[asterisk-users] Asterisk not starting up due to database problems
When I try and start asterisk I get the following, however I have commented out the data the connections in res_mysql.conf and res_pgsql.conf. I am not sure therefore why I am getting these errors. Do I have to change something else to turn this off? Thanks Asterisk 1.4.21.2~dfsg-3, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer 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. = This package has been modified for the Debian GNU/Linux distribution Please report all bugs to http://bugs.debian.org/asterisk = [ Booting... [ Reading Master Configuration ] [ Initializing Custom Configuration Options ] [May 3 13:29:14] NOTICE[7477]: cdr.c:1373 do_reload: CDR simple logging enabled. [May 3 13:29:14] NOTICE[7477]: loader.c:859 load_modules: 175 modules will be loaded. ..[May 3 13:29:14] NOTICE[7477]: res_odbc.c:235 load_odbc_config: Adding ENV var: INFORMIXSERVER=my_special_database [May 3 13:29:14] NOTICE[7477]: res_odbc.c:235 load_odbc_config: Adding ENV var: INFORMIXDIR=/opt/informix [May 3 13:29:14] NOTICE[7477]: res_odbc.c:530 odbc_obj_connect: Connecting asterisk [May 3 13:29:14] WARNING[7477]: res_odbc.c:541 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [May 3 13:29:14] WARNING[7477]: res_odbc.c:463 ast_odbc_request_obj: Failed to connect to asterisk [May 3 13:29:14] NOTICE[7477]: res_odbc.c:316 load_odbc_config: Registered ODBC class 'asterisk' dsn-[asterisk] [May 3 13:29:14] NOTICE[7477]: res_odbc.c:716 load_module: res_odbc loaded. .[May 3 13:29:14] WARNING[7477]: res_smdi.c:1335 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener. .[May 3 13:29:14] NOTICE[7477]: config.c:1274 ast_config_engine_register: Registered Config Engine odbc . Loading [Sub]Agent Module .[May 3 13:29:14] WARNING[7477]: res_config_pgsql.c:683 parse_config: Postgresql RealTime: No database user found, using 'asterisk' as default. [May 3 13:29:14] WARNING[7477]: res_config_pgsql.c:691 parse_config: Postgresql RealTime: No database password found, using 'asterisk' as default. [May 3 13:29:14] WARNING[7477]: res_config_pgsql.c:699 parse_config: Postgresql RealTime: No database host found, using localhost via socket. [May 3 13:29:14] WARNING[7477]: res_config_pgsql.c:707 parse_config: Postgresql RealTime: No database name found, using 'asterisk' as default. [May 3 13:29:14] WARNING[7477]: res_config_pgsql.c:715 parse_config: Postgresql RealTime: No database port found, using 5432 as default. [May 3 13:29:14] WARNING[7477]: res_config_pgsql.c:725 parse_config: Postgresql RealTime: No database socket found, using '/tmp/pgsql.sock' as default. [May 3 13:29:14] ERROR[7477]: res_config_pgsql.c:782 pgsql_reconnect: Postgresql RealTime: Failed to connect database server asterisk on . Check debug for more info. [May 3 13:29:14] WARNING[7477]: res_config_pgsql.c:605 load_module: Postgresql RealTime: Couldn't establish connection. Check debug. [May 3 13:29:14] NOTICE[7477]: config.c:1274 ast_config_engine_register: Registered Config Engine pgsql ..[May 3 13:29:14] WARNING[7477]: res_config_mysql.c:542 parse_config: MySQL RealTime: No database user found, using 'asterisk' as default. [May 3 13:29:14] WARNING[7477]: res_config_mysql.c:549 parse_config: MySQL RealTime: No database password found, using 'asterisk' as default. [May 3 13:29:14] WARNING[7477]: res_config_mysql.c:556 parse_config: MySQL RealTime: No database host found, using localhost via socket. [May 3 13:29:14] WARNING[7477]: res_config_mysql.c:563 parse_config: MySQL RealTime: No database name found, using 'asterisk' as default. [May 3 13:29:14] WARNING[7477]: res_config_mysql.c:570 parse_config: MySQL RealTime: No database port found, using 3306 as default. [May 3 13:29:14] WARNING[7477]: res_config_mysql.c:577 parse_config: MySQL RealTime: No database socket found, using '/tmp/mysql.sock' as default. [May 3 13:29:14] ERROR[7477]: res_config_mysql.c:629 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on (err 2002). Check debug for more info. [May 3 13:29:14] WARNING[7477]: res_config_mysql.c:476 load_module: MySQL RealTime: Couldn't establish connection. Check debug. [May 3 13:29:14] NOTICE[7477]: config.c:1274 ast_config_engine_register: Registered Config Engine mysql ..[May 3 13:29:14] NOTICE[7477]: pbx_ael.c:4131 pbx_load_module: Starting AEL load process. [May 3 13:29:14] NOTICE[7477]: pbx_ael.c:4138
