./configure --with-crypto --with-ssl --with-srtp --with-pjproject-bundled
The culprit of this behavior is option --with-ssl
Version 15.5 does not have this problem.
26.09.2018 16:46, George Joseph пишет:
On Tue, Sep 25, 2018 at 2:18 PM Dmitriy Serov <mailto:serov@gmail.com>&
Hello.
After successful compilation 15.6.1 (bundled pjsip) and start asterisk i
has error Symbol pjsip_tls_transport_start2 not found.
/main/libasteriskpj.exports does not containg pjsip_tls_transport_start2
and pjsip_tls_transport_start.
More:
* All versions before (including 15.5) has
-rc1. This is very sad,
because encryption should be mandatory in the modern world.
Dmitriy Serov.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk communi
Hello.
A little sub from my dialplan:
[sub-Read]
exten => s,1,NoOp(Read)
same => n,Set(LOCAL(tmp_record_file)=/tmp/asterisk-in/${EPOCH})
same => n,Monitor(wav16,${tmp_record_file},o)
same => n,Read(tmp_ext,${ARG2},${ARG3},${ARG4},${ARG5},${ARG6})
same => n,StopMonitor()
same =>
AMI action Originate has param "Codecs". I think it helps.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+ManagerAction_Originate
22.11.2017 13:24, Kseniya Blashchuk пишет:
Hi all!
Asterisk 13.1.0 Ubuntu 16.04, all latest.
Can anybody explain this to me - I run Originate command from
[sub-out-do-dial]
exten => s,1,NoOp(Dial)
same => n,NoOp(FirstChannel: ${CHANNEL})
same => n,Dial(,60,gF)
same => n,NoOp(SecondChannel: ${CHANNEL})
same => n,Return()
[some]
exten => s,1,GoSub(sub-out-do-dial,s,1)
In case of the destination channel hangs up in log i see:
Joshua, issue has been filed. Thank you!
https://issues.asterisk.org/jira/browse/ASTERISK-26689
03.01.2017 20:58, Joshua Colp пишет:
On Mon, Dec 19, 2016, at 06:36 AM, Dmitriy Serov wrote:
Yes, this means the remote end was not sending any audio stream.
But it shouldn't.
According to [1
11:33, Jean Aunis пишет:
This means the remote end was not sending any audio stream, or the
audio stream was not received by Asterisk. The problem may have many
different reasons, but often it is a network-related issue.
Le 16/12/2016 à 21:19, Dmitriy Serov a écrit :
Today I faced a problem
Today I faced a problem. Please help to solve this problem.
Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware
v2.06(AAGJ.9)C1
Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip trunk).
Call using early media (183 Session in progress) and rtp_timeout=10.
After 10
06.09.2016 17:08, George Joseph пишет:
On Tue, Sep 6, 2016 at 7:58 AM, Dmitriy Serov <serov@gmail.com
<mailto:serov@gmail.com>> wrote:
06.09.2016 16:42, George Joseph пишет:
On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov
<serov@gmail.com <mailto:s
of modules. All the necessary modules
written manually in modules.conf.
At the end of the installation it reported the extra modules (thanks)
and they removed me as "garbage".
On 6 September 2016 at 14:42, George Joseph <gjos...@digium.com> wrote:
On Tue, Sep 6, 2016 at 7:3
06.09.2016 16:42, George Joseph пишет:
On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov <serov@gmail.com
<mailto:serov@gmail.com>> wrote:
Hello.
Several months server working on asterisk 13.7 and pjproject 2.5
(installed separately). Once a day the server cras
Hello.
Several months server working on asterisk 13.7 and pjproject 2.5
(installed separately). Once a day the server crashes or hangs and is
familiar sores that written watchdogs.
Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5).
Solved all the problems with
There is a separate app for recording voice (app_record) or dtmf input
(app_read).
But there is no way to allow the user to choose to enter by voice or by
keypad in same time.
app_record analyzes the dtmf input, but only the # and * (to quit).
Nothing stored in variables :(
Is there a
asterisk 13.8.7, PJSIP.
One VoIP provider requires a specific value in the field "contact" of a
INVITE.
What setting does indicate the value will be in this field (instead
"asterisk")?
