[asterisk-users] no outgoing calls with Digium B410P
Hi all! Sorry for my poor english, i'm italian. I installed Digium B410P on my asterisk server. I followed the official installation instructions found on digium site. These instruction, in my opinion, are not clear, so i tried with other ways (found on the trixbox site). I found this: http://www.trixbox.org/forums/trixbox-forums/trunks/howto-install-script-digium-b410p Now i can receive calls from pstn, but i can't do any call to pstn. I attach my /etc/misdn-init.conf/ and also my etc/asterisk/misdn.conf file for clarity. Thanks - Daniele misdn.conf Description: Binary data misdn-init.conf Description: Binary data ___ --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] no outgoing calls with Digium B410P
Hi Daniele, Could you please tell us what exactly happens? Are your able to see some error in the log/console? On Jan 7, 2008 11:53 AM, daniele visaggio [EMAIL PROTECTED] wrote: Hi all! Sorry for my poor english, i'm italian. I installed Digium B410P on my asterisk server. I followed the official installation instructions found on digium site. These instruction, in my opinion, are not clear, so i tried with other ways (found on the trixbox site). I found this: http://www.trixbox.org/forums/trixbox-forums/trunks/howto-install-script-digium-b410p Now i can receive calls from pstn, but i can't do any call to pstn. I attach my /etc/misdn-init.conf/ and also my etc/asterisk/misdn.conf file for clarity. Thanks - Daniele ___ --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] no outgoing calls with Digium B410P
Hi Daniele, Could you please tell us what exactly happens? Are your able to see some error in the log/console? Thanks for your answer. I'm managing the asterisk server from a windows client via ssh (putty client), so i can't paste here the output of the asterisk CLI, but when i try to do a call to the PSTN i see a lot o f messages, but no one of them looks like an error. They are of this type: -- Executing [EMAIL PROTECTED]:1] Macro (SIP/501-0827c9e82, dialout-trunk|1|0266200xxx||) in new stack 0266200xxx is the number i'm trying calling to, 501 is my extension number (i have one SIP hard-phone); the number 1 before 0266200xxx is part of the dial patterns i created, because i want to dial 1 before the outgoing number. Anyway, on the official digium documentation, it's written that in order to call out over a port, the Dial () command has to be formatted as follows: Dial (misdn/g:myoutsidelines/ $ {EXTEN}). Have i to edit my extension.confand insert in a dial command to do an outgoing call? Thanks a lot - Daniele ___ --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] no outgoing calls with Digium B410P
daniele visaggio wrote: Thanks for your answer. I'm managing the asterisk server from a windows client via ssh (putty client), so i can't paste here the output of the asterisk CLI, but when i try to do a call to the PSTN i see a lot o f messages, but Putty does support copying all the the clipboard. It's in the pull down menu. I usually paste it into a notepad document and remove unwanted text. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no outgoing calls with Digium B410P
Daniele, you need an external calls rule in your extension.conf, that is 1 to call using PSTN line. Please send your extension and we can take a look to find your problem. p.s. I'm Italian too. On Jan 7, 2008 3:03 PM, daniele visaggio [EMAIL PROTECTED] wrote: Hi Daniele, Could you please tell us what exactly happens? Are your able to see some error in the log/console? Thanks for your answer. I'm managing the asterisk server from a windows client via ssh (putty client), so i can't paste here the output of the asterisk CLI, but when i try to do a call to the PSTN i see a lot o f messages, but no one of them looks like an error. They are of this type: -- Executing [EMAIL PROTECTED]:1] Macro (SIP/501-0827c9e82, dialout-trunk|1|0266200xxx||) in new stack 0266200xxx is the number i'm trying calling to, 501 is my extension number (i have one SIP hard-phone); the number 1 before 0266200xxx is part of the dial patterns i created, because i want to dial 1 before the outgoing number. Anyway, on the official digium documentation, it's written that in order to call out over a port, the Dial () command has to be formatted as follows: Dial (misdn/g:myoutsidelines/ $ {EXTEN}). Have i to edit my extension.confand insert in a dial command to do an outgoing call? Thanks a lot - Daniele ___ --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] no outgoing calls with Digium B410P
On Mon, Jan 07, 2008 at 03:03:18PM +0100, daniele visaggio wrote: I'm managing the asterisk server from a windows client via ssh (putty client), so i can't paste here the output of the asterisk CLI, Huh??? * You can record history in putty. * You can manually copy text from putty( just mark the text) * You can save text as a local file (e.g: copy from the log and edit) and copy it via sftp (e.g: winscp). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no outgoing calls with Digium B410P
2008/1/7, map [EMAIL PROTECTED]: Daniele, you need an external calls rule in your extension.conf, that is 1 to call using PSTN line. Please send your extension and we can take a look to find your problem. p.s. I'm Italian too. Ok, i attach my extension.conf. Thank you very much, i'm very happy of finding another italian asterisk user. Ciao e grazie! extension.conf.tar.gz Description: GNU Zip compressed data ___ --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] no outgoing calls with Digium B410P
