Re: [Asterisk-Users] Snom 190 - dhcp - settings_server
I am using the same idea.
But, you don't want to put {mac} in the file name.
Just use snom200.htm.
What the phone does, it first reads snom200.htm
and then automatically proceeds to read a file of form
snom200-000413xx.htm
Put lines for all phones in snom200.htm and the rest in the file with
mac-address.
However I would use a more specific path for a web-server ;-) Something
like:
option tftp-server-name http://192.168.0.9/snom/snom200.htm
Best regards Pertti
Stefan Tichy wrote:
Hi,
in the Snom FAQ I found the following information:
After staring up, the phone tries the URL given in the Setting
URL of the phone. ... BTW this setting can also be set via DHCP.
option tftp-server-name http://192.168.0.9/snom200{mac}.htm;
The documents used:
FAQ-04-06-14-sf.pdf Setting up DHCP for snom phones
FAQ-04-03-24-sf.pdf How can I update a snom phone?
The phone used is a snom 190 (snom190-SIP 3.52e).
If I use the webinterface to insert the URL it works fine, but I am
not able to set this URL using dhcp.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails
Prestige 2000W is the same BCM phone that was earlier referred as Wifi-600 in this list. http://www.bcm.com.tw/product/pdf/pdf1/Spec-WiFi600_2003_1103.pdf It has the same problem. If you enable WEP encryption ( 104 bit ), the voice becomes very choppy. Almost unusable. Without WEP it is fine. I wonder if anybody has better results with WEB enabled and with latest software releases ? -- Pertti Lars Boegild Thomsen wrote: I have noticed this one and I have also informed ZyXEL, but their response was vague to say the least. It is correct that the ZyXEL phone does not send a SIP Cancel when you disconnect an outgoing call that has not yet been picked up by the remote end. I have several times asked ZyXEL to put a formal bug report procedure in place with proper tracking but to no avail. Regards, Lars... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dominique Kull Sent: 02 June 2004 22:46 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying since it always leaves an empty VM. thanks Dominique ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails
Thanks ! I must get the phone back for test then ;-) What software release and codec are you using ? Best regards Pertti Markus Engelbrecht wrote: Hello Pertti, I'm running the ZyXEL with WEP (128BIT) here at home and I don't have problems with the voice quality. Best Regards, Markus -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Pertti Pikkarainen Gesendet: Donnerstag, 3. Juni 2004 08:26 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails Prestige 2000W is the same BCM phone that was earlier referred as Wifi-600 in this list. http://www.bcm.com.tw/product/pdf/pdf1/Spec-WiFi600_2003_1103.pdf It has the same problem. If you enable WEP encryption ( 104 bit ), the voice becomes very choppy. Almost unusable. Without WEP it is fine. I wonder if anybody has better results with WEB enabled and with latest software releases ? -- Pertti Lars Boegild Thomsen wrote: I have noticed this one and I have also informed ZyXEL, but their response was vague to say the least. It is correct that the ZyXEL phone does not send a SIP Cancel when you disconnect an outgoing call that has not yet been picked up by the remote end. I have several times asked ZyXEL to put a formal bug report procedure in place with proper tracking but to no avail. Regards, Lars... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dominique Kull Sent: 02 June 2004 22:46 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying since it always leaves an empty VM. thanks Dominique ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura-spa2000
I had the same problem with a Siemens dect once ( and with Sipura ). The problem was solved by adding flash hook time. This is a configurable parameter in many dect phones. I added several hundreds of ms and the button started to work ( or actually - Sipura was able to 'see' the action ). -- Pertti Simon Chappell wrote: thanks for the reply, i thought it may be a stupid question but if i hit either hook buttons i do not get any result when in a call. if i press the hangup button it hangs up, press the pick up button and nothing happens :-( that is why i thought I was doing something silly or not understanding something. It is a panasonic dect phone Simon Richard Neese wrote: the off hook / hangup switch should act as a flash button also... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom 200 and hold
