Re: [asterisk-users] hi
Try /usr/sbin/asterisk. Also, copy the list. Don't email me privately. aster...@opensourcesolution.in wrote: hi friend, i gave that command which u told i.e asterisk -V. the output is below [r...@localhost ~]# cd /etc/ [r...@localhost etc]# asterisk -v bash: asterisk: command not found [r...@localhost etc]# asterisk -V bash: asterisk: command not found thx -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to check version of asterisk
Mr. aster...@opensourcesolution, if you had googled for how to know the asterisk version you would have found the solution right away. CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, November 08, 2009 11:04 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] how to check version of asterisk On Sun, Nov 08, 2009 at 06:20:46AM +, aster...@opensourcesolution.in wrote: hi all, i had installed asterisk under /etc. now i want to know by command which version of asterisk i had installed. how to know the version plz tell me. asterisk -V -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hi
Ciao, try as below (in bold the command) *[r...@dhcppc0 asterisk]# pwd /etc/asterisk* [r...@dhcppc0 asterisk]# *asterisk -vr* *Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer **marks...@digium.com* marks...@digium.com* Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Connected to Asterisk 1.4.24 currently running on dhcppc0 (pid = 2711) Verbosity is at least 8 dhcppc0*CLI* 2009/11/8 Alex Balashov abalas...@evaristesys.com Try /usr/sbin/asterisk. Also, copy the list. Don't email me privately. aster...@opensourcesolution.in wrote: hi friend, i gave that command which u told i.e asterisk -V. the output is below [r...@localhost ~]# cd /etc/ [r...@localhost etc]# asterisk -v bash: asterisk: command not found [r...@localhost etc]# asterisk -V bash: asterisk: command not found thx -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giancarlo Lombardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Failure of user registration with XLITE
Dear all, I'm setting up a connection via XLITE softphone and asterisk 1.4 but I get the error: *Registration error: 404 Not found* Here my configuration file of asterisk: *[r...@dhcppc0 asterisk]# vi sip.conf [gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* *[r...@dhcppc0 asterisk]# vi extensions.conf [tutorial] exten = 1234,1,Dial(SIP,gianca)* *exten = 12345,1,Dial(SIP,giusy) * Here the XLITE user data: *Display Name: gianca* *Username: 1234* *Password: pwd_gianca* *Authorization User Name: 1234* *Domain: 192.168.1.100* ** Here the output of wireshark in between Xlite client and asterisk server: ** *0040 2e 31 30 30 20 53 49 50 2f 32 2e 30 0d 0a 56 69 .100 SIP /2.0..Vi 0050 61 3a 20 53 49 50 2f 32 2e 30 2f 55 44 50 20 31 a: SIP/2 .0/UDP 1 0060 39 32 2e 31 36 38 2e 31 2e 31 31 36 3a 35 34 30 92.168.1 .116:540 0070 35 30 3b 62 72 61 6e 63 68 3d 7a 39 68 47 34 62 50;branc h=z9hG4b 0080 4b 2d 64 38 37 35 34 7a 2d 32 34 32 38 38 65 37 K-d8754z -24288e7 0090 32 38 32 36 64 30 31 32 38 2d 31 2d 2d 2d 64 38 2826d012 8-1---d8 00a0 37 35 34 7a 2d 3b 72 70 6f 72 74 0d 0a 4d 61 78 754z-;rp ort..Max 00b0 2d 46 6f 72 77 61 72 64 73 3a 20 37 30 0d 0a 43 -Forward s: 70..C 00c0 6f 6e 74 61 63 74 3a 20 3c 73 69 70 3a 31 32 33 ontact: sip:123 00d0 34 40 31 39 32 2e 31 36 38 2e 31 2e 31 31 36 3a *...@192.16*4...