Displaying 20 results from an estimated 300 matches similar to: "Auto Answer"
2006 Nov 14
1
How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?
My SIP phones can dial out through Sipura SPA3k to POTS for local and
911 calls _but_ incoming POTS calls are being swallowup somehow.
Am I on the right track with the code snippit below?
sip.conf:
---------
In sip.conf the following code is _supposed_ to ring the SIP phones when
a POTS line call comes in through Sipuara to Asterisk.
[spa3k-pstn-in] ; Pots-line-in from Sipura
; If
2020 May 01
4
Length of dial string
Hi all
as per the new release notice for 13.33.0 received today - can anyone advise
me the max limit of the string to the Dial Command - see
* [ASTERISK-27946
<BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] -
dial (API): Storage of dialed target uses AST_MAX_EXTENSION
when it shouldn't
I have been fighting with this issue for months trying to find a solution I
2020 May 01
1
Length of dial string
Hi Dovid
Yes was one of the options but as the required list is dynamic becomes very
messy - and all combinations problem - where as "call all workers job xxx"
is what is needed so the ability to call 20+ numbers is what is needed - agi
does a database search for all jobx workers and constructs a dialstring with
SIP, DAHDI and Local devices.
Can someone tell me where to find maximum
2014 Nov 17
1
motif and other xmpp
Hi list, I have a big doubt!, I have some users with ejabberd and am
using motif to make some calls to extensions, here works fine, the
problem is when I want to send a message to another user on ejabberd
and asterisk take this message as part him, like a sip message , the
other user does not receive this message xmpp
User A xmpp == Chat to == User B xmpp (not receive the message)
look cli
2015 Mar 27
2
Gateway Eurotech
Hi, I know there are people with much experience in asterisk, and I
want to ask if anyone had experiance with this gw
http://www.eurotech-communication.com/products/voip-gateways/VoIP-32-CHANNELS-2E1-PRI-1U/
I'm having trouble getting connect with asterisk
anyone has any production?
regardss
--
rickygm
http://gnuforever.homelinux.com
2015 Mar 04
2
hangup call gw FXO
I'm having some problems with a vega sangoma, if a call comes into my
ivr and hangs up, the call continues to ring and leaves hanging the
channel, I have to restart Asterisk and everything works Ok
my sangoma is a vega 50 , 4 FXO .
I tried different tone of countries and does not work,
this is the trace of which is for hanging up the channel:
http://pastebin.com/y410Rhzt
I was thinking
2015 Mar 12
2
GXP 1405 and asterisk
Hi list, someone has successfully change different ringtone from
dialpan with asterisk with this model Granstream?
for example:
exten => 0,1,Playback(pls-wait-connect-call)
same=> n,SIPAddHeader(Alert-Info:;info=ring3)
same=> n,Dial(SIP/310&SIP/318,30,t)
can not get it to work
any idea o tips?
regardss
--
rickygm
http://gnuforever.homelinux.com
2018 Apr 26
2
cluster of 3 nodes and san
Hi list, I need a little help, I currently have a cluster with vmware
and 3 nodes, I have a storage (Dell powervault) connected by FC in
redundancy, and I'm thinking of migrating it to proxmox since the
maintenance costs are very expensive, but the Doubt is if I can use
glusterfs with a san connected by FC? , It is advisable? , I add
another data, that in another site I have another cluster
2011 Jun 17
1
Missed calls and groups
Is there a SIP header I can set (for Snom and Yealink phones if that's
relevant) or any other mechanism to tell a phone to ignore a particular
call from it's missed call list?
I have bits of the dialplan that ring groups of phones eg:
exten => 200,1,Dial(Sip/112&SIP/113&SIP/114)
and I don't want such calls being recorded by the phone as a missed
call.
Calls to the
2014 Oct 01
1
JABBER_STATUS CODE 7
Hi all,I hope to find a solution with the help of the list, I'm trying
to get the status of my extensions with ejabberd , the idea is to
visualize my users ejabberd incoming calls or missed.
