sdp Content-Length: 337 v=0 o=budgeTone-PubIP 8000 8000 IN IP4 64.7.189.14 s=SIP Call c=IN IP4 64.7.189.14 t=0 0 m=audio 5004 RTP/AVP 2 8 4 18 15 97 9 a=sendrecv a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:15 G728/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 --- (13 headers 16 lines)--- Using INVITE request as basis request - 2143389df4877360@64.7.189.14 Sending to 64.7.189.14 : 5060 (non-NAT) Found peer 'budgeTone-PubIP' Reliably Transmitting (no NAT) to 64.7.189.14...
2005 Feb 20
0
SIP to SIP calls have no audio until put on hold and taken back off
...nt-Type: application/sdp Content-Length: 354 v=0 =1019 0 8000 IN IP4 192.168.201.111 s=SIP Call c=IN IP4 192.168.201.111 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 9 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=rtpmap:9 G722/8000 a=ptime:20 13 headers, 17 lines Using latest request as basis request Sending to 192.168.201.111 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio f...
2009 Oct 23
3
SIREN14 call setup and record/playback
...8000. a=rtpmap:98 SIREN14/16000. a=fmtp:98 bitrate=32000. a=rtpmap:97 SIREN14/16000. a=fmtp:97 bitrate=24000. a=rtpmap:102 G7221/16000. a=fmtp:102 bitrate=32000. a=rtpmap:101 G7221/16000. a=fmtp:101 bitrate=24000. a=rtpmap:103 G7221/16000. a=fmtp:103 bitrate=16000. a=rtpmap:9 G722/8000. a=rtpmap:15 G728/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=sendrecv. m=video 12388 RTP/AVP 109 34 96 31. b=TIAS:384000. a=rtpmap:109 H264/90000. a=fmtp:109 profile-level-id=42800d; max-mbps=40000; max-fs=1792; max-br=1025. a=rtpmap:34 H263/90000. a=fmtp:34 CIF4=...
2006 Feb 01
6
Receiving faxes with spandsp - strange problem
Hello, I'm trying to receive faxes with asterisk. My configuration is like this: PSTN fax -> ISDN -> Cisco router with VoIP module -> Asterisk When I try to send a fax from PSTN fax I got the standard fax signal, Asterisk starts rxfax application and then call ends and there is no tif anywhere. On the fax display there is still one message: Calling... Part of my extensions.conf:
2003 Jun 23
0
Budgetone + remote call pickup
...OPTIONS Content-Type: application/sdp Content-Length: 314 v=0 o=225 0 0 IN IP4 192.168.1.235 s=- c=IN IP4 192.168.1.235 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 12 headers, 15 lines Using latest request as basis request Sending to 192.168.1.235 : 5060 (non-NAT) Capabilities: us - 4, them - 269, combined - 4 Non-codec capabilities: us - 1, them - 1, combined - 1 Reliably Transmitting (no NAT): SIP/2.0...
2005 Jun 06
1
Issue with SIP inter-op
...ntent-Type: application/sdp Content-Length: 386 v=0 o=- 8000 1 IN IP4 69.xx.xx.xx s=- c=IN IP4 69.xx.xx.xx t=0 0 m=audio 31060 RTP/AVP 4 18 0 8 2 15 99 101 a=sendrecv a=rtpmap:4 G723/8000/3 a=rtpmap:18 G729/8000/3 a=rtpmap:0 PCMU/8000/3 a=rtpmap:8 PCMA/8000/3 a=rtpmap:2 G726-32/8000/3 a=rtpmap:15 G728/8000/3 a=rtpmap:99 iLBC/8000/3 a=fmtp:99 mode=20 a=ptime:60 a=rtpmap:101 telephone-event/8000/3 a=fmtp:101 0-11 --- (11 headers 18 lines)--- Using INVITE request as basis request - bc1e6d746b7c0e4df@192.168.1.3 Sending to 69.xx.xx.xx : 5060 (NAT) Found peer 'sip-devices' Reliably Transmit...