@192.16 * 8.1.116: 00e0 35 34 30 35 30 3b 72 69 6e 73 74 61 6e 63 65 3d 54050;ri nstance= 00f0 36 33 61 39 66 64 62 62 62 62 39 64 30 33 62 30 63a9fdbb bb9d03b0 0100 3e 0d 0a 54 6f 3a 20 22 67 69 61 6e 63 61 22 3c ..To: gianca 0110 73 69 70 3a 31 32 33 34 40 31 39 32 2e 31 36 38 sip:1234 @192.168 0120 2e 31 2e 31 30 30 3e 0d 0a 46 72 6f 6d 3a 20 22 .1.100. .From: 0130 67 69 61 6e 63 61 22 3c 73 69 70 3a 31 32 33 34 gianca sip:1234 0140 40 31 39 32 2e 31 36 38 2e 31 2e 31 30 30 3e 3b @192.168 .1.100; 0150 74 61 67 3d 65 34 35 64 65 35 36 62 0d 0a 43 61 tag=e45d e56b..Ca 0160 6c 6c 2d 49 44 3a 20 4e 47 49 33 59 6a 49 7a 4d ll-ID: N GI3YjIzM 0170 6a 49 79 4e 47 49 77 5a 54 6b 77 4d 54 63 35 5a jIyNGIwZ TkwMTc5Z 0180 47 49 77 4d 57 51 33 4d 57 5a 69 4f 57 4a 6b 4e GIwMWQ3M WZiOWJkN 0190 44 59 2e 0d 0a 43 53 65 71 3a 20 31 20 52 45 47 DY...CSe q: 1 REG 01a0 49 53 54 45 52 0d 0a 45 78 70 69 72 65 73 3a 20 ISTER..E xpires: 01b0 33 36 30 30 0d 0a 41 6c 6c 6f 77 3a 20 49 4e 56 3600..Al low: INV 01c0 49 54 45 2c 20 41 43 4b 2c 20 43 41 4e 43 45 4c ITE, ACK , CANCEL 01d0 2c 20 4f 50 54 49 4f 4e 53 2c 20 42 59 45 2c 20 , OPTION S, BYE, 01e0 52 45 46 45 52 2c 20 4e 4f 54 49 46 59 2c 20 4d REFER, N OTIFY, M 01f0 45 53 53 41 47 45 2c 20 53 55 42 53 43 52 49 42 ESSAGE, SUBSCRIB 0200 45 2c 20 49 4e 46 4f 0d 0a 55 73 65 72 2d 41 67 E, INFO. .User-Ag 0210 65 6e 74 3a 20 58 2d 4c 69 74 65 20 72 65 6c 65 ent: X-L ite rele 0220 61 73 65 20 31 31 30 33 6b 20 73 74 61 6d 70 20 ase 1103 k stamp 0230 35 33 36 32 31 0d 0a 43 6f 6e 74 65 6e 74 2d 4c 53621..C ontent-L 0240 65 6e 67 74 68 3a 20 30 0d 0a 0d 0a ength: 0 * * 00 17 c4 59 b6 56 00 0a e6 23 92 6b 08 00 45 00 ...Y.V.. .#.k..E. 0010 01 ea 0b 80 00 00 40 11 e9 5a c0 a8 01 64 c0 a8 ..@. .Z...d.. 0020 01 74 13 c4 d3 22 01 d6 10 53 53 49 50 2f 32 2e .t. .SSIP/2. 0030 30 20 34 30 34 20 4e 6f 74 20 66 6f 75 6e 64 0d 0 404 No t found. 0040 0a 56 69 61 3a 20 53 49 50 2f 32 2e 30 2f 55 44 .Via: SI P/2.0/UD 0050 50 20 31 39 32 2e 31 36 38 2e 31 2e 31 31 36 3a P 192.16 8.1.116: 0060 35 34 30 35 30 3b 62 72 61 6e 63 68 3d 7a 39 68 54050;br anch=z9h 0070 47 34 62 4b 2d 64 38 37 35 34 7a 2d 32 34 32 38 G4bK-d87 54z-2428 0080 38 65 37 32 38 32 36 64 30 31 32 38 2d 31 2d 2d 8e72826d 0128-1-- 0090 2d 64 38 37 35 34 7a 2d 3b 72 65 63 65 69 76 65 -d8754z- ;receive 00a0 64 3d 31 39 32 2e 31 36 38 2e 31 2e 31 31 36 3b d=192.16 8.1.116; 00b0 72 70 6f 72 74 3d 35 34 30 35 30 0d 0a 46 72 6f rport=54 050..Fro 00c0 6d 3a 20 22 67 69 61 6e 63 61 22 3c 73 69 70 3a m: gian casip: 00d0 31 32 33 34 40 31 39 32 2e 31 36 38 2e 31 2e 31 **1...@192*1...@192 * .168.1.1 00e0 30 30 3e 3b 74 61 67 3d 65 34 35 64 65 35 36 62 00;tag= e45de56b 00f0 0d 0a 54 6f 3a 20 22 67 69 61 6e 63 61 22 3c 73 ..To: g iancas 0100 69 70 3a 31 32 33 34 40 31 39 32 2e 31 36 38 2e ip:1234@ 192.168. 0110 31 2e 31 30 30 3e 3b 74 61 67 3d 61 73 36 35 35 1.100;t ag=as655 0120 64 36 66 31 32 0d 0a 43 61 6c 6c 2d 49 44 3a 20 d6f12..C all-ID: 0130 4e 47 49 33 59 6a 49 7a 4d 6a 49 79 4e 47 49 77 NGI3YjIz MjIyNGIw 0140 5a 54 6b 77 4d 54 63 35 5a 47 49 77 4d 57 51 33 ZTkwMTc5 ZGIwMWQ3 0150 4d 57 5a 69 4f 57 4a 6b 4e 44 59 2e 0d 0a 43 53 MWZiOWJk NDY...CS 0160 65 71 3a 20 31 20 52 45 47 49 53 54 45 52 0d 0a eq: 1 RE GISTER.. 0170 55 73 65 72 2d 41 67 65 6e 74 3a 20 41 73 74 65
Re: [asterisk-users] Failure of user registration with XLITE
