[asterisk-users] Top posting - there is no rule.
What's with the occasional "Un-Top-posting", there is no rule that says you can't, http://www.asterisk.org/community/rules My preference is top posting, as you see the answer at a quick glance, instead of reaching for the scroll bar (or whatever key stokes are required) to get to the bottom, to find that the answer isn't there yet. Note: Flaming is not an acceptable behaviour :) Alec Davis PS. Sorry to the asterisk-dev list that have seen this already, posted in wrong forum. -- _ -- 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] Is it possible to dial an automated message when Asterisk receives an email?
Steve, Thanks for your advice! You have an interesting point of view. I shall discuss this with my office partners on Monday. Thank you both. PS: In case you wonder, I'm from Argentina, hence my native language is Spanish On Sat, Apr 2, 2011 at 3:57 PM, Steve Edwards wrote: > Un-top-posting... > > > On Sat, Apr 2, 2011 at 3:39 PM, John Kiniston wrote: >> > > You could use a procmail recipe to create a call file and then move >> it >> to the /var/spool/asterisk/outgoing directory. >> Below is a untested example .procmailrc: >> >> :0: >> * ^to.trig...@example.com >> | /usr/local/bin/callout.sh >> >> where callout.sh would look like this perhaps: >> >> !/bin/bash >> sleep 5 >> CALL="callout.call"; >> echo channel: LOCAL/NUMBER@pstn-local > /tmp/$CALL; >> echo context: ivr-call-out >> /tmp/$CALL; >> echo exten: s >> /tmp/$CALL; >> echo priority: 1 >> /tmp/$CALL; >> >> echo mv /tmp/$CALL /var/spool/asterisk/outgoing >> done >> >> Again all untested writing by the seat of my pants type stuff. >> > > On Sat, 2 Apr 2011, Rafael Bermúdez wrote: > > John, >> > > Thanks for your reply. I will test this script. >> > > A couple of comments of the top of my head: > > ) If you construct the call file name using the PID you can accomodate more > than a single event at the same (or really close) time. > > ) The 'mv' command has an extraneous 'echo' > > ) Pay attention to permissions. Sudo can help if needed. Personally, I > prefer a restrictive sudo to the blunt hammer of wide open permissions :) > > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _ > -- 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 > -- Bermúdez Rafael -- _ -- 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] Best Scripting Language
On Fri, 2011-04-01 at 13:27 +0100, Roger Burton West wrote: > On Fri, Apr 01, 2011 at 05:27:20PM +0530, Gopalakrishnan A.N wrote: > >Can anyone suggest which is the best scripting language for Asterisk or any > >telecom device? > > Depends on the other parameters. Perl is great for rapid development, > but I wouldn't run it per-call on a box taking hundreds of calls per > second. (Ditto Ruby and Python.) C will be much faster, but it's more > effort to write and debug. > You can develop in perl (great language, sort of swiss-armyknife under computer languages, around for many years) And if you are satisfied with the result, you can translate it into binary, thus having the best of both (if not any) worlds. hw -- _ -- 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] Registration from '"000000" x 1000
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Saturday, April 02, 2011 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Registration from '"00" x 1000 On 04/02/2011 02:08 PM, Steve Davies wrote: > On 2 April 2011 09:46, Jonas Kellens wrote: > >> Hello list, >> >> I often see the following in my message log : >> >> [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" >> ' failed for '184.106.109.168' - No matching peer found >> [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" >> ' failed for '184.106.109.168' - No matching peer found >> [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" >> ' failed for '184.106.109.168' - No matching peer found >> [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" >> ' failed for '184.106.109.168' - No matching peer found >> [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" >> ' failed for '184.106.109.168' - No matching peer found >> >> And there are hundreds of them... >> >> >> Is there a setting so I can make Asterisk not respond to SIP PEER >> registrations which are not in my sip.conf or my realtime MySQL DB ?? >> > Yes, you add a rule to your firewall! Even better, get it filtered > further out so that it does not waste your inbound Internet bandwidth, > because in my experience, once those SIP spammers start, they continue > for weeks at the very least. > > IIRC, the way SIP registrations works basically requires than an > failed/un-authorised attempt is responded to, so that the other party > knows to authenticate. If you stop sending that response, no-one can > authenticate. > > Hope that helps. > Steve So in short, there is no way of throwing away registrations that are not in sip.conf. The only thing I can do is check the messages file now and then to see if there were bad registrations, and then blacklist them. Kind regards, Jonas. Search the archive for Fail2Ban, it is what you are looking for. -- _ -- 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] Is it possible to dial an automated message when Asterisk receives an email?
