2015-07-08 13:11 GMT-06:00 Joshua Colp jc...@digium.com:
You probably want to add rewrite_contact=yes to your endpoint. This will
cause it to reuse the existing connection established from the phone.
Generally the port provided by the phone is not reachable.
Hi Joshua , I add the option you
2015-07-08 13:09 GMT-06:00 Ryan, Travis ry...@oscarwinski.com:
Asterisk13 can do native tls with each phone? Nice.
any example?
rickygm
http://gnuforever.homelinux.com
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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 ?:
2015-06-05 14:29 GMT-06:00 Luca Bertoncello lucab...@lucabert.de:
I think it is a problem on Asterisk for OpenWRT... :(
Regards
Luca Bertoncello
(lucab...@lucabert.de)
compilation problems with the module srtp , check the module
module show like srtp
--
rickygm
2015-06-05 12:21 GMT-06:00 Luca Bertoncello lucab...@lucabert.de:
Hi list!
I'm trying to configure my Asterisk to accept SIP-TLS connections, too.
I followed this HowTo:
http://remiphilippe.fr/sips-on-asterisk-sip-security-with-tls/
But as soon I try to connect to my Asterisk
2015-03-27 10:52 GMT-06:00 Carlos Rojas crt.ro...@gmail.com:
I Ricky
I have worked with this gateway few years ago, it's good product, they have
gateways with PRI connectors and SIP.
The quality is good, and it woks good with asterisk or regular PBXs.
Hi carlos , thank for your advice, I
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
2015-03-23 11:08 GMT-06:00 ricky gutierrez xserverli...@gmail.com:
Hi , I'm having some problems with functions enable auto answer in
some Grandstream GXP 1405 , I have enabled this feature in the snom
821 phone and work gr8 , in the gandstream not work, I enable the
function on the phone
Hi , I'm having some problems with functions enable auto answer in
some Grandstream GXP 1405 , I have enabled this feature in the snom
821 phone and work gr8 , in the gandstream not work, I enable the
function on the phone
Allow Auto Answer by Call-Info: yes
Dialplan:
exten =
2015-03-18 12:54 GMT-06:00 ricky gutierrez xserverli...@gmail.com:
I'm confused this is not a patch, it's just garbage ;), I'm making a
connection xmpp with asterisk and not connected, at the cli shows me
the message every second:
RROR[2545]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP
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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rickygm
http://gnuforever.homelinux.com
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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
2015-03-12 13:07 GMT-06:00 Bryant Zimmerman brya...@zktech.com:
SIPAddHeader(Alert-Info:\;info=ring3)
In the phone config add the value ring3 and select Account # / Call
Settings / Match Incoming Caller ID (Matching Rule)
In the first rule place the word ring3 and select your ring tone.
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/310SIP/318,30,t)
can not get it to work
any idea o tips?
2015-03-05 6:11 GMT-06:00 Steve Davies davies...@gmail.com:
Looking at the pastebin, the Vega device sends a CANCEL with reason:
Reason: Q.850 ;cause=16.
Cause 16 is normal clearing and suggests that the original caller has
disconnected. I would take a look at the Vega's logs
I tried to
On Wednesday, March 4, 2015, ricky gutierrez xserverli...@gmail.com wrote:
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
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
Hi list , I have a question with account codes, all my outgoing calls
are authenticated, but now the boss wants to monitor these calls with
the codes.
example: maria has an extension 110, but peter was in place and use
the phone maria , maria then says that she did not make that call to
that
2015-03-02 3:44 GMT-06:00 A J Stiles asterisk_l...@earthshod.co.uk:
Ah. *Incoming* calls are not something that is within your control; they have
already been routed onto a line by your telco. So you will need to speak to
someone at your telco about doing this.
Hi Aj, I call to telco and
2015-02-27 10:25 GMT-06:00 A J Stiles asterisk_l...@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 =
2015-02-25 18:23 GMT-06:00 John Kiniston johnkinis...@gmail.com:
I'd recommend using DEVICE_STATE
On your extension 101, Check the DEVICE_STATE of peer SIP/101, If it's not
'NOT_INUSE' then dial it, Otherwise dial SIP/102
exten =
2015-02-26 10:45 GMT-06:00 A J Stiles asterisk_l...@earthshod.co.uk:
You just need to use call groups.
In your chan_extra.conf (if it's an OpenVox) or chan_dahdi.conf, add
something like
group=1
to the definition for each span.