Hello, Try this in X-Lite config section: /Display Name: gianca/ /Username: //gianca/ /Password: pwd_gianca/ /Authorization User Name: //gianca/ /Domain: 192.168.1.100 / Ahmed Ossama giancarlo lombardo wrote: Dear all, I'm setting up a connection via XLITE softphone and asterisk 1.4 but I get the error: /Registration error: 404 Not found/ Here my configuration file of asterisk: /[r...@dhcppc0 asterisk]# vi sip.conf [gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial/ /[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial/ /[r...@dhcppc0 asterisk]# vi extensions.conf [tutorial] exten = 1234,1,Dial(SIP,gianca)/ /exten = 12345,1,Dial(SIP,giusy) / Here the XLITE user data: /Display Name: gianca/ /Username: 1234/ /Password: pwd_gianca/ /Authorization User Name: 1234/ /Domain: 192.168.1.100/ Here the output of wireshark in between Xlite client and asterisk server: // /0040 2e 31 30 30 20 53 49 50 2f 32 2e 30 0d 0a 56 69 .100 SIP /2.0..Vi 0050 61 3a 20 53 49 50 2f 32 2e 30 2f 55 44 50 20 31 a: SIP/2 .0/UDP 1 0060 39 32 2e 31 36 38 2e 31 2e 31 31 36 3a 35 34 30 92.168.1 .116:540 0070 35 30 3b 62 72 61 6e 63 68 3d 7a 39 68 47 34 62 50;branc h=z9hG4b 0080 4b 2d 64 38 37 35 34 7a 2d 32 34 32 38 38 65 37 K-d8754z -24288e7 0090 32 38 32 36 64 30 31 32 38 2d 31 2d 2d 2d 64 38 2826d012 8-1---d8 00a0 37 35 34 7a 2d 3b 72 70 6f 72 74 0d 0a 4d 61 78 754z-;rp ort..Max 00b0 2d 46 6f 72 77 61 72 64 73 3a 20 37 30 0d 0a 43 -Forward s: 70..C 00c0 6f 6e 74 61 63 74 3a 20 3c 73 69 70 3a 31 32 33 ontact: sip:123 00d0 34 40 31 39 32 2e 31 36 38 2e 31 2e 31 31 36 3a //4...@192.16/ mailto:4...@192.16/ 8.1.116: 00e0 35 34 30 35 30 3b 72 69 6e 73 74 61 6e 63 65 3d 54050;ri nstance= 00f0 36 33 61 39 66 64 62 62 62 62 39 64 30 33 62 30 63a9fdbb bb9d03b0 0100 3e 0d 0a 54 6f 3a 20 22 67 69 61 6e 63 61 22 3c ..To: gianca 0110 73 69 70 3a 31 32 33 34 40 31 39 32 2e 31 36 38 sip:1234 @192.168 0120 2e 31 2e 31 30 30 3e 0d 0a 46 72 6f 6d 3a 20 22 .1.100. .From: 0130 67 69 61 6e 63 61 22 3c 73 69 70 3a 31 32 33 34 gianca sip:1234 0140 40 31 39 32 2e 31 36 38 2e 31 2e 31 30 30 3e 3b @192.168 .1.100; 0150 74 61 67 3d 65 34 35 64 65 35 36 62 0d 0a 43 61 tag=e45d e56b..Ca 0160 6c 6c 2d 49 44 3a 20 4e 47 49 33 59 6a 49 7a 4d ll-ID: N GI3YjIzM 0170 6a 49 79 4e 47 49 77 5a 54 6b 77 4d 54 63 35 5a jIyNGIwZ TkwMTc5Z 0180 47 49 77 4d 57 51 33 4d 57 5a 69 4f 57 4a 6b 4e GIwMWQ3M WZiOWJkN 0190 44 59 2e 0d 0a 43 53 65 71 3a 20 31 20 52 45 47 DY...CSe q: 1 REG 01a0 49 53 54 45 52 0d 0a 45 78 70 69 72 65 73 3a 20 ISTER..E xpires: 01b0 33 36 30 30 0d 0a 41 6c 6c 6f 77 3a 20 49 4e 56 3600..Al low: INV 01c0 49 54 45 2c 20 41 43 4b 2c 20 43 41 4e 43 45 4c ITE, ACK , CANCEL 01d0 2c 20 4f 50 54 49 4f 4e 53 2c 20 42 59 45 2c 20 , OPTION S, BYE, 01e0 52 45 46 45 52 2c 20 4e 4f 54 49 46 59 2c 20 4d REFER, N OTIFY, M 01f0 45 53 53 41 47 45 2c 20 53 55 42 53 43 52 49 42 ESSAGE, SUBSCRIB 0200 45 2c 20 49 4e 46 4f 0d 0a 55 73 65 72 2d 41 67 E, INFO. .User-Ag 0210 65 6e 74 3a 20 58 2d 4c 69 74 65 20 72 65 6c 65 ent: X-L ite rele 0220 61 73 65 20 31 31 30 33 6b 20 73 74 61 6d 70 20 ase 1103 k stamp 0230 35 33 36 32 31 0d 0a 43 6f 6e 74 65 6e 74 2d 4c 53621..C ontent-L 0240 65 6e 67 74 68 3a 20 30 0d 0a 0d 0a ength: 0 / / 00 17 c4 59 b6 56 00 0a e6 23 92 6b 08 00 45 00 ...Y.V.. .#.k..E. 0010 01 ea 0b 80 00 00 40 11 e9 5a c0 a8 01 64 c0 a8 ..@. .Z...d.. 0020 01 74 13 c4 d3 22 01 d6 10 53 53 49 50 2f 32 2e .t. .SSIP/2. 0030 30 20 34 30 34 20 4e 6f 74 20 66 6f 75 6e 64 0d 0 404 No t found. 