[asterisk-users] Development on top of freePbx Gui and AsteriskNow
Dear all, I have to develop and integrate an own application with AsteriskNOW. So create table, access them, do some action from asterisk freepbx GUI and use my data inside dialplan (e.g: to choice if a number can be dialed or no) Can someone suggest which technologies are available or link some documentation ? Thanks in advance. -- 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] looking for an Asterisk supervision (status viewer) tool
Ciao, may be is enough the Free PBX admi web installed with standard Asterisk now distribution. Real time status is for sure available. 2009/11/11 Klaus Darilion klaus.mailingli...@pernau.at Hi Christina! From documentation it seems it only supports queues and agents. I do not have a single queue nor agents. Does it also support real-time status for normal SIP-SIP calls? regards klaus Christina Casey schrieb: Hi Klaus, Yes all the below is possible/easy with the OrderlyStats call centre management and reporting tool. It's a free download - please see http://www.orderlyq.com/orderlystats.html Kind regards, Christina Casey Accounts Manager Orderly Software Ltd. Subject: [asterisk-users] looking for an Asterisk supervision (status viewer) tool From: Klaus Darilion klaus.mailingli...@pernau.at Date: Tue, 10 Nov 2009 14:04:16 +0100 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi! I am looking for a tool (application or webinterface) which shows me the current status of an Asterisk server, e.g.: - Status of the SIP peers (registered/offline) - current incoming and outgoing calls - start-time, numbers, some history - history (calls stopped in the last 15 minutes, who hang up?) - should be possible to link those calls to the relevant SIP peers - kill calls Before coding it myself, is there something you can recommend to me? The thing should be complete auto configured, e.g. no configuration file which peers/channels to be displayed, just fetch all the configuration from Asterisk and display it. thanks klaus ___ -- 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 -- 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] Bad quality of call
Hi all, I did some call using an asterisk 1.4 PBX and 2 softphone in a private network; call is up, but with bad quality. Someone knows how to debug this problem ? Thanks in advance for any help. -- 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] Fwd: Bad quality of call
Dear all, I'm not sure that mail was correctly delivered. Best Regards Giancarlo Lombardo -- Forwarded message -- From: giancarlo lombardo gianclomba...@gmail.com Date: 2009/11/11 Subject: Bad quality of call To: asterisk-users@lists.digium.com Hi all, I did some call using an asterisk 1.4 PBX and 2 softphone in a private network; call is up, but with bad quality. Someone knows how to debug this problem ? Thanks in advance for any help. -- Giancarlo Lombardo -- 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] Fwd: user extension in asterisk GUI
Hi all, I'm not sure that this mail was received. Thanks again -- Forwarded message -- From: giancarlo lombardo gianclomba...@gmail.com Date: 2009/11/10 Subject: user extension in asterisk GUI To: asterisk-users@lists.digium.com Hi all, I just configured some user in sip.conf and extensions.conf; they works fine. Now I'm trying to do the same with Extensions feature of FreePBXAdministration, but I cannot see what I have done manualy. Is gui working with other an source, How can I access such data ? Thanks in advance for any suggestion. -- Giancarlo Lombardo -- 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] Bad quality of call
Dear all, thanks for the suggestion; it seems to be a problem of my speaker; in fact I have same problem using skype 2009/11/11 Michael Wyres mwy...@cdm.com.au The reasons for poor call quality are many and varied. As another poster suggested, the headset you are using might be poorly configured, or just a poor example. An under-spec server could also do it – I use two simple, low-spec Virtual Machines in my dev lab that I bring up when I want to test dialplan variations, without interfering with the live systems. Quality is usually terrible because they are underpowered, but I’m only quickly testing dialplans, so I’m not concerned about the quality in those instances. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *giancarlo lombardo *Sent:* Thursday, 12 November 2009 01:02 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Bad quality of call Hi all, I did some call using an asterisk 1.4 PBX and 2 softphone in a private network; call is up, but with bad quality. Someone knows how to debug this problem ? Thanks in advance for any help. -- Giancarlo Lombardo IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. ___ -- 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] Call declined