2005 Jul 07
1
Calls with oh323 with no sound
...k) ; G72316K3 - G.723.1(6.3k) ; G72315K3 - G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G726 - G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K - G.726(32k) ; G72640K - G.726(40k) ; G728 - G.728 ; G729 - G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; codec=G...
2003 Jul 08
0
codec problems with asterisk
...000 bps g711ulaw G.711 u Law 64000 bps g723ar53 G.723.1 ANNEX-A 5300 bps g723ar63 G.723.1 ANNEX-A 6300 bps g723r53 G.723.1 5300 bps g723r63 G.723.1 6300 bps g726r16 G.726 16000 bps g726r24 G.726 24000 bps g726r32 G.726 32000 bps g728 G.728 16000 bps g729br8 G.729 ANNEX-B 8000 bps g729r8 G.729 8000 bps
2003 Dec 05
2
asterisk codec sizes, data plus overhead
Hello. I have been searching the archives for a simple, clear listing of the available codecs with total size, plus the data and overhead sizes. Does anyone have this handy, and can it be added somewhere, even the wiki. Regards...Martin -- The system will be down for 10 days for preventive maintenance.
2005 May 11
0
Vegastream assistance?
...ntCode=H323 context=sip [register] alias=ASTERIX [codecs] ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3 - G.723.1(6.3k) ; G72315K3 - G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G728 - G.728 ; G729 - G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 codec=G729 codec=G7231 codec=G711A codec=G711U
2010 Sep 02
0
NCS - Cablemodem
...- 'aaln/1 at 0-13-11-82-bd-a.ssw.dominio.net in 'alberti' is idle ssw*CLI> mgcp audit endpoint aaln/1 at 0-13-11-82-bd-a.ssw.dominio.net Posting Request: AUEP 3 aaln/1 at 0-13-11-82-bd-a.ssw.dominio.net MGCP 1.0 NCS 1.0 F: A to 10.30.15.254:2427 MGCP read: 200 3 OK A: a:PCMU;PCMA;G728;G729;G729E;G726-16;G726-24;G726-32;G726-40, p:10-30, b:19-100, e:on, t:1, s:off, v:L;fxr;rg;xal;x-xl;fm;lcs;sst;x-jc;x-pol;xrm, m:sendrecv;sendonly;recvonly;inactive;netwloop;netwtest;replcate;confrnce, dq-gi, sc-rtcp: 81/70;81/71;82/70;82/71;80/70;80/71, sc-rtp: 62/51;62/50;64/51;64/50;60/51;...
2006 Mar 28
0
codec translation problem???
2004 Oct 07
2
openphone & Asterisk
What is the configuration of H323.conf and openphone in order to run openphone and asterisk together ?
2003 May 28
0
calls between SIP and H.323 clients
...on may have a "frames" option ; associated with it. ; Valid values for the "codec" option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3 - G.723.1(6.3k) ; G72315K3 - G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G728 - G.728 ; G729 - G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; codec=GSM0610 frames=4 codec=G711A frames=20 ;codec=G7231 -------------- n...
2005 May 25
0
oh323 problems - Solved
...k) ; G72316K3 - G.723.1(6.3k) ; G72315K3 - G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G726 - G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K - G.726(32k) ; G72640K - G.726(40k) ; G728 - G.728 ; G729 - G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; ;codec=...
2005 Jul 27
1
H323 Configuration file
Folks! I would appreciate if someone could send me a simple working h323 configuration file oh323.conf that is part of asterisk@home installation. I have tried to use the oh323.conf content listed on WIKI but it is just not working as my H323 endpoint ( PA168 based ATCOM Phone) cannot register. I need a working example of this file for similar phone. Seshu
2003 Sep 22
2
how to dial a h323 destination ?
Hi all, i have big problems to make a h323 call over the gatekeeper from my provider. The provider demanded following account data: H323 ID: XXX-XXX-XX-X DetinationNumer: XXXXXXXXXXX I have configured the oh323.conf following: gatekeeper=XX.XX.XXX.XXX alias=XXX-XXX-XX-X Isx the alias equal to the h323id ? And how i have to make a call with the dial app ? I have following config: exten