0040 0a 56 69 61 3a 20 53 49 50 2f 32 2e 30 2f 55 44 .Via: SI P/2.0/UD 0050 50 20 31 39 32 2e 31 36 38 2e 31 2e 31 31 36 3a P 192.16 8.1.116: 0060 35 34 30 35 30 3b 62 72 61 6e 63 68 3d 7a 39 68 54050;br anch=z9h 0070 47 34 62 4b 2d 64 38 37 35 34 7a 2d 32 34 32 38 G4bK-d87 54z-2428 0080 38 65 37 32 38 32 36 64 30 31 32 38 2d 31 2d 2d 8e72826d 0128-1-- 0090 2d 64 38 37 35 34 7a 2d 3b 72 65 63 65 69 76 65 -d8754z- ;receive 00a0 64 3d 31 39 32 2e 31 36 38 2e 31 2e 31 31 36 3b d=192.16 8.1.116; 00b0 72 70 6f 72 74 3d 35 34 30 35 30 0d 0a 46 72 6f rport=54 050..Fro 00c0 6d 3a 20 22 67 69 61 6e 63 61 22 3c 73 69 70 3a m: gian casip: 00d0 31 32 33 34 40 31 39 32 2e 31 36 38 2e 31 2e 31 //1...@192/ mailto:1...@192/ .168.1.1 00e0 30 30 3e 3b 74 61 67 3d 65 34 35 64 65 35 36 62 00;tag= e45de56b 00f0 0d 0a 54 6f 3a 20 22 67 69 61 6e 63 61 22 3c 73 ..To: g iancas 0100 69 70 3a 31 32 33 34 40 31 39 32 2e 31 36 38 2e ip:1234@ 192.168. 0110 31 2e 31 30 30 3e 3b 74 61 67 3d 61 73 36 35 35 1.100;t ag=as655 0120 64 36 66 31 32 0d 0a 43 61 6c 6c 2d 49 44 3a 20 d6f12..C all-ID: 0130 4e 47 49 33 59 6a 49 7a
[asterisk-users] CDR userfield not written into DB
Hi everybody, i've been googling for quite some time now but can't find an answer to my problem... I'm using Asterisk 1.2.12.1 with mysql as the cdr backend. In the dialplan i've written exten = 1234,n,Set(CDR(userfield)=blah) exten = 1234,n,Answer() exten = 1234,n,Queue(.) exten = 1234,n,Hangup() When I'm doing a call I can see that the statement is executed. But when the call finishes, a cdr is written into the DB with an empty 'userfield'. I'm sure, I'm missing something but can't figure out, what... Thanks Norbert ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hi
Hi, Un-top-posting, aster...@opensourcesolution.in wrote: On Sun, Nov 08, 2009 at 01:04:31PM +0100, giancarlo lombardo wrote: i gave that command which u told i.e asterisk -V. the output is below [r...@localhost ~]# cd /etc/ [r...@localhost etc]# asterisk -v bash: asterisk: command not found [r...@localhost etc]# asterisk -V bash: asterisk: command not found try as below (in bold the command) *[r...@dhcppc0 asterisk]# pwd /etc/asterisk* [r...@dhcppc0 asterisk]# *asterisk -vr* My guess for the output of that command: bash: asterisk: command not found But if I strictly followed your advice: $ *asterisk -vr* bash: *asterisk: command not found If this is installed from source: ./main/asterisk -V in the source directory. Just in case it was not installed. In this case, ask yourself why wasn't it installed. What is the output of: echo $PATH What Linux distributribution is it? If not Linux: What OS is it? What version? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-stat! - help needed (once again due to mailserver problem)
Thanks Matt,
Now it works:) - I just switch off notices. When the notices where on, the
program didn't work. Beside the obvious lines with notices, whatever I
entered in the form it always displayed Coming Soon page instead of the
calls data.
Thanks again,
pepesz
On Fri, Nov 6, 2009 at 7:15 AM, Matt Riddell li...@venturevoip.com wrote:
On 26/10/09 3:47 AM, Lukasz Pakula wrote:
Dear all,
I'm trying to install Asterisk-stat (ASTERISK CDR ANALYSER) following:
http://www.voip-info.org/wiki/index.php?page=Asterisk+CDR+Areski+GUI
however it fails to run properly - lots of lines like:
*Notice*: Undefined variable: s in
*/home/lukasz/DATA/www/asterisk-stat/cdr.php* on line *26*
*Notice*: Undefined variable: t in
*/home/lukasz/DATA/www/asterisk-stat/cdr.php* on line *27*
That's not an error - it's a notice - it means you have error_reporting
set to E_ALL in php.ini.
Depending on which version of Linux you use the file could be in a few
places.
If you are using Debian it would be in:
/etc/php5/apache2/php.ini
You'll need to restart Apache after changing the setting.
