Hi Dan,
Your best bet for looking at RTP media specifics is the standards that
define RTP.
Wikipedia has some really good resources on RTP and a list of the
various RFC standards that relate:
https://en.wikipedia.org/wiki/Real-time_Transport_Protocol
On 8/28/23 11:16, Dan Cropp wrote:
I am working on a project that uses Asterisk ARI ExternalMedia request to
stream the RTP audio from Asterisk to an UDP/RTP receiver project.
Using slin16 format.
1) I believe I am seeing is a 12-byte header followed by 640 bytes of data. Is
this correct?
2) Is there some place I can find a
> On 18 Aug 2023, at 04:50, Federico wrote:
>
> I am looking for a decent provider of SIP Trunks but it has to pass the Stir
> Shaken token to the next carrier. Does anybody know about any? Sipstation
> from Sangoma, does not support Stir Shaken. ( Case #01466843 /
> 001300G8PLG / MAIN /
Thanks. I have accounts with both companies and both have issues.
From: asterisk-users On Behalf Of
Dovid Bender
Sent: Friday, August 18, 2023 11:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about Sip Trunks who support Stir Shaken
Telnyx, 382com, voicetel and as others mentioned BandWidth. I have contacts
at 382 and voicetel if you want an intro.
On Thu, Aug 17, 2023 at 11:50 PM Federico
wrote:
> I am looking for a decent provider of SIP Trunks but it has to pass the
> Stir Shaken token to the next carrier. Does anybody
t; *Sent:* Thursday, August 17, 2023 11:49 PM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' <
> asterisk-users@lists.digium.com>
> *Subject:* [asterisk-users] Question about Sip Trunks who support Stir
> Shaken
>
>
>
> I am looking for a decent pr
Check out Twilio
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On
Behalf Of Federico
Sent: Thursday, August 17, 2023 11:49 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Question about Sip Trunks who support Stir Shaken
I am looking for a decent provider of SIP Trunks but it has to pass the Stir
Shaken token to the next carrier. Does anybody know about any? Sipstation
from Sangoma, does not support Stir Shaken. ( Case #01466843 /
001300G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ])
Although it's
On 5/29/2023 4:12 PM, Steve Matzura wrote:
On 5/28/2023 2:27 PM, Naveen Albert wrote:
However, you can also pass audio without supervising (early media).
You typically need to Progress() first to allow this, e.g. for SIP,
or audio won't pass at all.
...
If you want it to ring once and do
On 5/28/2023 2:27 PM, Naveen Albert wrote:
However, you can also pass audio without supervising (early media).
You typically need to Progress() first to allow this, e.g. for SIP, or
audio won't pass at all.
...
If you want it to ring once and do something else, you could simply do:
exten
On 5/28/23 14:20, Steve Matzura wrote:
Who controls how many times an incoming call from an external (DID)
provider will ring before Asterisk picks up the call and handles it
internally
Asterisk and this is defined with your timeout on the dial command, mine
is 26 seconds so around 5 rings.
Who controls how many times an incoming call from an external (DID)
provider will ring before Asterisk picks up the call and handles it
internally--the provider or Asterisk? If it's the DID provider, I'll
work on that with them; if it's Asterisk, I didn't find anything
anywhere that looks like
Please disregard, I figured out what I was doing wrong.
Dan
From: Dan Cropp
Sent: Friday, January 20, 2023 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Question on ARI externalMedia
A couple years ago, I know I had ARI externalMedia working. Trying to figure
A couple years ago, I know I had ARI externalMedia working. Trying to figure
out what I'm doing wrong today.
https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
My ari.conf
[general]
enabled = yes
pretty = no
allowed_origins = *
[MyApp]
type = user
read_only = no
On Mon, Sep 5, 2022 at 9:16 AM Mark Murawski
wrote:
> On 8/4/22 20:32, Jerry Geis wrote:
> > I am running Asterisk 13.30.0
> > 40 core CPU (VM) VMware.
> > CentOS 7
> > 32 G ram
> > 10G vmx network
> >
> > Should be plenty of room for anything...
> >
> > Yes asterisk is running 270% CPU...
> >
On 8/4/22 20:32, Jerry Geis wrote:
I am running Asterisk 13.30.0
40 core CPU (VM) VMware.
CentOS 7
32 G ram
10G vmx network
Should be plenty of room for anything...
Yes asterisk is running 270% CPU...
Is it not taking advantage of the 40 cores ?