Un-top-posting... On Sat, Apr 2, 2011 at 3:39 PM, John Kiniston wrote: You could use a procmail recipe to create a call file and then move it to the /var/spool/asterisk/outgoing directory. Below is a untested example .procmailrc: :0: * ^to.trig...@example.com | /usr/local/bin/callout.sh where callout.sh would look like this perhaps: !/bin/bash sleep 5 CALL="callout.call"; echo channel: LOCAL/NUMBER@pstn-local > /tmp/$CALL; echo context: ivr-call-out >> /tmp/$CALL; echo exten: s >> /tmp/$CALL; echo priority: 1 >> /tmp/$CALL; echo mv /tmp/$CALL /var/spool/asterisk/outgoing done Again all untested writing by the seat of my pants type stuff. On Sat, 2 Apr 2011, Rafael Bermúdez wrote: John, Thanks for your reply. I will test this script. A couple of comments of the top of my head: ) If you construct the call file name using the PID you can accomodate more than a single event at the same (or really close) time. ) The 'mv' command has an extraneous 'echo' ) Pay attention to permissions. Sudo can help if needed. Personally, I prefer a restrictive sudo to the blunt hammer of wide open permissions :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] Is it possible to dial an automated message when Asterisk receives an email?
On Sat, 2 Apr 2011, Rafael Bermúdez wrote: I have a server that sends a preformatted email when an event occur. What I need is that when Asterisk receives this email automatically dial a pre-recorded message. It doesn't have to dial ride away, maybe a scheduled cron job will be enough. Procmail, call files and a little scripting would be one approach. Can you take a step back up the chain and have your server: ) Execute the script to create the call file? ) Create the call file on a shared drive? ) Connect to Asterisk via AMI to originate the call? Thanks, and sorry for my lousy English Probably much better than my ish. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] Is it possible to dial an automated message when Asterisk receives an email?
John, Thanks for your reply. I will test this script. Once again, thank you! On Sat, Apr 2, 2011 at 3:39 PM, John Kiniston wrote: > You could use a procmail recipe to create a call file and then move it > to the /var/spool/asterisk/outgoing directory. > Below is a untested example .procmailrc: > > :0: > * ^to.trig...@example.com > | /usr/local/bin/callout.sh > > where callout.sh would look like this perhaps: > > !/bin/bash > sleep 5 > CALL="callout.call"; > echo channel: LOCAL/NUMBER@pstn-local > /tmp/$CALL; > echo context: ivr-call-out >> /tmp/$CALL; > echo exten: s >> /tmp/$CALL; > echo priority: 1 >> /tmp/$CALL; > > echo mv /tmp/$CALL /var/spool/asterisk/outgoing > done > > > Again all untested writing by the seat of my pants type stuff. > > 2011/4/2 Rafael Bermúdez : > > Hello. > > > > I have a server that sends a preformatted email when an event occur. What > I > > need is that when Asterisk receives this email automatically dial a > > pre-recorded message. It doesn't have to dial ride away, maybe a > scheduled > > cron job will be enough. > > > > Is that possible? Any hint? What should I be looking for? > > > > > > Thanks, and sorry for my lousy English > > > > -- > > Bermúdez Rafael > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Bermúdez Rafael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it possible to dial an automated message when Asterisk receives an email?
You could use a procmail recipe to create a call file and then move it to the /var/spool/asterisk/outgoing directory. Below is a untested example .procmailrc: :0: * ^to.trig...@example.com | /usr/local/bin/callout.sh where callout.sh would look like this perhaps: !/bin/bash sleep 5 CALL="callout.call"; echo channel: LOCAL/NUMBER@pstn-local > /tmp/$CALL; echo context: ivr-call-out >> /tmp/$CALL; echo exten: s >> /tmp/$CALL; echo priority: 1 >> /tmp/$CALL; echo mv /tmp/$CALL /var/spool/asterisk/outgoing done Again all untested writing by the seat of my pants type stuff. 2011/4/2 Rafael Bermúdez : > Hello. > > I have a server that sends a preformatted email when an event occur. What I > need is that when Asterisk receives this email automatically dial a > pre-recorded message. It doesn't have to dial ride away, maybe a scheduled > cron job will be enough. > > Is that possible? Any hint? What should I be looking for? > > > Thanks, and sorry for my lousy English > > -- > Bermúdez Rafael > > -- > _ > -- 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
[asterisk-users] Is it possible to dial an automated message when Asterisk receives an email?