I wonder what I'm doing differently with 2.05e because for me the R-hold/transfer works. I'm not using transfer myself very often but now I tried this several times with a couple of snom200s ( all 2.05e ). I pressed R ( hold ), the calling end started to hear MOH, I dialed a new number ( + ok ) and hit xfer when the other end answered. The call got connected fine - every time. I'm using CVS head the 13th of May. PS Someone mentioned about some other problems with 2.05e. What kind of problems are they ? For me it would be important to know. Best regards Pertti Lars Boegild Thomsen wrote: Well - on a Snom 200 it's pretty easy. I reckon the R button should work actually, but you can also just press one of the other line buttons and immediately get a new dial-tone. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Swan Sent: 21 May 2004 01:29 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] snom 200 and hold Hi, I've looked through the archives and seen references to placing calls on hold on a snom 200 (any version of the firmware but we have the latest: 2.05e.) Basically, we can't place calls on hold on the snom 200! The manual talks about the Flash button (which is really the R button, as far as I can tell.) Pressing the R button will immediately disconnect the incoming call. Another poster to this list indicated one could just choose another line and the current line will be put on hold. This is not true on our phone: again, the original call is immediately disconnected. We've been all over the settings in the snom 200 and have tweaked a bunch of parameters. So: how does one place an incoming call on hold on a snom 200 so that we can do attended transfer? Michael Swan Neon Software, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 200
We are using SNOM200 with *. And we are very happy with it ( specially with the latest sw 2.05a ). I believe some of the missing advanced features become available when chan_sip2 is used. Best regards Pertti Hermann Wecke wrote: Sorry to ask this here but I believe that it is the best place to receive a feedback... I would like to know if anyone is using SNOM 100 / SNOM 200 phones with *, and the overall impression about these phones... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MWI indicator on SNOM200 doesn't disappear
I have also complained about the change in MWI to SNOM. My 2.03o phones still work with Asterisk but 2.04 versions do not. However, you can turn off the MWI by pressing the MWI button but not remotely ( NOTIFY ). I once got the example under from SNOM ( Asterisk version is under it ). According to SNOM this is an example of the format the phone is expecting in order to get MWI turned off. The relevant difference really looks like to be the 'Message-Account'. NOTIFY sip:[EMAIL PROTECTED]:5060;line=jet7pbic SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK-7c9c323d4898e621adb7244baa8cab62.1 Via: SIP/2.0/UDP 192.168.0.8:5062;branch=z9hG4bK-zt7bd9vxqo74 Record-Route: sip:intern.snom.de:5060;maddr=192.168.0.1;lr From: sip:[EMAIL PROTECTED]:5062;tag=vn8jb3vkko To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 23 NOTIFY Max-Forwards: 69 Contact: sip:[EMAIL PROTECTED]:5062 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 85 Message-Waiting: no Message-Account: sip:[EMAIL PROTECTED]:5062 Voice-Message: 0/0 This is what Asterisk is sending at the moment. And this is ok with 2.03o. Does chan_sip2 send somehow different NOTIFY ? NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.1.15.30:5060;branch=z9hG4bK3f99907b From: Asterisk sip:[EMAIL PROTECTED];tag=as243abda7 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 -- Pertti Olle E. Johansson wrote: Geert Nijpels wrote: Ian White wrote: On recent releases of the snom200 firmware, the MWI indicator will turn on, but won't turn off when the message has been checked. It works on firmware 2.03o, but not in 2.04g or newer. I filed a bug report with snom, but they're claiming it is an asterisk issue and that it should have been resolved. They suggested that I ask on the list. Anyway, Asterisk had a bug where it didn't send the NOTIFY correctly to turn off the MWI. The message doesn't contain the line so the phone doesn't know which line to apply the messages to. Basically the NOTIFY message should contain something like the following: NOTIFY sip:[EMAIL PROTECTED];line=34n34jed SIP/2.0 There was a bugfix for this in Asterisk for this problem, do you have that applied? I am running the current CVS version, and don't see anything in the code that looks like this has been touched, and I haven't seen reference to it on this list. They are right in that the line information isn't being sent, looking at the SIP debugs on both ends. Anybody have ideas? Ian This is a problem I have been digging into a bit. In my case asterisk did not send out the NOTIFY with the header Content-Type: application/simple-message-summary, but with Content-Type: text/plain, so the NOTIFY is treated as a txt message. In result, when I pressed the MWI button, I saw the text from asterisk stating the amount of messages I have. I changed it to work, and now asterisk calls the extension the message is sent from ([EMAIL PROTECTED]). After calling this the MWI indication disappears, I'm not sure if it also disappears after calling from another phone. I'm using chan_sip2 and I changed some stuff, so I'm not sure if this is also a problem with standard chan_sip (the txt vs vm issue). Chan_sip2 handles Contact: differently than chan_sip and works better with Snom phones. It's actually where the whole chan_sip2 project started... :-) /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PC based Switchboard application files??