Thanks.
currect settings (with templates):
[srv_d22778](srv-auth)
username=999
password=secret
Today was another attempt to upgrade to version 13.9 (git).
1. The result was https://issues.asterisk.org/jira/browse/ASTERISK-25970
Had to temporarily block this contact and look forward to advice of how
to fix it.
2. Also, an unpleasant surprise was the increase in CPU usage from
10-50%
At the moment I plan to migrate from asterisk 13.7 to 13.8.
Because of relatively frequent updates I am building asterisk from a
directory that is updated via git switch to the desired branch.
1. Will be updated pjproject patches with "git pull"?
2. Will update himself pjproject?
3. And what
ry
to use a stun (did not seem to help, and it only works in the case of
RTP) or proxy?
Thanks.
21.03.2016 23:32, George Joseph пишет:
On Mon, Mar 21, 2016 at 11:58 AM, Dmitriy Serov <serov@gmail.com
<mailto:serov@gmail.com>> wrote:
Good day.
Aste
. But, like, it's a
feature of code that may already be fixed.
2. deleting a contact much earlier than the 90 seconds specified during
the registration
Would be grateful for any clues.
Dmitriy Serov.
expiration settings:
[common-aor](!)
type=aor
qualify_frequency=60
default_expiration=120
Please help find the cause of strange behavior res_pjsip.
Making outgoint call to other sip server (CommuniGatePro), my asterisk
suddenly sends BYE after picking up!
Partial log of an outgoing call with full debug is attached and on web:
http://pastebin.com/tLNCpx4d
No diagnostic messages
guessing, but that side of the
call isn't in your log.
If it is from an internal extension, I think a SIP trace on that side
would help.
On Wed, Feb 10, 2016, 3:20 PM Dmitriy Serov <serov@gmail.com
<mailto:serov@gmail.com>> wrote:
Please help find the cause of stra
of them.
Thanks.
Dmitriy Serov.
[2016-01-31 14:44:56] VERBOSE[1950] res_pjsip_logger.c: <---
Transmitting SIP response (874 bytes) to UDP:109.60.222.xxx:49912 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
109.60.222.xxx:49912;rport=49912;received=109.60.222.xxx;
Hello.
Asterisk 13.7 (branch 13), res_pjsip
Log file contains a lot of lines:
[2016-01-11 15:33:00] ERROR[2862] res_pjsip/pjsip_configuration.c:
Unable to create ast_sip_contact_status for contact
17378/sip:17...@87.255.225.:5060
[2016-01-11 15:33:00] VERBOSE[2703] res_pjsip_registrar.c:
for any clues.
Dmitriy Serov.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk
for a proxy to adequately advise
you any deeper.
Good luck. I hope some of the above is helpful to you.
Bryant
*From*: "Dmitriy Serov" <serov@gmail.com>
*Sent*: Monday, November 2, 2015 9:10 AM
*To*:
teful if tell me how to get the User agent.
Thanks
Dmitriy Serov.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
htt
Hello.
Recently I installed voipmonitor and voipmonitor-gui trial version.
After examining it, I was amazed at the abundance of useful information
that can and should be obtained from the work of Asterisk.
1. The cost voipmonitor-gui too expensive and not justified in my case.
For this
06.10.2015 1:22, Joshua Colp пишет:
On 15-10-05 05:58 PM, Dmitriy Serov wrote:
05.10.2015 23:24, Joshua Colp пишет:
On 15-10-05 05:22 PM, Dmitriy Serov wrote:
Hello. Do I understand correctly that the current implementation
res_pjsip does not support ZRTP?
http://lists.digium.com/pipermail
06.10.2015 16:08, Matthew Jordan пишет:
I know this is shocking to hear, but this is an open source project.
That means anyone can fix something. Anyone can add something. Even
you! You have all the power to affect your system.
It also means that no one is under any obligation to do it for
05.10.2015 23:24, Joshua Colp пишет:
On 15-10-05 05:22 PM, Dmitriy Serov wrote:
Hello. Do I understand correctly that the current implementation
res_pjsip does not support ZRTP?
http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html
ZRTP is not supported in Asterisk itself
lacking, and much of
the promise failed. Dmitriy Serov.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http
te: Digest
realm="ruvoip.net",nonce="1441594874/5741fb37496404a4aa5cf0e53a129867",opaque="7441b8c64eddc67a",algorithm=md5,qop="auth"
Server: ruVoIP.net PBX Content-Length: 0 Dmitriy Serov.
