[asterisk-users] Problem installing B410P BRI card for asterisk
Dear all, I know this is more a Digium hardware than an Asterisk issue. Already posted a question at Digium, however also like to see whether anyone in the Asterisk community has encountered the following situation: I installed a Digium B410P BRI PCI card on my new asterisk server, following the steps specified in the manual. I can see the PCI card is available using the lspci command: ... 04:00.0 Ethernet controller [0200]: Intel Corporation 82574L Gigabit Network Connection [8086:10d3] 05:00.0 Ethernet controller [0200]: Intel Corporation 82574L Gigabit Network Connection [8086:10d3] 08:00.0 PCI bridge [0604]: ASPEED Technology, Inc. AST1150 PCI-to-PCI Bridge [1a03:1150] (rev 02) 09:00.0 VGA compatible controller [0300]: ASPEED Technology, Inc. ASPEED Graphics Family [1a03:2000] (rev 10) 0a:01.0 ISDN controller [0204]: Digium, Inc. Wildcard B410 quad-BRI card [d161:b410] (rev 01) ... I specified the following in my system.conf in /etc/dahdi: loadzone = nl defaultzone = nl span = 1,1,0,ccs,ami bchan = 1,2 hardhdlc = 3 I loaded the driver using sudo modprobe wcb4xxp. Next I ran dahdi_cfg -vv which returns: DAHDI Tools Version - 2.5.0.2 DAHDI Version: 2.5.0.2 Echo Canceller(s): HWEC Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: none) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: none) (Slaves: 02) Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03) 3 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) I'm in doubt about the DAHDI_SPANCONFIG failed on span 1: No such device or address (6). Next, if i execute sudo dmesg as specified by the manual it returns a huge trace: [ 376.082907] Wrote 0x0 to register 0x1ab but got back 0x4 [ 376.594754] Wrote 0x0 to register 0x1ab but got back 0x4 [ 377.106605] Wrote 0x0 to register 0x1ab but got back 0x4 [ 377.618423] Wrote 0x0 to register 0x1ab but got back 0x4 [ 378.130266] Wrote 0x0 to register 0x1ab but got back 0x4 [ 378.642088] Wrote 0x0 to register 0x1ab but got back 0x4 [ 1202.812870] show_signal_msg: 21 callbacks suppressed [ 1202.812876] dahdi_tool[1277]: segfault at 3fc378fa0 ip 004021ac sp 7fff131dd930 error 4 in dahdi_tool[40+3000] And a lot of Wrote 0x0 to register 0x1ab but got back 0x4 statements. If i run dahdi_tools it fails with a segmentation fault. Any suggestions are appreciated! Kind regards, Marco Mooijekind. -- _ -- 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 1.4.42 NOTIFY replies ignore NAT setting
Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even with Symmetric NAT on). The reason for this is that the NOTIFY reply does not attempt to lookup the SIP peer and check its NAT flags. I've seen some nasty From: header string parsing code + find_peer() that does this, but I was wondering if there's an easier way. Thanks. -- James -- _ -- 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] Best practise for operator
Hi, My question in short: what is the best practise for a small company (30-50 users) with one operator that answers each call? I am a developer and now working on the implementation of Asterisk in our company. With the definitive guide next to me, the scripting of extensions and configuring trunks isn't that hard from what I have read until now. What worries me is what the best way would be to get our operator to work nicely / efficiently. On one location we have a receptionist who takes the calls to our main number. She would transfer when needed. We also have about 7 other locations (retail stores) which we are going to connect to the same Asterisk box. We will switch to voip completely using a voip provider, so everything will be SIP. First of all: to have seperate lines, would one usually use seperate SIP accounts? I have seen enterprise SIP phones who say they support 6 lines, and then also support 6 SIP accounts. Does this mean a line equals a SIP account? Are there people here who have Asterisk running in their company already with an operator? What software or phone do you use? What is the setup (extension config) in Asterisk? I have read about the Shared Line stuff (SLA right?), but this would only be needed if several people need to see a light when a line is busy, and able to exchange this line. Why would one do that if you have the ability to transfer calls, use queues, and park calls? Would the best setup be to create a receptionist queue, with only one member/agent in it? Or could I better use several lines on one SIP account (although I still haven't figured out how to do that). Or maybe six SIP accounts on the enterprise device and connect them so an extension using Dial(sip1sip2sip3etc...)? Any recommendations are more than welcome! I am interested in how other companies solved this. I have checked out FOP2 (by installing it), but by the looks of it now I'd have to change the buttons config each time extensions and/or users change. Would be better if this came straight out of Asterisk. Any other software operator/switch boards that really work. Maybe there's only a few that really stand out? I have also checked Voice Operator Panel on the website. Any reviews on that? I hope somebody would like to share their setup. Thanks in advance! Kind regards, Roland. -- _ -- 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] Problem installing B410P BRI card for asterisk