The procedure was changed. I'm sending that directly. We'll need to know who actually downloads that. If anybody else needs it, please contact me off-list. Best regards Pertti Steven Elliott wrote: On 22/04/04 8:50, Pertti Pikkarainen [EMAIL PROTECTED] wrote: Good day all I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard) But in the pdf howto they speak about a swb.txt and I dont have that file The swb.txt is there but where did you find the SwB.war file? Steven ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PC based Switchboard application files??
I'm so sorry. The file is now there. Please download it. Thanks ! Best regards Pertti Altus Snyman wrote: Good day all I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard) But in the pdf howto they speak about a swb.txt and I dont have that file Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PC based Switchboard application files??
You are ok already. It should work even if that privilege line is missing. But to be sure later you can easily fix that. The error is due to a typo in the end of the file Run the first GRANT command again with 'asterisksettings' and not 'asterikssettings' I just fixed the download file. Best regards Pertti Altus Snyman wrote: Is this error ok? When I insert txt file into the db,Im loged in as postgres CREATE TABLE INSERT 16984 1 CREATE TABLE CREATE TABLE INSERT 17003 1 CREATE TABLE CREATE TABLE CREATE TABLE INSERT 17020 1 INSERT 17021 1 NOTICE: CREATE TABLE will create implicit sequence 'cdr_acctid_seq' for SERIAL column 'cdr.acctid' CREATE TABLE CREATE TABLE NOTICE: ALTER TABLE / ADD PRIMARY KEY will create implicit index 'cdr_pkey' for table 'cdr' ALTER TABLE setval 454 (1 row) ERROR: Relation asterikssettings does not exist GRANT GRANT GRANT GRANT GRANT GRANT GRANT GRANT On Thu, 2004-04-22 at 09:16, Pertti Pikkarainen wrote: I'm so sorry. The file is now there. Please download it. Thanks ! Best regards Pertti Altus Snyman wrote: Good day all I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard) But in the pdf howto they speak about a swb.txt and I dont have that file Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ** Nordic LANWAN Communication Oy Pertti Pikkarainen vp of engineering WWW: http://www.lanwan.fi E-Mail: [EMAIL PROTECTED] tel: +358-9-4243 fax: +358-9-5023840 gsm: +358-500-511467 Sinikalliontie 16 02630 Espoo FINLAND ** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PC based Switchboard application files??
