Hello.
Is there an analog option outofcall_message_context for pjsip?
or: how to determine that the call is an outbound text message?
Dmitriy Serov.
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I can assure you that asterisk is crashing, as when I try to reconnect I see
it reloading again.
Could be that something is deleting the core ! is there a way to find the
path to where the core files are stored?
My system is Lubuntu , Linux #41 SMP PREEMPT Tue Nov 11 16:35:58 CST 2014
armv7l
In a word, no.
PRI service providers will generally only allow the caller ID to be set to
one of the numbers in the range that you have for inbound with them.
On 18 Mar 2015 11:30, Rizwan H Qureshi rizwanhas...@gmail.com wrote:
Hi All,
I have to forward incoming call on PRI back out to PRI but
On Wednesday 18 Mar 2015, Rizwan H Qureshi wrote:
Hi All,
I have to forward incoming call on PRI back out to PRI but I need the
original Callerid to passthrough. Is it possible with DAHDI PRI cards
without involving the service provider?
Thanks
It depends who your service provider is!
Any
If you take a look at the safe_asterisk shell script, usually located at
/usr/sbin/safe_asterisk (for CentOS at least), you'll be able to find where
the core files are located. If it's not located there, then you'll need to
look at the Asterisk init script for the scripts location. I hope this
On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome jl...@me.com wrote:
Hey guys,
have issues with reinvite, no matter what endpoint is calling asterisk
always tries switch simple_bridge to native_rtp
Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge
technology to
Hi All,
I have to forward incoming call on PRI back out to PRI but I need the
original Callerid to passthrough. Is it possible with DAHDI PRI cards
without involving the service provider?
Thanks
--
Best Ragards
Rizwan H Qureshi
V: +971 (0) 528272154
linkedin.com/in/rhqureshi
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Hi Guys
I have a 4 port PRI card that I need to setup each port in their own
group.
In chan_dahdi.conf I have the following which works for one port
How do I add the rest of the ports in their own groups so that I can have
different signaling on each?
[channels]
language=en
On Wed, Mar 18, 2015 at 4:43 AM, Dmitriy Serov serov@gmail.com wrote:
Hello.
Is there an analog option outofcall_message_context for pjsip?
or: how to determine that the call is an outbound text message?
The 'message_context' endpoint option [1] should provide what you're
looking for.
I have a 4 port PRI card that I need to setup each port in their own group.
In chan_dahdi.conf I have the following which works for one port
How do I add the rest of the ports in their own groups so that I can have different signaling
on each?
[channels]
language=en
switchtype=euroisdn
Thanks AJ and David,
We were actually using GSM gateways by setting busy forward number on the
SIMs and just giving busy signal on every incoming call, telco took care of
the forwarding and the line was free within seconds. Now we need to scale
up the setup but GSM gateways a very very expensive
4 Port PRI sangoma a104
From: jg [mailto:webaccounts...@jgoettgens.de]
Sent: Wednesday, March 18, 2015 2:09 PM
To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 4 Port PRI
I have a 4 port PRI card that I need to setup each port in their
Hey guys,
have issues with reinvite, no matter what endpoint is calling asterisk always
tries switch simple_bridge to native_rtp
Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge
technology to native_rtp
in endpoints table “direct_media” sets to “no” on all endpoints
When parsing the config file, all the current settings are applied when the
'channel = ' directive is encountered. So something like this will make
the three remaining groups and set signalling on ports 1 3 as pri_cpe and
ports 2 4 as pri_net.
; setting specific to Group 2
group=2
On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome jl...@me.com wrote:
Well, it breaks audio for all NAT endpoints, how can I fix this?
Local (packet to packet) bridging should not do that. Remote (direct
media) can do that.
Can you confirm - by looking at a verbose level 4 log - how Asterisk
is
Well, it breaks audio for all NAT endpoints, how can I fix this?
On 18 Mar 2015, at 15:48, Matthew Jordan mjor...@digium.com wrote:
On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome jl...@me.com
mailto:jl...@me.com wrote:
Hey guys,
have issues with reinvite, no matter what endpoint is
Attached is my safe_asterisk script, it is moving the core to some dumpdrop
directory that does not seem to exist.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support
Sent: Wednesday, March 18, 2015
My, how embarrassing. I of course meant that as a personal message to
Don. But if anyone else knows the answer, I'm interested! lol
Cheers,
j
On 03/18/2015 10:02 AM, Jeff LaCoursiere wrote:
Hey Don,
How are you? I may be heading your way in the next month or so. Have
to meet with a
Hey Don,
How are you? I may be heading your way in the next month or so. Have to
meet with a guy in Eden Prairie, and stop off at my
brother/sisterm-in-law's as well.
Got a question for you - with TBCT, who pays for the call once it is
transferred? Still me as the owner of the trunk?
This depends on what you mean by “not involving the service provider.”
If you are literally forwarding calls that come in on the PRI back out on the
PRI, the most efficient way is with Two B-Channel Transfer (TBCT). Check it out
in the wiki.
You need to make sure your carrier supports
The way it's expected to work:
Inbound call to our toll-free number, we pay for the call TO US until
terminated
Inbound call to our local number, caller pays for the call TO US until
terminated (if long distance charges apply for caller)
In either case, we pay for the outbound call if
Hi list , this is a bug?
ERROR[361]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection
available when trying to connect client
regardss
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http://gnuforever.homelinux.com
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Kindly guide with debugging TLS issue in asterisk 11.16. Compiled from
source and works all ok !
Added the below to sip.conf
tlsenable=yes
tlsbindaddr=0.0.0.0:5061
However asterisk doesn't even listen to port 5061
sudo netstat -anp
Kindly guide
Thanks
Best,
Chirag A.
--
2015-03-18 10:52 GMT-06:00 ricky gutierrez xserverli...@gmail.com:
Hi list , this is a bug?
ERROR[361]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection
available when trying to connect client
regardss
Hi , I'm trying to apply this patch from the source asterisk
asterisk-11.16.0
2015-03-18 11:13 GMT-06:00 ricky gutierrez xserverli...@gmail.com:
Hi , I'm trying to apply this patch from the source asterisk
asterisk-11.16.0 and when I apply it shows me this message
asterisk-11.16.0]#patch -p0 refs
patch: Only garbage was found in the patch input.
is the
Hello!
As I see there is dsp_drop_silence switch in confbridge.
Could you tell me how asterisk detects silence?
Is it possible to change silence level,
so, let's say some not loud enough background noises will be recognized
as silence
and only loud enough human voice will be recognized as
Hello.
Voice quality when calling - this is one of the most important in the PBX.
You need to record the quality parameters for each call to improve.
Because the overall quality of a call can only be determined upon
completion, I did it in the HangUp handler and wrote in custom fields of
CDR.
Jonas Kellens wrote:
Hello
i have the following field (text string) in a MySQL database :
${KNUMMER} ${phone_number_to} ${phone_number_from} ${CHANNEL:4}
I read this string form the database and want to have the dialplan
variables to be replaced with the correct content.
Sounds like you
Amber and Sarosh wrote:
Hi
I am in need of information about how to configure the sip.conf and
extensions.conf for subscribers to
support the dialog-info event package rfc 4235. I am using Asterisk 11.7.0.4
version. Also please inform
if the phone must have the support for this too?
There
Dennis Guse wrote:
Hey,
I am running default Asterisk 11.16.0 on a FreeBSD-Machine.
I need to register to several other SIP-Services (actually 3):
short sip.conf
register = XX@a
register = XX@b
register = XX@c
If I remember correctly this worked quite well, but I now checked the
system again
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