Hello. I have a server that sends a preformatted email when an event occur. What I need is that when Asterisk receives this email automatically dial a pre-recorded message. It doesn't have to dial ride away, maybe a scheduled cron job will be enough. Is that possible? Any hint? What should I be looking for? Thanks, and sorry for my lousy English -- Bermúdez Rafael -- _ -- 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] Registration from '"000000" x 1000
On 04/02/2011 02:08 PM, Steve Davies wrote: On 2 April 2011 09:46, Jonas Kellens wrote: Hello list, I often see the following in my message log : [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" ' failed for '184.106.109.168' - No matching peer found [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" ' failed for '184.106.109.168' - No matching peer found [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" ' failed for '184.106.109.168' - No matching peer found [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" ' failed for '184.106.109.168' - No matching peer found [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" ' failed for '184.106.109.168' - No matching peer found And there are hundreds of them... Is there a setting so I can make Asterisk not respond to SIP PEER registrations which are not in my sip.conf or my realtime MySQL DB ?? Yes, you add a rule to your firewall! Even better, get it filtered further out so that it does not waste your inbound Internet bandwidth, because in my experience, once those SIP spammers start, they continue for weeks at the very least. IIRC, the way SIP registrations works basically requires than an failed/un-authorised attempt is responded to, so that the other party knows to authenticate. If you stop sending that response, no-one can authenticate. Hope that helps. Steve So in short, there is no way of throwing away registrations that are not in sip.conf. The only thing I can do is check the messages file now and then to see if there were bad registrations, and then blacklist them. Kind regards, Jonas. -- _ -- 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] Problem getting TDM400P clone card to go off-hook and dial
I am having problems getting a Nicherons TDM400P wildcard clone to dial out.
Everything appears to be configured correctly, but although I see call
progress, it never seems to actually pick up the phone.
(The following is a test of 911 emergency, where I substitute 811 [repair
service] as the actual number dialed.)
*CLI>
-- Executing [911@from-internal:1] Goto("SIP/101-",
"nineoneone,s,1") in new stack
-- Goto (nineoneone,s,1)
-- Executing [s@nineoneone:1] Set("SIP/101-",
"SET_EMERG_FLAG=0") in new stack
-- Executing [s@nineoneone:2] ChanIsAvail("SIP/101-", "DAHDI/4")
in new stack
-- Hanging up on 'DAHDI/4-1'
-- Hungup 'DAHDI/4-1'
-- Executing [s@nineoneone:3] Set("SIP/101-",
"GLOBAL(EMERGENCY)=1") in new stack
== Setting global variable 'EMERGENCY' to '1'
-- Executing [s@nineoneone:4] Set("SIP/101-",
"SET_EMERG_FLAG=1") in new stack
-- Executing [s@nineoneone:5] Dial("SIP/101-", "DAHDI/4/811") in
new stack
-- Called 4/811
-- DAHDI/4-1 is ringing
== Extension Changed 101[from-internal] new state Idle for Notify User 101
== Extension Changed 101[from-internal] new state Idle for Notify User 103
-- Hanging up on 'DAHDI/4-1'
-- Hungup 'DAHDI/4-1'
== Spawn extension (nineoneone, s, 5) exited non-zero on
'SIP/101-'
-- Executing [h@nineoneone:1] GotoIf("SIP/101-", "1?3") in new
stack
-- Goto (nineoneone,h,3)
-- Executing [h@nineoneone:3] Set("SIP/101-",
"GLOBAL(EMERGENCY)=0") in new stack
== Setting global variable 'EMERGENCY' to '0'
*CLI>
When "DAHDI/4-1 is ringing" appears I indeed hear ringing progress tones,
but they appear to be coming from Asterisk, as the card does not pick up the
phone at this point, or ever. I'm using jack #4 on the board, which is
supposed an FXO port,
Here's the output from various relevant tools & config files:
--
*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret
BlockedState
pseudodefault
default In Service
1from-pstn
default In Service
2from-pstn
default In Service
3from-internal
default In Service
4from-internal
default In Service
---
dahdi-channels.conf:
; Autogenerated by /usr/sbin/dahdi_genconf on Fri Apr 1 06:52:48 2011
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is
intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global
settings
;
; Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)
;;; line="1 WCTDM/4/0 FXSKS (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default
;;; line="2 WCTDM/4/1 FXSKS (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 2
callerid=
group=
context=default
;;; line="3 WCTDM/4/2 FXOKS (SWEC: MG2)"
signalling=fxo_ks
callerid="Channel 3" <4003>
mailbox=4003
group=5
context=from-internal
channel => 3
callerid=
mailbox=
group=
context=default
;;; line="4 WCTDM/4/3 FXOKS (SWEC: MG2)"
signalling=fxo_ks
callerid="Channel 4" <4004>
mailbox=4004
group=5
context=from-internal
channel => 4
callerid=
mailbox=
group=
context=default
; Span 2: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: HRtimer) 1"
--
chan_dahdi.conf
...