On 12/30/2011 02:44 AM, Marco Mooijekind wrote: Dear all, I know this is more a Digium hardware than an Asterisk issue. Already posted a question at Digium, however also like to see whether anyone in the Asterisk community has encountered the following situation: I installed a Digium B410P BRI PCI card on my new asterisk server, following the steps specified in the manual. I can see the PCI card is available using the lspci command: ... 04:00.0 Ethernet controller [0200]: Intel Corporation 82574L Gigabit Network Connection [8086:10d3] 05:00.0 Ethernet controller [0200]: Intel Corporation 82574L Gigabit Network Connection [8086:10d3] 08:00.0 PCI bridge [0604]: ASPEED Technology, Inc. AST1150 PCI-to-PCI Bridge [1a03:1150] (rev 02) 09:00.0 VGA compatible controller [0300]: ASPEED Technology, Inc. ASPEED Graphics Family [1a03:2000] (rev 10) 0a:01.0 ISDN controller [0204]: Digium, Inc. Wildcard B410 quad-BRI card [d161:b410] (rev 01) ... I specified the following in my system.conf in /etc/dahdi: loadzone = nl defaultzone = nl span = 1,1,0,ccs,ami bchan = 1,2 hardhdlc = 3 I loaded the driver using sudo modprobe wcb4xxp. Next I ran dahdi_cfg -vv which returns: DAHDI Tools Version - 2.5.0.2 DAHDI Version: 2.5.0.2 Echo Canceller(s): HWEC Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: none) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: none) (Slaves: 02) Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03) 3 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) I'm in doubt about the DAHDI_SPANCONFIG failed on span 1: No such device or address (6). Next, if i execute sudo dmesg as specified by the manual it returns a huge trace: [ 376.082907] Wrote 0x0 to register 0x1ab but got back 0x4 [ 376.594754] Wrote 0x0 to register 0x1ab but got back 0x4 [ 377.106605] Wrote 0x0 to register 0x1ab but got back 0x4 [ 377.618423] Wrote 0x0 to register 0x1ab but got back 0x4 [ 378.130266] Wrote 0x0 to register 0x1ab but got back 0x4 [ 378.642088] Wrote 0x0 to register 0x1ab but got back 0x4 [ 1202.812870] show_signal_msg: 21 callbacks suppressed [ 1202.812876] dahdi_tool[1277]: segfault at 3fc378fa0 ip 004021ac sp 7fff131dd930 error 4 in dahdi_tool[40+3000] And a lot of Wrote 0x0 to register 0x1ab but got back 0x4 statements. If i run dahdi_tools it fails with a segmentation fault. Any suggestions are appreciated! Either your B410P card is misbehaving (i.e. broken), or there is some sort of PCI compatibility issue between it and the system you have it installed in. This is a hardware issue, and should be pursed with Digium Support. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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
Re: [asterisk-users] Asterisk 1.4.42 NOTIFY replies ignore NAT setting
On 12/30/2011 04:07 AM, James Lamanna wrote: Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even with Symmetric NAT on). The reason for this is that the NOTIFY reply does not attempt to lookup the SIP peer and check its NAT flags. I've seen some nasty From: header string parsing code + find_peer() that does this, but I was wondering if there's an easier way. Since Asterisk does not initiate subscriptions, these NOTIFY requests arriving to the Asterisk system must be 'unsolicited'. As such, they don't have an associated SIP dialog structure, so there's no simple way to know whether they are associated with a known peer or not. You say that Asterisk's behavior is 'incorrect', but it's only 'incorrect' because you believe it should be looking up any associated peer and using that peer's NAT setting; Asterisk's behavior as you've quoted is *correct* according to the RFC3261 rules for how replies should be sent, assuming that the top-most Via header does not have an 'rport' parameter present in it. The *proper* way to solve this problem is to have the UA sending the NOTIFY request include the 'rport' parameter in the top-most Via header of the request; if that is done, then whatever UA receives the request will be able to properly respond, even if the request crosses a NAT. Another way to solve it, if the sending UA cannot be changed to emit proper SIP requests, is to modify Asterisk to attempt a peer lookup when it is going to reply to request that it cannot associate with any known dialog, and then have the peer configured with 'nat=yes' (in the case of 1.4.42). A third option is to set 'nat=yes' in the [general] section of sip.conf, so that Asterisk will reply using rport-style behavior regardless of whether the request could be associated with a peer or not. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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
Re: [asterisk-users] Block Specific Number on Inbound
Here's what I do... Changed some variables for obscurity. 911 is the inbound
#... exten 6000 rings to SIP/TEST
exten = 911,1,GotoIf(${BLACKLIST()}?blacklisted)
exten = 911,n,Macro(stdexten,6000,SIP/test)
exten = 911,n,Playback(transfer,skip)
exten = 911,n(blacklisted),Goto(blacklisted,s,1)
Blacklisted context uses zapateller and plays the intercept message
[blacklisted]
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,Zapateller
exten = s,4,Zapateller
exten = s,5,Playback(ss-noservice)
exten = s,6,Hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stuart Sheldon
Sent: Thursday, December 29, 2011 9:06 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Block Specific Number on Inbound
Check out the X Boy/Girl friend feature.
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf
Around the middle of the page.
Stu
-Original Message-
From: Kevin Oravits korav...@rcolegal.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Subject: [asterisk-users] Block Specific Number on Inbound
Date: Fri, 30 Dec 2011 01:39:46 +
Greetings,
Is there a way to block a specific inbound number? I’ve found code online for
blocking all nocallerid and all 800, etc. but nothing for a specific number. My
company is wanting me to block a specific number. Is this possible in Asterisk
1.4 and 1.6 or do I need to go through my Service Provider?
