On Wednesday 29 May 2019 at 22:01:11, Jason wrote:
> I am communicating
How?
> with Asterisk 13.18.3 over the AMI and issue the command:
>
> ActionID: 11
> Action: command
> Command: core show calls
>
> And the response I get is:
>
> Response: Follows
> Privilege: Command
> ActionID: 11
> --E
On Friday 12 April 2019 at 15:35:04, Gokan Atmaca wrote:
> > The phone does work, you do need to TFTP the configuration files to the
> > phone though. Doesn't look like custom firmware is required.
>
> very thanks...
>
> the following ?
> cisco ip phone: 8905 , 8950, 8450
I am unable to ident
s why I couldn't find a model 3950 phone :)
Well, it uses SIP, so it should work as well as any Cisco phone does on
Asterisk.
Antony.
> On Fri, Apr 12, 2019 at 3:58 PM Antony Stone wrote:
> > On Friday 12 April 2019 at 14:42:57, Gokan Atmaca wrote:
> > > Hello
&
On Friday 12 April 2019 at 14:42:57, Gokan Atmaca wrote:
> Hello
>
> Can I use Cisco 3950 on Asterisk ?
Please give us a link to a datasheet for that device.
Regards,
Antony
--
Schrödinger's rule of data integrity: the condition of any backup is unknown
until a restore is attempted.
On Thursday 04 April 2019 at 17:59:07, Karsten Wemheuer wrote:
> Am Sonntag, den 10.03.2019, 12:46 +0300 schrieb Gokan Atmaca:
> >
> > Mynum: 6001 , Othernum: 6002.
> >
> > I can record as follows. But I do not enter individual records for
> > each internal required. I want to do it more smoothl
On Thursday 04 April 2019 at 14:28:15, Brian J. Murrell wrote:
> # echo -e "Action: Login\r\nUsername: myasterisk\r\nPassword: a\r\n\r\n"
It's not "Password", it's "Secret" :)
Antony.
--
I don't know, maybe if we all waited then cosmic rays would write all our
software for us. Of course it m
On Thursday 21 March 2019 at 21:59:51, Darryl Moore wrote:
> For a paging system? No you don't. A number of SNOM PA1's and a few
> grandstream phones and you're golden.
Are you suggesting using standard telephones (presumably in auto-answer
speakerphone mode) as paging devices?
Depending on the
Hi.
I'm reading https://wiki.asterisk.org/wiki/display/AST/Function_CDR
and wondering what "r - Searches the entire stack of CDRs on the channel."
means.
Can anyone help with understanding this?
Thanks,
Antony.
--
Normal people think "If it ain't broke, don't fix it".
Engineers think "If
On Tuesday 19 March 2019 at 21:36:53, Mike Diehl wrote:
> Hi all,
>
> I have a user who is reporting one-way audio, but only when a call is made
> to or from particular PSTN (cell) numbers.
I'm assuming you're using a PSTN trunking provider to connect to those numbers
(ie: you don't have your o
On Friday 15 March 2019 at 15:18:04, sean darcy wrote:
> From my provider I get extensions of:
>
> +1<10digit number>
> 1<10 digit number>
> <10 digit number>
>
> seemingly randomly.
>
> What I'd like to do is
>
> exten=_!1234567890,1,Answer()
> Any suggestions?
exten => _+1XX,Goto(
On Tuesday 12 March 2019 at 01:19:37, Jerry Geis wrote:
> Lets say I have to make 40 phone calls quickly.
>
> If I use the AMI interface to originate a call, close the connection, open
> another connection etc...
> This works. but is slow...
How about using call files instead?
https://wiki.aste
.wav,ab)
> exten => 6XXX,n,Dial(SIP/${ARG2}/${ARG1},20) (*)
> exten => 6XXX,n,StopMixMonitor()
> exten => 6XXX,n,Hangup()
>
> On Sun, Mar 10, 2019 at 12:59 PM Antony Stone wrote:
> > On Sunday 10 March 2019 at 10:46:24, Gokan Atmaca wrote:
> > > Hello
> >
d also makes things a lot simpler if you decide you need to
change the basic pattern match at some time in the future).