...
[channels]
#include /etc/asterisk/dahdi-channels.conf
...
-
root@Trixie:/etc/asterisk# dahdi_hardware
pci::02:01.0 wctdm+ e159:0001 Wildcard TDM400P REV I
---
root@Trixie:~# lsdahdi
### Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)
1 FXOFXSKS (In use) (SWEC: MG2)
2 FXOFXSKS (In use) (SWEC: MG2)
3 FXSFXOKS (In use) (SWEC: MG2)
4 FXSFXOKS (In use) (SWEC: MG2)
### Span 2: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: HRtimer) 1"
root@Trixie:~#
-
extensions.conf (excerpt)
; Global variables
[globals]
; Stuff for 911
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/4
; Change this for production use:
;EMERGENCY_NUM=some_test_phone_number
EMERGENCY_NUM=811
;EMERGENCY_NUM=911
...
; Which trunk to use for any DAHDI (PSTN-'Hard Line'-AKA POTS) type stuff
POTSTRUNK=DAHDI/4
...
; Emergency -- DO NOT REMOVE!
exten => 911,1,Goto(nineoneone,s,1)
...
; EMERGENCY! See http://www.voip-info.org/wiki-Asterisk+tips+911 for
details.
[nineoneone]
exten => s,1,Set(SET_EMERG_FLAG=0)
exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten =>
[asterisk-users] tarnsfer automatically
Hello list, i have one question related to transfer call i have 2 number for the inbound and i want to configure asterisk like that. When the customer call the first number 0522XX the call will be transfer automatically to anther number 0520xx Any help please. Thanks and regards. -- _ -- 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] Registration from '"000000" x 1000
On 2 April 2011 09:46, Jonas Kellens wrote: > Hello list, > > I often see the following in my message log : > > [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" > ' failed for '184.106.109.168' - No matching peer found > [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" > ' failed for '184.106.109.168' - No matching peer found > [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" > ' failed for '184.106.109.168' - No matching peer found > [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" > ' failed for '184.106.109.168' - No matching peer found > [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" > ' failed for '184.106.109.168' - No matching peer found > > And there are hundreds of them... > > > Is there a setting so I can make Asterisk not respond to SIP PEER > registrations which are not in my sip.conf or my realtime MySQL DB ?? Yes, you add a rule to your firewall! Even better, get it filtered further out so that it does not waste your inbound Internet bandwidth, because in my experience, once those SIP spammers start, they continue for weeks at the very least. IIRC, the way SIP registrations works basically requires than an failed/un-authorised attempt is responded to, so that the other party knows to authenticate. If you stop sending that response, no-one can authenticate. Hope that helps. Steve -- _ -- 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] Registration from '"000000" x 1000
Hello list, I often see the following in my message log : [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" ' failed for '184.106.109.168' - No matching peer found [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" ' failed for '184.106.109.168' - No matching peer found [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" ' failed for '184.106.109.168' - No matching peer found [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" ' failed for '184.106.109.168' - No matching peer found [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" ' failed for '184.106.109.168' - No matching peer found And there are hundreds of them... Is there a setting so I can make Asterisk not respond to SIP PEER registrations which are not in my sip.conf or my realtime MySQL DB ?? Kind regards, Jonas. -- _ -- 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] automixmon output file location and exec command options
Hi all, I have 2 quick question regarding the file location and post record command of the recording using automixmon in features.conf. With the normal monitor/mixmonitor applications you can change the location of where the recordings will be stored, by changing the MONITOR_FILENAME variable. I tried changing the TOUCH_MIXMONITOR_OUTPUT variable to include a path but it sill puts the recorded file in /var/spool/asterisk/monitor. Is there any way I can change this? The second question, is, is it possible to execute a command after one touch mixmonitor has completed? With the mixmonitor application this is possible, I was wondering if the option was available for the automixmon feature in features.conf I've been doing some googling around to see if I can find any info on the above, but I seem to be coming up short, or I'm looking in the wrong places. Any tips/suggestions would be appreciated. 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
