Le 25/01/2023 à 17:56, Antony Stone a écrit :
On Wednesday 25 January 2023 at 16:46:14, Daniel wrote:
On Sunday 01 January 2023 at 17:30:03, Antony Stone wrote:
The [globals] section of that dialplan includes:
Kphones=SIP/KC470IP&SIP/KSnom870
Sphones=SIP/SYealinkT38G
; 20 and there is no problem.
BTW you should move to asterisk community lots more people there.
--
Daniel
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Check out the new Asterisk community forum at: h
Hi there,
I can confirm that this is indeed the problem.
If you follow the advise below you will be sorted.
From my mobile phone
On 23 Jul 2021, 8:44 am, at 8:44 am, Jean Aunis wrote:
>Le 22/07/2021 à 18:32, Carlos Chavez a écrit :
>> I started noticing a few days ago that whenever I dia
and the Asterisk is not crashing.
I looked at the source code and it seems that I am sending all the right
parameters with a valid JSON format.
Anyone has an idea what has been change in the Asterisk code and how can I
adopt my code to the this
the PAI headers.
Regards,
Daniel Friedman
Trixton LTD.
Tel: 972.72.2557000
Mobile: 972.50.6655579
Email: d...@3xton.com
Website: http://www.3xton.com
-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of asterisk-users-requ
On Thu, Oct 11, 2018 at 05:18:24PM +, Ivan Demkovitch wrote:
> Where problem comes in - if person not at the desk - his cell phone shows
> call from OFFICE number and there is no way to tell who is really calling.
> We use Callcentric as a trunk if it makes any difference.
> I'd like to add in
On Fri, Oct 12, 2018 at 07:59:52AM -0400, Telium Support Group wrote:
> I have an Asterisk system with 2 trunks (as shown below). I need to be able
> to disable a trunk at runtime. I may not change the dialplan but I can
> change sip.conf and reload.
>
> Any attempt to dial in the dialplan uses t
On Wed, Aug 29, 2018 at 11:37:34AM -0400, Jerry Geis wrote:
> I have a connection to a cisco all manager SIP trunk. The first call coming
> across CCM to the asterisk server works fine... Then when I do a second
> call from CCM to asterisk I am getting a SIP 401 unauthorized.
>
> My definition is
> It does seem like a bug. However, you have a complicated dialplan with a lot
> of pieces happening at
> once so it may not actually be an Asterisk bug but a problem with your
> dialplan. To unravel this is
> going to take some bookkeeping on your part.
Hi Richard,
Thanks for the detailed re
> Doing some more tests, this reads like a bug to me.
> Using a hanguphandler with DumpChan in the dialplan context that executes
> the Queue, I can see that DYNAMIC_FEATURES is set.
> After the attended transfer when the call is ended, the hanguphandler still
> shows that DYNAMIC_FEATURES is set.
> Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp.
> AgentA answers and is able to use that feature code.
> If AgentA performs an attended transfer of a call from a queue to AgentB, the
> feature code no longer works.
>
> It only doesn't work when using Queue() and an Attend
Hi,
I think I've identified an issue and just want to check before completing a bug
report.
Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. AgentA
answers and is able to use that feature code.
If AgentA performs an attended transfer of a call from a queue to AgentB, the
On Fri, Aug 03, 2018 at 04:24:06PM +0200, Daniel Tryba wrote:
> redirect_method=uri_pjsip works as expected with regard to the header
> manipulation stuff.