Thanks,
Kevin Oravits
Phone Sys Admin
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[asterisk-users] Recompile Asterisk
I need to recompile asterisk to install some apps. Does anyone has a good tutorial on how to do this ? I've asterisk 1.4.13 installed should I upgrade to 1.6 ? will I benifit from video improvements if I upgrade ? Will I loose asterisk settings if I recompile ? If you have previously recompiled asterisk please let me know your experience! Any information about this will be very welcome. Thanks! Regards, LL -- _ -- 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] Recompile Asterisk
LL wrote: I need to recompile asterisk to install some apps. Does anyone has a good tutorial on how to do this From the statement above, you're giving conflicting information. You first sounded like the installation was compiled from source, but then you're asking for a tutorial on how to compile. If the installation was installed by your distro's package manager, then you won't want to install from source. Can you tell us how it was installed? 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 -- 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] wait_for_answer: Unable to write frametype: 2 on Asterisk *CLI
Hi, I get this warning [Dec 30 19:59:20] WARNING[17666]: app_dial.c:1353 wait_for_answer: Unable to write frametype: 2 on Asterisk *CLI Any clue ? Please let me know if anyone needs any additional information or details of configuration files. Regards, Kaushal -- _ -- 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 1.4.42 NOTIFY replies ignore NAT setting
On Fri, Dec 30, 2011 at 6:02 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 12/30/2011 04:07 AM, James Lamanna wrote: Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even with Symmetric NAT on). The reason for this is that the NOTIFY reply does not attempt to lookup the SIP peer and check its NAT flags. I've seen some nasty From: header string parsing code + find_peer() that does this, but I was wondering if there's an easier way. Since Asterisk does not initiate subscriptions, these NOTIFY requests arriving to the Asterisk system must be 'unsolicited'. As such, they don't have an associated SIP dialog structure, so there's no simple way to know whether they are associated with a known peer or not. You say that Asterisk's behavior is 'incorrect', but it's only 'incorrect' because you believe it should be looking up any associated peer and using that peer's NAT setting; Asterisk's behavior as you've quoted is *correct* according to the RFC3261 rules for how replies should be sent, assuming that the top-most Via header does not have an 'rport' parameter present in it. The *proper* way to solve this problem is to have the UA sending the NOTIFY request include the 'rport' parameter in the top-most Via header of the request; if that is done, then whatever UA receives the request will be able to properly respond, even if the request crosses a NAT. Another way to solve it, if the sending UA cannot be changed to emit proper SIP requests, is to modify Asterisk to attempt a peer lookup when it is going to reply to request that it cannot associate with any known dialog, and then have the peer configured with 'nat=yes' (in the case of 1.4.42). A third option is to set 'nat=yes' in the [general] section of sip.conf, so that Asterisk will reply using rport-style behavior regardless of whether the request could be associated with a peer or not. Thanks Kevin. I'll have to turn rport on on all my Linksys/Cisco phones and give it a shot. -- James -- _ -- 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 fail2ban filters - show us yours
On 12/29/2011 01:55 PM, Bruce B wrote:
Hi,
I Have added this line for asterisk 1.8 (i have allowguest=yes and
context=default in sip.conf):
NOTICE.* .*: Call from '.*' (HOST) to extension '.*' rejected
because extension not found in context 'default'.
Em 29-12-2011 13:03, Patrick Lists escreveu:
Hi,
In the thread Interesting attack tonight fail2ban them Bruce
B mentioned it would be nice to have input from the Community to
come up with the best set of fail2ban filters. That's a great
idea. So let's start with Bruce's filters (thanks!) and take it
from there. Anyone have any improvements and/or additions?
Apologies for the line wrap. No idea how to prevent that in
Thunderbird. The filters are also at http://pastebin.com/6T9M1W3F
Not sure but it may be possible that logging has changed between
Asterisk 1.4, 1.6, 1.8 and 10 so please mention the asterisk
version with your filters.
For Asterisk 1.8:
failregex = Registration from '.*' failed for
'HOST(:[0-9]{1,5})?' - Wrong password
Registration from '.*' failed for
'HOST(:[0-9]{1,5})?' - No matching peer found
Registration from '.*' failed for
'HOST(:[0-9]{1,5})?' - Device does not match ACL
Registration from '.*' failed for
'HOST(:[0-9]{1,5})?' - Username/auth name mismatch
Registration from '.*' failed for
'HOST(:[0-9]{1,5})?' - Peer is not supposed to register
NOTICE.* HOST failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' (from HOST)
NOTICE.* .*: Host HOST failed MD5 authentication
for '.*' (.*)
VERBOSE.* logger.c: -- .*IP/HOST-.* Playing
'ss-noservice' (language '.*')
There are 2 lines that I have which are not in this list:
NOTICE.* .*: Registration from '.*' failed for 'HOST' - ACL
error (permit/deny)
NOTICE.* .*: Failed to authenticate user .*@HOST.*
How about those (no idea for which Asterisk version they are)?
Regards,
Patrick
Thanks Patrick. This is a great initiative. Let's all build the
strongest and most detailed filter possible. I actually looked at mine
and now see that it has weaknesses due Asterisk 1.8.8x giving
different type of logs or maybe FreePBX. Let's test, fix and append to
the end of the filter. Everyone is welcome to contribute.