After restarting postgres you need to stop and start $CATALINA_HOME/bin/shutdown.sh $CATALINA_HOME/bin/startup.sh Or something is wrong with the postgre access rights. Did you remember modify /usr/local/pgsql/data/pg_hba.conf If a new start doesn't help please, send me $CATALINA_HOME/logs/catalina.out Best regards Pertti Altus Snyman wrote: It comes up with the index page but when you login with admin,admin it says error logging on to database,postgresql is running as postgres user and the db has been added with the txt file,I did the change in tomcat folder. Must postgresql run as postgres user? Any Ideas Thanks Altus On Thu, 2004-04-22 at 09:31, Altus Snyman wrote: Is this error ok? When I insert txt file into the db,Im loged in as postgres CREATE TABLE INSERT 16984 1 CREATE TABLE CREATE TABLE INSERT 17003 1 CREATE TABLE CREATE TABLE CREATE TABLE INSERT 17020 1 INSERT 17021 1 NOTICE: CREATE TABLE will create implicit sequence 'cdr_acctid_seq' for SERIAL column 'cdr.acctid' CREATE TABLE CREATE TABLE NOTICE: ALTER TABLE / ADD PRIMARY KEY will create implicit index 'cdr_pkey' for table 'cdr' ALTER TABLE setval 454 (1 row) ERROR: Relation asterikssettings does not exist GRANT GRANT GRANT GRANT GRANT GRANT GRANT GRANT On Thu, 2004-04-22 at 09:16, Pertti Pikkarainen wrote: I'm so sorry. The file is now there. Please download it. Thanks ! Best regards Pertti Altus Snyman wrote: Good day all I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard) But in the pdf howto they speak about a swb.txt and I dont have that file Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PC based Switchboard application files??
in /usr/local/pgsql/data/pg_hba.conf this kind of information is given # To allow TCP/IP access, even from localhost, the postmaster must also be # started with the -i option or the option TCPIP_SOCKET must be set in # /etc/postgresql/postgresql.conf. In the end of my/usr/local/pgsql/data/postgresql.conf # TCP/IP access is allowed by default, but the default access given in # pg_hba.conf will permit it only from localhost, not other machines. tcpip_socket = 1 I have postgres and tomcat in the same computer. --Pertti Altus Snyman wrote: Does it any difference that I'm using a already running tomcat? Here is the output Thanks again for your help DBCP borrowObject failed: Connection refused. Check that the hostname and port are correct and that the postmaster is accepting TCP/IP connections. Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources init INFO: Initializing, config='org.apache.struts.taglib.bean.LocalStrings', returnNull=true Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources init INFO: Initializing, config='org.apache.struts.util.LocalStrings', returnNull=true Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources init INFO: Initializing, config='org.apache.struts.taglib.html.LocalStrings', returnNull=true Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources init INFO: Initializing, config='org.apache.struts.taglib.html.LocalStrings', returnNull=true Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources init INFO: Initializing, config='org.apache.struts.taglib.html.LocalStrings', returnNull=true Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources init INFO: Initializing, config='org.apache.struts.taglib.html.LocalStrings', returnNull=true Apr 22, 2004 11:28:58 AM org.apache.struts.util.PropertyMessageResources init INFO: Initializing, config='org.apache.struts.taglib.html.LocalStrings', returnNull=true DBCP borrowObject failed: Connection refused. Check that the hostname and port are correct and that the postmaster is accepting TCP/IP connections. Stopping service Tomcat-Standalone On Thu, 2004-04-22 at 14:30, Pertti Pikkarainen wrote: After restarting postgres you need to stop and start $CATALINA_HOME/bin/shutdown.sh $CATALINA_HOME/bin/startup.sh Or something is wrong with the postgre access rights. Did you remember modify /usr/local/pgsql/data/pg_hba.conf If a new start doesn't help please, send me $CATALINA_HOME/logs/catalina.out Best regards Pertti Altus Snyman wrote: It comes up with the index page but when you login with admin,admin it says error logging on to database,postgresql is running as postgres user and the db has been added with the txt file,I