>
> Also I can't remember why, in the past, I decided to not use uri_pjsip
> other than having the redirected host in CD
On Thu, Aug 02, 2018 at 05:29:23PM +0200, Daniel Tryba wrote:
> With chan_sip there is the variable SIP_MAX_FORWARDS to set
> Max-Forwards. This counter is persistant after a redirect. I can't find
> the equivalent for PJSIP, so I went the way of header manipulation. Only
> to
On Thu, Aug 02, 2018 at 02:40:48PM +1000, Patrick Wakano wrote:
> In my opinion, Asterisk should at fail the Dial and proceed with whatever
> was configured in the dialplan I tried some other 4XX SIP codes, but
> the only one I found not behaving properly is the 400 one
I think you are rig
On Thu, Aug 02, 2018 at 05:29:23PM +0200, Daniel Tryba wrote:
> With chan_sip there is the variable SIP_MAX_FORWARDS to set
> Max-Forwards. This counter is persistant after a redirect. I can't find
> the equivalent for PJSIP, so I went the way of header manipulation. Only
> to
With chan_sip there is the variable SIP_MAX_FORWARDS to set
Max-Forwards. This counter is persistant after a redirect. I can't find
the equivalent for PJSIP, so I went the way of header manipulation. Only
to find out that any headers added to the outbound leg are lost after a
redirect (with redirec
On Tue, Jun 19, 2018 at 07:38:12PM +0200, Olivier wrote:
> I've just discovered chan_sip's ignoresdpversion setting.
> Do you use it ?
> If positive which kinnd of issue could you solve with it ?
IIRC I used to enable this option when talking to some Ericsson SBC. It
solved a problem concerning a
On Fri, Jun 15, 2018 at 05:32:30PM +0200, Olivier wrote:
> In my testing, I saw that Asterisk always included a REFER value in each
> INVITE's Allow header, no matter how allowtransfer/allow_tranfer was set.
>
> Is there a way to remove this REFER value entirely either globally or
> specifically f
On Tue, Jun 05, 2018 at 11:34:51AM +0200, Olivier wrote:
> 1.According SIP RFCs, is possible/recommended to have different values in
> From and P-Asserted-Id fields ?
> For instance, From field showing 123456789 and P-Asserted-Id showing
> 987654321 (beside privacy considerations) ?
Yes, most obvi
On Tue, May 29, 2018 at 08:32:39PM -0700, David P wrote:
> We would like to use 20-char extension values that use dashes and alphanums
> after the first four digits. In order to handle these via pattern-matching,
> how can I define a pattern that allows dashes? There seems to be no option
> at http
On Thu, May 17, 2018 at 12:27:17PM -0400, sean darcy wrote:
> > WARNING.* .*: fail2ban=''
> >
> ># Option: ignoreregex
> ># Notes.: regex to ignore. If this regex matches, the line is ignored.
> ># Values: TEXT
> >#
> >ignoreregex =
> >
> >
> Thanks. Very useful as a tutorial for fai
On Wed, May 16, 2018 at 04:51:49PM +0200, Olivier wrote:
> 1. When Asterisk receives a SIP call coming from PSTN, is there a time
> frame within which Asterisk must reply something to keep caller from
> canceling the call ? Where does this limit come from ? From SIP RFC ? From
> local regulation bo
On Wed, May 16, 2018 at 11:01:53AM -0400, Mike Diehl wrote:
> I have a user who would like to stream their favorite radio station from
> iHeart radio for their music on hold.
>
> It this TECHNICALLY possible?
Yes.
> If so, any pointers would be appreciated.
https://www.voip-info.org/asterisk-co
On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote:
> I receive an INVITE/SDP containing:
>
> m=audio 11310 RTP/AVP 3 0 101
>
> which I interpret as gsm, ulaw, rfc2833.
