Displaying 20 results from an estimated 2000 matches similar to: "pjsip insecure=port,invite"
2017 Dec 03
2
PJSIP OPTIONS
If understand correctly type=identify is more for sip trunk
configuration ?
;[mytrunk]
;type=identify
;endpoint=mytrunk
;match=198.51.100.1
;match=198.51.100.2
In chan_sip it was just reply 200 OK on keepalive packet without need
define trunks.
volga629
On Sun, 3 Dec, 2017 at 10:45 AM, Joshua Colp <jcolp at digium.com> wrote:
> On Sun, Dec 3, 2017, at 10:42 AM, volga629 at
2017 Dec 03
2
PJSIP OPTIONS
Right now it reply 401 Unauthorized with message in log "No matching
endpoint ..."
on Content 0 should reply 200 OK I guess
<--- Received SIP request (376 bytes) from UDP:10.30.100.41:5060 --->
OPTIONS sip:10.30.100.27:5080 SIP/2.0
Via: SIP/2.0/UDP
10.30.100.41;branch=z9hG4bKf5eb.1ac76487000000000000000000000000.0
To: <sip:10.30.100.27:5080>
From:
<sip:vprx00 at
2015 Aug 24
3
PJSIP add
I am trying to set add a SIP Header to a call before adding it to the Queue.
The dial plan sends the call to my macro to perform the work. When I use chan_sip, everything works as expected. When I use PSJIP support, it's not adding the SIP header.
Looking at the output, I see the macro is called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs
2013 May 28
1
DTMF recognized after call establishment
Hi,
I am receiving DTMF without any reason after call establishment.
The log as follows, and I suspect something related to directmedia,
[May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is
making progress passing it to SIP/MAN-000a4b48
[May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
answered SIP/MAN-000a4b48
[May 17 00:33:35] DTMF[4238] channel.c:
2020 Mar 14
2
congested/busy on trunk?
greetings asterisk users :)
ive just deployed version 17 and migrated as best I can to pjsip. I can
receive calls, and get to my mailbox prompt, however placing calls seems
impossible with the following error on dial:
Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid = 517890)
dunkel*CLI>
dunkel*CLI>
== Setting global variable 'SIPDOMAIN' to
2009 Feb 26
3
call-limit on a per destination basis
Hello,
I use asterisk to to IAX2 trunking between London POP & Reunion Island pop.
I would like to know if it's possible to do a kind of call-limit (i.e.
restrict to XX) channels but on a per dialcode and / or destination basis.
For example:
[trunk]
; reunion proper, i want to send no more than 24 channels
exten => _0262XXXXXX,1,Dial(IAX2/mytrunk/${EXTEN})
; reunion mobile, i want
2016 May 12
2
pjsip module reload problem
Hi!
Installing new asterisk server and decided to use chan_pjsip.
While module reload I get:
y 12 15:33:04] ERROR[21137]: config_options.c:715 aco_process_var: Could
not find option suitable for category '3567' named 'inband_progress' at
line 867 of
[May 12 15:33:04] ERROR[21137]: res_sorcery_config.c:317
sorcery_config_internal_load: Could not create an object of type
2016 Jul 20
3
PJSIP_DIAL_CONTACTS issue
Hi,
I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS.
When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial
command breaks and the call control go to hangup block instead of next
priority. The error in CLI says "*Dial requires an argument
(technology/resource)*".
This error seems legit as there are no contacts for an offline endpoint.
The dialplan
2019 Jan 26
3
INVITE from DID: No matching endpoint found but completes the call anyway
I have a trunk set up for the DID from my provider:
[my_provider]
type=registration
outbound_auth=my_provider
server_uri=sip:sip.example.com
client_uri=sip:my_username at sip.example.com
retry_interval=60
[my_provider]
type=auth
auth_type=userpass
password=123456
username=my_username
[my_provider]
type=aor
contact=sip:sip.example.com:5060
[my_provider]
type=endpoint
context=from-my_provider
2005 Mar 16
5
Asterisk Capabilities
I am new to Asterisk and currently work mainly with Cisco Callmanager.