Thanks !! it works 2009/11/10 Michael Wyres mwy...@cdm.com.au Try: *[tutorial]** exten = 1234,1,Dial(SIP/gianca,10,t)* *exten = 12345,1,Dial(SIP/giusy,10,t*) You want a “/” between SIP and the name of the phone, not an “,”. The “10” refers to the number of seconds you want the phone to ring. The “t” allows the channel to be transferred after pickup – not strictly needed, but I tend to put it in in most instances as generally you’ll want it. For more information on the Dial application, see http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *giancarlo lombardo *Sent:* Tuesday, 10 November 2009 09:03 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Call declined Dear all, I'm in basic setup of my network: I try to do a call from a softphone to an other one but I got the error 603 Declined. Below the 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* extension.conf: *[tutorial]** exten = 1234,1,Dial(SIP,gianca)* *exten = 12345,1,Dial(SIP,giusy*) Below the output of SIP debug of IP caller (192.168.1.116) in asterisk *dhcppc0*CLI** --- SIP read from 192.168.1.116:14862 --- INVITE sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport Max-Forwards: 70 Contact: sip:gia...@192.168.1.116:14862 To: 12345sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 From: giancasip:gia...@192.168.1.100 sip%3agia...@192.168.1.100 ;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 265* *v=0** o=- 6 2 IN IP4 192.168.1.116 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.116 t=0 0 m=audio 5960 RTP/AVP 107 0 8 101 a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv* *-** --- (12 headers 11 lines) --- Sending to 192.168.1.116 : 14862 (NAT) Using INVITE request as basis request - NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.* *--- Reliably Transmitting (no NAT) to 192.168.1.116:14862 ---** SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;received=192.168.1.116;rport=14862 From: giancasip:gia...@192.168.1.100 sip%3agia...@192.168.1.100 ;tag=db428348 To: 12345sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 ;tag=as29d2b71c Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY upported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=42ebb35e Content-Length: 0* *** Scheduling destruction of SIP dialog 'NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.' in 32000 ms (Method: INVITE) Found user 'gianca' dhcppc0*CLI --- SIP read from 192.168.1.116:14862 --- ACK sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport To: 12345sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 ;tag=as29d2b71c From: giancasip:gia...@192.168.1.100 sip%3agia...@192.168.1.100 ;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 ACK Content-Length: 0* *-** --- (7 headers 0 lines) --- dhcppc0*CLI --- SIP read from 192.168.1.116:14862 --- INVITE sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;rport Max-Forwards: 70 Contact: sip:gia...@192.168.1.116:14862 To: 12345sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 From: giancasip:gia...@192.168.1.100 sip%3agia...@192.168.1.100 ;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username=gianca,realm=asterisk,nonce=42ebb35e,uri= sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 ,response=8d00b3e1b28ed2e40681a3a9ee410046,algorithm=MD5 User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 265* *v=0** o=- 6 2 IN IP4 192.168.1.116 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.116 t=0 0 m=audio 5960 RTP/AVP 107 0 8 101 a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv
[asterisk-users] user extension in asterisk GUI
Hi all, I just configured some user in sip.conf and extensions.conf; they works fine. Now I'm trying to do the same with Extensions feature of FreePBXAdministration, but I cannot see what I have done manualy. Is gui working with other an source, How can I access such data ? Thanks in advance for any suggestion. -- 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] Call declined
;rport=14862 From: giancasip:gia...@192.168.1.100 sip%3agia...@192.168.1.100 ;tag=db428348 To: 12345sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 Content-Length: 0* * -- Executing [12...@tutorial:1] Dial(SIP/gianca-088b96e0, SIP|giusy) in new stack == Spawn extension (tutorial, 12345, 1) exited non-zero on 'SIP/gianca-088b96e0' Scheduling destruction of SIP dialog 'NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.' in 32000 ms (Method: INVITE)* *--- Reliably Transmitting (no NAT) to 192.168.1.116:14862 --- SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;received=192.168.1.116;rport=14862 From: giancasip:gia...@192.168.1.100 sip%3agia...@192.168.1.100 ;tag=db428348 To: 12345sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 ;tag=as12cbf532 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0* * dhcppc0*CLI --- SIP read from 192.168.1.116:14862 --- ACK sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;rport To: 12345sip:12...@192.168.1.100 sip%3a12...@192.168.1.100 ;tag=as12cbf532 From: giancasip:gia...@192.168.1.100 sip%3agia...@192.168.1.100 ;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 2 ACK Content-Length: 0 * ** -- 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] 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
