Re: [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones
Hi Carlos, On 04/28/2013 10:56 PM, Carlos Alvarez wrote: We have a new customer with a lot of old phones like the 9133i. They won't register, and we see some very strange behavior with them. If the SIP peer exists, they simply fail silently, with no error in the CLI or the messages log. Nothing works, but no errors. If the peer does not exist, it's clear that it's registering improperly: [2013-04-28 13:34:31] NOTICE[3058] chan_sip.c: Registration from 'abc123 sip:abc123@' failed for '68.2.x.x' - No matching peer found Typically of course we'd expect to see: sip:abc123@server We're running the latest available firmware, but it's from 2009. Any ideas on this before we just trash all the older phones? I reviewed one of those a long time ago. I'm afraid all I can remember is that it had its fair share of issues. I did a lot of factory resets and had to set the config through tftp *or* the UI but not mix both. Did you try doing factory resets (page 9-11 in admin guide) and/or downgrading to an older firmware release? Main info page: http://www.aastra.ca/document-library.htm?curr_fam=Aastra+9000icurr_nav=2prod_id=6441 The 1.4.0 firmware: https://www.voipon.co.uk/products/download/aastra_9133i.zip The 1.4.3 firmware: http://www.aastra.ca/cps/rde/aareddownload?file_id=6441-6449-_P07_XMLdsproject=www-aastratelecom-commtype=zip Release notes of 1.4.3: http://www.aastra.ca/cps/rde/aareddownload?file_id=6441-6427-_P07_XMLdsproject=www-aastratelecom-commtype=pdf Manual http://www.aastra.ca/cps/rde/aareddownload?file_id=6441-6425-_P07_XMLdsproject=www-aastratelecom-commtype=pdf Regards, Patrick -- _ -- 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
[asterisk-users] Asterisk 11.3.0 - Mask for new file not correct
Hello, I'm facing a rights issue on with Asterisk 11.3.0 running on CentOS release 5.8. Asterisk process is running with asterisk since it is define in asterisk.conf as following: runuser = asterisk rungroup = asterisk You can see asterisk proccess here: ps aux |egrep 'python|asterisk' root 11581 0.0 0.1 65940 600 ?SApr17 0:00 /bin/sh /usr/sbin/safe_asterisk asterisk 11583 0.0 3.8 341560 19440 ?Sl Apr17 2:02 /usr/sbin/asterisk -f -vvvg -c The fact is we want to use the RECORDED_FILE function from Application_Record module and create a file with 666 permissions. But when I check the created file, rights are not what I expected. [root@STD1-SRVASTSVI-03 pseudos]$ ll -rw-r--r-- 1 asterisk asterisk 51244 mars 29 16:04 Pseudo_2_.wav I checked the doc on https://wiki.asterisk.org/wiki/display/AST/Application_Record but I didn't find anything about umask permissions. I checked Doxygen, I can see file creation permissions is set to 666 #define AST_FILE_MODE 0666 http://doxygen.asterisk.org/trunk/asterisk_8h.html#a6293b2dae52a2b470494df672a26c42 What can I do to fix that or debug? Ludovic BOUÉ -- _ -- 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] Can't register to Asterisk 1.6 with old Aastra phones
Well the solution turned out to be putting the Asterisk server name in the Proxy field as well as in the server field. Then it properly formatted the SIP registration request. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Asterisk 11.3.0 - Mask for new file not correct
On Apr 29, 2013, at 10:51 AM, Ludovic Boué wrote: The fact is we want to use the RECORDED_FILE function from Application_Record module and create a file with 666 permissions. But when I check the created file, rights are not what I expected. [root@STD1-SRVASTSVI-03 pseudos]$ ll -rw-r--r-- 1 asterisk asterisk 51244 mars 29 16:04 Pseudo_2_.wav I checked the doc on https://wiki.asterisk.org/wiki/display/AST/Application_Record but I didn't find anything about umask permissions. I checked Doxygen, I can see file creation permissions is set to 666 #define AST_FILE_MODE 0666 http://doxygen.asterisk.org/trunk/asterisk_8h.html#a6293b2dae52a2b470494df672a26c42 What can I do to fix that or debug? The AST_FILE_MODE works by the same rules as mode parameter in open(2): The effective permissions are modified by the process's umask in the usual way: The permissions of the created file are (mode ~umask).[1] My guess is that the umask of your asterisk process is 022, which is very typical. You'll have to play around with your umask settings and file permissions to get things the way you want them. [1]: http://linux.die.net/man/2/open Ludovic BOUÉ -- David M. Lee Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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
[asterisk-users] Gateway?
This is going to sound like a dumb-ass question: The device that allows you to bridge Asterisk (or any other PBX) into the pstn.. What is that called? So, I guess, not a SIP trunk, but the device that actually IS the SIP trunk. Am I making sense? Thanks -- _ -- 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] Gateway?
Here are your answers: 1st question: Anything that makes sense. 2nd question: Maybe Please, explain your setup. jg -- _ -- 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