I am bring around 300 SIP endpoints in a muted
On Thu, Sep 1, 2022 at 1:32 PM Dan Cropp wrote:
> Using AMI, we send an Originate with EarlyMedia: true setting
>
>
>
> If the other end sends a 183, Asterisk
>
> When the 183 is received, Asterisk indicates the ChannelState: 6 and
> ChannelStateDesc: Up values.
>
> All is fine up to this point.
Using AMI, we send an Originate with EarlyMedia: true setting
If the other end sends a 183, Asterisk
When the 183 is received, Asterisk indicates the ChannelState: 6 and
ChannelStateDesc: Up values.
All is fine up to this point.
It may take the caller several seconds before the called party
Hi,
Am Donnerstag, dem 04.08.2022 um 20:32 -0400 schrieb Jerry Geis:
> I am running Asterisk 13.30.0
> 40 core CPU (VM) VMware.
> CentOS 7
> 32 G ram
> 10G vmx network
>
> Should be plenty of room for anything...
>
> Yes asterisk is running 270% CPU...
> Is it not taking advantage of the 40
Doesn’t that mean, effectively that you are using the equivalent of 100% of 2.7
CPUs?
--Don
From: asterisk-users On Behalf Of
Jerry Geis
Sent: Thursday, August 4, 2022 7:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question
I am running Asterisk 13.30.0
40 core CPU (VM) VMware.
CentOS 7
32 G ram
10G vmx network
Should be plenty of room for anything...
Yes asterisk is running 270% CPU...
Is it not taking advantage of the 40 cores ?
I am bring around 300 SIP endpoints in a muted audio conference (so one
way) and this
On Wed, Jul 27, 2022 at 11:02 AM Dan Cropp wrote:
> Looking at the Asterisk wiki
>
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Geolocation+Implementation
>
Just FYI, I'm in the process of clarifying and adding more info. Should be
done Friday.
>
>
> I see the dial plan support the
Looking at the Asterisk wiki
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Geolocation+Implementation
I see the dial plan support the GeolocProfileCreate and there is support for
GEOLOC_PROFILE settings to be set on the dial plan.
We currently use AMI Originate support. We may have
On Tue, Oct 12, 2021 at 2:54 PM Dan Cropp wrote:
> We tell asterisk to use the slin format for ExternalMedia. However, the
> unicast channel is selecting ulaw formatand the RTP data is indicating it’s
> ulaw format.
>
>
>
> Anyone know why ulaw format would be on chosen?
>
What do your ARI
We tell asterisk to use the slin format for ExternalMedia. However, the
unicast channel is selecting ulaw formatand the RTP data is indicating it's
ulaw format.
Anyone know why ulaw format would be on chosen?
[10/12 16:13:39.396] DEBUG[1665] http.c: HTTP Request URI is
Thanks Joshua
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Friday, February 14, 2020 1:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question on pjsip.conf and aors
On Fri, Feb 14, 2020 at 3:04 PM Dan Cropp
mailto:d...@amtelco.com
On Fri, Feb 14, 2020 at 3:04 PM Dan Cropp wrote:
> I have the following configuration…
>
>
>
> [aor3]
>
> type = aor
>
> max_contacts = 1
>
> remove_existing = yes
>
>
>
> [auth3]
>
> type = auth
>
> username = 1004
>
> password = SuperSecretProbation
>
>
>
> [1004]
>
> type = endpoint
>
>
I have the following configuration...
[aor3]
type = aor
max_contacts = 1
remove_existing = yes
[auth3]
type = auth
username = 1004
password = SuperSecretProbation
[1004]
type = endpoint
context = IS
transport = transport1
auth = auth3
aors = aor3
accountcode = 3
dtmf_mode = rfc4733
Hello,
Reading this old thread, isn't there also an error in [1] as It also
mentions a tlscafile setting.
Cheers
[1]
https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone
Le ven. 7 déc. 2018 à 16:41, Kevin Harwell a écrit :
> On Fri, Dec 7, 2018 at 9:11 AM Dan
like they have been modifying settings on their end
without informing us.
Dan
-Original Message-
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Monday, July 15, 2019 1:31 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Question on calculating PJSIP md5
On Fri, Jul 12, 2019, at 5:10 PM, Dan Cropp wrote:
>
> Just tracked down the code for the chan_sip MD5 REGISTER and have been
> able to verify that chan_sip is calculating the HA1 same as I am
> calculating the md5_cred for PJSIP
>
> 3016:a...@xyz.com:3016
>
>
> Both chan_sip and PJSIP
terisk-users
mailto:asterisk-users-boun...@lists.digium.com>>
On Behalf Of Dan Cropp
Sent: Wednesday, July 10, 2019 10:48 AM
To: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>
Subject: [asterisk-users] Question on calculating the md5_sum
Using chan_sip, we ar
-users@lists.digium.com
Subject: [asterisk-users] Question on calculating the md5_sum
Using chan_sip, we are able to register with an NEC switch. When I try to
REGISTER with PJSIP, the authentication is being rejected. Traces show it's
using md5 authentication.