> On Sat, Mar 9, 2019 at 6:50 PM Doug Lytle wrote:
> > On Sat, Mar 9, 2019 at 4:25 PM Antony Stone wrote:
> >
> > a) work for recording incoming / outgoing ca
On Saturday 09 March 2019 at 14:19:19, Gokan Atmaca wrote:
> Hello
>
> How can I record voice between internalities? I can record voice in
> incoming and outgoing calls, but I can't make it between the internal.
> Would you support this?
Show us the parts of your dial plan which:
a) work for re
On Tuesday 05 March 2019 at 17:22:16, Gokan Atmaca wrote:
> > exten => _13XXX,1,dial(${OPERATOR},20)
1. What is the content of ${OPERATOR}?
2. What do you have for this connection in sip.conf?
3. What number/s have you been assigned by your upstream SIP provider?
Antony.
> On Tue, Mar 5,
On Thursday 28 February 2019 at 18:00:54, Ivan Demkovitch wrote:
> Antony,
> Ok, I see what you are saying. Yes, than NAT occuring on our router.
> Asterisk server is on internal IP (192.168..)
> # Now that I read what you say I think there might be 2 issues. "Randomness"
> is one, but I am not e
On Thursday 28 February 2019 at 17:40:28, Ivan Demkovitch wrote:
> Noone connects to Asterisk box/server from outside. Callcentric SIP trunk
> configured and Asterisk maintains connection to it itself.
Okay, I didn't actually mean "does anyone connect *inbound* to your Asterisk
server" - I was
On Thursday 28 February 2019 at 00:26:17, Ivan Demkovitch wrote:
> Asterisk is NOT exposed to internet, noone connects to Asterisk
> from internet. We use Callcentric for VOIP trunk.
That's the point where you lost me.
Callcentric is out on the Internet. How does it connect to your Asterisk
se
On Wednesday 27 February 2019 at 23:10:33, Ivan Demkovitch wrote:
> Hello,
> This is not technical post,
Hm, no?
> just looking for suggestions on what to check.I have asterisk for long time,
Which version?
> no updates, just maintain OS updates. I use SPA504G phones.
Tell us about your netw
On Friday 22 February 2019 at 18:05:26, hw wrote:
> Hi,
>
> the ecnryption tutorial[1] says to add 'encryption=yes' into sip.conf
> for a peer to use SRTP.
>
> I have all the account information in a mysql database in a table called
> `sippeers` asterisk uses. The table doesn't seem to have a c
On Monday 11 February 2019 at 14:14:16, Susan Ruiz wrote:
> how to configure sip did from with asterisk?
1. Create a trunk in sip.conf, tell it the context to send calls to.
2. Create that context in extensions.conf, telling it how you want to handle
the calls.
If you need further details or g
On Wednesday 06 February 2019 at 13:54:44, Mark Wiater wrote:
> These two phones are not using the same extension, are they?
If you shut down the softphone, does the hardware phone then ring?
Antony.
> On 2/6/2019 8:49 AM, basti wrote:
> > both phones are registered. and the hardware phone can
On Wednesday 06 February 2019 at 13:49:39, basti wrote:
> both phones are registered. and the hardware phone can also make calls.
> but an incoming call is not displayed and also not hearing.
Are both the hardware phone and the softphone on the same network?
If not, is one behind NAT and the oth
uot; as the
extension. It depends on what is feeding into you "o2-in" context, of course.
Antony.