Now in the [globals] section of your dialplah, have
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 =
Hi list, I write on the list looking for help, buy a openvox gw gsm
for four channels and I'm a little disappointed with the support
openvox, for some reason , The call doesn´t get trough
support tells me it was my asterisk server, but does not really work
me and my internal calls are working
: inmaconsa-Voice-Sip-ipbx
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0
2015-01-19 10:24 GMT-06:00 ricky gutierrez xserverli...@gmail.com:
Hi
2015-01-09 9:05 GMT-06:00 Tech Support aster...@voipbusiness.us:
Hello;
Did you remember to uncomment the dateformat in
/etc/asterisk/logger.conf? That's necessary for fail2ban to work.
Logger.conf
[general]
dateformat=%F %T
Hi , I'll show my logger
dateformat=%F %T ; ISO 8601
2015-01-09 3:53 GMT-06:00 Stefan Gofferje li...@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
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:
2014-12-29 4:51 GMT-06:00 Doug Lytle supp...@drdos.info:
I use monit, but I only watch the pid
check process asterisk with pidfile /var/run/asterisk/asterisk.pid
start program = /usr/sbin/service asterisk start
stop program = /usr/sbin/service asterisk stop
Doug
work fine my friend , thnk
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
Hi, here again, I'm going around with this problem and can not find a
solution, but I put different context within xmpp.conf, asterisk
believes xmpp messages between users are SIP message.
any idea?
2014-11-17 16:56 GMT-06:00 ricky gutierrez xserverli...@gmail.com:
Hi list, I have a big doubt
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
I think asterisk does not respect this, I have added several within
xmpp.conf buddy
Client: ejabberd
Buddy: alci...@xmpp.domain.com
Resource: asterisk-xmpp
node: http://www.asterisk.org/xmpp/client/caps
version:
2014-10-13 14:44 GMT-06:00 Matthew Jordan mjor...@digium.com:
The error message is pretty explicit about what you asked it to look for:
{quote}
acf_jabberstatus_read: Resource alcides of buddy operadora@172.16.8.59
was not found.
strange, I put the fqdn to ejabberd, and now , not shows the
domain in jabberid.
Regards,
Marcelo H. Terres
mhter...@gmail.com
IM: marc...@jabber.mundoopensource.com.br
http://www.mundoopensource.com.br
http://offtopicsandfun.blogspot.com
http://biertasters.blogspot.com
http://twitter.com/mhterres
On Mon, Oct 13, 2014 at 6:06 PM, ricky gutierrez
anyone here?
2014-10-01 8:09 GMT-06:00 ricky gutierrez xserverli...@gmail.com:
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
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
Hi, I've been trying to talk xmpp with asterisk with ICE-UDP, but
still does not work
2014-07-18 7:26 GMT-06:00 Daniel Pocock dan...@pocock.pro:
I have a multi-homed machine (quite a few IP addresses on one of the
interfaces)
For SIP I found that using externaddr in sip.conf would make it
I have configured support for ice in sip.conf, and made a connection
with motif to jingle, but does not work for me
[Jul 15 12:03:32] ERROR[21758]: chan_motif.c:1955
jingle_interpret_ice_udp_transport: Received ICE-UDP transport
information on session '8b4hdffbt37vg' but ICE support not
#rpm -qa | grep uuid
uuid-1.6.1-10.el6.x86_64
libuuid-2.17.2-12.14.el6_5.x86_64
uuid-devel-1.6.1-10.el6.x86_64
and res_rtp_asterisk was added in the compilation
rtp.conf
rtpstart=1
rtpend=2
icesupport=yes
2014-07-15 12:19 GMT-06:00 Joshua Colp jc...@digium.com:
ricky gutierrez wrote
.
https://wiki.asterisk.org/wiki/display/~jcolp/ICE,+STUN,+and+TURN+Support
2014-07-15 12:41 GMT-06:00 ricky gutierrez xserverli...@gmail.com:
#rpm -qa | grep uuid
uuid-1.6.1-10.el6.x86_64
libuuid-2.17.2-12.14.el6_5.x86_64
uuid-devel-1.6.1-10.el6.x86_64
and res_rtp_asterisk was added
chan_jingle2 is supported in Asterisk 11?
2014-07-15 13:28 GMT-06:00 ricky gutierrez xserverli...@gmail.com:
I'm reading the wiki and says that by default is active, I have it set
in sip.conf and rtp.conf
icesupport=yes
Usage
By default ICE support is enabled in res_rtp_asterisk. It can
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