The packets looks almost identical
Using chan_sip, we are able to register with an NEC switch. When I try to
REGISTER with PJSIP, the authentication is being rejected. Traces show it's
using md5 authentication.
The packets looks almost identical. The one area that I suspect is causing the
problem is the md5_cred for my
Mike,
Are you using port mirroring or is VoipMonitor running on the same box? If
the latter I would run tcpdump and compare what VM says it has to what you
see in your wireshark dump. If you are sniffing via port mirroring your
switch maybe dropping packets (we had that when we tried to mirror
Hi all,
I'm not sure this is the place to ask, but here goes...
I'm using voipmonitor to gather call statistics such as packet counts, average
jitter, etc.
Eventually, I want to use those stats to detect and alert on poor call quality.
However, I'm finding that the packet counts for each leg
On Fri, Dec 7, 2018 at 9:11 AM Dan Cropp wrote:
> In the asterisk wiki instructions for Configuring Asterisk for WebRTC
> clients…
>
>
>
>
> https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients
>
>
>
> “To communicate with websocket clients, Asterisk uses its
In the asterisk wiki instructions for Configuring Asterisk for WebRTC clients...
https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients
"To communicate with websocket clients, Asterisk uses its built-in HTTP daemon.
Configure /etc/asterisk/http.conf as follows:
On Fri, Apr 27, 2018, at 11:13 AM, Olivier wrote:
> Hello,
>
> I don't know if this list is the best place to ask such question but here
> it is, anyway.
>
> In page [1], I can read in PJSIP's endpoint section configuration reference:
> identify_by username,location Way(s) for
Hello,
I don't know if this list is the best place to ask such question but here
it is, anyway.
In page [1], I can read in PJSIP's endpoint section configuration reference:
identify_by username,location Way(s) for Endpoint to be
identified
Then clicking over identify_by text, you
On 1 July 2015 at 04:03, Jerry Geis ge...@pagestation.com wrote:
I see in my log file this:
Jun 30 21:44:26] NOTICE[42192][C-02f3] chan_sip.c: Call from '' (
5.189.144.120:5076) to extension '011972592675431' rejected because
extension not found in context 'default'.
which is great its
I see in my log file this:
Jun 30 21:44:26] NOTICE[42192][C-02f3] chan_sip.c: Call from '' (
5.189.144.120:5076) to extension '011972592675431' rejected because
extension not found in context 'default'.
which is great its rejected - however
in my sip.conf file I have
deny=0.0.0.0
Hello,
on previous versions of asterisk, extension h and H make us know who
ended a call (caller or callee). In the last * versions, seems that only
h extension is used, as stated here
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
In the last versions, how do we know which
Starting with Asterisk 13.1 we are seeing this WARNING
messages a lot in our logs and console:
WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type
frames with SIP write)
We found that line in function sip_write inside chan_sip.c.
In our previous
thank you, we are using the same configuration files in 13, same setup, just
different asterisk version. we just dont see the msgs in the console/logs, it
is the same exact voice traffic on both asterisk versions
is that something that you set on/off? if that is the case how can it be done?
On Fri, Feb 6, 2015 at 5:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
On 6 February 2015 at 07:54, Olli Heiskanen ohjelmistoarkkite...@gmail.com
wrote:
Hello,
Got a question regarding custom announcements in Asterisk.
My goal is to allow my users record their own queue announcements
On 6 February 2015 at 07:54, Olli Heiskanen ohjelmistoarkkite...@gmail.com
wrote:
Hello,
Got a question regarding custom announcements in Asterisk.
My goal is to allow my users record their own queue announcements and
choose which announcements they want to use in each queue. I have
Oops, quite right, how typoful of me!
Thanks for the excellent points, I'll look into gluster and puppet and see
may way onwards from there.
cheers,
Olli
2015-02-06 12:32 GMT+02:00 Gareth Blades mailinglist+aster...@dns99.co.uk:
On 06/02/15 07:54, Olli Heiskanen wrote:
My goal is to allow
On 06/02/15 07:54, Olli Heiskanen wrote:
My goal is to allow my users record their own queue announcements and
choose which announcements they want to use in each queue. I have
several Asterisk servers and a Kamailio server which dispatches call
traffic between the Asterisks. Question is, is
Hello,
Got a question regarding custom announcements in Asterisk.