> On 31.01.19 11:26, Antony Stone wrote:
> > On Thursday 31 January 2019 at 10:59:01, basti wrote:
> >> Hello I use this dial paln:
> >>
> >> [o2-in]
&
On Thursday 31 January 2019 at 10:59:01, basti wrote:
> Hello I use this dial paln:
>
> [o2-in]
> exten => o2,1,Answer
> exten => o2,n,Playback(hello-world)
> exten => o2,n,Ringing
> exten => o2,n,Dial(SIP/10&SIP/20&Local/s@no-op,25,rt)
> exten => o2,n,Playtones(425/1000,0/4000)
> exten => o2,n,W
y idea about it is that it's more of an after-the-fact SIP
analysis tool, not a real-time "what is my Asterisk server doing *now* " tool.
Regards,
Antony.
> > Date: Sat, 15 Dec 2018 17:30:37 +0100
> > From: Antony Stone
> > To: asterisk-users@lists.digium.
Hi.
Does anyone have any recommendations for a *really* real-time monitoring
solution for Asterisk?
I'm thinking that something like Grafana (which I've played with for another
purpose, but don't really use yet) can do a good job of displaying the data
with very little latency, but I'm wonderi
On Thursday 06 December 2018 at 17:49:25, hw wrote:
> On 12/05/2018 05:00 PM, Antony Stone wrote:
> > On Wednesday 05 December 2018 at 15:31:38, hw wrote:
> >> I don't see a table for that.
> >
> > You need to create that for yourself.
> >
> > Ast
On Wednesday 05 December 2018 at 15:31:38, hw wrote:
> On 12/05/2018 01:19 PM, Antony Stone wrote:
> > On Wednesday 05 December 2018 at 13:04:57, hw wrote:
> >> On 12/04/2018 07:07 PM, Antony Stone wrote:
> >>> On Tuesday 04 December 2018 at 16:11:39, hw wrote:
On Wednesday 05 December 2018 at 13:04:57, hw wrote:
> On 12/04/2018 07:07 PM, Antony Stone wrote:
> > On Tuesday 04 December 2018 at 16:11:39, hw wrote:
> >> On 12/01/2018 05:30 PM, Marcelo Terres wrote:
> >>> Queue_log
> >>
> >> Thanks!
> &g
On Tuesday 04 December 2018 at 16:11:39, hw wrote:
> On 12/01/2018 05:30 PM, Marcelo Terres wrote:
> > Queue_log
>
> Thanks!
>
> That's not really it; however, how do I make it so that asterisk writes
> this information right away into a mariadb database instead of into a
> file so that I could
Hi.
I'm trying to use the VM_INFO function to discover the number of messages in a
user's mailbox, for reporting purposes outside of Asterisk.
I'm using Asterisk 13.14.1~dfsg-2+deb9u4 on Debian 9 "Stretch".
${VM_INFO(${User}@Voicemail,count,INBOX)} tells me the number of unheard
messages in th
On Tuesday 23 October 2018 at 12:51:56, Doug Lytle wrote:
> >>> No, it's not a firewall problem; I've currently allowed connections to
> >>> 5038
>
> Antony,
>
> Do you have any deny/permit section in the manager.conf that would need to
> be adjusted?
No, and since I posted this, I've found the
Hi.
I have three servers running corosync and pacemaker, to maintain a floating
address between them. This is working fine, and I can, for example, SSH to the
floating address and get to whichever server has the address at the time.
I am trying to connect to the same server (using the same add
On Tuesday 16 October 2018 at 19:04:42, Ivan Demkovitch wrote:
> Thanks all,
> I did contact Callcentric about it and their tech support helped meget
> those headers established. They even helped to troubleshoot Asterisk
> dialplan. A the end all works as it should.
For the benefit of others who
On Friday 12 October 2018 at 10:38:49, Jonas Kellens wrote:
> Hello
>
> thank you for your answer.
>
> This does not happen all the time. It happens about once every 4 months.