If you're brave you could surround the lines creating the problem with:
if (isset($s)) {
// Do something with $s
}
(replacing the commented line // with the line in question)
--
Cheers,
Matt Riddell
Director
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Re: [asterisk-users] CDR userfield not written into DB
Norbert Zawodsky schrieb: I'm using Asterisk 1.2.12.1 with mysql as the cdr backend. In the dialplan i've written exten = 1234,n,Set(CDR(userfield)=blah) exten = 1234,n,Answer() exten = 1234,n,Queue(.) exten = 1234,n,Hangup() When I'm doing a call I can see that the statement is executed. But when the call finishes, a cdr is written into the DB with an empty 'userfield'. I'm sure, I'm missing something but can't figure out, what... /etc/asterisk/cdr_mysql.conf : [global] userfield=1 ... Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hi
Ciao, try /etc/asterisk/asterisk -vr 2009/11/8 Alex Balashov abalas...@evaristesys.com Try /usr/sbin/asterisk. Also, copy the list. Don't email me privately. aster...@opensourcesolution.in wrote: hi friend, i gave that command which u told i.e asterisk -V. the output is below [r...@localhost ~]# cd /etc/ [r...@localhost etc]# asterisk -v bash: asterisk: command not found [r...@localhost etc]# asterisk -V bash: asterisk: command not found thx -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giancarlo Lombardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR userfield not written into DB
Norbert Zawodsky wrote: Hi everybody, i've been googling for quite some time now but can't find an answer to my problem... My first hit on Google: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg147997.html exten = 1234,n,Set(CDR(userfield)=blah) This is version 1.4.x format. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR userfield not written into DB
Philipp Kempgen schrieb: Norbert Zawodsky schrieb: I'm using Asterisk 1.2.12.1 with mysql as the cdr backend. In the dialplan i've written exten = 1234,n,Set(CDR(userfield)=blah) exten = 1234,n,Answer() exten = 1234,n,Queue(.) exten = 1234,n,Hangup() When I'm doing a call I can see that the statement is executed. But when the call finishes, a cdr is written into the DB with an empty 'userfield'. I'm sure, I'm missing something but can't figure out, what... /etc/asterisk/cdr_mysql.conf : [global] userfield=1 ... Philipp Kempgen Oh god! That simple Sorry!! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help in installing asterisk
aster...@opensourcesolution.in schrieb: when i am compiling asterisk-1.4.26.3, i am getting errors of dependency. Well, install the dependencies first. :-) What exactly does it complain about? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failure of user registration with XLITE
Ciao, the problem is still present, does anyone have some other suggestion ? Below the output of CLI with debug option on XLITE IP and show peers command: *dhcppc0*CLI --- SIP read from 192.168.1.116:14166 --- REGISTER sip:192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14166 ;branch=z9hG4bK-d8754z-4d4ced5bca35b64c-1---d8754z-;rport Max-Forwards: 70 Contact: sip:1...@192.168.1.116:14166;rinstance=c18a16f442f17333 To: giancasip:1...@192.168.1.100 sip%3a1...@192.168.1.100 From: giancasip:1...@192.168.1.100 sip%3a1...@192.168.1.100 ;tag=be7e8a36 Call-ID: YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 0* *- --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.1.116 : 14166 (NAT)* *--- Transmitting (NAT) to 192.168.1.116:14166 --- SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.1.116:14166 ;branch=z9hG4bK-d8754z-4d4ced5bca35b64c-1---d8754z-;received=192.168.1.116;rport=14166 From: giancasip:1...@192.168.1.100 sip%3a1...@192.168.1.100 ;tag=be7e8a36 To: giancasip:1...@192.168.1.100 sip%3a1...@192.168.1.100 ;tag=as0194534b Call-ID: YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE. CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0* * Scheduling destruction of SIP dialog 'YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE.' in 32000 ms (Method: REGISTER) Really destroying SIP dialog 'YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE.' Method: REGISTER dhcppc0*CLI sip show peers Name/username HostDyn Nat ACL Port Status giusy/giusy(Unspecified)D 0Unmonitored gianca/gianca (Unspecified)D 0Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] dhcppc0*CLI* 2009/11/8 Ahmed Ossama ah...@master-zone.net Hello, Try this in X-Lite config section: /Display Name: gianca/ /Username: //gianca/ /Password: pwd_gianca/ /Authorization User Name: //gianca/ /Domain: 192.168.1.100 / Ahmed Ossama giancarlo lombardo wrote: Dear all, I'm setting up a connection via XLITE softphone and asterisk 1.4 but I get the error: /Registration error: 404 Not found/ Here my configuration file of asterisk: /[r...@dhcppc0 asterisk]# vi sip.conf [gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial/ /[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial/ /[r...