--
_
-- Bandwidth and
,GotoIf($[${pseudodid} = 088]?internal,13,1:13)
-Andrew Galdes
On Thu, Apr 2, 2015 at 3:46 PM, Dmitriy Serov serov@gmail.com
mailto:serov@gmail.com wrote:
This is one of the chronic problems. Try this option in sip.conf:
match_auth_username=yes
Carefully read the description
This is one of the chronic problems. Try this option in sip.conf:
match_auth_username=yes
Carefully read the description, it is better to test in after hours.
02.04.2015 2:50, Andrew Galdes пишет:
Hello all,
I have an Asterisk server (Asterisk 10.12.4) with multiple sip
accounts with the
Hello.
I have plain text password for endpoint with outbound registration
(someone else's server).
My aim is to write it in pjsip.conf.
md5 means that I know realm. I do not always know it.
Is where any way?
Dmitriy Serov
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.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
into a separate table and join them may be an option.
But do not want to resort to such a decision
How do you solve this problem?
Dmitriy Serov.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
rewrite_contact=yes
allow=!all,alaw
Dmitriy Serov
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org
07.03.2015 1:21, Kevin Harwell пишет:
On Fri, Mar 6, 2015 at 3:46 PM, Dmitriy Serov serov@gmail.com
mailto:serov@gmail.com wrote:
07.03.2015 0:24, Kevin Harwell пишет:
On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov
serov@gmail.com mailto:serov@gmail.com wrote
identification by sip uri.
I think it will be usefull.
P.S. i hope issues will be rejected:
https://issues.asterisk.org/jira/browse/ASTERISK-22306 and SWP-6069
Dmitriy Serov
--
_
-- Bandwidth and Colocation Provided by http
07.03.2015 0:24, Kevin Harwell пишет:
On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov serov@gmail.com
mailto:serov@gmail.com wrote:
Hello.
Asterisk 13.2.
I transfer configs from chan_sip to res_pjsip.
In chan_sip i have match_auth_username=yes and have nothing
.
It will be better ACL has optional link to Endpoint.
Or you can offer other solution?
Dmitriy Serov
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
Not in
use0 of inf
OutAuth: srv_d228/talk37.ru
InAuth: srv_d228/talk37.ru
Aor: srv_d228 10
Contact: srv_d228/sip:sipnet.ru:5060 Avail 9.858
Thanks!
Dmitriy Serov
Sorry, i found the source of problem.
https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels
dialing via pjsip have to change dialplan syntax :(
May be it will be usefull sombody else.
04.03.2015 21:54, Dmitriy Serov пишет:
Hello.
I am using asterisk and chan_sip a lot of years
Hello all.
Dialing with tT options and function Read (to prompt number) has a
trouble for me.
Can I temporarily features off during Read?
features.conf:
[featuremap]
blindxfer = ## ; Blind transfer (default is #)
atxfer = **; Attended transfer
I try:
01.04.2013 23:52, Paul Belanger пишет:
On 13-04-01 03:16 PM, Dmitriy Serov wrote:
31.03.2013 23:15, Barry Flanagan ?:
On 31 March 2013 18:11, Dmitriy Serov serov@gmail.com
mailto:serov@gmail.com wrote:
Hi, asterisk admin and users.
I need to SIP INVITE uri with domain via peer
31.03.2013 23:15, Barry Flanagan ?:
On 31 March 2013 18:11, Dmitriy Serov serov@gmail.com
mailto:serov@gmail.com wrote:
Hi, asterisk admin and users.
I need to SIP INVITE uri with domain via peer. And uri domain
differ then peer domain.
dialplan:
exten = s,n
@domain1.com
I need: num...@domain2.com
I can't use SIP uri dial, i need authorization (peer1)
I think asterisk can't do that. Is where work around?
Dmitriy Serov.
--
_
-- Bandwidth and Colocation Provided by http://www.api
48 matches
Mail list logo