> I just can not pinpoint WHEN exactly it occurs. I just see in the
> verbose logfile that it occurs after a MYSQL inser
On Thursday 11 October 2018 at 22:11:10, Ivan Demkovitch wrote:
> Abdul,
> Added code like you proposed, I see it in logs but still don't see caller
> ID coming in:
> -- Goto (internal,101,1)
> -- Executing [101@internal:1] NoOp("SIP/callcentric13-06d1", "Call
> ID: "DEMKOVITCH,IVAN" <1555
On Tuesday 09 October 2018 at 18:52:59, Yves wrote:
> Am 08.10.2018 um 13:02 schrieb Antony Stone:
> > On Monday 08 October 2018 at 12:44:43, Yves wrote:
> >> I am looking for an easy way to execute any AGI Command directly from
> >> the dialplan without the need
On Monday 08 October 2018 at 12:44:43, Yves wrote:
> I am looking for an easy way to execute any AGI Command directly from the
> dialplan without the need to call an external script.
The whole point of AGI is that it calls an external script in order to replace
commands in the dialplan.
Executi
On Thursday 04 October 2018 at 17:10:01, Jonas Kellens wrote:
> Hello
>
> using Asterisk 1.8.32.
Ooh, vintage :)
> I notice that there is a spontaneous reboot of the Asterisk system from
> time to time.
>
> When I look in the logs (verbose file) I noticed that every time this
> occurs it's at
you run into problems come back and let us knwo what you did and
what's not working and we can try to help.
Antony.
> On Mon, Sep 17, 2018 at 7:18 PM Antony Stone wrote:
> > On Monday 17 September 2018 at 15:42:50, Priyaranjan Nayak wrote:
> > > Hi All,
>
On Monday 17 September 2018 at 15:42:50, Priyaranjan Nayak wrote:
> Hi All,
>
> I am using Asterisk 15 server and wanted to configure IVR call simulation.
What do you mean by "simulation"?
> My configuration scenario is
> 1. A subscriber will register to Asterisk server and start a call.
> 2. T
On Monday 10 September 2018 at 21:54:33, Marcelo Terres wrote:
> I have think it should be
>
> context=0705680837
>
> Not
>
> context=[0705680837]
Ha! You're right... so simple :)
Antony.
> On Mon, 10 Sep 2018, 20:43 , wrote:
> > On 2
On Saturday 08 September 2018 at 22:38:19, aster...@a-domani.nl wrote:
> Hi all
>
> some how I'm getting confused: it seems I clobbered incoming calls from
> my sip provider.
> I can not imagine that my upgrade from 15.3 to 15.5 could be related
>
> I'm certain that dialling my own number, resul
On Saturday 01 September 2018 at 22:12:50, sean darcy wrote:
> 13.21.0
>
> Every 2-3 minutes:
Does it really vary, or is it more like "every 150 seconds"?
> Sep 1 16:00:57] WARNING[150257]: res_stun_monitor.c:140
> stun_monitor_request: STUN poll got no response. Re-evaluating STUN
> server ad
On Wednesday 22 August 2018 at 23:49:29, Ahmed Chohan wrote:
> Hi,
>
> I would like to know how can I achieve merge 2 conference rooms in same
> asterisk server. For example 10 users joined bridge A and max user limit is
> set to 10. If more than 10 users try to join this bridge A, 11th user
> sh
On Sunday 19 August 2018 at 14:20:35, Khalil Khamlichi wrote:
> Thanks for your response, this works but we cannot hardcode this in the
> dialplan, we need this to be done from an external application connected
> either via manager or stasis.
Have you considered using Asterisk Realtime to store (
On Wednesday 15 August 2018 at 23:10:21, Dovid Bender wrote:
> Hi,
>
> I am trying to install wanpipe
Tell us how you are trying to install things.
> with dahdi on a CentOS7 box and I am running in to a few issues. My setup.
>
> CentOS 7
> asterisk-15.5.0
> libpri-1.6.0
> dahdi linux and dahdi
On Wednesday 08 August 2018 at 22:30:52, Saint Michael wrote:
> I am trying to install Asterisk 11
Why?