@dhcppc0 asterisk]# vi extensions.conf [tutorial] exten = 1234,1,Dial(SIP,gianca)/ /exten = 12345,1,Dial(SIP,giusy) / Here the XLITE user data: /Display Name: gianca/ /Username: 1234/ /Password: pwd_gianca/ /Authorization User Name: 1234/ /Domain: 192.168.1.100/ Here the output of wireshark in between Xlite client and asterisk server: // /0040 2e 31 30 30 20 53 49 50 2f 32 2e 30 0d 0a 56 69 .100 SIP /2.0..Vi 0050 61 3a 20 53 49 50 2f 32 2e 30 2f 55 44 50 20 31 a: SIP/2 .0/UDP 1 0060 39 32 2e 31 36 38 2e 31 2e 31 31 36 3a 35 34 30 92.168.1 .116:540 0070 35 30 3b 62 72 61 6e 63 68 3d 7a 39 68 47 34 62 50;branc h=z9hG4b 0080 4b 2d 64 38 37 35 34 7a 2d 32 34 32 38 38 65 37 K-d8754z -24288e7 0090 32 38 32 36 64 30 31 32 38 2d 31 2d 2d 2d 64 38 2826d012 8-1---d8 00a0 37 35 34 7a 2d 3b 72 70 6f 72 74 0d 0a 4d 61 78 754z-;rp ort..Max 00b0 2d 46 6f 72 77 61 72 64 73 3a 20 37 30 0d 0a 43 -Forward s: 70..C 00c0 6f 6e 74 61 63 74 3a 20 3c 73 69 70 3a 31 32 33 ontact: sip:123 00d0 34 40 31 39 32 2e 31 36 38 2e 31 2e 31 31 36 3a //4...@192.16/ mailto:4...@192.16/ 8.1.116: 00e0 35 34 30 35 30 3b 72 69 6e 73 74 61 6e 63 65 3d 54050;ri nstance= 00f0 36 33 61 39 66 64 62 62 62 62 39 64 30 33 62 30 63a9fdbb bb9d03b0 0100 3e 0d 0a 54 6f 3a 20 22 67 69 61 6e 63 61 22 3c ..To: gianca 0110 73 69 70 3a 31 32 33 34 40 31 39 32 2e 31 36 38 sip:1234 @192.168 0120 2e 31 2e 31 30 30 3e 0d 0a 46 72 6f 6d 3a 20 22 .1.100. .From: 0130 67 69 61 6e 63 61 22 3c 73 69 70 3a 31 32 33 34 gianca sip:1234 0140 40 31 39 32 2e 31 36 38 2e 31 2e 31 30 30 3e 3b @192.168 .1.100; 0150 74 61 67 3d 65 34 35 64 65 35 36 62 0d 0a 43 61 tag=e45d e56b..Ca 0160 6c 6c 2d 49 44 3a 20 4e 47 49 33 59 6a 49 7a 4d ll-ID: N GI3YjIzM 0170 6a 49 79 4e 47 49 77 5a 54 6b 77 4d 54 63 35 5a jIyNGIwZ TkwMTc5Z 0180 47 49 77 4d 57 51 33 4d 57 5a 69 4f 57 4a 6b 4e GIwMWQ3M WZiOWJkN 0190 44 59 2e 0d 0a 43 53 65 71 3a 20 31 20 52 45 47 DY...CSe q: 1 REG 01a0 49 53 54 45 52 0d 0a 45 78 70 69 72 65 73 3a 20 ISTER..E xpires: 01b0 33 36 30 30 0d 0a 41 6c 6c 6f 77 3a 20 49 4e 56 3600..Al low: INV 01c0 49
Re: [asterisk-users] Help with concurrent VoIP calls
Hi Joe, What is the app that generates your bandwidth table shown below? Joe Greco wrote: By fast I mean the best Business DSL Bellsouth has to offer: Up to 6.0 Mbps downstream - Up to 512 Kbps upstream That almost sounds like an invitation to check out what business service your cableco offers. One thing to be aware of with DSL and cable modems is that there can be various ill effects as your line gets closer to its rated capacity; do not expect that you'll get a reliable 512Kbps upstream. VoIP is sensitive to loss, latency, and jitter. You may be able, for example, to only get 384Kbps reliably out of the link (before packet loss/jitter/etc wreck its suitability for VoIP). That's a good time to look seriously at a gateway package like pfSense that can prioritize certain classes of traffic while also limiting overall bandwidth. As an example, we noticed on the local business cable offering (2Mbps up) Shaped PL min avg max stddev 2.2M3 6.4 251 557 176 2.1M1 7.8 350 584 134 2.0M3 6.4 271 535 132 1.9M1 7 254 527 131 1.8M0 6 79 339 90 1.75M 0 5.9 14 92 11 1.7M0 5.4 13 77 10 1.65M 0 4.9 11 69 7 1.6M0 5.4 13 55 9 1.5M0 5.3 11 59 7 1.4M0 5 11 57 7 1.3M0 4.9 11 54 6 1.2M0 4.9 11 52 7 1.1M0 4.8 14 53 11 The max starts trending up after 1.6M (helps to graph it) and pretty much everything goes to hell after 1.75M. ... JG ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Modem card
I have a conexant modem card a-link internal pci-card Will it work with asterisk? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outbound routing
I have 2 questions: 1. Can I make outbound route rule based on the Source Channel? 2. Can I auto change the outbound route based on time/Day of week? Any help very appreciated.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound routing
-- Sent from mobile device On Nov 8, 2009, at 2:13 PM, B.Masoud @ SH i...@saudihome.com wrote: I have 2 questions: 1. Can I make outbound route rule based on the Source Channel? Yes. 2. Can I auto change the outbound route based on time/Day of week? Yes. See GotoIfTime(). Any help very appreciated.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failure of user registration with XLITE