So far we have:
*For Asterisk 1.8:*
failregex = Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Wrong password
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
No matching peer found
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Device does not match ACL
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Username/auth name mismatch
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Peer is not supposed to register
NOTICE.* HOST failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' (from HOST)
NOTICE.* .*: Host HOST failed MD5 authentication for '.*'
(.*)
VERBOSE.* logger.c: -- .*IP/HOST-.* Playing
'ss-noservice' (language '.*') *#Outdated?*
#*Situation:* allowguest=yes and context=default in sip.con
- *Tested by **Diego Aguirre?*
NOTICE.* .*: Call from '.*' (HOST) to extension '.*' rejected
because extension not found in context 'default'
The following are what I found to be insecure but need escaping and
fine tuning to work with filter:
*Asterisk 1.8 + FreePBX:*
*Situation:* When target is coming in from unknown DID -
Needs character escaping
Executing [unknown@from-sip-external:1] NoOp(SIP/10.0.0.6-0001,
Received incoming SIP connection from unknown peer to unknown) in
new stack
*Situation:* Same as above except for an extension is called. Above
was just IP call. Extension 011x doesn't exist.
Executing [011566@from-sip-external:1]
NoOp(SIP/10.0.0.6-0003, Received incoming SIP connection from
unknown peer to 011566) in new stack
*Situation: *Same as above except for extension 101 does exist but
system still rejects calls due to no guest allowed?!
Executing [101@from-sip-external:1] NoOp(SIP/10.0.0.6-0005,
Received incoming SIP connection from unknown peer to 101) in new stack
*All of above have this following which can be used as a universal
filter: *Executing [s@from-sip-external:8]
Playback(SIP/10.0.0.6-0005, ss-noservice) in new stack *
*
*
***Notice how this ss-noservice is difference from current the
outdated filter one:
*VERBOSE.* logger.c: -- .*IP/HOST-.* Playing 'ss-noservice'
(language '.*')*
-Bruce
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Re: [asterisk-users] Asterisk fail2ban filters - show us yours
Had one of my systems hit this morning too. Asterisk 1.8 branch+FreePBX 2.9 no anonymous. 260 call attemps in 2 minutes. Here is part of the logs. I am updating my filter to see if it helps, THANKS Bruce!!! I am trying to get this working for FreePBX as I think they are more vulnerable than the vanilla Asterisk setups. Some of the charecters might have to be escaped and I am not an expert in Python but trying to learn it so I will post back my findings. In the meanwhile it would be great if others share their findings as well. -- _ -- 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] Recompile Asterisk
I'm sorry if I gave conflicting information but I didn't setup this specific freepbx/asterisk and I'm not sure if it was compiled from source, probably not, but I'll have to double check it the person that installed it. I know that I currently running asterisk 1.4.13 + FreePBX 2.9.0.7 on CentOS 5 x64. I need to compile app_mp4 and I've read that I need to compile asterisk from source in order to achieve this, does this make any sense ? I'm following a tutorial on how to compile app_mp4 at http://web.archive.org/web/20090322060930/http://sip.fontventa.com/content/view/15/44/, but I'm afraid to go ahead and lose my current freepbx/asterisk settings. I'm pretty new to asterisk, based on this, some of my questions may not the the million dollar questions..., basically, what I need is to install the app_mp4 and several others apps and make sure the system config (extensions, trunks, etc) is preserved. any good advises ? Thank you all, regards LL On 12/30/2011 2:21 PM, Doug Lytle wrote: LL wrote: I need to recompile asterisk to install some apps. Does anyone has a good tutorial on how to do this From the statement above, you're giving conflicting information. You first sounded like the installation was compiled from source, but then you're asking for a tutorial on how to compile. If the installation was installed by your distro's package manager, then you won't want to install from source. Can you tell us how it was installed? Doug -- _ -- 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 1.4.42 NOTIFY replies ignore NAT setting
On Fri, Dec 30, 2011 at 8:35 AM, James Lamanna jlama...@gmail.com wrote: On Fri, Dec 30, 2011 at 6:02 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 12/30/2011 04:07 AM, James Lamanna wrote: Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even with Symmetric NAT on). The reason for this is that the NOTIFY reply does not attempt to lookup the SIP peer and check its NAT flags. I've seen some nasty From: header string parsing code + find_peer() that does this, but I was wondering if there's an easier way. Since Asterisk does not initiate subscriptions, these NOTIFY requests arriving to the Asterisk system must be 'unsolicited'. As such, they don't have an associated SIP dialog structure, so there's no simple way to know whether they are associated with a known peer or not. You say that Asterisk's behavior is 'incorrect', but it's only 'incorrect' because you believe it should be looking up any associated peer and using that peer's NAT setting; Asterisk's behavior as you've quoted is *correct* according to the RFC3261 rules for how replies should be sent, assuming that the top-most Via header does not have an 'rport' parameter present in it. The *proper* way to solve this problem is to have the UA sending the NOTIFY request include the 'rport' parameter in the top-most Via header of the request; if