>
> and I reply with an OK/SDP containing:
>
> m=audio 15884 RTP/AVP 0 3 101
>
> which I interpret as
On Tue, May 08, 2018 at 03:04:55PM -0500, Jeff LaCoursiere wrote:
> Thats till doesn't change the SIP header.?? Basically they want to send a RE
> INVITE and authenticate my DID number.?? But my DID number does not have a
> peer or user entry in sip.conf.?? Perhaps I am answering my own question,
>
On Wed, Apr 11, 2018 at 12:04:18PM -0400, Telium Technical Support wrote:
> Maybe proxy is the wrong word I chose. Asterisk is something like a peer to
> the legacy PBX. I thought about setting up individual SIP accounts on the
> Asterisk box to connect to the legacy PBX, or maybe a SIP trunk to
On Tue, Apr 10, 2018 at 09:22:02PM -0400, Telium Technical Support wrote:
> I need to create a SIP proxy to be placed in front of a legacy PBX. When a
> phone registers with the proxy, I would like Asterisk to register with the
> PBX behind it. (To tell the PBX to send calls to the proxy and then
On Fri, Apr 06, 2018 at 02:27:31PM -0500, Brent Davidson wrote:
> I have multiple Asterisk instances set up in different locations and would
> like to modify the callerID of inbound calls to identify which instance the
> call is coming from. I knew how to do that with the old sip format, but
> can
On Wed, Mar 28, 2018 at 08:16:26PM -0600, Carlos Chavez wrote:
> ?? I thought I had found and answer to this question by using
> CALLERID(ani) but it seems that only works on versions prior to 12.?? On
> Asterisk 13 setting CALLERID(num) before dialing to an external trunk always
> changes CDR(
On Mon, Mar 19, 2018 at 12:59:47PM -0300, Joshua Colp wrote:
> > To try to reproduce the problem with our SBC, is there a way to tell
> > the asterisk, preferably PJSIP, to directly answer with 180 ringing
> > without prior 100 trying?
>
> The PJSIP channel driver has no option or ability to do th
On Wed, Mar 07, 2018 at 10:08:52PM +, Thomas Peters wrote:
> You did indeed warn me. I've made progress, gotten the dhcp option 242 to
> work, and finally gotten the phone to the point where it asks for a username
> and password. I defined these on the Asterisk server. I entered them on the
On Tue, Mar 06, 2018 at 05:36:04PM +, Thomas Peters wrote:
> But please don't tell me the only way to program up each phone is via
> the craft interface?
>
> Every other phone I've ever used requires a configuration file, which
> has the MAC address of the phone as its name. The Avaya phones m
On Tue, Mar 06, 2018 at 09:05:25AM +0100, Markus Weiler wrote:
> 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?
I'd call it something like "br
On Thu, Mar 01, 2018 at 02:46:31PM +, Thomas Peters wrote:
> Right-- I've seen the Avaya document you cite below. It says "To
> administer DHCP option 242, make a copy of an existing option 176" but
> I don't have any example of option 176 or 242 to copy, and don't know
> what to do to /etc/dhc
On Wed, Feb 28, 2018 at 08:48:38PM +, Thomas Peters wrote:
> I'd like to start configuring my Avaya 9608G phones for use on
> Asterisk / FreePBX / PBX-In-a-Flash. I'm using a variety of other
> phones on my system without major issues.
>
> I've read the discussion back in March, May and August
On Wed, Jan 17, 2018 at 03:16:04PM +0200, Atux Atux wrote:
[asterisk dialplan mysql]
> I would like to ask if there is a way to implement this easily in my
> dialplan, please.
The answer is: yes
If you'd search for "asterisk dialplan mysql", you'get something like
https://www.voip-info.org/wiki/v
On Tue, Jan 16, 2018 at 06:19:30PM +0100, Paul Neuwirth wrote:
> Thank you both. That was (most likely) what I was looking for - but
> still some worries about sending plaintext passwords...
The AMI interface can use a Challenge-Response mechanisme for logins,
if you are this concerned you should
On Mon, Jan 15, 2018 at 10:41:27PM +, David Klaverstyn wrote:
> port1 < Calling Number (len=12) [ Ext: 0 TON: National Number (2) NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> port1 < Presentation: Presentation allowed of
> network provided number (3) '
On Fri, Dec 08, 2017 at 06:11:47PM +0100, Olivier wrote:
> 1. Is this a bug in debian-debug repo ? If positive, should I file a bug
> report ?
>
> 2. Is correct to understand that to get DONT_OPTIMZE, BETTER_BACKTRACE and
> so on options compiled in, I must recompile anyway ?
As far as I know the
Hi,
In my dialplans, I'm currently using PJSIP_AOR to check the status of a contact
before dialling so that I can route the call differently if the endpoint is
offline.
But PJSIP_AOR seems to take about 0.9 seconds to return. If I'm checking 10
endpoints, that can cause a significant delay.
Is
On Fri, Sep 29, 2017 at 12:27:53PM -0300, Joshua Colp wrote:
> > "git checkout -b 13" appears to fix this.
