Re: [asterisk-users] Problem with OPTIONS requests.

return a 200 instead of 404. It doesn't matter what the 's' extension does, so it can just call Hangup. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- ___

Re: [asterisk-users] problem with logger: syslog vs. file

In article <88f96e46-e6bb-a7ef-bebb-5588ef6cd...@gmx.ch>, Fourhundred Thecat <400the...@gmx.ch> wrote: > > On 2020-06-02 17:48, Tony Mountifield wrote: > > In article <94191802-6c9c-bdab-615b-001786a2a...@gmx.ch>, > > Fourhundred Thecat <400the...@gmx.

Re: [asterisk-users] problem with logger: syslog vs. file

agree with you that it is strange the two logging types are different. But someone with a different opinion than yours might well say "Why did they decide to omit the line number and function from the file logging? It's very useful information!" The beauty of open source is of cour

Re: [asterisk-users] Perl AGI: read variable with quotes

line you quoted: same => n,NoOp(PAI=${PAI}) Then turn on verbose logging and try the call. Look at the logged NoOp line and see if it contains just the 'John' or the whole value '"John Doe" ' If it contains the whole value, then the problem is in the AGI library reading the variable. If

Re: [asterisk-users] pre-dial handler, how to access variables from calling channel?

cy the __ when reading the variable, just use ${PAI} as before. See https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org --

Re: [asterisk-users] possible bug in Asterisk 16

:881] 3. Set(CALLERID(num)=044111) [extensions.conf:882] -= 2 extensions (5 priorities) in 1 context. =- Notice that the "n" converted to "4" correctly even though the extension changed. I wasn't aware it wou

Re: [asterisk-users] Delays on conferences

o Asterisk or from Asterisk. Do you have internal_timing set in asterisk.conf? What timing module are you using? Does it always happen, or just sometimes? Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk

Re: [asterisk-users] Load issues using AGI

Nevertheless, we will try what you just posted. Even if you put "exit 0" at the top of the script, the perl interpreter will still need to compile the whole script (and any modules it uses) before it executes the "exit 0". Try commenting out or removing the rest of the scri

Re: [asterisk-users] asterisk-users Digest, Vol 181, Issue 3

ndle "qualify". So in your [trunkinbound] context, just add a line like this: exten => s,1,Hangup And leave everything else in that context unchanged. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mount

[asterisk-users] Who speaks the en_GB sounds?

Who is the male voice artist who recorded the en_GB sounds for Asterisk? Would be useful to know in case of the need to get additional matching sounds recorded. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http

Re: [asterisk-users] ARI libraries?

In article , Jean-Denis Girard wrote: > Le 20/07/2019 à 12:21, Tony Mountifield a écrit : > > What is the bug with channel variables? Do you have a fix for it? > > Channels variables caused an error, my fix is in aioswagger11/client.py > (line 80) : >

Re: [asterisk-users] ARI libraries?

In article <301a2e78-d490-3805-e30f-41b668aac...@sysnux.pf>, Jean-Denis Girard wrote: > > Hi Tony, > > Le 20/07/2019 à 06:29, Tony Mountifield a écrit : > > Are there any other languages/libraries I should be considering? > > Same here, after years of AGI /

[asterisk-users] ARI libraries?

languages/libraries I should be considering? Thanks for any advice! Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth

Re: [asterisk-users] Cannot originate to extension unless /etc/hosts is edited constantly?

obviously is fatal anyway as I got lots of phones on which I want to > playback recordings and editing /etc/hosts for each phone is impossible if > two phones want to listen to different recordings at the same time- > /etc/hosts can only contain one "local"). > > How can I

Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp)

to propagate the answer, busy or other failure from the destination channel back to the originating channel. Is it possible that the setup part of the call (between initiation and answer) is recorded in a separate CDR? Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co

Re: [asterisk-users] Recompiling Ast results in a binary with differing SHA256 sums?

ecompile, then compare the first binary with the recompiled one? At the simplest level use "cmp -l". Or maybe convert each binary to a hexdump with "hexdump -C", and then use diff or vimdiff to compare them. Cheers Tony -- Tony Mo

Re: [asterisk-users] AMI manager logins - omitting from logging output?

nt it to log login attempts - or, > to put it another way, is there any way to tell the logger module to ignore > AMI? Look in /etc/asterisk/manager.conf for the option "displayconnects = yes/no". It can be set globally in [general] or individually in [ServiceCheck] (for exa

Re: [asterisk-users] Dial to FastAGI application appears as 1-second CDR - how do I fix?