> on debian 9
Have you tried installing https://packages.debian.org/jessie/asterisk from
Debian 8 to see if it'll go onto Debian 9?
Antony.
--
Programming is a Dark Art, and it will alwa
On Tuesday 31 July 2018 at 12:38:04, Raimundo Pérez Nieves wrote:
> Hi guys, I sent a dial to asterisk
Which verson?
> with a specific timeout, I want to increase it for some users if it is
> approaching to the end, but when I send AbsoluteTimeout action
Show us what command you are sending?
>
On Wednesday 25 July 2018 at 19:53:47, Saint Michael wrote:
> I need to launch a remote process at the machine that has the dialer. I
> could hard-code the IP address in a global variable, but It would be much
> more elegant if the dialplan would have a "manager" object where I could
> read "mana
On Friday 13 July 2018 at 15:53:47, Dovid Bender wrote:
> Hi,
>
> Is there any way of telling Asteirsk to withhold answer subversion on a
> call till I call Answer.
Please could you express differently what you are trying to do?
I do not understand the above question.
> My DP looks like this:
On Thursday 07 June 2018 at 10:44:15, Tony Mountifield wrote:
> In article <201806070119.51560>, Antony Stone wrote:
> >
> > Is there any way to tell AMI that I don't want it to log login attempts -
> > or, to put it another way, is there any way to tell the
Hi.
Is there any way to eliminate AMI manager logins from the logging output
(without just turning the log level down and thereby losing lots of other stuff
as well)?
I'm running Asterisk 13.14.1 as a backend service to LVS/IPVS, and using the
AMI login as the "service alive" check to see whic
On Wednesday 06 June 2018 at 16:30:08, Dovid Bender wrote:
> On Wed, Jun 6, 2018 at 6:18 AM, Antony Stone wrote:
> > On Wednesday 06 June 2018 at 12:02:38, Dovid Bender wrote:
> > > Hi,
> > >
> > > I have tested ControlPlayback and grabbed files via
On Wednesday 06 June 2018 at 12:02:38, Dovid Bender wrote:
> Hi,
>
> I have tested ControlPlayback and grabbed files via an apache server with
> no issue.
ControlPlayback is an Asterisk dialplan function.
How have you integrated this with Apache?
> I want to be able to grab files via aws S3 wh
On Tuesday 05 June 2018 at 08:33:26, David P wrote:
> We're using Asterisk 14.7.6 and I have a dialplan that ends like this:
>
> same => n,Dial(SIP/${EXTEN:0:4}@peer1)
> same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH})
> same => n,Hangup()
>
> When peer1 hangsup, the priorities after the Dial a
On Thursday 31 May 2018 at 15:52:53, Jonas Kellens wrote:
> Hello list
>
> is there a way to limit the number of dns lookups for 1 and the same host?
>
> I see on Asterisk CLI a flood of :
>
> [May 31 15:45:37]> doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
> [May 31 15:45:37]>
On Thursday 12 April 2018 at 18:17:10, Hermann Wecke wrote:
> On Thu, Apr 12, 2018 at 11:43 AM, Antony Stone wrote:
> >
> > Are you by any chance running fail2ban, without the IP address of this
> > location in a whitelist?
>
> fail2ban: yes
> whitelist: yes
Have
On Thursday 12 April 2018 at 17:22:39, Hermann Wecke wrote:
> I'm trying to solve a mystery for the last couple of days.
>
> I have a mix of D70, D50 and D40 behind NAT. Server is in a
> colocation, not a VPS.
>
> For several years, everything was working fine, no issues. A few days
> ago I star
On Wednesday 04 April 2018 at 00:30:00, Richard Mudgett wrote:
> On Tue, Apr 3, 2018 at 4:57 PM, Matt Fredrickson wrote:
> > On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield
> > > Both the 32-bit and 64-bit were fresh installs of the latest CentOS 6.9
> > > from online repositories using a kicks
On Wednesday 04 April 2018 at 00:20:29, Dave Platt wrote:
> Also, check the wiring. Check each individual RJ-45 jumper, *and* the
> in-house wiring, with a proper tester that can verify that the
> individual pairs are hooked up correctly.