Re: [asterisk-users] Asterisk not starting up due to database problems
I am just trying to start asterisk from the console by typing: asterisk -c What part of the config do you require me to post? I am not wanting to store any of the config in a database and have the connections commented out. Thanks -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: 03 May 2009 14:11 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk not starting up due to database problems On Sun, May 03, 2009 at 01:30:49PM +0100, Charlie Grosvenor wrote: When I try and start asterisk I get the following, however I have commented out the data the connections in res_mysql.conf and res_pgsql.conf. I am not sure therefore why I am getting these errors. Do I have to change something else to turn this off? What are you trying to do? What fails? What are the relevant parts of the configuration files? Thanks Asterisk 1.4.21.2~dfsg-3, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer 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. = This package has been modified for the Debian GNU/Linux distribution Please report all bugs to http://bugs.debian.org/asterisk = [ Booting... [ Reading Master Configuration ] [ Initializing Custom Configuration Options ] [May 3 13:29:14] NOTICE[7477]: cdr.c:1373 do_reload: CDR simple logging enabled. [May 3 13:29:14] NOTICE[7477]: loader.c:859 load_modules: 175 modules will be loaded. ..[May 3 13:29:14] NOTICE[7477]: res_odbc.c:235 load_odbc_config: Adding ENV var: INFORMIXSERVER=my_special_database [May 3 13:29:14] NOTICE[7477]: res_odbc.c:235 load_odbc_config: Adding ENV var: INFORMIXDIR=/opt/informix [May 3 13:29:14] NOTICE[7477]: res_odbc.c:530 odbc_obj_connect: Connecting asterisk [May 3 13:29:14] WARNING[7477]: res_odbc.c:541 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [May 3 13:29:14] WARNING[7477]: res_odbc.c:463 ast_odbc_request_obj: Failed to connect to asterisk [May 3 13:29:14] NOTICE[7477]: res_odbc.c:316 load_odbc_config: Registered ODBC class 'asterisk' dsn-[asterisk] [May 3 13:29:14] NOTICE[7477]: res_odbc.c:716 load_module: res_odbc loaded. .[May 3 13:29:14] WARNING[7477]: res_smdi.c:1335 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener. .[May 3 13:29:14] NOTICE[7477]: config.c:1274 ast_config_engine_register: Registered Config Engine odbc . Loading [Sub]Agent Module .[May 3 13:29:14] WARNING[7477]: res_config_pgsql.c:683 parse_config: Postgresql RealTime: No database user found, using 'asterisk' as default. [May 3 13:29:14] WARNING[7477]: res_config_pgsql.c:691 parse_config: Postgresql RealTime: No database password found, using 'asterisk' as default. [May 3 13:29:14] WARNING[7477]: res_config_pgsql.c:699 parse_config: Postgresql RealTime: No database host found, using localhost via socket. [May 3 13:29:14] WARNING[7477]: res_config_pgsql.c:707 parse_config: Postgresql RealTime: No database name found, using 'asterisk' as default. [May 3 13:29:14] WARNING[7477]: res_config_pgsql.c:715 parse_config: Postgresql RealTime: No database port found, using 5432 as default. [May 3 13:29:14] WARNING[7477]: res_config_pgsql.c:725 parse_config: Postgresql RealTime: No database socket found, using '/tmp/pgsql.sock' as default. [May 3 13:29:14] ERROR[7477]: res_config_pgsql.c:782 pgsql_reconnect: Postgresql RealTime: Failed to connect database server asterisk on . Check debug for more info. [May 3 13:29:14] WARNING[7477]: res_config_pgsql.c:605 load_module: Postgresql RealTime: Couldn't establish connection. Check debug. [May 3 13:29:14] NOTICE[7477]: config.c:1274 ast_config_engine_register: Registered Config Engine pgsql ..[May 3 13:29:14] WARNING[7477]: res_config_mysql.c:542 parse_config: MySQL RealTime: No database user found, using 'asterisk' as default. [May 3 13:29:14] WARNING[7477]: res_config_mysql.c:549 parse_config: MySQL RealTime: No database password found, using 'asterisk' as default. [May 3 13:29:14] WARNING[7477]: res_config_mysql.c:556 parse_config: MySQL RealTime: No database host found, using localhost via socket. [May 3 13:29:14] WARNING[7477]: res_config_mysql.c:563 parse_config: MySQL RealTime: No database name found, using 'asterisk' as default. [May 3 13:29:14] WARNING[7477]: res_config_mysql.c:570 parse_config
[asterisk-users] Compiling Asterisk With ZapTel?