In article <7d8dc02f-0fce-4d47-72d9-604994c33...@palosanto.com>, Alex Villací­s Lasso wrote: > El 29/05/18 a las 05:24, Tony Mountifield escribió: > > In article <3a005ff6-19a4-215b-4751-bee616ec7...@palosanto.com>, > > Alex Villací­s Lasso wrote: > >&g

Re: [asterisk-users] Dial to FastAGI application appears as 1-second CDR - how do I fix?

nt ast_do_masquerade(struct ast_channel } exchange; struct ast_channel *clonechan, *chans[2]; struct ast_channel *bridged; +#ifdef I_THINK_THIS_IS_WRONG /* Tony Mountifield, 2018-03-29. Removing this code fixes lost CDRs with masquerade */ struct ast_cdr *cdr; +#endi

Re: [asterisk-users] Strange problem with PRI on 64-bit?

In article <20180404133024.kpidrkuiyjoqd...@xorcom.com>, Tzafrir Cohen <tzafrir.co...@xorcom.com> wrote: > On Wed, Apr 04, 2018 at 11:28:33AM +, Tony Mountifield wrote: > > In article > > <cald46g3wz8kajro4_nde211osv8paxfdszw-g6t2qxjlk-y...@mail.gmail.co

Re: [asterisk-users] Strange problem with PRI on 64-bit?

In article <pa2c9u$1pa$1...@softins.softins.co.uk>, Tony Mountifield <t...@softins.co.uk> wrote: > In article >

Re: [asterisk-users] Strange problem with PRI on 64-bit?

ed that all should be ok, otherwise the executables would fail to run (I initially discovered this when dahdi_cfg couldn't find libtonezone). Would there be any subtle issues with the 64-bit libraries being loaded from /usr/lib instead of /usr/lib64? Should Asterisk and DAHDI builds also be u

Re: [asterisk-users] Strange problem with PRI on 64-bit?

In article

Re: [asterisk-users] Audio Dropouts During Call

n, things can degrade to half-duplex trying to talk to full-duplex, resulting in lots of collisions and packet loss when there is any kind of significant traffic. Your description would be consistent with the firewall introducing lots of LAN collisions when busy, in the central gigabit switch, even i

Re: [asterisk-users] Strange problem with PRI on 64-bit?

In article

[asterisk-users] Strange problem with PRI on 64-bit?

have any clues why there would be a difference in PRI behaviour between 32-bit and 64-bit builds? Has anyone else run into anything similar? Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

Re: [asterisk-users] DIALSTATUS vs HANGUPCAUSE

er mechanisms for failure (such as failure to create a channel within Asterisk, or an attempt to send to an unreachable peer), that may set DIALSTATUS without setting HANGUPCAUSE. So HANGUPCAUSE should be considered as extra detail, rather than a replacement or alternative to DIALSTATUS. Cheers Tony --

Re: [asterisk-users] SIP trunks going to the wrong context

] > exten => 11,1,Dial(SIP/officephone,120,m) > > [secondline] > exten => 22,1,Dial(SIP/livingroomphone,120,m) > > [thirdline] > exten => 33,1,Dial(SIP/bedroomphone,120,m) But because you have all three of your trunk peers pointing to the same context,

Re: [asterisk-users] SIP trunks going to the wrong context

e) is not used to select or match the inbound SIP peer. When the call comes in from sipgate, it probably doesn't have a fromuser. The fromuser can be used to select the peer based on matching the [string] that names the peer. Otherwise, when Asterisk is looking for a matching peer section, I believe it

Re: [asterisk-users] ERROR during high volume MoH dialplan

av, ulaw, alaw, gsm and g729. They will also sound better than transcoding from the gsm versions. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- ___

Re: [asterisk-users] IAX port 4569

you don't have verbose listed for console, then AFAIK, you won't see verbose messages however many -v options you give it! Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mou