>
> This did not create a branch from 13. This created a branch named "13"
> from the branch you were on, which was most likely master. That is why
> your "git review" is not working as you e
I'm trying to figure out how to commit some code for review. Following:
https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage
Created a ssh alias.
Cloned using: "git clone ssh://asterisk/asterisk"
Set name and email.
Installed the gerrit commit hook: "git review -s"
Try to change to asterisk 13
On Thu, Aug 31, 2017 at 05:54:43PM +, Joseph Smith wrote:
>
> So I am looking for a better way to allow several thousand callers to listen
> to this IVR menu at the same time.
>
An alternative that comes to mind is to have 1 conference with 1 channel
playing MoH in it and then add callers i
On Thu, Aug 17, 2017 at 07:28:00AM +, Grant Bagdasarian wrote:
> Is there an option to give to the Dial command, or another variable to set,
> to make Asterisk copy such information to the B Leg?
> Or do I have to program this out myself?
In chan_sip there are the trustrpid and sendrpid optio
On Fri, Aug 04, 2017 at 03:27:40PM -0400, Jerry Geis wrote:
> Audio packets are running...
>
> 961 16.150421076 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU,
> SSRC=0x6A3E0AF1, Seq=28402, Time=73280
> 962 16.170411284 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU,
> SSRC=0
room to do playback etc)
I guess the only option we have would be to patch the source to
generate a custom AMI event which would be difficult to test given the
it happens perhaps 1-2 times per 10,000-20,000 calls!
Thanks
Daniel
>This is a normal thing that can happen when doing B-channel select
-- Hungup 'DAHDI/0:58-1'
Any ideas why this could be happening?
I believe these messages are coming from chan_dahdi.c and the
"pri_fixup_principle" function.
--
Cheers,
Daniel
--
_
-- Bandwidth and Colocat
On Fri, Jul 07, 2017 at 07:44:26PM +0530, Rahul MathuR wrote:
> Could you please let me know whether the latest Asterisk has a support for
> inbound UPDATE ?
>
> In my case, the carrier is sending an UPDATE to change the codec which is
> replied by 5xx from Asterisk 11.17.1.
Asterisk 13/PJSIP sup
> > Can't find a way to control the dtmf mode on a per session basis with
> > pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any
> > hints on how to do this?
>
> There is no current way, but a community member has recently posted a
> change[1] for review which implements this.
>
On Thu, Jun 29, 2017 at 11:55:51AM -0500, Richard Mudgett wrote:
> > To me this looks like a bug in asterisk. Either asterisk should use the
> > same rtp payloads for telephone-events on both call legs during inital
> > callsetup or asterisk should come to the conclusion there is an
> > incompatibl
Can't find a way to control the dtmf mode on a per session basis with
pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any
hints on how to do this?
--
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-- Bandwidth and Colocation Provided by http://www.api-digit
While trying to use direct_media I'm seeing RTP payload mismatches after
succesful reinvites.
Initial INVITE from endpoint A to asterisk has rfc4733 DMTF
m=audio 35648 RTP/AVP 9 8 111 96
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
>From asterisk to upstream U:
m=audio 14338 RTP/AVP 9 8 111 18
On Mon, Jun 19, 2017 at 11:47:04AM -0400, Tech Support wrote:
> I know that there are probably several solutions to this problem, but
> what I am trying to do is provide some redundancy for my customers CDR data.
> I know that doing simple backups of MySQL is probably the easiest way to go,
> b
On Fri, Jun 16, 2017 at 08:38:59AM +0100, J Montoya or A J Stiles wrote:
> > Whatever has been done, if anything, isn't working effectively. At this
> > point I'd like to see some response from the mailing list admin about any
> > root-cause efforts, AFAIC this is starting to smear the Digium/Aste
On Thu, Jun 15, 2017 at 08:56:29PM -0400, Christopher van de Sande wrote:
> I just setup an anonymous endpoint in pjsip.conf and a context that
> forwards to $EXTEN and when I setup the correct SRV records, it seems
> that any SIP client that's smart enough can just dial my SIP/email
> address. Is
On Thu, Jun 15, 2017 at 12:11:36PM +0200, Benoit Panizzon wrote:
> Or does anyone have an idea over what the asterisk is stumbling?