I have tried to compile asterisk with zaptel: ./configure --with-zaptel=/usr/src/zaptel-1.4.0 make make install however when I run asterisk it says that the zap command is missing. What am I doing wrong? I have compiled and installed zaptel fine and it is recognizing my card. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] X100P - zttools says red status
The card that I have got only has one port. I assume there are two versions? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: 27 January 2007 00:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] X100P - zttools says red status Charlie Grosvenor wrote: Yes the line is connected, a standard phone works fine when connected to the line. There're 2 ports on the card. Which port are you using? One of the ports is for connecting another phone in parallel to the card. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] X100P - zttools says red status
I have an X100P which I have set up as per the guidelines: http://www.x100p.com/support/doc/quick_start_fxo.php The card is recognized by the system: Zaptel Version: 1.4.0 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. However when I run zttools it says its status is red, which my understanding of is that it has not detected the line. I am in the uk and using a standard BT line (with ADSL). Any suggestions? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] X100P - zttools says red status
Yes the line is connected, a standard phone works fine when connected to the line. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 26 January 2007 23:45 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] X100P - zttools says red status On Fri, Jan 26, 2007 at 08:42:36PM -, Charlie Grosvenor wrote: I have an X100P which I have set up as per the guidelines: http://www.x100p.com/support/doc/quick_start_fxo.php The card is recognized by the system: Zaptel Version: 1.4.0 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. However when I run zttools it says its status is red, which my understanding of is that it has not detected the line. I am in the uk and using a standard BT line (with ADSL). Any suggestions? Is the line plugged in? Can you connect a standard phone to the same line? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] X100P how do i recieve incomming calls?
There is only one x100p card in the system thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 22 January 2007 23:40 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] X100P how do i recieve incomming calls? On Mon, Jan 22, 2007 at 08:08:16PM -, Charlie Grosvenor wrote: I have just purchased a 2nd hand X100P, Is there another X100P card in the same system? if I do a ztcfg -vv I get: Zaptel Version: 1.4.0 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. My understanding of the above is that the zaptel driver has detected the card. What do I now need to do, in order to get an incoming call to work with asterisk? I assume I need to make some sort of change to /etc/asterisk/zapata.conf in order to tell asterisk about the card? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] X100P how do i recieve incomming calls?
I have just purchased a 2nd hand X100P, if I do a ztcfg -vv I get: Zaptel Version: 1.4.0 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. My understanding of the above is that the zaptel driver has detected the card. What do I now need to do, in order to get an incoming call to work with asterisk? I assume I need to make some sort of change to /etc/asterisk/zapata.conf in order to tell asterisk about the card? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] X100P how do i recieve incomming calls?
Dmesg reports: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.0 Echo Canceller: MG2 ACPI: PCI interrupt :00:0c.0[A] - GSI 16 (level, low) - IRQ 185 wcfxo: DAA mode is 'FCC' Found a Wildcard FXO: Wildcard X100P Capability LSM initialized When I run zttool it says that alarms: red. The card is connected to the phone line. Anybody any idea what the problem is? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charlie Grosvenor Sent: 22 January 2007 20:08 To: asterisk-users@lists.digium.com Subject: [asterisk-users] X100P how do i recieve incomming calls? I have just purchased a 2nd hand X100P, if I do a ztcfg -vv I get: Zaptel Version: 1.4.0 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. My understanding of the above is that the zaptel driver has detected the card. What do I now need to do, in order to get an incoming call to work with asterisk? I assume I need to make some sort of change to /etc/asterisk/zapata.conf in order to tell asterisk about the card? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Escalate Call To Mobile
I am using Voip Talk and have my extensions.conf set up to make outgoing
calls:
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
What I want to do is have extension 2000 first try and call me on
extension 5000 then if nobody has answered after five rings call me on
my mobile (number 07944123123 (not my real number)) if nobody then
answers after 3 rings goto voicemail.
exten = 2000,2,Dial(SIP/5000,5)
exten = 2000,2,Dial(IAX2/[EMAIL PROTECTED]/447944123123,3)
exten = 2000,3,Voicemail,b5000
exten = 2000,103,Voicemail,u5000
This is not working. Can somebody tell me what I am doing wrong?