My goal is to allow my users record their own queue announcements and
choose which announcements they want to use in each queue. I have several
Asterisk servers and a Kamailio server which dispatches call traffic
between the
I have a machine with three IP addresses.
NIC eth0
NIC eth1
and a virtual address on ETH1
All my devices work normally communicating to the virtual address on eth1.
My question is just for mulitcast.
The end device has an option for allowed source so I put in the virtual
address
from my server.
Hi,
upto asterisk 1.8 you used to get this error if there were more than 1
m= line in an invite... Asterisk was just telling you it was declining
the second. I belive from 10.0 onwards asterisk now just replies back
with port 0 to the stream it isn't interested in...
You can ignore it - if its
I get tons of these messages
chan_sip.c:10088 process_sdp: Declining non-primary audio stream:
audio 30660 RTP/AVP 4 101 13
What does it mean and does it show a problem like one-way audio?
Thanks for your help.
--
_
-- Bandwidth
It supposed to be like this Dial(SIP/${EXTEN}#ip.add.re.ss)
Regards
On Fri, Aug 15, 2014 at 6:20 AM, CDR vene...@gmail.com wrote:
In channel PJSIP I use this format
Dial(PJSIP/endpoint/sip:${EXTEN}@ip.add.re.ss)
what would be the equivalent of this format in old SIP?
I tried
In channel PJSIP I use this format
Dial(PJSIP/endpoint/sip:${EXTEN}@ip.add.re.ss)
what would be the equivalent of this format in old SIP?
I tried
Dial(SIP/peer/${EXTEN}@ip.add.re.ss)
but it does not work. I just cannot embed the IP address in the peer's
definition, but I need to use some other
I found that PJSIP allows only one asterisk per box. I tried to start
several asterisks with the parameter -C and PJSIP only worked on the
first process. In the other processes, the command pjsip reload was
absent. Each pjsip transport in the second and subsequent processes
was bound to a
On Mon, Jul 21, 2014 at 7:00 PM, CDR vene...@gmail.com wrote:
I found that PJSIP allows only one asterisk per box. I tried to start
several asterisks with the parameter -C and PJSIP only worked on the
first process. In the other processes, the command pjsip reload was
absent. Each pjsip
Okay - maybe I'm just suffering from a moment of horrible ADD - but, I'm a
little lost.
I see that Asterisk 12 has a nice REST API - very nice - something I can
use. However, and this is gonna sound dumb - but all the CLI commands are
different now. What did I miss?
Can anyone, please, anyone
:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question about Asterisk 12
Okay - maybe I'm just suffering from a moment of horrible ADD - but, I'm
a little lost.
I see that Asterisk 12 has a nice REST API - very nice - something I can
use. However
I looked on http://www.voip-info.org - maybe I missed it?
The Digium/Asterisk site - I see all sorts of cool things about the REST
API, but CLI - maybe I missed it!!?? - again, I could be looking in the
wrong place?
https://wiki.asterisk.org/wiki/display/AST/Home
To my knowledge the
CDR wrote:
I am trying to identify the module (*.so) that contains the Asterisk
Management Interface, so as to set noload=XXX.so in modules.conf. Any
idea?
There is no module, it's provided as core functionality. Disabling it
can be done in manager.conf
--
Joshua Colp
Digium, Inc. | Senior
I am trying to identify the module (*.so) that contains the Asterisk
Management Interface, so as to set noload=XXX.so in modules.conf. Any
idea?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Jonas Kellens wrote:
Hello,
short question : does Asterisk reserve RTP ports for every IP-phone that
is being called ?
It uses 2 ports per channel under normal circumstances, 1 for RTP and 1
for RTCP.
If for instance an incoming call makes 10 IP-phones ring, does this mean
that Asterisk
Hello,
short question : does Asterisk reserve RTP ports for every IP-phone that
is being called ?
If for instance an incoming call makes 10 IP-phones ring, does this mean
that Asterisk preserves 10 x 2 RTP ports for audio ?
I guess Asterisk sends in the SIP INVITE an SDP body with an RTP
On 10/29/2013 05:14 PM, Joshua Colp wrote:
Jonas Kellens wrote:
Hello,
short question : does Asterisk reserve RTP ports for every IP-phone that
is being called ?