What if you set the maxdata in asterisk to a value lower than the other
side? e.g. sip.conf:
t38pt_udptl = yes,fec,maxdatagram=400
--
_
On Wed, Jun 14, 2017 at 10:18:19AM -0400, Mike wrote:
> I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP.
> PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has
> its own callerid values and presence. I pass on those calls to PBX_B via
> SI, and I'm trying
On Mon, Jun 12, 2017 at 05:00:31PM +0200, Hans-Peter Jansen wrote:
> is somebody attending, that wants to share his outgoing dial rules of
> extension.conf, like used in typical(?) german pbx setups?
>
> * zero prefix for outside calls
> * zero zero or plus prefix for international calls
> * h
On Mon, Jun 12, 2017 at 09:07:31AM +0200, Olivier wrote:
> Lately, I'm receiving emails asking me to re-enable my list subscription
> due to "excessive bouncing".
>
> What does this exactly mean and why am I receiving this ?
> Beside re-enabling my subscription, what can I do to improve things ?
On Sun, Jun 11, 2017 at 01:16:10PM +0200, Michael Maier wrote:
> Let's go into details:
> I'm starting at the point where asterisk / fax client receives the T.38
> reininvite from the server from the provider 195.185.37.60:5060 for the
> fax client (extension 91):
I'm running Asterisk 11 on my fax
On Fri, Jun 09, 2017 at 11:40:01AM -0300, Joshua Colp wrote:
> What seems to be happening is that the session is being set up and the
> user=phone parameter added. It's only after that the values are updated
> to be Anonymous and the user=phone parameter is left there. Please file
> an issue[1] wit
With pjsip (asterisk 13.14.1) I see the problem that an anonymous from
header gets user=phone appendend to the URI if user_eq_phone=yes is
specified:
On the incoming leg:
From: anonymous
;tag=Q5zBj7BMnvI6Fe6O2866fox3ZHmn-smt
Get transformed to
From: "Anonymous"
;tag=fa3cb748-6af9-485f-8a70-a2b9
On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote:
> extensions.conf:
> [home]
> exten = 102,1,Answer()
> same = n,Wait(1)
If this is copy and paste, then your dialplan is broken (= should be =>)
But to debug, enable logging (core set verbose 5), when needed debugging
(core set debug
On Tue, Jun 06, 2017 at 08:23:33AM -0400, James B. Byrne wrote:
> > The reports are there to tell you something isn't right (like on this
> > mailing list). Disabling them is only hiding the problem, people might
> > be replying with the correct answer to a problem, but the OP might
> > never gets
On Tue, Jun 06, 2017 at 12:40:21AM +0200, Hans-Peter Jansen wrote:
> > Yes, something like if they can't fix the R-URI:
> > exten => X_.,n,Set(TO=${CUT(SIP_HEADER(To),@,1)})
> > exten => X_.,n,Set(TO=${CUT(TO,:,2)})
> > exten => X_.,n,Goto(somewhereelsetopreventloops${TO},1)
>
> Sorry for the sill
On Mon, Jun 05, 2017 at 06:10:50PM +0200, Hans-Peter Jansen wrote:
> ; matches 12345678099, too
> exten => _1234567800,1,Dial(SIP/int)
>
> Except from SIP invite with tcpdump:
>
> INVITE sip:12345678@provider:5060 SIP/2.0
> From: ;tag=as6bc7cbbc
> To:
12345678099 doesn't match _1234567800.
