[asterisk-users] Asterisk 1.6 Overlap dialling timeout?
Hello, I'm experimenting with Overlap Dialling in asterisk 1.6. I've enabled this in sip.conf and on the SNOM 300 phone. My problem is that asterisk dials out as soon as it matches an extension without waiting to see if the user is going to type in more digits. Is there a way to set a timeout per channel or globally? I'd like Asterisk to wait for a few seconds once its found a match in case the user needs to key in more digits. Thank You. Regards, Veselin K -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Minimum hardware requirements for 10 concurrent calls?
Any advise? Thank you. Veselin K On Wed, Nov 04, 2009 at 04:51:49PM +, vese...@campbell-lange.net wrote: Hello, I'm considering an Asterisk box for up to 10-15 concurrent calls. Incoming PSTN/ISDN/IAX2, outgoing PSTN/ISDN/IAX2. Could someone roughly suggest the minimal hardware requirements for this kind of setup? Trying to come up with the cheapest solution. Thank you. Veselin K ___ -- 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] Caching Asterisk SIP useragent info?
Apologies that I did not make myself clear the first time, I meant the process of Asterisk saving useragent data for its users. Is it configurable via asterisk or is it just the re-register settings on the SNOM phone? Thanks again Paul. Veselin K On Tue, Nov 18, 2008 at 10:21:14AM +1100, Paul Hales wrote: The process for upgrading would greatly depend on how Asterisk was installed in the first place. If Asterisk was installed from source, then a fresh download of source followed by the usual configure/make/etc commands would do the trick. PaulH Veselin K wrote: Hello Paul, thanks for the reply. Could you please tell me what is the process called so I can research it further. Thank you. Veselin K On Mon, Nov 17, 2008 at 10:47:47AM +1100, Paul Hales wrote: This process has been greatly improved in the latest versions of Asterisk - might be time to upgrade. PaulH [EMAIL PROTECTED] wrote: Hello, I'm running an Asterisk 1.4.14 on a linux machine. Serving SIP Snom users. I've noticed that each time Asterisk is restarted, for the first 5-10 minutes, the SIP users can dial but cannot be dialed until each phone re-registers itself against the server. So only after the Saved useragent...for peer 111 line appears on the Asterisk console, then the 111 user can be reached. What exactly is this process? Is it that the phones send their extension/password details to the server at specific intervals or does the server send a broadcast message, looking for phones? Is there any way to cache/save this SIP useragent information so in case the server is restarted, the user need not wait for their phone to re-register? Also I believe that it is sufficient for the user to just pickup their handset in order to force their phone to re-register quicker. However I'd like to avoid asking the users to do that. Thank you much. Veselin K ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Caching Asterisk SIP useragent info?
Hello Paul, thanks for the reply. Could you please tell me what is the process called so I can research it further. Thank you. Veselin K On Mon, Nov 17, 2008 at 10:47:47AM +1100, Paul Hales wrote: This process has been greatly improved in the latest versions of Asterisk - might be time to upgrade. PaulH [EMAIL PROTECTED] wrote: Hello, I'm running an Asterisk 1.4.14 on a linux machine. Serving SIP Snom users. I've noticed that each time Asterisk is restarted, for the first 5-10 minutes, the SIP users can dial but cannot be dialed until each phone re-registers itself against the server. So only after the Saved useragent...for peer 111 line appears on the Asterisk console, then the 111 user can be reached. What exactly is this process? Is it that the phones send their extension/password details to the server at specific intervals or does the server send a broadcast message, looking for phones? Is there any way to cache/save this SIP useragent information so in case the server is restarted, the user need not wait for their phone to re-register? Also I believe that it is sufficient for the user to just pickup their handset in order to force their phone to re-register quicker. However I'd like to avoid asking the users to do that. Thank you much. Veselin K ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