Tzafrir Cohen wrote: On Mon, Jan 07, 2008 at 03:03:18PM +0100, daniele visaggio wrote: I'm managing the asterisk server from a windows client via ssh (putty client), so i can't paste here the output of the asterisk CLI, Huh??? * You can record history in putty. * You can manually copy text from putty( just mark the text) * You can save text as a local file (e.g: copy from the log and edit) and copy it via sftp (e.g: winscp). PuTTY IS the way to manage Asterisk. One can scroll back, can set the scrollback buffer to many thousands of lines if need be, cut and paste with ease directly into an E-mail and the suite also comes with PSFTP to do file transfer. The ONLY issue I have had with PuTTY is ( my ) inability to run make menuselect. regardless of how I set PuTTY, it complains about terminal size. Otherwise I have no reason to even use the console on the Asterisk box. John Novack -- Dog is my co-pilot ___ --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] no outgoing calls with Digium B410P
On Mon, Jan 07, 2008 at 10:19:11AM -0500, John Novack wrote: The ONLY issue I have had with PuTTY is ( my ) inability to run make menuselect. regardless of how I set PuTTY, it complains about terminal size. Hmm.. I have never encountered this. Sounds like a bug. Can you point me to a bug report or to paste here the exact message? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no outgoing calls with Digium B410P
2008/1/7, map [EMAIL PROTECTED]: Hi Daniele, Please send a snapshot of your Putty Asterisk log. Go to Putty configuration - Window - Lines of scrollback and put a number greater than 200 :-). I suggest 10. Sorry, i'm using a Linux version of putty (i'm running Ubuntu) and the configuration is different. I can't find Lines of scrollback and modify the scrollback number. The putty-linux sw-structure is probably different from the putty-windows one. ___ --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] no outgoing calls with Digium B410P
Hi Daniele, Please send a snapshot of your Putty Asterisk log. Go to Putty configuration - Window - Lines of scrollback and put a number greater than 200 :-). I suggest 10. On Jan 7, 2008 4:00 PM, daniele visaggio [EMAIL PROTECTED] wrote: 2008/1/7, map [EMAIL PROTECTED]: Daniele, you need an external calls rule in your extension.conf, that is 1 to call using PSTN line. Please send your extension and we can take a look to find your problem. p.s. I'm Italian too. Ok, i attach my extension.conf. Thank you very much, i'm very happy of finding another italian asterisk user. Ciao e grazie! ___ --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] no outgoing calls with Digium B410P
On Jan 7, 2008, at 8:53 AM, daniele visaggio wrote: 2008/1/7, map [EMAIL PROTECTED]: Hi Daniele, Please send a snapshot of your Putty Asterisk log. Go to Putty configuration - Window - Lines of scrollback and put a number greater than 200 :-). I suggest 10. Sorry, i'm using a Linux version of putty (i'm running Ubuntu) and the configuration is different. I can't find Lines of scrollback and modify the scrollback number. The putty-linux sw-structure is probably different from the putty-windows one. you can use a regular terminal and issue the script command all the output will be saved to a file, ___ --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] no outgoing calls with Digium B410P
daniele visaggio wrote: 2008/1/7, map [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi Daniele, Please send a snapshot of your Putty Asterisk log. Go to Putty configuration - Window - Lines of scrollback and put a number greater than 200 :-). I suggest 10. Sorry, i'm using a Linux version of putty (i'm running Ubuntu) and the configuration is different. I can't find Lines of scrollback and modify the scrollback number. The putty-linux sw-structure is probably different from the putty-windows one. ___ --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 Since you are using linux, open an Xterm window and issue the following command: ssh -l user id name or ip address of * This should prompt you to verify the ssh key the first time and then ask for your password. Cut and paste works from an XTerm window. Lyle ___ --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] no outgoing calls with Digium B410P
On Mon, Jan 07, 2008 at 04:53:03PM +0100, daniele visaggio wrote: 2008/1/7, map [EMAIL PROTECTED]: Hi Daniele, Please send a snapshot of your Putty Asterisk log. Go to Putty configuration - Window - Lines of scrollback and put a number greater than 200 :-). I suggest 10. Sorry, i'm using a Linux version of putty (i'm running Ubuntu) and the configuration is different. I can't find Lines of scrollback and modify the scrollback number. The putty-linux sw-structure is probably different from the putty-windows one. Nag nag nag :-) The menu in that version of putty is available by holding the ctrl key and pressing the right mouse key. The right mouse key alone extends the selection, like in xterm. In that menu you have copy all, as well as the ability to change settings (scroll-buffer, log file, whatever). Or jut select text with the mouse and it will be automatically copied. Or use ssh from any other terminal, as others here have mentioned. I normally use .ssh/config for maintaning aliases and special settings. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users