Hi, I using PJSIP as sip driver, I wound like to get source IP on inbound calls
from my peers,
tried use Function_CHANNEL like
${CHANNEL(pjsip,type,remote_addr)}
but it returns only empty string, what I doing wrong?
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request dies in source so I could patch it to response 200 OK ?
> On 20 Apr 2015, at 15:08, Nick Awesome wrote:
>
> Hi guys, have really annoying problem with trunks when I calling over voip
> provider..
>
>
> after awhile provider sends INFO packages but for some reason Aster
o() instead of Stasis()?
>
> On Fri, May 22, 2015 at 11:25 AM, Matthew Jordan <mailto:mjor...@digium.com>> wrote:
> On Fri, May 22, 2015 at 4:41 AM, Nick Awesome <mailto:jl...@me.com>> wrote:
> > Can anyone tell me how can I create echo test using ARI stasis application?
Can anyone tell me how can I create echo test using ARI stasis application?
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what may cause this, and how can I fix it ?
WARNING[23010]: pjsip:0 : tsx0x7f24f41b2 ..Failed to send Request msg
NOTIFY/cseq=15293 (tdta0x7f2480001a70)! err=171064 (Unsuitable transport
selected (PJSIP_ETPNOTSUITABLE))--
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Hi guys, have really annoying problem with trunks when I calling over voip
provider..
after awhile provider sends INFO packages but for some reason Asterisk doesn’t
answer on it.
after 8 packagers provider just drops the call, here is the package
<--- Received SIP request (555 bytes) from UDP:
5004
-- Channel PJSIP/304-0022 left 'native_rtp' basic-bridge
-- Channel PJSIP/99-0023 left 'native_rtp' basic-bridge
-- AGI Script /pbx/agi.php completed, returning 4
> On 18 Mar 2015, at 18:26, Matthew Jordan wrote:
>
> On Wed, Mar 18, 2015 at
Well, it breaks audio for all NAT endpoints, how can I fix this?
> On 18 Mar 2015, at 15:48, Matthew Jordan wrote:
>
> On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <mailto:jl...@me.com>> wrote:
>> Hey guys,
>>
>> have issues with reinvite, no matt
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
Stuck with TLS transport,
I have 2 phones (both in local network for tests)
one connected with up second with tls
when I calling TLS to UPD everything is fine, but when UDP calls TLS I getting
an error
ERROR[44230]: pjsip:0 : tlsc0x7f143012 TLS connect() error: Connection
refused [code=1
t; On 03 Mar 2015, at 20:14, Nick Awesome wrote:
>
> Hey guys,tried to make tls work with pjsip on asterisk 13.2.0
>
> have compiled pjsip with ssl,
>
> added transport
>
> [tls]
> type=transport
> cert_file=/pbx/keys/server.crt
> ca_list_file=/pbx/keys/ca
Hey guys,tried to make tls work with pjsip on asterisk 13.2.0
have compiled pjsip with ssl,
added transport
[tls]
type=transport
cert_file=/pbx/keys/server.crt
ca_list_file=/pbx/keys/ca.key
priv_key_file=/pbx/keys/server.key
protocol=tls
bind=192.168.1.4:5061
local_net=192.168.1.0/24
external_me
success! just replaced MeetMe to Bridge in softkey.xml and conf works now with
the latest fw!
On Feb 26, 2015, at 9:00 AM, Nick Awesome wrote:
>
> I have not working 3way conference, when I trying to connect second call,
> phone says “unable to set up conference”
> and sending so
another issues with cisco 7975
I have phone registered on asterisk
have 2 different issues on different versions of firmware,
on 9-4-2-1S I have not working 3way conference, when I trying to connect second
call, phone says “unable to set up conference”
and sending some cisco xml data to asteri
Oh god it works !
to switch cisco to upd I used config:
2
with udp it works well, thanks for your help :)
> On 24 Feb 2015, at 17:02, Joshua Colp wrote:
>
> If you use UDP with force_rport=no it'll work.
> If you use TCP then set rewrite_contact=yes so it'll reuse the established
> TCP connec
=25333 (tdta0x7f1aa00ad810)! err=120111 (Connection
refused)
> On 24 Feb 2015, at 15:05, Joshua Colp wrote:
>
> Nick Awesome wrote:
>> Hay guys, got trouble with registration with cisco 7975
>
> The "force_rport" option is incompatible with Cisco, it needs to be
&
Hay guys, got trouble with registration with cisco 7975
Here is the debug :
<--- Received SIP request (576 bytes) from UDP:192.168.1.61:49533 --->
REGISTER sip:192.168.1.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK35076381
From: ;tag=0c8525a68961001f44d503e2-d9359bd3
To:
Call-ID:
Works, thank you!