Can you provide a link to such a tester?
> The problem i
On Tuesday 27 March 2018 at 15:25:25, Tech Support wrote:
> All;
>
> We do a lot of programming and customizations for Asterisk and
> normally, we do everything in Perl. For that, we use the CPAN module
> Asterisk::AMI, and it works great. However, we have several programs that
> would benefi
On Monday 26 March 2018 at 23:16:07, Benjamin Marty wrote:
> 2018-03-26 23:01 GMT+02:00 Antony Stone:
> > That sounds like a pretty big challenge for Asterisk.
>
> As far as it at least claim to record some sort of h264 it doesn't sound as
> a to big challenge for me
On Monday 26 March 2018 at 22:33:25, Benjamin Marty wrote:
> Hi,
>
> I'm using the Record dialplan Application in an Context. My goal is to get
> a single screenshot of the h264 media stream per call.
That sounds like a pretty big challenge for Asterisk.
> same => n,Record(/tmp/test.wav,0,10,q
On Monday 26 March 2018 at 10:58:01, Administrator TOOTAI wrote:
> Hi all,
>
> we are running some Asteriks (version 13 on Debian Stretch) as VM/kvm in
> datacenter and have trunks with other Asterisk (v1.8 11 or 13) instances
> behind FW. Problem we face is that when we reboot the DC Asterisks,
On Tuesday 06 March 2018 at 09:05:25, Markus Weiler wrote:
> Hi Group,
>
> we're just wondering, in German we call the different types of phone-numbers
> (Geographic,mobile,national,VoIP...) Rufnummerngassen (phone number alleys
> ;-) )
> Is there an english word for this?
No.
It's just another
On Monday 05 March 2018 at 12:06:51, Atux Atux wrote:
> Hi. I have an Asterisk Server (A) where it acts as the main gateway to
> offer services.
> There are different asterisk servers (B -D) that connect as extensions to
> the Server A.
Why not use IAX?
> I would like to implement TLS and SRTP f
On Thursday 01 March 2018 at 14:02:37, Atux Atux wrote:
> Hi. I would like to protect my system from failed attempts. I would like to
> ask if there is a way to do a blacklist for certain amount of time
> consecutive attempts from the same IP.
fail2ban
> For example if we have an IP that gets a
On Thursday 22 February 2018 at 23:44:43, Carlos Chavez wrote:
> On 2/22/18 4:40 PM, Carlos Chavez wrote:
> > On 2/22/18 3:54 PM, Carlos Chavez wrote:
> >> On 2/22/18 3:46 PM, Antony Stone wrote:
> >>> On Thursday 22 February 2018 at 21:41:41, Carlos Chavez wr
On Thursday 22 February 2018 at 21:41:41, Carlos Chavez wrote:
> On 2/22/18 1:07 PM, Antony Stone wrote:
> > On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote:
> >> Usually phone companies set the outgoing CallerID for you but
> >>
> >> recen
On Thursday 22 February 2018 at 20:07:47, Antony Stone wrote:
> On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote:
> > Usually phone companies set the outgoing CallerID for you but
> >
> > recently we got control over that and are now setting the ou
On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote:
> Usually phone companies set the outgoing CallerID for you but
> recently we got control over that and are now setting the outgoing
> Calleir ID ourselves. My problem now is that the CDR will put the
> outgoing CID in the CDR in
On Tuesday 20 February 2018 at 14:09:05, Marcus Kvarsell wrote:
> Hi,
>
> I am experimenting with getting hold of the sip cause and sip response from
> outgoing call. How could i make a userevent printing the sip cause and/or
> sip response. I have tried using hangupcause, sip_cause and such , bu
On Monday 12 February 2018 at 12:25:00, Uzma Anjum wrote:
> Hello...
