[asterisk-users] SIP disconnects after 20 seconds behind NAT
Hi, I have an asterisk server sitting behind a pfsense firewall, I have successfully configured pfsense for NAT traversal, and clients from the internet can call clients inside the network of asterisk, as well as other clients registered with this asterisk server on the internet. The problem now is when a client from the internet do a call, the call disconnects in 10~20 seconds, but during this period the call goes fine and voice is heard on both ends; But when a client on the same network of asterisk calls another client registered from the internet, the call is established without any issues, and it doesn't disconnect. I have also noticed that when internet clients do calls, and the call is established on both ends, if one of the two parties hang up, the other end isn't notified and the call stays opened at this end. I could provide config files if needed. Please advice about resolving this issue. Ahmed Ossama -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to register a sip account with x-lite
Hi,
Tim Culhane wrote:
Hi,
I'm running Asterisk 1.6.2.2 on a linux red hat machine.
I'm attempting to connect to asterisk using x-lite, but always get the
'401 unauthorized' error back from the server.
As far as I know I have my sip.conf and extensions.conf configured
correctly and I've tried changing lots of things but am still getting
nowhere.
In my sip.conf I have:
[user1]
type=friend
secret=user1
callerid=user1
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=internal
disallow=all
allow=gsm
allow=ulaw
allow=alaw
This got no issues.
Note I've tried nat=no and qualify=no but it doesn't change anything.
Extensions.conf has:
[internal]
exten = user1,1,Dial(SIP/user1,10,r)
exten = 1359,1,Dial(SIP/user1,10,r)
exten = user1,hint,SIP/user1
If I could even get some debug from Asterisk showing me why the user is
unauthorized that would be great. I have added verbose and debug to the
messages line in logger.conf and also set 'sip set debug on'. But I
can't find any debug output anywhere. The messages file just contains
start up messages for asterisk and no debug output. I can't find any sip
debug output anywhere.
Anybody know what I can do to try and isolate my configuration problems?
How can I get debug trace from Asterisk showing the connection coming in
from x-lite?
As for x-lite, you should use user1 in User name, Password and
Authorization user name
Many thanks,
Tim
-
Tim Culhane,
Critical Path Ireland,
42-47 Lower Mount Street,
Dublin 2.
Direct line: 353-1-2415107
phone: 353-1-2415000
tim.culh...@criticalpath.net
http://www.criticalpath.net
Critical Path
a global leader in digital communications
--
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Re: [asterisk-users] Denying call transfer to certain extensions
But this won't help if 100 or 101 wants to call 102. What I want is, if a call coming from a trunk 100 rings, and if the caller wants to be transfered to 101, the transfer is denied. In other words, 101 can't get transfered calls. Danny Nicholas wrote: Follow-me will most likely be your best bet for this trick. Say you have extensions 100, 101 and 102. 100 is the receptionist, 101 is sales and 102 is the boss, who doesn't want to be disturbed. If you set up followme on 102 to go to voicemail or whatever, 102 won't ring. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Ossama Sent: Monday, February 22, 2010 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Denying call transfer to certain extensions Hi all, Is there a way to deny call transfers to certain extensions? Thanks, Ahmed Ossama -- _ -- 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] Denying call transfer to certain extensions
Hi all, Is there a way to deny call transfers to certain extensions? Thanks, Ahmed Ossama -- _ -- 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] chan_mobile Voice setting
Hello all, I have successfully paired my mobile with asterisk, and chan_mobile already run very well, but sometimes when i restart asterisk chan_mobile fails to initialize with the error: chan_mobile.c: Incorrect voice setting for adapter toshiba, it must be 0x0060 - see 'man hciconfig' for details. I have tried several bluetooth adapters, as well as setting Class in /etc/bluetooth/main.conf to: 0x3e0100, 0x000100 and 0x006000 but the issue still happens, it usually pairs, but sometimes fail with the error mentioned. Thanks in advance, Ahmed Ossama ___ -- 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
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