On Mon, Jun 05, 2017 at 01:08:17PM -0400, James B. Byrne wrote:
> This is likely the issue surrounding mailing lists rewriting headers
> and/or modifying messages bodies or simply re-transmitting messages as
> the original sender from an unapproved domain. This was discussed at
> length on the ITEF
Having enabled a strict DMARC setup I noticed everytime I send a message
here I get all these reports of messages which fail DMARC. Since I don't
want people to miss my wise thoughts maybe the maintainers of this list
could look into DKIM signing (or any of the other ways to work around
spf and dma
On Fri, Jun 02, 2017 at 02:36:38PM +0200, Jonas Kellens wrote:
> [Jun 2 14:29:28] == DTLS ECDH initialized (secp256r1), faster PFS enabled
> [Jun 2 14:29:28] ERROR[27360][C-0ae5]: res_rtp_asterisk.c:1441
> ast_rtp_dtls_set_configuration: Specified certificate file
> '/etc/letsencrypt/live/w
On Thu, Jun 01, 2017 at 09:06:25PM +0200, Loic Chabert wrote:
> [gotoexternal]
> exten => _X.,1,Dial(SIP/${EXTEN}@provider)
>
> When my SIP provider return to asterisk a 404 SIP error code, asterisk
> return to phone a 503 error code.
>
> Why 503 error code has been returned, and not the original
On Wed, May 31, 2017 at 01:39:25PM -0700, Steve Edwards wrote:
> >What bugs you about the output format?
>
> It's been a while, but as I recollect, it included the date/timestamp in the
> file name of the 'ring buffer' which meant that each time the host was
> rebooted, dumpcap didn't know the fil
On Wed, May 31, 2017 at 12:36:47PM -0700, Steve Edwards wrote:
> I want to capture all SIP messages.
>
> I have about 30 hosts in about 6 colos.
>
> My first thought was dumpcap, but the output file name format bugs me.
>
> What do you use for long term SIP capture?
What bugs you about the outp
On Wed, Apr 26, 2017 at 06:25:43PM +0200, Daniel Tryba wrote:
> Whoever when a terminating call comes in from the uplink provider, a
> sip request is send to a redirector. The redirector has
> redirect_method=uri_core configured (the only method that works for
> me).
[...]
> The r
> Hello
> I have the following scenario:
> [mynicecontext]
> exten => 2000,1,Dial(SIP/deviceA&SIP/deviceB&SIP/deviceC)
> As expected, by dialing 2000, all three devices will ring. And that's fine.
> However, there are situations where I only want "deviceA" and "deviceB"
to ring. I would like to
Hi,
Is it possible to set up a feature code to move both a caller and callee to a
meeting room?
If yes, what should I be looking at?
Bonus question, is it possible to then automatically dial a 3rd person and
invite them to the meeting room?
The client wants to do this with the push of a couple
> > Anybody got an idea why the last scenario fails to work?
>
> If you turn up core debug (core set debug 2) and ensure it is going to
> the CLI then the bridge_native_rtp module will tell you why exactly it
> can't native bridge. You might also want to do a core show channel on
> both channels t
I'm trying to implement direct_media between multiple peers and an
uplink provider, all of whom have direct_media=yes configures.
For originating calls to the uplink provider direct_media=yes works like
expected. SIP flows through asterisk, rtp doesn't
SIP: enduser <-> SBC <-> asterisk 13 <-> upl
how, to cut it straight, standard distro kernel
worked fine for the deployments I was involved in.
Cheers,
Daniel
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com
Kamailio World Conference - May 8
> Hello
> as you can read in my original post "moh reload" and "module reload
> res_musiconhold.so" does nothing.
> Only at restart the new files are available.
> Is this a bug ?? How can I get more debugging for this problem ??
Just spotted that you are using Asterisk 1.8.32.3.
The bug I'm think
> Hello
> as you can read in my original post "moh reload" and "module reload
> res_musiconhold.so" does nothing.
> Only at restart the new files are available.
> Is this a bug ?? How can I get more debugging for this problem ??
I think there is currently a bug with MOH. For now, if you add a fil
Ø Set the ATTENDED_TRANSFER_COMPLETE_SOUND channel variable to the sound
file to play on a transfer.
>Does that have to be set in the Dial handler in order to get set on the
>dialled channel or can it be inheritted?
Never mind. Tested it. Working great! Thanks.
--
__
Ø Set the ATTENDED_TRANSFER_COMPLETE_SOUND channel variable to the sound
file to play on a transfer.
Does that have to be set in the Dial handler in order to get set on the dialled
channel or can it be inheritted?