that is done, then whatever UA receives the request will be able to properly respond, even if the request crosses a NAT. Another way to solve it, if the sending UA cannot be changed to emit proper SIP requests, is to modify Asterisk to attempt a peer lookup when it is going to reply to request that it cannot associate with any known dialog, and then have the peer configured with 'nat=yes' (in the case of 1.4.42). A third option is to set 'nat=yes' in the [general] section of sip.conf, so that Asterisk will reply using rport-style behavior regardless of whether the request could be associated with a peer or not. Thanks Kevin. I'll have to turn rport on on all my Linksys/Cisco phones and give it a shot. Hi Kevin, That doesn't appear to work correctly: The response does not come back to 34972 even though rport is in the Via. U xxx.234:34972 - yyy..7:5060 NOTIFY sip:yyy.7 SIP/2.0..Via: SIP/2.0/UDP 10.132.38.19:6957;branch=z9hG4bK-25ea41f0;rport..From: 1316 sip:1316@yyy.7;tag=80f427ae9e884ado0..To: sip:yyy .7..Call-ID: 4fa38a62-b7d76...@10.132.38.19..cseq: 1 NOTIFY..Max-Forwards: 70..Contact: 1316 sip:1316@10.132.38.19:6957..Event: keep-alive..User-Agent: Linksys/SPA942-6.1.3( a)..Content-Length: 0 # U yyy.7:5060 - xxx.234:6957 SIP/2.0 481 No subscription..Via: SIP/2.0/UDP 10.132.38.19:6957;branch=z9hG4bK-25ea41f0;received=xxx.234;rport=34972..From: 1316 sip:1316@yyy.7;tag=80f427ae9e884 ado0..To: sip:yyy.7;tag=as07ad17b5..Call-ID: 4fa38a62-b7d76...@10.132.38.19..cseq: 1 NOTIFY..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY, INFO..Supported: replaces..Content-Length: 0 -- _ -- 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] Recompile Asterisk
The safest (IMO) method for installing new settings would be to download 1.4.13 (if you can find that old of an archive) and build it on another machine, then copy the needed .so modules to /usr/local/asterisk/modules on your running machine. You should be able to see if it was compiled or built from an rpm by doing show version from CLI. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of LL Sent: Friday, December 30, 2011 11:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Recompile Asterisk I'm sorry if I gave conflicting information but I didn't setup this specific freepbx/asterisk and I'm not sure if it was compiled from source, probably not, but I'll have to double check it the person that installed it. I know that I currently running asterisk 1.4.13 + FreePBX 2.9.0.7 on CentOS 5 x64. I need to compile app_mp4 and I've read that I need to compile asterisk from source in order to achieve this, does this make any sense ? I'm following a tutorial on how to compile app_mp4 at http://web.archive.org/web/20090322060930/http://sip.fontventa.com/content/v iew/15/44/, but I'm afraid to go ahead and lose my current freepbx/asterisk settings. I'm pretty new to asterisk, based on this, some of my questions may not the the million dollar questions..., basically, what I need is to install the app_mp4 and several others apps and make sure the system config (extensions, trunks, etc) is preserved. any good advises ? Thank you all, regards LL On 12/30/2011 2:21 PM, Doug Lytle wrote: LL wrote: I need to recompile asterisk to install some apps. Does anyone has a good tutorial on how to do this From the statement above, you're giving conflicting information. You first sounded like the installation was compiled from source, but then you're asking for a tutorial on how to compile. If the installation was installed by your distro's package manager, then you won't want to install from source. Can you tell us how it was installed? Doug -- _ -- 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 -- _ -- 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 1.4.42 NOTIFY replies ignore NAT setting
On 12/30/2011 12:29 PM, James Lamanna wrote: On Fri, Dec 30, 2011 at 8:35 AM, James Lamannajlama...@gmail.com wrote: On Fri, Dec 30, 2011 at 6:02 AM, Kevin P. Flemingkpflem...@digium.com wrote: On 12/30/2011 04:07 AM, James Lamanna wrote: Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even with Symmetric NAT on). The reason for this is that the NOTIFY reply does not attempt to lookup the SIP peer and check its NAT flags. I've seen some nasty From: header string parsing code + find_peer() that does this, but I was wondering if there's an easier way. Since Asterisk does not initiate subscriptions, these NOTIFY requests arriving to the Asterisk system must be 'unsolicited'. As such, they don't have an associated SIP dialog structure, so there's no simple way to know whether they are associated with a known peer or not. You say that Asterisk's behavior is 'incorrect', but it's only 'incorrect' because you believe it should be looking up any associated peer and using that peer's NAT setting; Asterisk's behavior as you've quoted is *correct* according to the RFC3261 rules for how replies should be sent, assuming that the top-most Via header does not have an 'rport' parameter present in it. The *proper* way to solve this problem is to have the UA sending the NOTIFY request include the 'rport' parameter in the top-most Via header of the request; if that is done, then whatever UA receives the request will be able to properly respond, even if the request crosses a NAT. Another way to solve it, if the sending UA cannot be changed to emit proper SIP requests, is to modify Asterisk to attempt a peer lookup when it is going to reply to request that it cannot associate with any known dialog, and then have the peer configured with 'nat=yes' (in the case of 1.4.42). A third option is to set 'nat=yes' in the [general] section of sip.conf, so that Asterisk will reply using rport-style behavior regardless of whether the request could be associated with a peer or not. Thanks Kevin. I'll have to turn rport on on all my Linksys/Cisco phones and give it a shot. Hi Kevin, That doesn't appear to work correctly: The response does not come back to 34972 even though rport is in the Via. U xxx.234:34972 - yyy..7:5060 NOTIFY sip:yyy.7 SIP/2.0..Via: SIP/2.0/UDP 10.132.38.19:6957;branch=z9hG4bK-25ea41f0;rport..From: 1316 sip:1316@yyy.7;tag=80f427ae9e884ado0..To:sip:yyy .7..Call-ID: 4fa38a62-b7d76...