>
> I'm running asterisk-13 and international calls are not working in it.How
> can I make it work.Can anyone please tell me.
We are sorry, but all our telepaths and clairvoyants are busy dealing with
other queries right now.
On Tuesday 23 January 2018 at 14:43:21, Tech Support wrote:
> All;
>
> I had someone ask me if they received an incoming phone call and it was
> forwarded "off pbx" to their cell phone, would the call be strictly between
> the caller and the cell phone, or would it between the caller, the pbx
On Saturday 20 January 2018 at 18:45:49, Jonathan H wrote:
> Oh, what a good idea! That's exactly the kind of lateral thinking I
> was hoping someone would come up with.
>
> I thought it was called MixMonitor, and tried to wrap my head around
> it but couldn't.
MixMonitor is related, but differe
On Wednesday 17 January 2018 at 11:59:21, Paul Neuwirth wrote:
> Hello group,
>
> I tried a lot to enlarge the frequency (i.e. more announces, low wait
> between). according to config, every 30 seconds the announcement should
> take place. In fact, the first periodic announce is done after 2
> mi
On Tuesday 16 January 2018 at 18:19:30, Paul Neuwirth wrote:
> On Tue, 16 Jan 2018 18:18:18 +0200 Tzafrir Cohen wrote:
>
> > Anyway, as mentioned before: you should probably use AMI.
>
> Thank you both. That was (most likely) what I was looking for - but
> still some worries about sending plainte
On Thursday 04 January 2018 at 01:27:59, bilal ghayyad wrote:
> Hello
> It will be amazing if possible to do sip trunk with any of social media
> providers like: whatsapp, facebook, imo, viber, ... etc
To the best of my knowledge none of the services you mention either operate
over SIP or provid
On Sunday 31 December 2017 at 00:49:17, sean darcy wrote:
> I've been getting a lot of timeouts on non-critical invite transactions.
> So how is someone on a Dutch ISP using my server to mess with a US DoD
> ip address ?
What's your setting for "allowguest" (under [general]) in
/etc/asterisk/si
On Sunday 12 November 2017 at 18:27:56, Tzafrir Cohen wrote:
> On Sun, Nov 12, 2017 at 04:45:45PM +0000, Antony Stone wrote:
> > Hi.
> >
> > https://www.voip-info.org/wiki/view/Asterisk+manager+API says that "There
> > are a finite (but extendable) set of
Hi.
https://www.voip-info.org/wiki/view/Asterisk+manager+API says that "There are
a finite (but extendable) set of actions available to the client, determined by
the modules presently loaded in the Asterisk engine."
Can anyone point me at some appropriate documentation for adding custom
comman
On Thursday 02 November 2017 at 16:33:04, Tech Support wrote:
> I have a customer who is looking for a particular DID. (I dialed it and
> it is not in service). I searched through my preferred upstream provider's
> list but I came up empty. I wrote them, and this is their reply.
>
> "We curre
On Wednesday 01 November 2017 at 12:15:08, Michael Maier wrote:
> On 11/01/2017 at 10:14 AM Antony Stone wrote:
> > On Wednesday 01 November 2017 at 08:10:36, Michael Maier wrote:
> >>
> >> I'm facing the following scenario:
> >>
> >> - I
On Wednesday 01 November 2017 at 08:10:36, Michael Maier wrote:
> Hello!
>
> I'm facing the following scenario:
>
> - Initial call opened to asterisk: SDP g722,alaw,ulaw
>
> - Outgoing call to provider started with Invite / SDP alaw, g726 and
> g729.
So, you're claiming to the provider that
Hi.
Does anyone have some recommendations for measuring total end-to-end latency
(by which I mean: the time between person A saying something and person B
hearing it) when there are both SIP and PSTN/analogue/mobile legs in the call
path?