did the change in tomcat folder. Must postgresql run as postgres user? Any Ideas Thanks Altus On Thu, 2004-04-22 at 09:31, Altus Snyman wrote: Is this error ok? When I insert txt file into the db,Im loged in as postgres CREATE TABLE INSERT 16984 1 CREATE TABLE CREATE TABLE INSERT 17003 1 CREATE TABLE CREATE TABLE CREATE TABLE INSERT 17020 1 INSERT 17021 1 NOTICE: CREATE TABLE will create implicit sequence 'cdr_acctid_seq' for SERIAL column 'cdr.acctid' CREATE TABLE CREATE TABLE NOTICE: ALTER TABLE / ADD PRIMARY KEY will create implicit index 'cdr_pkey' for table 'cdr' ALTER TABLE setval 454 (1 row) ERROR: Relation asterikssettings does not exist GRANT GRANT GRANT GRANT GRANT GRANT GRANT GRANT On Thu, 2004-04-22 at 09:16, Pertti Pikkarainen wrote: I'm so sorry. The file is now there. Please download it. Thanks ! Best regards Pertti Altus Snyman wrote: Good day all I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard) But in the pdf howto they speak about a swb.txt and I dont have that file Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http
Re: [Asterisk-Users] Snom 200 Admin Password
There is a way. Right after reboot, and when you see the first text, hit any key and you will get a 'boot menu'. Give the phone an ip-address and define a tftp-server. The bootfile must be named snom200.bin ( e.g rename the latest snom sw ). After you have succesfully got it to download the code, the phone is also resetted to factory defaults. You will see erasing flash etc. If the download fails the phone will use the sw it has got and there will be no change in the config either. --Pertti Chris Orme wrote: Hi. Did you buy the phone or get it second hand ? If second hand do you have any paperwork from the person you bought it from and did they buy it through official distribution? If you got it through distribution I would am fairly sure your vendor might be able to help ? I have a rough idea of how it would be possible but I would think you'll probably have to prove ownership as this password is how carriers lock their phones. If you got it from a carrier I imagine you might possibly have to pay them an unlock charge so you can change carriers. Or did you accidently set the admin password? Chris On Sat, 17 Apr 2004, WipeOut wrote: Hi, I have a Snom 200 that has had admin mode switched off and I have no idea when the admin password has been set to.. Does anyone know of a way to reset the phone to factory defaults?? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PC based Switchboard application
We have switchboard application ( PC+browser+Java ) with quite a rich feature set. It talks to * via manager port. Works as a call center too. However, it is not open source. If you are interested in, please contact me directly. Best regards Pertti Keith D'Atrio wrote: Hello All I am looking for a PC based switchboard application. Cisco CallManager has a web attendant console that allows you to use the PC to transfer calls and the like and I was wondering if there was a similar program compatible with *. Thank you in advance Keith D'Atrio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BCM Wireless SIP Phone
I had fairly good experince at first in the lab. But when I configured the phone to use 104-bit WEP-key, like most of the production networks, the quality degraded a lot. You can still talk but the quality is bad ( choppy ). With 40bit key or without WEP, the quality was fine. I tested this a few months ago. So, I don't know if later versions are any better. I would be interested in to know. --Pertti Miguel Cavazos wrote: the phone works for the wlan600 its a great phone poor battery but even palms with wifi use ALOT more battery when wifi is on and considering this phone has the wifi ON all the time the 23 stand by hours and 3 hr talk is ok it registers with asterisk just fine, try get it from pulver Miguel On Wed, 2004-03-10 at 04:51, Steven Thomas wrote: Hi, Has anyone tried this Wireless SIP phone with Asterisk? If so, any limitations? Thanks. http://www.bcm.com.tw/product/productIS.htm Regards, Steven Thomas Network Integration Services IBM Australia Ph: 0404 099 262 NH011, IBM Centre, 601 Pacific Hwy, St Leonards, 2065. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell 1750 server and Asteriks...