/[r...@dhcppc0 asterisk]# vi extensions.conf [tutorial] exten = 1234,1,Dial(SIP,gianca)/ /exten = 12345,1,Dial(SIP,giusy) / Here the XLITE user data: /Display Name: gianca/ /Username: 1234/ /Password: pwd_gianca/ /Authorization User Name: 1234/ /Domain: 192.168.1.100/ http://192.168.1.100/ Your XLITE user name should be the same as the sip account name(gianca not 1234). And the extensions.conf should be: exten = 1234,1,Dial(SIP/gianca) giancarlo lombardo wrote: Ciao, the problem is still present, does anyone have some other suggestion ? Below the output of CLI with debug option on XLITE IP and show peers command: /dhcppc0*CLI --- SIP read from 192.168.1.116:14166 http://192.168.1.116:14166 --- REGISTER sip:192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14166;branch=z9hG4bK-d8754z-4d4ced5bca35b64c-1---d8754z-;rport Max-Forwards: 70 Contact: sip:1...@192.168.1.116:14166;rinstance=c18a16f442f17333 To: giancasip:1...@192.168.1.100 mailto:sip%3a1...@192.168.1.100 From: giancasip:1...@192.168.1.100 mailto:sip%3a1...@192.168.1.100;tag=be7e8a36 Call-ID: YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 0/ /- --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.1.116 : 14166 (NAT)/ /--- Transmitting (NAT) to 192.168.1.116:14166 http://192.168.1.116:14166 --- SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.1.116:14166;branch=z9hG4bK-d8754z-4d4ced5bca35b64c-1---d8754z-;received=192.168.1.116;rport=14166 From: giancasip:1...@192.168.1.100 mailto:sip%3a1...@192.168.1.100;tag=be7e8a36 To: giancasip:1...@192.168.1.100 mailto:sip%3a1...@192.168.1.100;tag=as0194534b Call-ID: YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE. CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0/ / Scheduling destruction of SIP dialog 'YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE.' in 32000 ms (Method: REGISTER) Really destroying SIP dialog 'YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE.' Method: REGISTER dhcppc0*CLI sip show peers Name/username HostDyn Nat ACL Port Status giusy/giusy(Unspecified)D 0 Unmonitored gianca/gianca (Unspecified)D 0 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] dhcppc0*CLI/ 2009/11/8 Ahmed Ossama ah...@master-zone.net mailto:ah...@master-zone.net Hello, Try this in X-Lite config section: /Display Name: gianca/ /Username: //gianca/ /Password: pwd_gianca/ /Authorization User Name: //gianca/ /Domain: 192.168.1.100 / Ahmed Ossama giancarlo lombardo wrote: Dear all, I'm setting up a connection via XLITE softphone and asterisk 1.4 but I get the error: /Registration error: 404 Not found/ Here my configuration file of asterisk: /[r...@dhcppc0 asterisk]# vi sip.conf [gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial/ /[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial/ /[r...@dhcppc0 asterisk]# vi extensions.conf [tutorial] exten = 1234,1,Dial(SIP,gianca)/ /exten = 12345,1,Dial(SIP,giusy) / Here the XLITE user data: /Display Name: gianca/ /Username: 1234/ /Password: pwd_gianca/ /Authorization User Name: 1234/ /Domain: 192.168.1.100/ http://192.168.1.100/ Here the output of wireshark in between Xlite client and asterisk server: // /0040 2e 31 30 30 20 53 49 50 2f 32 2e 30 0d 0a 56 69 .100 SIP /2.0..Vi 0050 61 3a 20 53 49 50 2f 32 2e 30 2f 55 44 50 20 31 a: SIP/2 .0/UDP 1 0060 39 32 2e 31 36 38 2e 31 2e 31 31 36 3a 35 34 30 92.168.1 .116:540 0070 35 30 3b 62 72 61 6e 63 68 3d 7a 39 68 47 34 62 50;branc h=z9hG4b 0080 4b 2d 64 38 37 35 34 7a 2d 32 34 32 38 38 65 37 K-d8754z -24288e7 0090 32 38 32 36 64 30 31 32 38 2d 31 2d 2d 2d 64 38 2826d012 8-1---d8 00a0 37 35 34 7a 2d 3b 72 70 6f 72 74 0d 0a 4d 61 78 754z-;rp ort..Max 00b0 2d 46 6f 72 77 61 72 64 73 3a 20 37 30 0d 0a 43 -Forward s: 70..C 00c0 6f 6e 74 61 63 74 3a 20 3c 73 69 70 3a 31 32 33 ontact: sip:123 00d0 34 40 31 39 32 2e 31 36 38 2e 31 2e 31 31 36 3a //4...@192.16/ mailto:4...@192.16 mailto:4...@192.16/ 8.1.116: 00e0 35 34 30 35 30 3b 72 69 6e 73 74 61 6e 63 65 3d 54050;ri nstance= 00f0 36 33 61 39 66 64 62 62 62 62 39
Re: [asterisk-users] outbound routing
Can you tell me how on the first question? Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Sunday, November 08, 2009 10:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] outbound routing -- Sent from mobile device On Nov 8, 2009, at 2:13 PM, B.Masoud @ SH i...@saudihome.com wrote: I have 2 questions: 1. Can I make outbound route rule based on the Source Channel? Yes. 2. Can I auto change the outbound route based on time/Day of week? Yes. See GotoIfTime(). Any help very appreciated.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound routing
Contexts. Put the 'Source channels' in different contexts. Lyle B.Masoud @ SH wrote: Can you tell me how on the first question? Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Sunday, November 08, 2009 10:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] outbound routing -- Sent from mobile device On Nov 8, 2009, at 2:13 PM, B.Masoud @ SH i...@saudihome.com wrote: I have 2 questions: 1. Can I make outbound route rule based on the Source Channel? Yes. 2. Can I auto change the outbound route based on time/Day of week? Yes. See GotoIfTime(). Any help very appreciated.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failure of user registration with XLITE