Thanks
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RE: [asterisk-users] Escalate Call To Mobile
Thanks, this it seems was what the problem was. Is it possible to
specify number of rings instead of timeout in seconds?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: 24 December 2006 14:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Escalate Call To Mobile
Also the timeout to Dial is in SECONDS, not RINGS. In the USA a ring
cycle is about 6 seconds.
C F wrote:
You Dont Have A Priority 1 And You Have Priority 2 Twice
On 12/24/06, Charlie Grosvenor [EMAIL PROTECTED] wrote:
I am using Voip Talk and have my extensions.conf set up to make
outgoing
calls:
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
What I want to do is have extension 2000 first try and call me on
extension 5000 then if nobody has answered after five rings call me
on
my mobile (number 07944123123 (not my real number)) if nobody then
answers after 3 rings goto voicemail.
exten = 2000,2,Dial(SIP/5000,5)
exten = 2000,2,Dial(IAX2/[EMAIL PROTECTED]/447944123123,3)
exten = 2000,3,Voicemail,b5000
exten = 2000,103,Voicemail,u5000
This is not working. Can somebody tell me what I am doing wrong?
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RE: [asterisk-users] VoipTalk unable to accept calls at present?
I have managed to resolve this. If anybody is interested my machine is multihomed. I set IAX.conf and SIP.conf just to listen on one ip address, this seemed to solve the problem. Regards From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charlie Grosvenor Sent: 14 December 2006 22:38 To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoipTalk unable to accept calls at present? I am trying to get asterisks to work with http://www.voiptalk.org 's IAX service. I have configured asterisks as per their instructions and am using the x-lite soft phone. When I get an incoming call the softphone rings but the caller (from pstn) gets a recorded message saying the number is unable to accept calls at present. Does anybody know what might be causing this? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial 9 For Outside Line?
[default]
Some extensions defined
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
I have the above defined in extensions.conf. This enables me to make
outgoing calls but would like to make it so you have to dial 9 to do
this. Could somebody let me know what I need to change for it to do
this?
Thanks
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[asterisk-users] VoipTalk unable to accept calls at present?
I am trying to get asterisks to work with http://www.voiptalk.org 's IAX service. I have configured asterisks as per their instructions and am using the x-lite soft phone. When I get an incoming call the softphone rings but the caller (from pstn) gets a recorded message saying the number is unable to accept calls at present. Does anybody know what might be causing this? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PortSip and Astericks new install
Thanks for you reply, it does not output anything on the console when i make the call. However i have turned on SIP Debug and get the following: -- SIP read from 192.168.2.3:8099: REGISTER sip:192.168.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:8099;rport;branch=z9hG4bK32333 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:8099 Max-Forwards: 70 User-Agent: PortSIP softphone 2.0 Expires: 150 Content-Length: 0 --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.2.3 : 8099 (NAT) Transmitting (NAT) to 192.168.2.3:8099: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.3:8099;branch=z9hG4bK32333;received=192.168.2.3;rport=8099 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (NAT) to 192.168.2.3:8099: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.3:8099;branch=z9hG4bK32333;received=192.168.2.3;rport=8099 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060;tag=as3d203e7d Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5d41ffa1 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms [Kserver1*CLI -- SIP read from 192.168.2.3:8099: REGISTER sip:192.168.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:8099;rport;branch=z9hG4bK41 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER Contact: sip:[EMAIL PROTECTED]:8099 Authorization: Digest username=4289, realm=asterisk, nonce=5d41ffa1, uri=sip:192.168.1.1:5060, response=c3f43fd747f5ef51168bbfa2401b680b, algorithm=MD5 Max-Forwards: 70 User-Agent: PortSIP softphone 2.0 [Kserver1*CLI Expires: 150 Content-Length: 0 --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.2.3 : 8099 (NAT) Transmitting (NAT) to 192.168.2.3:8099: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.3:8099;branch=z9hG4bK41;received=192.168.2.3;rport=8099 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- [Kserver1*CLI 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.2.3:8099: OPTIONS sip:[EMAIL PROTECTED]:8099 