Yes, works ok with TE410P and E400P. The server has both slot types. -- Pertti Robert R. Randall wrote: Has anyone tried the Dell 1750 server as an Asterisks server with one of the 4 port Digium cards? I'm just looking for a reference point on this. Thanks. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hide outgoing CallerId on Zap interface
Mayby you missed my reply as well. Here it is again ... When I need to hide callerid ( sip phones ), I will configure this in sip.conf. You need to include restrictcid=yes for each user that needs to be hidden. -- Pertti Mickey Binder wrote: There seems to be some trouble with either the maillist or my client. I haven't received any of the posted replys on this topic, but found the replys through the asterisk.linkx.net search engine. But anyways here is the reply on the mail from: James H dot Cloos Jr. cloos at jhcloos.com If it is a pri I'd give SetCallerID() a try in the dialplan. It is a PRI and I've tried the SetCallerId() which displays my PRI main number, like the other experiments I've tried. I talked to my Telco provider which said that it isn't possible to hide the number via shortcodes but that it should be done via my PABX. -Original Message- From: Mickey Binder [mailto:[EMAIL PROTECTED] Sent: 13. februar 2004 12:14 To: Asterisk maillist Subject: [Asterisk-Users] Hide outgoing CallerId on Zap interface Hi there I know I have asked a somehow similar question earlier but since then I've tried some different things which isn't working. I want to completely hide my outgoing CallerId when dialing out on my Zap interface. I've tried a lot of different settings in sip.conf and hoped that zap would hide the CallerId if sip was told to do so, but that wasn't the case. Then I've tried to set hidecallerid=yes in zapata.conf (and restarted *) but this only results in my main number CallerId being displayed. Is it somehow possible to completely hide the CallerId, like when someone from a secret number is calling and the display on my mobile says Secret number ? And if that is possible, is it then possible to do it on a per-user basis configured via sip.conf? regards, Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hide outgoing CallerId on Zap interface
This is excactly what restrictcid=yes does in sip.conf. Eg. when it is used you'll see this in pri debug: Calling Number (len=12) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan ( E.164/E.163) (1) Presentation: Presentation prohibited, user number passed network screening (33) 12345 12345 is your number but it will not be passed to the other party that is being called. -- Pertti Alfred R. Nurnberger wrote: The correct way to hide your callerid on a PRI interface is to set the presentation indicator. Some CO switches do a basic sanity check on the callerid they receive. If you set the number string to empty but the presentation indicator to allow the number they will replace the number string by your main number. I do not know how or if possible to change the presentation indicator on * but a look in libpri should give some clues. - Alfred. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mickey Binder Sent: Monday, February 16, 2004 6:13 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Hide outgoing CallerId on Zap interface There seems to be some trouble with either the maillist or my client. I haven't received any of the posted replys on this topic, but found the replys through the asterisk.linkx.net search engine. But anyways here is the reply on the mail from: James H dot Cloos Jr. cloos at jhcloos.com If it is a pri I'd give SetCallerID() a try in the dialplan. It is a PRI and I've tried the SetCallerId() which displays my PRI main number, like the other experiments I've tried. I talked to my Telco provider which said that it isn't possible to hide the number via shortcodes but that it should be done via my PABX. -Original Message- From: Mickey Binder [mailto:[EMAIL PROTECTED] Sent: 13. februar 2004 12:14 To: Asterisk maillist Subject: [Asterisk-Users] Hide outgoing CallerId on Zap interface Hi there I know I have asked a somehow similar question earlier but since then I've tried some different things which isn't working. I want to completely hide my outgoing CallerId when dialing out on my Zap interface. I've tried a lot of different settings in sip.conf and hoped that zap would hide the CallerId if sip was told to do so, but that wasn't the case. Then I've tried to set hidecallerid=yes in zapata.conf (and restarted *) but this only results in my main number CallerId being displayed. Is it somehow possible to completely hide the CallerId, like when someone from a secret number is calling and the display on my mobile says Secret number ? And if that is possible, is it then possible to do it on a per-user basis configured via sip.conf? regards, Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hide outgoing CallerId on Zap interface