User-Age nt: Aste 0180 72 69 73 6b 20 50 42 58 0d 0a 41 6c 6c 6f 77 3a risk PBX ..Allow: 0190 20 49 4e 56 49 54 45 2c 20 41 43 4b 2c 20 43 41INVITE, ACK, CA 01a0 4e 43 45 4c 2c 20 4f 50 54 49 4f 4e 53 2c 20 42 NCEL, OP TIONS, B 01b0 59 45 2c 20 52 45 46 45 52 2c 20 53 55 42 53 43 YE, REFE R, SUBSC 01c0 52 49 42 45 2c 20 4e 4f 54 49 46 59 0d 0a 53 75 RIBE, NO TIFY..Su 01d0 70 70 6f 72 74 65 64 3a 20 72 65 70 6c 61 63 65 pported: replace 01e0 73 0d 0a 43 6f 6e 74 65 6e 74 2d 4c 65 6e 67 74 s..Conte nt-Lengt 01f0 68 3a 20 30 0d 0a 0d 0ah: 0 * Any help is welcome. -- 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] 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] 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] 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
Re: [asterisk-users] Location
Italy Milan 2009/11/7 Thomas Perron thomas.per...@gmail.com Where is everyone located? I am in Washington DC. ___ -- 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] Tutorial for SIP user
Ciao, I installed Xlite on Windows Vista, the IP connection (ping) is working, shall I check something else ? Thanks in advance. 2009/11/1 Farooq Hussain farooqhussain...@gmail.com Dear Giancarlo, On which OS your are installing XLITE. If you are trying to connect XLITE using Winodws XP please make a entry in your firewall. I think that would solve your problem On Sun, Nov 1, 2009 at 10:27 AM, giancarlo lombardo gianclomba...@gmail.com wrote: Dear all, I'm trying to setup a SIP call with XLITE using my asterisk PBX, but I have trouble, I see on XLITE console: Registration Error: 503 - Service unavailable. Someone have a tutorial or a step by step description how to do that ? Thanks in advance -- 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 -- Thanks Farooq Hussain ___ -- 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] Tutorial for SIP user
Dear all, I'm trying to setup a SIP call with XLITE using my asterisk PBX, but I have trouble, I see on XLITE console: Registration Error: 503 - Service unavailable. Someone have a tutorial or a step by step description how to do that ? Thanks in advance -- 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] Software for PC-PC voice comunication
Thanks, it sounds good. 2009/10/27 giancarlo lombardo gianclomba...@gmail.com I just installed an Asterisknow server can someone suggest a software to be used for a PC - PC voice comunication to test in easy way the functionalities of my server. Thanks in advance for the help -- 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] Software for PC-PC voice comunication
I just installed an Asterisknow server can someone suggest a software to be used for a PC - PC voice comunication to test in easy way the functionalities of my server. Thanks in advance for the help ___ -- 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] GUI for asterix management
Ciao, at the moment machine is standalone, so i need to start GUI from console. 2009/10/24 Warren Selby wcse...@selbytech.com Have you tried accessing the IP address of your server from another computer's web browser? --Warren Selby On Fri, Oct 23, 2009 at 10:19 AM, giancarlo lombardo gianclomba...@gmail.com wrote: Dear all, I just installed asterixnow, but no graphical interface start automaticaly neither linux nor some other, just command line. Shall I do something or shall I install something more ? Many Thanks in advance for any help. Giancarlo ___ -- 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] GUI for asterix management
Ciao,
ok I'll try to install X and firefox. I understood that is not in standard
distribution of Asterisk;
Thanks for the help and sorry for trivial question.
2009/10/24 Alan Lord (News) alansli...@gmail.com
On 24/10/09 11:05, Steve Howes wrote:
On 24 Oct 2009, at 10:52, giancarlo lombardo wrote:
at the moment machine is standalone, so i need to start GUI from
console.
There is no Asterisk GUI that runs like that. You could install X and
Firefox but thats just a bit retarded. Plug in a network cable! Its
not like its not going to need one eventually.
If it's a true server with no X you could just use a text browser like
Lynx or Links. Then browse to http://localhost{:port}/gui url
But to be frank, the Op will learn more by editing the configuration
files directly.
HTH
Alan
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[asterisk-users] GUI for asterix management
Dear all, I just installed asterixnow, but no graphical interface start automaticaly neither linux nor some other, just command line. Shall I do something or shall I install something more ? Many Thanks in advance for any help. Giancarlo ___ -- 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