Re: [asterisk-users] OT: Want to capture all SIP messages

-w $FILE udp port 5060 /dev/null 2>&1 & --- I start it in /etc/rc.d/rc.local for want of anywhere better. Being in /var/tmp, cron.daily/tmpwatch deletes files older than 30 days. I could just have easily put them somewhere else and used the -W option to tcpdump to remove old fi

Re: [asterisk-users] Automatically dial a number, then an extension

e no way of telling when that extension has been answered and it is safe to play the message. That is why I added the comment "how long???", as it is just a guess. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk -

Re: [asterisk-users] asterisk name in mysql

le/dongle0-010002' That means that your copy of asterisk does not have the "app_mysql" module. If you compiled asterisk yourself, you need to go into menuselect and make sure app_mysql is selected, and then recompile. You will probably find app_mysql under "Addons". If it

Re: [asterisk-users] Asterisk dahai install centos 7

.0-327 The difference is significant, and suggests that you are actually running an older kernel than the latest. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _

Re: [asterisk-users] What could be stopping "Disconnect Call" feature from working (set in features.txt)

those > channels? Yes, append /n to the local channel: same => n,Dial(Local/s@dial-test/n,3,L(354:6)) Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- __

Re: [asterisk-users] Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)

In article

Re: [asterisk-users] Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)

of these!). When matching an extension being dialled, Asterisk is only concerned about priority 1, so that's the only priority you need to double. You should be able to use ! safely in priority 2 upwards: exten => _X,1,NoOp(Matching single digit) exten => _X.,1,NoOp(Matching multip

Re: [asterisk-users] Multiple phones when one is unregistered

e am I missing? The ExecIf command is provided in the module app_exec, which is usually located at /usr/lib/asterisk/modules/app_exec.so Maybe you had turned off app_exec in the menuconfigi when building, or maybe your modules.conf has a noload => app_exec.so C

Re: [asterisk-users] SIP trunk

m.com, and that should match the "s" extension: exten => s,1,NoOp(Didn't get a number) Maybe that's what is happening in your case, so try adding an "s" extension. Hope this helps, Tony -- Tony Mountifield

Re: [asterisk-users] Mobiles not detecting as BUSY until Dial() timeout completes

line has Call Waiting enabled, and the waiting call has been ignored by the recipient until it times out. See https://en.wikipedia.org/wiki/Call_waiting and try *#43# on the mobile in question to check whether call waiting is active. Use #43# to try deactivating it and see if that

Re: [asterisk-users] Compile error with libpri 1.4.15

not timezone.h And it is being looked for in /usr/include/dahdi/tonezone.h, not in the dahdi source directory. You need to have installed DAHDI before compiling libpri or asterisk. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.u

Re: [asterisk-users] Compile error with libpri 1.4.15

n't tried 2.11.0. On my boxes with 2.10.0.1, /usr/include/dahdi/tonezone.h does exist. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _

Re: [asterisk-users] is there some blocking in 11.21.0

In article

Re: [asterisk-users] is there some blocking in 11.21.0

In article

Re: [asterisk-users] is there some blocking in 11.21.0

In article

[asterisk-users] SNMP order of channel types

to vary between reboots or rebuilds? 3. Is there a way I can make the order predictable and fixed? modules.conf? 4. Or alternatively, a way I can make MRTG find the correct index dynamically? Thanks for any advice, Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play

Re: [asterisk-users] FastAGI not working

n describe how you have set up your FastAGI server, and how it invokes DatabaseQuery.agi, that would help us to help you! Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org --

Re: [asterisk-users] SNMP on Asterisk 11

none none access MyRWGroup "" any noauthexact allallall which makes everything under .1 visible. Not sure if that is the default or not. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http:/

[asterisk-users] Dialing a call back out on same SIP trunk as it came in

that I should. The siptrunk entry contains canreinvite=no and directmedia=no. The version of Asterisk on these boxes is 10.5.1, if that's relevant. Thanks for any insight! Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org

Re: [asterisk-users] Dialing a call back out on same SIP trunk as it came in

call was already answered. > > Can anyone explain why I need the Answer? It feels wrong that I should. > > The siptrunk entry contains canreinvite=no and directmedia=no. > > The version of Asterisk on these boxes is 10.5.1, if that's relevant. > > Thanks for any insight! >

Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining

/safe_asterisk, at the commented-out settings for SYSMAXFILES and MAXFILES, and try setting those. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk keeps complaining

, so it would be best to update to that version and see if the problem persists. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

[asterisk-users] Find out or log negotiated codec for SIP channel?

to do is to determine historically the usage of the G.729 licences installed in a system, but an answer to the more general question would be useful. Thanks Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

Re: [asterisk-users] How does chan_sip match an ACK?