@10.132.38.19..cseq: 1 NOTIFY..Max-Forwards: 70..Contact: 1316 sip:1316@10.132.38.19:6957..Event: keep-alive..User-Agent: Linksys/SPA942-6.1.3( a)..Content-Length: 0 # U yyy.7:5060 - xxx.234:6957 SIP/2.0 481 No subscription..Via: SIP/2.0/UDP 10.132.38.19:6957;branch=z9hG4bK-25ea41f0;received=xxx.234;rport=34972..From: 1316sip:1316@yyy.7;tag=80f427ae9e884 ado0..To:sip:yyy.7;tag=as07ad17b5..Call-ID: 4fa38a62-b7d76...@10.132.38.19..cseq: 1 NOTIFY..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY, INFO..Supported: replaces..Content-Length: 0 That would be a bug then; the 481 response was not sent to the proper port. It's strange though, because the rport parameter was properly updated with the 'perceived port', and the received parameter was added as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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] Asterisk 1.8.8.1 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.8.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.8.1 resolves a regression introduced in Asterisk 1.8.8.0 reported by the community, and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: * Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop causes the loop to exit prematurely. This causes a variety of negative side effects, which may include having Music On Hold failing during a SIP Hold. (closes issue ASTERISK-19095) Reported by: Stefan Schmidt For a full description of the changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.1 Thank you for your continued support of Asterisk! -- _ -- 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 1.4.42 NOTIFY replies ignore NAT setting
On Fri, Dec 30, 2011 at 11:55 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 12/30/2011 12:29 PM, James Lamanna wrote: On Fri, Dec 30, 2011 at 8:35 AM, James Lamannajlama...@gmail.com wrote: On Fri, Dec 30, 2011 at 6:02 AM, Kevin P. Flemingkpflem...@digium.com wrote: On 12/30/2011 04:07 AM, James Lamanna wrote: Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even with Symmetric NAT on). The reason for this is that the NOTIFY reply does not attempt to lookup the SIP peer and check its NAT flags. I've seen some nasty From: header string parsing code + find_peer() that does this, but I was wondering if there's an easier way. Since Asterisk does not initiate subscriptions, these NOTIFY requests arriving to the Asterisk system must be 'unsolicited'. As such, they don't have an associated SIP dialog structure, so there's no simple way to know whether they are associated with a known peer or not. You say that Asterisk's behavior is 'incorrect', but it's only 'incorrect' because you believe it should be looking up any associated peer and using that peer's NAT setting; Asterisk's behavior as you've quoted is *correct* according to the RFC3261 rules for how replies should be sent, assuming that the top-most Via header does not have an 'rport' parameter present in it. The *proper* way to solve this problem is to have the UA sending the NOTIFY request include the 'rport' parameter in the top-most Via header of the request; if that is done, then whatever UA receives the request will be able to properly respond, even if the request crosses a NAT. Another way to solve it, if the sending UA cannot be changed to emit proper SIP requests, is to modify Asterisk to attempt a peer lookup when it is going to reply to request that it cannot associate with any known dialog, and then have the peer configured with 'nat=yes' (in the case of 1.4.42). A third option is to set 'nat=yes' in the [general] section of sip.conf, so that Asterisk will reply using rport-style behavior regardless of whether the request could be associated with a peer or not. Thanks Kevin. I'll have to turn rport on on all my Linksys/Cisco phones and give it a shot. Hi Kevin, That doesn't appear to work correctly: The response does not come back to 34972 even though rport is in the Via. U xxx.234:34972 - yyy..7:5060 NOTIFY sip:yyy.7 SIP/2.0..Via: SIP/2.0/UDP 10.132.38.19:6957;branch=z9hG4bK-25ea41f0;rport..From: 1316 sip:1316@yyy.7;tag=80f427ae9e884ado0..To:sip:yyy .7..Call-ID: 4fa38a62-b7d76...@10.132.38.19..cseq: 1 NOTIFY..Max-Forwards: 70..Contact: 1316 sip:1316@10.132.38.19:6957..Event: keep-alive..User-Agent: Linksys/SPA942-6.1.3( a)..Content-Length: 0 # U yyy.7:5060 - xxx.234:6957 SIP/2.0 481 No subscription..Via: SIP/2.0/UDP 10.132.38.19:6957;branch=z9hG4bK-25ea41f0;received=xxx.234;rport=34972..From: 1316sip:1316@yyy.7;tag=80f427ae9e884 ado0..To:sip:yyy.7;tag=as07ad17b5..Call-ID: 4fa38a62-b7d76...@10.132.38.19..cseq: 1 NOTIFY..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY, INFO..Supported: replaces..Content-Length: 0 That would be a bug then; the 481 response was not sent to the proper port. It's strange though, because the rport parameter was properly updated with the 'perceived port', and the received parameter was added as well. Could this be because this is sent through a temporary' response, rather than the traditional allocation? (it uses transmit_response_using_temp) Thanks. -- James -- _ -- 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 1.4.42 NOTIFY replies ignore NAT setting