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK60c2e2eb;rport From: asterisk sip:[EMAIL PROTECTED];tag=as6aa7255c To: sip:[EMAIL PROTECTED]:8099 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 17 Nov 2006 20:47:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Transmitting (NAT) to 192.168.2.3:8099: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.3:8099;branch=z9hG4bK41;received=192.168.2.3;rport=8099 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060;tag=as3d203e7d Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 150 Contact: sip:[EMAIL PROTECTED]:8099;expires=150 Date: Fri, 17 Nov 2006 20:47:18 GMT Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms [Kserver1*CLI -- SIP read from 192.168.2.3:8099: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK60c2e2eb;rport=5060 From: asterisk sip:[EMAIL PROTECTED];tag=as6aa7255c To: sip:[EMAIL PROTECTED]:8099;tag=18467 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: PortSIP softphone 2.0 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER, UPDATE Content-Length: 0 --- (9 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' [Kserver1*CLI -- SIP read from 192.168.2.3:8099: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:8099;rport;branch=z9hG4bK31747 From: sip:[EMAIL PROTECTED]:5060;tag=30024 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 20 INVITE Contact: sip:[EMAIL PROTECTED]:8099 Max-Forwards: 70 User-Agent: PortSIP softphone 2.0 Subject: call Expires: 120 Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 325 v=0 o=- 2177823 2177823 IN IP4 192.168.2.3 s=PortSIP VOIP SDK 2.0 c=IN IP4 192.168.2.3 t=0 0 m=audio 51636 RTP/AVP 0 3 8 97 4 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:4 G723/8000
RE: [asterisk-users] PortSip and Astericks new install
I have tried the configuration exactly the same of yours and it's still not working. What could be wrong with my installation? I first of all tried the packages in Debian stable, this didn't work so I compiled from source but still the problem occurs. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: 16 November 2006 00:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PortSip and Astericks new install Charlie Grosvenor wrote: Thanks for your reply, I have tried what you have suggested and get the same issue. I have also tried express talk which connects then say: I just downloaded and installed PortSip2 I have an extension setup as [4289] type = friend host = dynamic username=4289 qualify=500 reinvite=no canreinvite=no nat=yes dtmfmode = rfc2833 context = managers callgroup=1 pickupgroup=1,2 mailbox = [EMAIL PROTECTED] secret=12345 disallow=all allow=ulaw allow=alaw allow=slin callerid = Doug Lytle 4289 PortSip2: Account: 4289 AuthName: 4289 Password: 12345 Server: YourAsteriskServer Port: 5060 When setup, it just worked. I'll have to guess that your Asterisk installation is a fault. 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PortSip and Astericks new install
What do you mean by asterisk console, I ran asterisk -r and entered sip show peers and this is what i got: server1*CLI sip show peers Name/username HostDyn Nat ACL Port Status 4289/4289 192.168.2.3 D N 8925 OK (2 ms) John (Unspecified)D 0Unmonitored 2 sip peers [2 online , 0 offline] This any good? Thanks From: [EMAIL PROTECTED] on behalf of Doug Lytle Sent: Thu 16/11/2006 21:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PortSip and Astericks new install Charlie Grosvenor wrote: I have tried the configuration exactly the same of yours and it's still not working. What could be wrong with my installation? I first of all Show us what being displayed on the Asterisk console, also show the output from sip show peers. 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PortSip and Astericks new install
I have just installed Asterisk and installed the sample configuration files. Asterisks appears to be working and I have added a SIP client: [John] type=friend secret=test host=dynamic allow=all I have been trying to dial the demo number 500 when using PortSip, Asterisks answers the phone but PortSip gives me the error: Call failed: codec not accepted 488. I have tried changing the enabled codecs in PortSip but this makes no difference. I have also tried various other SoftPhone but none of them seem to work. Anybody know what I have missed / doing wrong? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PortSip and Astericks new install
Thanks for your reply, I have tried what you have suggested and get the same issue. I have also tried express talk which connects then say: Initiated sip call to 500 Call has disconnected. Any other ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: 15 November 2006 21:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PortSip and Astericks new install Charlie Grosvenor wrote: [John] type=friend secret=test host=dynamic allow=all Try: disallow=all allow=alaw allow=ulaw allow=gsm 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