I'm testing with my operator extension with this code but only get the
missed call notification does not show me where the call is coming.
my piece of code
[operadora]
exten =>
2015 Jan 08
4
SEMI OFF-TOPIC - Fail2ban
Hi list , someone on the list has seen this type of connection
attempts in asterisk, fail2ban does not stop
2015-01-08 14:59:47] SECURITY[21515] res_security_log.c:
SecurityEvent="ChallengeSent",EventTV="1420750787-386840",Severity="Informational",Service="SIP",EventVersion="1",AccountID="sip:100 at
2015 Jul 08
6
tls on asterisk 13
Hi list , I'm doing some tests with asterisk 13.4 and tls, and failed
to make it work, all my terminals spa Cisco 5XX
look my cli
[Jul 8 11:09:16] ERROR[14733]: pjsip:0 <?>: tlsc0x7f539801 TLS
connect() error: Connection refused [code=120111]
[Jul 8 11:09:16] WARNING[14733]: pjsip:0 <?>: tsx0x7f53a8008 Failed
to send Request msg OPTIONS/cseq=48024 (tdta0x7f53c000dcb0)!
2015 Feb 25
5
situation with ivr and four-channel gateway
Hi list, I need your help ,I have an incoming call x the ivr and the
operator takes the call. ext "101" , If a second call reenters and the
operator is talking, I want to send to the extension 102 I use the
Variable DIALSTATUS , but not working
check IVR
[IVRINMA]
exten => s,1,Wait(1)
exten => s,n,Set(CHANNEL(language)=es)
same=> n,Set(TIMEOUT(digit)=4)
same=>
2018 Apr 27
0
cluster of 3 nodes and san
Hi, any advice?
El mi?., 25 abr. 2018 19:56, Ricky Gutierrez <xserverlinux at gmail.com>
escribi?:
> Hi list, I need a little help, I currently have a cluster with vmware
> and 3 nodes, I have a storage (Dell powervault) connected by FC in
> redundancy, and I'm thinking of migrating it to proxmox since the
> maintenance costs are very expensive, but the Doubt is if I can
2018 Apr 27
1
cluster of 3 nodes and san
>but the Doubt is if I can use glusterfs with a san connected by FC?
Yes, just format the volumes with xfs and ready to go
For a replica in different DC, be careful about latency. What is the
connection between DCs?
It can be doable if latency is low.
On Fri, Apr 27, 2018 at 4:02 PM, Ricky Gutierrez <xserverlinux at gmail.com> wrote:
> Hi, any advice?
>
> El mi?., 25 abr. 2018
2014 Dec 29
2
[OFF TOPIC] monit
Hi list , I'm trying to run monit with asterisk, starting as simple
# My PBX Asterisk
check process asterisk with pidfile /var/run/asterisk/asterisk.pid
start program = "/etc/init.d/asterisk start" with timeout 60 seconds
stop program = "/etc/init.d/asterisk stop" with timeout 60 seconds
if failed host 127.0.0.1 port 5038 then restart
if 5 restarts within 5 cycles then
2015 Mar 18
2
res_xmpp.c:3468 xmpp_client_reconnect:
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
--
rickygm
http://gnuforever.homelinux.com
2020 Jul 03
2
disk on vm with kvm
hi list, i am trying to change the input and output scheduler on my
disks, and it does not allow me ,
I have several virtualized vm over kvm, and when I try to make the
change it shows me this message:
echo "noop" > /sys/block/vda/queue/scheduler
-bash: echo: write error: Invalid argument
kernel version:
3.10.0-1127.13.1.el7.x86_64
any idea?
--
rickygm
2015 Feb 27
2
situation with ivr and four-channel gateway
2015-02-27 10:25 GMT-06:00 A J Stiles <asterisk_list at earthshod.co.uk>:
> O.K. So what does your existing Dial() statement in extensions.conf look
> like?
>
apology, put the gateway was sangoma but is a openvox ,
all my outgoing calls out for this context:
[my-mobile-out]
exten => _NXXXXXXX,n,Dial(SIP/1003/${EXTEN},55,rT)
exten =>
2015 Jan 09
2
SEMI OFF-TOPIC - Fail2ban
2015-01-09 3:53 GMT-06:00 Stefan Gofferje <lists at home.gofferje.net>:
>
> Do you really want to detect "ChallengeSent"? That should occur also on
> legitimate login processes...
>
Hi , strange thing is that I still have not this asterisk in
production and I see many attempts Connection.
Now keep in mind that when a connection of authentication is
successful the