In article mfbt6f$9rt$1...@softins.softins.co.uk, Tony Mountifield t...@softins.co.uk wrote: I am trying to debug a SIP issue, between an Asterisk 1.2.32 system that is behind a network device to which I don't have ready access, which is performing NAT with possibly some kind of SIP ALG

[asterisk-users] How does chan_sip match an ACK?

the address of the public endpoint): Mar 30 10:20:20 VERBOSE[5811] logger.c: Retransmitting #5 (no NAT) to 11.111.11.111:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 11.111.11.111:5060;branch=z9hG4bK6bee4b53;received=11.111.11.111 From: Tony Mountifield sip:2011@11.111.11.111;tag=as4ab948f7 To: sip:12345

Re: [asterisk-users] Dahdi ISDN logging

haven't yet tried doing so. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] DAHDI 2.10 on CentOS 5.11

on this system to change ATTRS back to SYSFS? I notice that it was changed to ATTRS in DAHDI 2.9. 2. Should /dev/dahdi/devices contain just @Board, or also 1/ and 2/? 3. Are the /dev/dahdi/devices entries used by Asterisk or anything else? Thanks for any advice! Cheers Tony -- Tony Mountifield Work: t

[asterisk-users] Other Allison prompts?

-- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

[asterisk-users] Anyone used WatchGuard SIP ALG?

it rewrites it. However, either they or WatchGuard will not accept there is a bug, despite my very detailed description of it. So if anyone else has any experience of using this product, I'd be very interested to hear from you. Thanks! Tony -- Tony Mountifield Work: t...@softins.co.uk - http

Re: [asterisk-users] Anyone used WatchGuard SIP ALG?

In article CAHE6+j3hb5d8mJfY69F73TVwZus9ZAQrDakt4+iW+tx58_uZ=g...@mail.gmail.com, Ishfaq Malik i...@pack-net.co.uk wrote: On 22 April 2014 16:24, Tony Mountifield t...@softins.co.uk wrote: Has anyone here used Asterisk inside a WatchGuard firewall, talking via the WatchGuard SIP

Re: [asterisk-users] Anyone used WatchGuard SIP ALG?

[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony Mountifield Sent: Tuesday, April 22, 2014 12:12 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Anyone used WatchGuard SIP ALG? In article CAHE6+j3hb5d8mJfY69F73TVwZus9ZAQrDakt4+iW+tx58_uZ=g...@mail.gmail.com

[asterisk-users] Asterisk to Microsoft Lync2013?

Are they any gotchas to be aware of in getting Asterisk and Lync 2013 talking to each other using SIP? Or is Lync a pretty standard implementation of SIP? Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

Re: [asterisk-users] Asterisk to Microsoft Lync2013?

In article CAHE6+j31hYcOmyS=kgkv+ge2p+zr-sgfyn2xcmnpmxew_hy...@mail.gmail.com, Ishfaq Malik i...@pack-net.co.uk wrote: On 11 April 2014 11:34, Tony Mountifield t...@softins.co.uk wrote: Are they any gotchas to be aware of in getting Asterisk and Lync 2013 talking to each other using SIP

Re: [asterisk-users] Asterisk to Microsoft Lync2013?

In article li8jbr$nbk$1...@softins.softins.co.uk, Tony Mountifield t...@softins.co.uk wrote: In article CAHE6+j31hYcOmyS=kgkv+ge2p+zr-sgfyn2xcmnpmxew_hy...@mail.gmail.com, Ishfaq Malik i...@pack-net.co.uk wrote: On 11 April 2014 11:34, Tony Mountifield t...@softins.co.uk wrote

Re: [asterisk-users] Need more meetme users -- hitting some limit

to mix. And yes, two pseudos per meetme - one for recording from and one for playing announcements into the conference. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

[asterisk-users] G729 - what happens if licences used up?

system), and also whether it is any different on later versions. Thanks, Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth

Re: [asterisk-users] G729 - what happens if licences used up?