It uses 2 ports per channel under normal circumstances, 1 for RTP and
1 for RTCP.
If for instance an incoming call makes 10
Jonas Kellens wrote:
So if I understand correct, you don't need to look at the amount of
concurrent calls to calculate the RTP range in rtp.conf, you need to
look at the amount of INVITES that are being send at one moment ?
The number of concurrent channels in existence which are using RTP.
Hi
A small question on Asterisk Manager. I use Perl Script for start a call:
my $response = $astman-sendcommand( Action = 'Originate',
Channel =
'SIP/ASTERISK/$Extension',
Exten = '200',
On Thu, Jul 4, 2013 at 12:24 AM, James B. Byrne byrn...@harte-lyne.cawrote:
I have this code in a dial plan:
exten = _417XX,n,GotoIf($[${CALLERID(num)}
SIP/41799]?notfromlocal)
exten = _417XX,n,GotoIf($[${CALLERID(num)}
SIP/41700]?notfromlocal)
The value of ${CALLERID(num)} appears to
On Thu, July 4, 2013 02:14, Satish Barot wrote:
${CALLERID(num)} should give you only number and not technology i.e.
41712.
Give this a shot,
exten = _417XX,n,Noop(CALLERIDNUM=${CALLERID(num)})
exten = _417XX,n,GotoIf($[$[${CALLERID(num)} 41799] |
$[${CALLERID(num)}
I need to block any audio before there is a connect, in SIP. How do I tell
the DIAL application to behave like that?
Yours
Philip
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
Is it me or Google just blocked Asterisk's chan_motif? I get violation of
terms of service audio message whenever I send a call.
Philip
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
CDR wrote:
Is it me or Google just blocked Asterisk's chan_motif? I get violation
of terms of service audio message whenever I send a call.
Works fine here. Their automated security system probably determined
your usage behavior was not consistent with normal usage and terminated
your
thanks asghar for your help and support and thanks ishfaq
2013/5/9 Asghar Mohammad asghar...@gmail.com
hi,
asterisk insert cdr when call is hangup and last dial statment,
i dont understatnd why you are using 2 dial statment on same extenstion?
if you you want dial to both extensions you
hello list,
i need your help about cdr ,i have installed the module cdr in my asterisk
1.4 .
for the inbound calls when i call my sip exten like below :
exten = 506,1,Dial(SIP/223, 10)
exten = 506,n,Dial(SIP/276, 10)
in CDR i have just one line with SIP /276 the last line but there is
no
On Thu, 2013-05-09 at 08:46 +, Salaheddine Elharit wrote:
hello list,
i need your help about cdr ,i have installed the module cdr in my
asterisk 1.4 .
for the inbound calls when i call my sip exten like below :
exten = 506,1,Dial(SIP/223, 10)
exten = 506,n,Dial(SIP/276, 10)
thanks i verify but i don't understanding if can someone give me an example
best regards
2013/5/9 Ishfaq Malik i...@pack-net.co.uk
On Thu, 2013-05-09 at 08:46 +, Salaheddine Elharit wrote:
hello list,
i need your help about cdr ,i have installed the module cdr in my
asterisk
hi,
asterisk insert cdr when call is hangup and last dial statment,
i dont understatnd why you are using 2 dial statment on same extenstion?
if you you want dial to both extensions you can use
506,1,Dial(SIP/223SIP/276) if you want dial both same time or if you want
to do failover the check Dial
On Mon, Mar 25, 2013 at 03:15:24PM +, Salaheddine Elharit wrote:
thank you so much
fo the upgrade from zptel to dahdi, if there is any possibility to upgrade
to dahdi without impacting my installation of asterisk and other
application already installed in my server.
if you can tell
On Mon, Mar 25, 2013 at 10:44:47AM +, Salaheddine Elharit wrote:
hello list,
i have a question related to zapata.conf,if i do any change in zapata.conf
i must restart asterisk or just i restart zapata ,and how to do .
“service zaptel restart” or there is any other command
ok thanks for your help and support i really appreciated
2013/3/26 Tzafrir Cohen tzafrir.co...@xorcom.com
On Mon, Mar 25, 2013 at 10:44:47AM +, Salaheddine Elharit wrote:
hello list,
i have a question related to zapata.conf,if i do any change in
zapata.conf
i must restart asterisk
hello list,
i have a question related to zapata.conf,if i do any change in zapata.conf
i must restart asterisk or just i restart zapata ,and how to do .