On 12/30/2011 03:09 PM, James Lamanna wrote: On Fri, Dec 30, 2011 at 11:55 AM, Kevin P. Flemingkpflem...@digium.com wrote: On 12/30/2011 12:29 PM, James Lamanna wrote: On Fri, Dec 30, 2011 at 8:35 AM, James Lamannajlama...@gmail.comwrote: On Fri, Dec 30, 2011 at 6:02 AM, Kevin P. Flemingkpflem...@digium.com wrote: On 12/30/2011 04:07 AM, James Lamanna wrote: Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even with Symmetric NAT on). The reason for this is that the NOTIFY reply does not attempt to lookup the SIP peer and check its NAT flags. I've seen some nasty From: header string parsing code + find_peer() that does this, but I was wondering if there's an easier way. Since Asterisk does not initiate subscriptions, these NOTIFY requests arriving to the Asterisk system must be 'unsolicited'. As such, they don't have an associated SIP dialog structure, so there's no simple way to know whether they are associated with a known peer or not. You say that Asterisk's behavior is 'incorrect', but it's only 'incorrect' because you believe it should be looking up any associated peer and using that peer's NAT setting; Asterisk's behavior as you've quoted is *correct* according to the RFC3261 rules for how replies should be sent, assuming that the top-most Via header does not have an 'rport' parameter present in it. The *proper* way to solve this problem is to have the UA sending the NOTIFY request include the 'rport' parameter in the top-most Via header of the request; if that is done, then whatever UA receives the request will be able to properly respond, even if the request crosses a NAT. Another way to solve it, if the sending UA cannot be changed to emit proper SIP requests, is to modify Asterisk to attempt a peer lookup when it is going to reply to request that it cannot associate with any known dialog, and then have the peer configured with 'nat=yes' (in the case of 1.4.42). A third option is to set 'nat=yes' in the [general] section of sip.conf, so that Asterisk will reply using rport-style behavior regardless of whether the request could be associated with a peer or not. Thanks Kevin. I'll have to turn rport on on all my Linksys/Cisco phones and give it a shot. Hi Kevin, That doesn't appear to work correctly: The response does not come back to 34972 even though rport is in the Via. U xxx.234:34972 -yyy..7:5060 NOTIFY sip:yyy.7 SIP/2.0..Via: SIP/2.0/UDP 10.132.38.19:6957;branch=z9hG4bK-25ea41f0;rport..From: 1316 sip:1316@yyy.7;tag=80f427ae9e884ado0..To:sip:yyy .7..Call-ID: 4fa38a62-b7d76...@10.132.38.19..cseq: 1 NOTIFY..Max-Forwards: 70..Contact: 1316 sip:1316@10.132.38.19:6957..Event: keep-alive..User-Agent: Linksys/SPA942-6.1.3( a)..Content-Length: 0 # U yyy.7:5060 -xxx.234:6957 SIP/2.0 481 No subscription..Via: SIP/2.0/UDP 10.132.38.19:6957;branch=z9hG4bK-25ea41f0;received=xxx.234;rport=34972..From: 1316sip:1316@yyy.7;tag=80f427ae9e884 ado0..To:sip:yyy.7;tag=as07ad17b5..Call-ID: 4fa38a62-b7d76...@10.132.38.19..cseq: 1 NOTIFY..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY, INFO..Supported: replaces..Content-Length: 0 That would be a bug then; the 481 response was not sent to the proper port. It's strange though, because the rport parameter was properly updated with the 'perceived port', and the received parameter was added as well. Could this be because this is sent through a temporary' response, rather than the traditional allocation? (it uses transmit_response_using_temp) I'm sure that is related, but it's still a bug :-) Unfortunately you've reported this against an Asterisk 1.4.x release, which is in security fix only mode, so even though it's a bug, there won't be a new 1.4.x release available with a fix for it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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] High verbose set at console effects the logger file Full - Why is that?
Hi everyone, I am playing around with Asterisk 1.8.8.0 from Digium repository. This is all there is to my logger.conf file: *[general]* *dateformat=%F %T* * * *[logfiles]* *full = notice,warning,error,debug,verbose,dtmf,fax* * * However, when I do, core set verbose 0 at CLI, Asterisk ceases to write to /var/log/asterisk/full file for some reason. When I type core set verbose 9 at CLI then it starts writing to /var/log/asterisk/full. Is this the correct behaviour or am I missing a config setting? Of course I want the /var/log/asterisk/full file to always keep the logs regardless of what the verbosity at CLI level is. 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] High verbose set at console effects the logger file Full - Why is that?
If you want to stop stuff from going to the console you can use the command logger mute and console will not get output but log file will. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 3:11 PM, Bruce B wrote: Hi everyone, I am playing around with Asterisk 1.8.8.0 from Digium repository. This is all there is to my logger.conf file: [general] dateformat=%F %T [logfiles] full = notice,warning,error,debug,verbose,dtmf,fax However, when I do, core set verbose 0 at CLI, Asterisk ceases to write to /var/log/asterisk/full file for some reason. When I type core set verbose 9 at CLI then it starts writing to /var/log/asterisk/full. Is this the correct behaviour or am I missing a config setting? Of course I want the /var/log/asterisk/full file to always keep the logs regardless of what the verbosity at CLI level is. 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 -- _ -- 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] High verbose set at console effects the logger file Full - Why is that?