When I need to hide callerid ( sip phones ), I will configure this in sip.conf. You need to include restrictcid=yes for each user that needs to be hidden. -- Pertti Jonathan Stanton @ Home wrote: Im in the UK and unless you dial a particular code first (141) before you dial the number the phonenumber will automatically stamp the call with your main number. I THINK that this setting just stops asterisk from sending the caller ID from the originiating extention down the line (and only if it was a digital line eg ISDN) Regards Jonathan - Original Message - From: Mickey Binder [EMAIL PROTECTED] To: Asterisk maillist [EMAIL PROTECTED] Sent: Friday, February 13, 2004 11:13 AM Subject: [Asterisk-Users] Hide outgoing CallerId on Zap interface Hi there I know I have asked a somehow similar question earlier but since then I've tried some different things which isn't working. I want to completely hide my outgoing CallerId when dialing out on my Zap interface. I've tried a lot of different settings in sip.conf and hoped that zap would hide the CallerId if sip was told to do so, but that wasn't the case. Then I've tried to set hidecallerid=yes in zapata.conf (and restarted *) but this only results in my main number CallerId being displayed. Is it somehow possible to completely hide the CallerId, like when someone from a secret number is calling and the display on my mobile says Secret number ? And if that is possible, is it then possible to do it on a per-user basis configured via sip.conf? regards, Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gsm + snom phones
About a month ago I made a test with snom200b. At least then it worked ok with *. At the moment I'm using mainly g711a. So, there is always a possibility something has changed. -- Pertti Matteo Brancaleoni wrote: Hi. I'm not using snom phones for a while, but now I want to test again them and I'm gonna buy a snom 200 105 . Some times ago I had a snom 100 , and gsm wasn't working with *. How's now the situation? the snom gsm works well with * ? Thanks for any info, Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gsm + snom phones
Yes, absolutely. sorry, I was unclear .. -- Pertti Matteo Brancaleoni wrote: Hi. About a month ago I made a test with snom200b. At least then it worked ok with *. At the moment I'm using mainly g711a. So, there is always a possibility something but you also tested gsm ? Greets,Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wifi600 problem
Thanks John, Can you check what version you are using ? I can start with the very same ( once I get it ). I have sent a request to BCM but haven't got any reply yet. -- Pertti John Todd wrote: At 8:49 PM +0200 11/17/03, Pertti Pikkarainen wrote: Some of you have got Wifi600 wireless SIP phone working with Asterisk. Specially John Todd ( nice review ). My phones register ok. They can also receive calls from other phones. But for some reason I can't make them call out ( anybody, ie. SIP or PSTN ). This seems to be due to the phone not understanding what it should do when it receives 'Proxy Authentication Required'. In my case it does nothing. Can someone tell me what Wifi600 software version was used when this phone was succesfully tested with Asterisk. Any other hint is also appreciated. -- Pertti I had the same problem initially. However, the vendor gave me a software update which fixed the authentication problem. I had hoped that it would have made it out to general distribution by now. Please contact your vendor to see if they have the software that they can give to you. If not, let me know who you're talking with and I'll see what I can do as far as information transfer to the company that sold you the phone so they start doing the right thing. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wifi600 problem
Some of you have got Wifi600 wireless SIP phone working with Asterisk. Specially John Todd ( nice review ). My phones register ok. They can also receive calls from other phones. But for some reason I can't make them call out ( anybody, ie. SIP or PSTN ). This seems to be due to the phone not understanding what it should do when it receives 'Proxy Authentication Required'. In my case it does nothing. Can someone tell me what Wifi600 software version was used when this phone was succesfully tested with Asterisk. Any other hint is also appreciated. -- Pertti ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Did you try to use *8 only instead of *8# ? Last time when I tried *8 picked the call with known results but I haven't tested any patches yet. I really hope call pickup now works. -- Pertti Rich Adamson wrote: Just submitted a patch for this on asterisk-dev. Quick fix add the following line above line 5022 in chan_sip.c ast_setstate(c,AST_STATE_DOWN); Just updated to current cvs a few minutes ago primarily to get the call pickup to function properly. Using C7960's and Snom 200 on RH9. All compiled and installed cleanly. Maybe I'm misunderstanding the call pickup functions; here's a couple of samples from my sip.conf: [3000] type=friend username=3000 secret=mypassword host=dynamic context=from-sip callgroup=2 pickupgroup=2 mailbox=3000 [3001] type=friend username=3001 secret=mypassword2 host=dynamic context=from-sip callgroup=2 pickupgroup=2 callgroup=2 mailbox=3001 [3002] type=friend username=3002 secret=mypassword3 host=dynamic context=from-sip callgroup=2 pickupgroup=2 mailbox=3002 If station 3002 calls 3001, I'm expecting the user at 3000 to hear the rining at 3001, and dial *8# to pick it up. When I try that, *8# does not pick up the call and only receives a busy. Are my expectations incorrect, my definitions, or what? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] exten sent with MWI??