“service zaptel restart” or there is any other command
Thanks and regards
--
it depends a little bit on the driver and asterisk version...
the safest way to become changes applied is to stop asterisk, reload the
driver and than start asterisk again.
regards,
yves
btw..:
zaptel ist outdated... you should definitely upgrade using dahdi drivers...
Am 25.03.2013 11:44,
i use asterisk 1.4, how i can do to reload dirver
1.service asterisk stop
2 CLI reload chan_zap.so
3 service asterisk start
that is right or i miss something ?
2013/3/25 Yves A. yves...@gmx.de
it depends a little bit on the driver and asterisk version...
the safest way to become changes
- Non-Commercial Discussion
Subject: Re: [asterisk-users] question about zapata.conf
i use asterisk 1.4, how i can do to reload dirver
1.service asterisk stop
2 CLI reload chan_zap.so
3 service asterisk start
that is right or i miss something ?
2013/3/25 Yves A. yves...@gmx.de
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] question about zapata.conf
i use asterisk 1.4, how i can do to reload dirver
1.service asterisk stop
2 CLI reload chan_zap.so
3 service asterisk start
that is right or i miss something ?
2013/3/25
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] question about zapata.conf
i use asterisk 1.4, how i can do to reload dirver
1.service asterisk stop
2 CLI reload chan_zap.so
3 service asterisk start
that is right or i miss something ?
2013/3/25
: [asterisk-users] question about zapata.conf
i use asterisk 1.4, how i can do to reload dirver
1.service asterisk stop
2 CLI reload chan_zap.so
3 service asterisk start
that is right or i miss something ?
2013/3/25 Yves A. yves...@gmx.de
it depends a little bit on the driver
The Dial events are created by app_dial. So long as you are using
app_dial to create your outbound channel, you should have that event.
Channel technology shouldn't matter.
I am using the same AMI method to start both calls.
Action: Originate
Channel: DAHDI/18/XX
or
Action: Originate
I just put a break at dial_exec_full (app/app_dial.c for Asterisk 11.0.2)
did my AMI call
Action: Originate
Async: yes
Channel: SIP/testsystem/XXX
(calls from my machine over SIP trunk to another 11.0.2 box that has
a PRI card to make a call out to my cell)
and did not get a break.
Why is
When I am monitoring the AMI I see the following event
for a call I just made over a SIP trunk.
Event: Newchannel
Privilege: call,all
Channel: SIP/testmachine-000d
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Exten:
Context: testmachine
Uniqueid:
Have you tried and looked up all events generated when you place the call?
some of them are bound to have the variable callerid set
On 24 January 2013 16:46, Jerry Geis ge...@pagestation.com wrote:
When I am monitoring the AMI I see the following event
for a call I just made over a SIP
.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada
Sent: Thursday, January 24, 2013 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] question on SIP trunk and AMI to place call
Have you tried and looked up all events generated when you place the call?
some of them are bound to have the variable callerid set
yes I have looked at all of them, CallerID is not set to the number I am
calling.
Jerry
Not the greatest solution, but since you are most likely using a script for the
AMI process, you could do an
Asterisk --rx core show channels verbose|grep SIP/testmachine-000d
And get the dialed number from that.
Actually you could issue the AMI command core show channels verbose.
there
On 01/24/2013 10:46 AM, Jerry Geis wrote:
When I am monitoring the AMI I see the following event
for a call I just made over a SIP trunk.
Event: Newchannel
Privilege: call,all
Channel: SIP/testmachine-000d
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
You probably want the Dial event. It is raised both at the beginning of
the Dial, as well as when the Dial completes.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial
Note that the Channel: field will contain the name initiating the Dial,
the Destination: field will
- Non-Commercial Discussion
Subject: Re: [asterisk-users] question on SIP trunk and AMI to place call
You probably want the Dial event. It is raised both at the beginning of
the Dial, as well as when the Dial completes.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11
On 01/24/2013 01:13 PM, Jerry Geis wrote:
You probably want the Dial event. It is raised both at the beginning of
the Dial, as well as when the Dial completes.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial
Note that the Channel: field will contain the name
Hello,
I have a question about directmedia or canreinvite, I have experience that
whatever I set directmedia=yes or no. After I run sip show settings.
all settings looks the same.
My question is how I could make sure from sip show settings that my
directmedia configuration is applied.
Thanks
I like the example of using that to add somebody to the conference, but
what I don't see is how the dialplan can know what conference the menu
item was called from. I was hoping that some variable might have been set,
but don't see it in the sources. Is the idea to do that outside of the
call to
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