--
_
-- Bandw
Hi,
During an attended transfer using the SIP phone feature buttons, I'm getting a
few complaints from recipients that they can't tell when the call they are
receiving has been transferred.
Is there any way (even if it's complicated) to generate a beep tone to the
recipient of the transferred c
The bug tracker includes several issues relating to Path (RFC 3327)
support. It is not clear which version actually included the patch and
which versions are working.
Could anybody update these issues in Jira with a brief comment about the
supported versions?
https://issues.asterisk.org/jira/b
Hi,
I recently had to reinstall Asterisk and FreePBX. asteirsk 11.20 and FreePBX 12.
This is running on Centos 6.7 32 bit.
When I use amportal start
It comes up with the errors below
Error in argument 1, char 2: option not found r
/usr/local/sbin/amportal: line 49: Usage:: command not found
WA
Ish,
I use the same version of Asterisk on CentOS 6.7. I wonder the same thing.
Hopefully we will find this out.
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Read README, check the requirements and get the google speech api key.
Then add a custom destination in FreePBX and edit your extensions_custom.conf.
> Am 22.02.2016 um 21:03 schrieb Daniel Chavez :
>
> Ok. Where I am now is, I copied the speech-recog.agi to the agi-bin directory.
>
Ok. Where I am now is, I copied the speech-recog.agi to the agi-bin directory.
Where do I go from here?
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I use FreePBX as well. There is no module for speech recognition. You have too
create a custom destination.
> Am 22.02.2016 um 20:53 schrieb Daniel Chavez :
>
> Thanks, this looks promising. I was wondering if there's an easier way to get
> this to work inside FreePBX?
&g
Thanks, this looks promising. I was wondering if there's an easier way to get
this to work inside FreePBX?
I have all of the dependencies installed for it, but now I want to know if
there's a mod I can use in FreePBX to get it setup?
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Daniel,
try this http://zaf.github.io/asterisk-speech-recog/.
I have tested it myself, it works very well.
Daniel
> Am 22.02.2016 um 19:34 schrieb Daniel Chavez :
>
> Thanks for the link.
> Are there no free alternatives for speech
Thanks for the link.
Are there no free alternatives for speech recognition?
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Hello list,
I was wondering if it were possible for asterisk to do a voice recognition type
IVR?
Like you know how most banks and stuff do, where they ask you to say your
selection or key it in?
If this is possible, how can I set this up? I'm using FreePBX 2.11 on Linux,
CentOS 6.7 32-bit, aster
down and
reboot server then asterisk doesn't come up.
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Daniel
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Hello,
My name is Daniel.
Very strange. Chan_Sip.so should be there. Maybe the cat didn't have enough cat
food? *smile*
I am a System Administrator, and have done plenty of Asterisk install's, so am
use to errors and such.
If you'd like, I can remote SSH in to troubleshoot your in
Bryant,
that sounds interesting. I am searching for a script which monitors and updates
the ip address. Does this your script? Can you share your script with us?
Thanks
Daniel
> Am 26.01.2016 um 16:39 schrieb Bryant Zimmerman :
>
> Daniel
>
> Thank you for your response. I
Bryant,
I have the same problem with dynamic public IPs and PJSIP. What is your idea to
solve the problem?
My suggestion would be to write a script that monitors the change,
pjsip.transports.conf updated and Asterisk restarts?
Daniel
> Am 26.01.2016 um 14:21 schrieb Joshua Colp :
>
&g
On top of the page: "Call pickup support added in Asterisk 11“
I think that is the problem. I do not know a solution for 1.8, but maybe
someone other.
> Am 29.12.2015 um 10:20 schrieb Luca Bertoncello :
>
> Daniel Heckl schrieb:
>
>> You are searching for „Call Pickup
You are searching for „Call Pickup“. It is implemented in Asterisk by default.
https://wiki.asterisk.org/wiki/display/AST/Call+Pickup
<https://wiki.asterisk.org/wiki/display/AST/Call+Pickup>
Take a look under section „Configuration Options“.
Daniel
> Am 29.12.2015 um 07:53 sch
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