In article CALLKq0RpimD05jz=osbgjydx-41uebohxmft_skwfjt51ko...@mail.gmail.com, Paul Belanger paul.belan...@polybeacon.com wrote: On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifield t...@softins.co.uk wrote: I haven't been able to find the answer online, and am not currently able to conduct

[asterisk-users] 11.5.0 - SIP registration not retrying after failures

to 0=forever). I couldn't see any other relevant settings. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth

Re: [asterisk-users] 11.5.0 - SIP registration not retrying after failures

In article 9519872.915.1383925949785.JavaMail.myoung@myoung-laptop, Michael L. Young myo...@acsacc.com wrote: From: Tony Mountifield t...@softins.co.uk To: asterisk-users@lists.digium.com Sent: Friday, November 8, 2013 10:39:25 AM Subject: [asterisk-users] 11.5.0 - SIP registration

Re: [asterisk-users] issue with dahdi_channels.conf

= qsig signalling = pri_net channel = 32-46,48-62 context = default group = 63 could you please help me What are the contents of /etc/dahdi/system.conf ? Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http

Re: [asterisk-users] issue with dahdi_channels.conf

to which Asterisk was talking had had its channel numbers misconfigured, resulting in a similar problem to what you have described. What happens if you swap the cables over between the two E1 ports on the card? Does the problem move to the second card (channels 32-46)? Cheers Tony -- Tony Mountifield

Re: [asterisk-users] is g729 codec free? or under license???

. It seems to be a solution to a historical problem that has largely gone away nowadays. You will get better quality with G.711 at least. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

Re: [asterisk-users] utils.c: fwrite() returned error: Broken pipe how to solve it ???

$resp,$line; close $s; # go on to display the results from @spans == Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

Re: [asterisk-users] utils.c: fwrite() returned error: Broken pipe how to solve it ???

checked), but it certainly sends \r\n to terminate lines that it outputs. IMHO, it's good to adhere to the same convention in both directions. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

Re: [asterisk-users] RTP port ranges

) It also means that you should allow at least twice as many ports as the number of simultaneous calls you want to handle. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

Re: [asterisk-users] 11.4.0: iax packets lost by amazon ec2

sure about that? I have found in the past that tcpdump sees inbound packets before they get to the iptables filter. What happens if you do: iptables -I INPUT 1 -p udp --dport 4569 -j ACCEPT Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t

Re: [asterisk-users] 11.4.0: iax packets lost by amazon ec2

In article l0fkfp$4ua$1...@ger.gmane.org, Sean Darcy seandar...@gmail.com wrote: On 09/07/2013 10:33 AM, Tony Mountifield wrote: In article 522a934d.8010...@gmail.com, Sean Darcy seandar...@gmail.com wrote: On 09/06/2013 07:08 PM, Steve Edwards wrote: On Fri, 6 Sep 2013, Sean Darcy wrote

Re: [asterisk-users] 11.4.0: iax packets lost by amazon ec2

if it's a new ssh connection, accept it. 5. Otherwise reject it. Nothing in there about accepting UDP, which is why you needed the extra rule to accept the IAX port. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http

Re: [asterisk-users] DAHDI wct4xxp high system CPU on idle?

In article kugfu0$5pj$1...@hp2.softins.co.uk, Tony Mountifield t...@softins.co.uk wrote: I have a system running CentOS 5.9 and DAHDI 2.6.2 with a 2-port E1 card using the wct4xxp driver (also using Asterisk 11.5.0, but that isn't relevant to the question). With DAHDI and Asterisk started

[asterisk-users] DAHDI wct4xxp high system CPU on idle?

, but I don't know whether that is just coincidence. The CPU is a X3450 with four cores and HT enabled. Any thoughts would be gratefully received! Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

[asterisk-users] Multi-homed SIP in Asterisk 11?

system, I wondered if anyone here has made such a setup, and whether there are any issues with getting SDP contents and media routing correct? Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

Re: [asterisk-users] Multi-homed SIP in Asterisk 11?