Okay, but I thought that the line console = is supposed to be for CLI and the line Full = is supposed to be for the file /var/log/asterisk/full. Why would the Full = be effected by core set verbose 0? Is this just bad assumption on the part of the developers? I would only assume that core set verbose 0 should only effect what I see at CLI level and not at my my /var/log/asterisk/full log file. Am I missing something? Thanks for the feedback. On Fri, Dec 30, 2011 at 6:19 PM, Jim Dickenson dicken...@cfmc.com wrote: If you want to stop stuff from going to the console you can use the command logger mute and console will not get output but log file will. -- Jim Dickenson mailto:dicken...@cfmc.com dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 3:11 PM, Bruce B wrote: Hi everyone, I am playing around with Asterisk 1.8.8.0 from Digium repository. This is all there is to my logger.conf file: *[general]* *dateformat=%F %T* * * *[logfiles]* *full = notice,warning,error,debug,verbose,dtmf,fax* * * However, when I do, core set verbose 0 at CLI, Asterisk ceases to write to /var/log/asterisk/full file for some reason. When I type core set verbose 9 at CLI then it starts writing to /var/log/asterisk/full. Is this the correct behaviour or am I missing a config setting? Of course I want the /var/log/asterisk/full file to always keep the logs regardless of what the verbosity at CLI level is. 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 -- _ -- 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 -- _ -- 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] High verbose set at console effects the logger file Full - Why is that?
Yes, you are missing the fact that the verbose setting controls what level of output will be generated in the first place. You can raise and lower the amount of stuff logged/printed on CLI. The lines in logger.conf control what types of lines go to which place. One can set the verbose level as well as the debug level. These control how much log information is generated at all not where it is being written. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 3:24 PM, Bruce B wrote: Okay, but I thought that the line console = is supposed to be for CLI and the line Full = is supposed to be for the file /var/log/asterisk/full. Why would the Full = be effected by core set verbose 0? Is this just bad assumption on the part of the developers? I would only assume that core set verbose 0 should only effect what I see at CLI level and not at my my /var/log/asterisk/full log file. Am I missing something? Thanks for the feedback. On Fri, Dec 30, 2011 at 6:19 PM, Jim Dickenson dicken...@cfmc.com wrote: If you want to stop stuff from going to the console you can use the command logger mute and console will not get output but log file will. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 3:11 PM, Bruce B wrote: Hi everyone, I am playing around with Asterisk 1.8.8.0 from Digium repository. This is all there is to my logger.conf file: [general] dateformat=%F %T [logfiles] full = notice,warning,error,debug,verbose,dtmf,fax However, when I do, core set verbose 0 at CLI, Asterisk ceases to write to /var/log/asterisk/full file for some reason. When I type core set verbose 9 at CLI then it starts writing to /var/log/asterisk/full. Is this the correct behaviour or am I missing a config setting? Of course I want the /var/log/asterisk/full file to always keep the logs regardless of what the verbosity at CLI level is. 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] High verbose set at console effects the logger file Full - Why is that?
One can set the verbose level as well as the debug level. These control how much log information is generated at all not where it is being written. What do you mean by above? Can I see something in the logger.conf that will keep it always at certain verbose level regardless of what command I issue at CLI? You see the problem I have is that Fail2ban reads the asterisk full log file. So, if I am playing on the CLI and then do core set verbose 0 and exit the box and forget to set it back to 9 then Fail2ban stops working because the log file hasn't logged the attack. I still think there is a way around this and I am missing a config. Why would anyone tie security logs to a mere CLI command? Thanks again -- _ -- 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] High verbose set at console effects the logger file Full - Why is that?
-- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 4:55 PM, Bruce B wrote: One can set the verbose level as well as the debug level. These control how much log information is generated at all not where it is being written. What do you mean by above? Can I see something in the logger.conf that will keep it always at certain verbose level regardless of what command I issue at CLI? No the verbose command controls how much verbose stuff is output. The debug command controls how much debug stuff is output. These are absolute controls of that information. As I said in my original email you can turn off stuff going to the CLI with the logger mute command. That way you do not adjust the verbose level at all. You see the problem I have is that Fail2ban reads the asterisk full log file. So, if I am playing on the CLI and then do core set verbose 0 and exit the box and forget to set it back to 9 then Fail2ban stops working because the log file hasn't logged the attack. I still think there is a way around this and I am missing a config. Why would anyone tie security logs to a mere CLI command? Thanks again -- _ -- 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 -- _ -- 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] High verbose set at console effects the logger file Full - Why is that?
So, based on what you are saying if I issue the command core set verbose 0 and then exit the system Fail2Ban will stop working for Asterisk (this is since Fail2ban works based on the log file entries). Can anyone else please confirm that as well. Thanks again for your input. On Fri, Dec 30, 2011 at 8:36 PM, Jim Dickenson dicken...@cfmc.com wrote: -- Jim Dickenson mailto:dicken...@cfmc.com dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 4:55 PM, Bruce B wrote: One can set the verbose level as well as the debug level. These control how much log information is generated at all not where it is being written. What do you mean by above? Can I see something in the logger.conf that will keep it always at certain verbose level regardless of what command I issue at CLI? No the verbose command controls how much verbose stuff is output. The debug command controls how much debug stuff is output. These are absolute controls of that information. As I said in my original email you can turn off stuff going to the CLI with the logger mute command. That way you do not adjust the verbose level at all. You see the problem I have is that Fail2ban reads the asterisk full log file. So, if I am playing on the CLI and then do core set verbose 0 and exit the box and forget to set it back to 9 then Fail2ban stops working because the log file hasn't logged the attack. I still think there is a way around this and I am missing a config. Why would anyone tie security logs to a mere CLI command? Thanks again -- _ -- 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 -- _ -- 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 -- _ -- 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