Examples:
Person A has a SIP phone registered to Aster
On Monday 02 October 2017 at 20:58:33, Steve Edwards wrote:
> I recently received a GoIP-32 for a client project -- primarily outbound
> calling.
>
> How should a GoIP be configured for Asterisk?
Have you tried http://www.hybertone.com/en/solutionsClass.asp?Id=78
Antony.
--
Police have found
Hi.
Does anyone know of a way to find out the ring time of a call as soon as it has
been answered (ie: without waiting for the call to be completed, when it's
part of the standard CDR record)?
I'm looking for a way to place a call, wait for it to be answered, and then
perform different actions
On Friday 01 September 2017 at 16:48:17, Dovid Bender wrote:
> On Fri, Sep 1, 2017 at 9:13 AM, Joshua Colp wrote:
> > On Fri, Sep 1, 2017, at 09:01 AM, Dave Topping wrote:
> > > http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/
> As Josh mentioned this is an issue with RTP and the SDP
On Thursday 31 August 2017 at 18:15:54, Joseph Smith wrote:
> I was hoping Asterisk would handle more than 4k simultaneous calls.
I know from experience that Asterisk can handle more than 4k simultaneous
calls, however it's an extreme case to have all of them playing music on hold.
I think that
On Saturday 29 July 2017 at 19:03:55, Joshua Colp wrote:
> On Sat, Jul 29, 2017, at 02:55 PM, O. Hartmann wrote:
> > Scenario:
> >
> > Our Asterisk 13 PBX (on network 192.168.254.0/24, bound to
> > 192.168.254.1:5060) is behind
> > a NAT, acting as a client to our ITSPs SIP server. But also, this
On Thursday 20 July 2017 at 20:46:30, Marcelo Terres wrote:
> I don't have much knowledge about freepbx, but if some day I had to use it,
> I would prefer to use the Asterisk compiled from source, unless it comes
> with an Asterisk package (rpm, supposing it is running CentOS).
FreePBX (as a dis
On Friday 14 July 2017 at 23:34:37, Motty Cruz wrote:
> Since the upgrade our remote users' conversions are choppy.
> Monitoring using CLI, I noticed the device always select ulaw
> for codec.
What's the device?
What are its codec settings?
What's your available & used bandwidth on the server'
On Saturday 08 July 2017 at 10:16:19, Antony Stone wrote:
> On Saturday 08 July 2017 at 07:15:08, Marcelo Terres wrote:
> > There are no sip show channels on AMI. Also, the output that you sent is
> > not a AMI output. Are u using AMI ou running commands on console?
>
>
u will probably achieve your goals
>
> https://wiki.asterisk.org/wiki/display/AST/AMI+Actions
Hm, I don't see anything there which will give me a list of the SIP channels
currently in use - what command should I be using for that?
Thanks,
Antony.
> On 7 Jul 2017 10:32 pm, Antony Stone wr
Hi.
I'm trying to get a list of the channels currently in use on an Asterisk server
(1.8.32.1 if it matters) over AMI.
I send the command "sip show channels", and I get back a response along the
lines of (* used to protect the innocent...):
Peer User/ANR Call ID Fo
On Friday 30 June 2017 at 19:11:08, Jonathan H wrote:
> I use a python AGI which pulls some info from a web service, which should
> take half a second.
>
> Sometimes, it takes 5-10 seconds which blocks the dialplan execution, but
> the dialplan should continue immediately as it's not dependent on
Hi.
I have some Nagios / Icinga monitoring plugins I've created for Asterisk, and
one of them checks the percentage of SIP accounts which are currently
registered on an Asterisk server.
It does this by running "sip show peers" via AMI and analysing the summary
line at the end:
1066 sip peers
On Monday 26 June 2017 at 18:01:22, J Montoya or A J Stiles wrote:
> On Monday 26 Jun 2017, Harel wrote:
> > Hello List,
> > I'm working on an autodialer project.
> > At the moment I use the Originate application then I "throw" it to an
> > extension where I Dial() the other party and then both le
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