In article 51f925f2.1040...@dns99.co.uk, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: On 31/07/13 15:32, Tony Mountifield wrote: Most of my experience until recently has been in Asterisk 1.2, and I am just starting to make use of Asterisk 11 for new systems. I have a question

Re: [asterisk-users] Multi-homed SIP in Asterisk 11?

the response! Tony -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony Mountifield Sent: Wednesday, July 31, 2013 10:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Multi-homed SIP

Re: [asterisk-users] What is my syntax error here?

am getting a message say there is no variable to check. So what I have done that is wrong? Is that step split into three lines in your dialplan? I think you might need to put it all on a single line. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t

Re: [asterisk-users] Troubleshooting TDMs (Packet capture like debugging)

provides the drivers with the ability to capture to pcap files, and a tool to control it. Presumably a recent version of Wiresharl will then be able to interpret the captured files. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http

Re: [asterisk-users] ISDN- E1 PRI module in network side signaling

can set this up using any pri card thats supported on Asterisk. And you may need to make an E1 crossover cable. These are different from Ethernet crossover cables. You need to cross pair 1-2 with pair 4-5. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t

Re: [asterisk-users] Set(CALLERID(num-pres)=prohib) / AGI version

, as I have used it to set other function-based values: SET VARIABLE CALLERID(num-pres) prohib Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

[asterisk-users] How to propagate NOANSWER up through a Local channel?

. So I understand the reasons for the above behaviours, but my question is: How can I propagate the NOANSWER status upwards from the inner Dial, so that the Local channel also returns NOANSWER? Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t

Re: [asterisk-users] How to propagate NOANSWER up through a Local channel?

In article 20121108092952.78cb6...@ws78.int.tlc, Chad Wallace cwall...@lodgingcompany.com wrote: On Thu, 8 Nov 2012 16:44:32 + (UTC) t...@softins.co.uk (Tony Mountifield) wrote: Here is a simplified example: [test] exten = _X.,1,Dial(Local/${EXTEN}@outbound) exten = _X.,n,NoOp

Re: [asterisk-users] i extension not triggering

for further details. There has also been discussion about this in the mailing list over the years. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

Re: [asterisk-users] Fully utilise all PRIs in a DAHDI group

-- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] AGI HANGUP PROBLEM

, Asterisk is the called party, and is therefore unable to clear down the line forcibly. This is not an Asterisk or AGI problem but a PSTN one. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

Re: [asterisk-users] AGI HANGUP PROBLEM

there is on SIP involved. The problem is that the PSTN will not drop the call when the called party on an analogue line hangs up, until after a long timeout. There is usually no solution to this. Cheers Tony On Tue, Sep 18, 2012 at 2:03 PM, Tony Mountifield t...@softins.co.ukwrote: In article

[asterisk-users] MixMonitor inserting extra 20ms packets of silence (1.4.43)

? Better still, seen it and solved it? The SIP phones are actually Soundwin ATAs. There is no zaptel or dahdi timing source in the system. Are there any known issues with MixMonitor that could cause this behaviour? Any pointers would be appreciated - thanks! Tony -- Tony Mountifield Work: t

Re: [asterisk-users] Record calls as BLOB into MySQL?

asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=- -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

Re: [asterisk-users] Basic GotoIf question

want to branch when the condition is false, you say: GotoIf(condition?:label2) I guess it could easily have been confusing because typically the condition is an expression contained within $[...], where the square brackets ARE part of the syntax! Hope this helps! Cheers Tony -- Tony Mountifield

Re: [asterisk-users] confbridge

/asterisk/sounds/en fuser * */* Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] confbridge

nothing this time. Not sure what I had done??? Anyway from above looks like the dsp.c tone is whats doing it. No, I think it's something else. Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org

Re: [asterisk-users] confbridge

, then the second call was giving me a busy. Thanks Tony! For all your help. Meetme must have been slightly smarter to say that device is already in the meetme so don't do a second call, where confbridge did not do that. Glad to help, and I'm pleased you tracked it down. Cheers Tony -- Tony

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