Thanks, it works !!! 2009/11/8 Lyle Giese l...@lcrcomputer.net /[r...@dhcppc0 asterisk]# vi extensions.conf [tutorial] exten = 1234,1,Dial(SIP,gianca)/ /exten = 12345,1,Dial(SIP,giusy) / Here the XLITE user data: /Display Name: gianca/ /Username: 1234/ /Password: pwd_gianca/ /Authorization User Name: 1234/ /Domain: 192.168.1.100/ Your XLITE user name should be the same as the sip account name(gianca not 1234). And the extensions.conf should be: exten = 1234,1,Dial(SIP/gianca) giancarlo lombardo wrote: Ciao, the problem is still present, does anyone have some other suggestion ? Below the output of CLI with debug option on XLITE IP and show peers command: *dhcppc0*CLI --- SIP read from 192.168.1.116:14166 --- REGISTER sip:192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14166 ;branch=z9hG4bK-d8754z-4d4ced5bca35b64c-1---d8754z-;rport Max-Forwards: 70 Contact: sip:1...@192.168.1.116:14166;rinstance=c18a16f442f17333sip:1...@192.168.1.116:14166;rinstance=c18a16f442f17333 To: giancasip:1...@192.168.1.100 sip%3a1...@192.168.1.100 From: giancasip:1...@192.168.1.100 sip%3a1...@192.168.1.100 ;tag=be7e8a36 Call-ID: YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 0* *- --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.1.116 : 14166 (NAT)* *--- Transmitting (NAT) to 192.168.1.116:14166 --- SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.1.116:14166 ;branch=z9hG4bK-d8754z-4d4ced5bca35b64c-1---d8754z-;received=192.168.1.116;rport=14166 From: giancasip:1...@192.168.1.100 sip%3a1...@192.168.1.100 ;tag=be7e8a36 To: giancasip:1...@192.168.1.100 sip%3a1...@192.168.1.100 ;tag=as0194534b Call-ID: YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE. CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0* * Scheduling destruction of SIP dialog 'YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE.' in 32000 ms (Method: REGISTER) Really destroying SIP dialog 'YTMxMzY0OTJiOTczNjlmNzZkNzEzMTE2N2FmM2E3NmE.' Method: REGISTER dhcppc0*CLI sip show peers Name/username HostDyn Nat ACL Port Status giusy/giusy(Unspecified)D 0Unmonitored gianca/gianca (Unspecified)D 0Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] dhcppc0*CLI* 2009/11/8 Ahmed Ossama ah...@master-zone.net Hello, Try this in X-Lite config section: /Display Name: gianca/ /Username: //gianca/ /Password: pwd_gianca/ /Authorization User Name: //gianca/ /Domain: 192.168.1.100 / Ahmed Ossama giancarlo lombardo wrote: Dear all, I'm setting up a connection via XLITE softphone and asterisk 1.4 but I get the error: /Registration error: 404 Not found/ Here my configuration file of asterisk: /[r...@dhcppc0 asterisk]# vi sip.conf [gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial/ /[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial/ /[r...@dhcppc0 asterisk]# vi extensions.conf [tutorial] exten = 1234,1,Dial(SIP,gianca)/ /exten = 12345,1,Dial(SIP,giusy) / Here the XLITE user data: /Display Name: gianca/ /Username: 1234/ /Password: pwd_gianca/ /Authorization User Name: 1234/ /Domain: 192.168.1.100/ Here the output of wireshark in between Xlite client and asterisk server: // /0040 2e 31 30 30 20 53 49 50 2f 32 2e 30 0d 0a 56 69 .100 SIP /2.0..Vi 0050 61 3a 20 53 49 50 2f 32 2e 30 2f 55 44 50 20 31 a: SIP/2 .0/UDP 1 0060 39 32 2e 31 36 38 2e 31 2e 31 31 36 3a 35 34 30 92.168.1 .116:540 0070 35 30 3b 62 72 61 6e 63 68 3d 7a 39 68 47 34 62 50;branc h=z9hG4b 0080 4b 2d 64 38 37 35 34 7a 2d 32 34 32 38 38 65 37 K-d8754z -24288e7 0090 32 38 32 36 64 30 31 32 38 2d 31 2d 2d 2d 64 38 2826d012 8-1---d8 00a0 37 35 34 7a 2d 3b 72 70 6f 72 74 0d 0a 4d 61 78 754z-;rp ort..Max 00b0 2d 46 6f 72 77 61 72 64 73 3a 20 37 30 0d 0a 43 -Forward s: 70..C 00c0 6f 6e 74 61 63 74 3a 20 3c 73 69 70 3a 31 32 33 ontact: sip:123 00d0 34 40 31 39 32 2e 31 36 38 2e 31 2e 31 31 36 3a //4...@192.16/ mailto:4...@192.16/ 8.1.116: 00e0 35 34 30 35 30 3b 72 69 6e 73 74 61 6e 63 65 3d 54050;ri nstance= 00f0 36 33 61 39 66 64 62 62 62 62 39 64 30 33 62 30 63a9fdbb bb9d03b0 0100 3e 0d 0a 54 6f 3a 20 22 67 69 61 6e 63 61 22 3c ..To: gianca 0110 73 69 70 3a 31 32 33 34 40 31 39 32 2e 31 36 38 sip:1234 @192.168 0120 2e 31 2e 31 30 30 3e 0d 0a 46 72 6f 6d 3a 20 22 .1.100. .From: 0130 67 69 61 6e 63 61 22 3c 73 69 70 3a 31 32
[asterisk-users] E1 connectivity problem (HDB3, CRC4MF, ISUP, V3)
Hi All; We are doing a configuration to link with another Simens switch via E1, they gave us these paramters to be setted, but we are facing an error at the trunks related to Asyn problem, look like something related to synchorinzation. The simens paramters are: Line Coding: HDB3 Country Protocl: ISUP Protocol Version: V3 CRC4: CRC4MF Interface Companding (like alaw and so on): a-LAW Impedance: 75 ohm Interworking Message (like PROGress or what): Basically support interworking messages as per V3 Actually, I do not know how to configure the ISUP and the V3 on my side? Any help? Also, what is the difference between CRC4MF and CRC4? Any specific thing need to be considered? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR userfield -
Set(CDR(userfield|r)=blah) This works for me on 1.4.24 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