With Callmanager I can setup partitions and call search spaces to
determine where a given phone can and can't dial. Does Asterisk offer
this type of functionality, and if so how?
Blake Parker
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2016 Nov 11
2
iaxmodem errors.
2013 Sep 24
1
PJSIP Identify Wiky
The Wiky needs to be updated
https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip#Configuringres_pjsip-IDENTIFY%28res_pjsip_endpoint_identifier_ip%29
This is the example shown:
"[6001]
endpoint=6001
match=203.0.113.1"
It should be:
"[6001]
type=identify
endpoint=6001
match=203.0.113.1"
2016 Sep 09
3
13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
09.09.2016 13:45, Joshua Colp ?????:
> Dmitry Melekhov wrote:
>> Hello!
>>
>>
>> Upgraded 13.10 to 13.11.1 today and now I see messages in log:
>>
>>
>> [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request
>> 'REGISTER' from '"3563" <sip:3563 at 192.168.32.254>' failed for
>>
2018 Sep 14
2
Re: live migration and config
14.09.2018 15:43, Jiri Denemark пишет:
> On Thu, Sep 13, 2018 at 19:37:00 +0400, Dmitry Melekhov wrote:
>>
>> 13.09.2018 18:57, Jiri Denemark пишет:
>>> On Thu, Sep 13, 2018 at 18:38:57 +0400, Dmitry Melekhov wrote:
>>>> 13.09.2018 17:47, Jiri Denemark пишет:
>>>>> On Thu, Sep 13, 2018 at 10:35:09 +0400, Dmitry Melekhov wrote:
2016 Oct 16
2
SIP on multiple ports
I have SIP (asterisk 11.23.0) running on port 5060 just fine. udp.
I have another SIP trunk thats wants to run on port 5068 (long story).
I have enabled tcpenable=yes in sip.conf and defined port=5068 in my trunk
definition. It does not seem that anything is listening on 5068?
How can I run SIP tcp on port 5068?
telnet localhost 5068
Trying 127.0.0.1...
telnet: connect to address 127.0.0.1:
2018 Sep 13
2
Re: live migration and config
13.09.2018 18:57, Jiri Denemark пишет:
> On Thu, Sep 13, 2018 at 18:38:57 +0400, Dmitry Melekhov wrote:
>>
>> 13.09.2018 17:47, Jiri Denemark пишет:
>>> On Thu, Sep 13, 2018 at 10:35:09 +0400, Dmitry Melekhov wrote:
>>>> After some mistakes yesterday we ( me and my colleague ) think that it
>>>> will be wise for libvirt to check config file existence
2016 Nov 15
2
iaxmodem errors.
2010 Feb 15
2
insecure=invite - not working for different dial plan
I'm using "insecure=invite" with two different dial plans, it it working with one dial plan but not with the other.
What other parameters could influence "insecure=invite"
In sip.conf below "insecure=invite" is working OK
[pstn-1270]
type=friend
secret=spa3k
username=voice-1270
mailbox=369
host=dynamic
insecure=invite
canreinvite=no
disallow=all
allow=ulaw
2014 Dec 25
2
originate , callerid
25.12.2014 15:46, Anthony Messina ?????:
> On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote:
>> I want to change call files, which has caller id in them, to call
>> originate from dial plan.
>> But I don't see such parameter here
>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate
>>
>> How can I pass callerid
2018 Sep 13
2
Re: live migration and config
13.09.2018 17:47, Jiri Denemark пишет:
> On Thu, Sep 13, 2018 at 10:35:09 +0400, Dmitry Melekhov wrote:
>> After some mistakes yesterday we ( me and my colleague ) think that it
>> will be wise for libvirt to check config file existence on remote side
> Which config file?
>
VM config file, namely qemu.
We forgot to mount shared storage (namely gluster volume), on which we