I just got this working. I added a line exten = asterisk,1,Goto,1001|1to my extensions.conf. Now when MWI button is pressed ( snom200 ) you get connected to voicemailmain. Even the word 'asterisk' can be modified. It is in the beginning of chan_sip.c. But I'm not sure if there are sideffects changing this. -- Pertti WipeOut . wrote: Hi, I have VoiceMailMain on extention 1001 so it would be nice to get that sent to the phone instead of [EMAIL PROTECTED] Address] when there is a message waiting.. Is it possible to change the extention that is sent to the phone when MWI is lit up?? Thanks.. -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 200 bugs
Stuart Hirst wrote: Does anyone have the same issues and is there any work arounds. I have a SNOM 200 which seems to work fine for so long but after an undetermined time when I make a call I hear no audio. If I reboot the SNOM all is fine again. The same here. Version sip-1.16w. You have to go down to 1.16b if you want to get a temporary solution. When the problem occurs, for some reason a RTP media stream is never disconnected ( SNOM to * ). The good news is that SNOM is aware of this and has promised a beta fix shortly. -- Pertti Also when I reboot the SNOM it only ever picks up the NTP time and registers correctly after the second reboot. Thanks for any info. Rgds, Stuart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to transfer a call??
I have it like this exten = 9998,1,Dial,SIP/9998|30|t 30 is a timeout value Check 'show application dial' WipeOut wrote: What is the correct syntax to use the 't' option?? This is the current line in my extensions.conf exten = 9998,1,Dial,SIP/9998 So would I change it to exten = 9998,1,Dial,SIP/9998,t Thanks. - Original Message - From: Pertti Pikkarainen [EMAIL PROTECTED] Date: Fri, 14 Mar 2003 13:50:21 +0200 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] How to transfer a call?? Negative side effect with 't' option: all the local SIP-to-SIP media streams travel trough Asterisk instead of going direct. Right now I'm using SNOM's transfer option instead. But now I can't use * call parking because of that. Using # is probably better if there are no scaling problems. Regards Pertti Steven Critchfield wrote: If you search the archives you would find that for IP phone you need to add a 't' option to the end of your dial command. The 't' option will let the user dial '#' to get the systems attention, then dial an extention for the transfer. On Fri, 2003-03-14 at 03:32, =?iso-8859-1?B?V2lwZU91dCCZ ?= wrote: Hi, Firstly let me start off by saying that asterisk is one of the most amazing pieces of open source I have seen, it rates right up there with Apache, OpenOffice, MySQL and even Linux itself.. Nice work!! I have just installed my first server, thanks to the astinstall script.. and I have read the Handbook (ver 1) and the white paper PDF's.. and I have managed to setup 2 extentions and make calls between them using MSN Messenger, nothing fantastic but its a start.. One answer is still missing.. How do I transfer a call to another ext?? I am looking at only using IP phones and so for the test system I am using MSN Messenger.. The final solution will probably use a linux softphone line gnophone or linphone.. All I have been able to find in the docs about call transfer is using a normal phone handset and hook-flash (not quite sure what that it, I am new to telephony).. So I guess what I am asking is what do I need to configure or do to be able to transfer a call from one IP ext to another?? Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- ** Nordic LANWAN Communication Oy Pertti Pikkarainen vp of engineering E-Mail: [EMAIL PROTECTED] tel: +358-9-5024100 fax: +358-9-5023840 gsm: +358-500-511467 Sinikalliontie 16 02630 Espoo FINLAND ** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