> On Feb 23, 2015, at 7:11 PM, Joshua Colp wrote:
>
> Nick Awesome wrote:
>> Hay guys, have question.
>>
>> When I do regular dial I use
>> $this->AGI->get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,tru
Hay guys, have question.
When I do regular dial I use
$this->AGI->get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true);
to get all contacts of current endpoint and so I dial to all phones at once,
but if I exec QUEUE, I have just one phone rings, seems like it take first one
as
Works! how I miss that… Thanks.
On 02 Oct 2014, at 17:05, Scott Griepentrog wrote:
> You can use the AGI command EXEC to execute a dialplan application, and the
> application UserEvent can be used to generate custom events that AMI clients
> can receive.
>
> https://wiki.asterisk.org/wiki/dis
exten => blablabla, 1,NoOp(*** ${PJSIP_HEADER(read,To)} *** ${CALLERID(num)}
> ***)
>
>
> Am 02.09.2014 um 20:11 schrieb Rainer Piper:
>> contact_user can be anything and calling an agi is no problem
>>
>>
>> Am 02.09.2014 um 19:49 schrieb Nick Awesome:
>&g
)
> exten => 4922897167168,1,Answer
> same => n,echo()
> same => n,Hangup()
> ; incoming VOIP 97167169 FAX
> ;exten => 4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>
>
> Regards
> Rainer
>
> Am 02.09.2014
hecking db for matches, in db I have table with all my
trunks information
On 02 Sep 2014, at 15:49, Joshua Colp wrote:
> Nick Awesome wrote:
>> Tried doing that, but
>>
>> first: AGI->exten is ’s’ for some reason. and second its not
>> practical, for example
sip :|
On 02 Sep 2014, at 15:32, A J Stiles wrote:
> On Tuesday 02 Sep 2014, Nick Awesome wrote:
>> Hello guys.
>>
>> Have 2 external numbers that required registration on provider server,
>>
>> trunk1: 73432260005@80.75.132.66
>> trunk2: 73432260050@80.75
yMmM0MWFiZTE.--be7c48325cdef400
From: ;tag=yddmzvcoi3waw24e.o
Call-ID: 22e7064301970213226722k41100rmwp
CSeq: 588 ACK
Content-Length: 0
On 02 Sep 2014, at 15:01, Joshua Colp wrote:
> Nick Awesome wrote:
>> Hello guys.
>
> Kia ora,
>
>> Have 2 external numbers that required registrati
So there is no way to do that with pjsip?
On 02 Sep 2014, at 11:35, Administrator TOOTAI wrote:
> Le 02/09/2014 08:47, Nick Awesome a écrit :
>> Hello guys.
>
> Hi
>
>>
>> Have 2 external numbers that required registration on provider server,
>>
&
Hello guys.
Have 2 external numbers that required registration on provider server,
trunk1: 73432260005@80.75.132.66
trunk2: 73432260050@80.75.132.66
Thing is I can’t figure out how to route them to different IVRs
by default Asterisk can’t match endpoint
Request from '' failed for '80.75.132.6
Probably you should use “Action: Park"
example:
Action: Park
Channel: SIP/1000-0003
Channel2: SIP/1000-0004
On 21 Jul 2014, at 17:00, mahdieh saeed wrote:
> Hi,
> I want to write API for doing some actions. I want to write function for hold
> special call via AMI.But I can not find any
Hi, after update on 12.4.0 asterisk crashes on MeetMe ending
on 12.3.2 it worked well.
Is some one else have this issues? should someone open a ticket?
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N
Hello guys,
I have trunk “1", Internal num “99" and MeetMe “1010"
now I calling 99 -> 89264959635 via 1
/pbx/agi.php: [agi_channel] => PJSIP/99-0012
/pbx/agi.php: [agi_callerid] => 99
/pbx/agi.php: [agi_calleridname] => 99
/pbx/agi.php:
oh.. its simple.
"[res_pjsip_endpoint_identifier_ip]" should be before
"identify=realtime,ps_endpoint_id_ips”, not "[res_pjsip]”
Thanks all for help :)
On 17 Jul 2014, at 11:05, Nick Awesome wrote:
> New information, as I said I’m using realtime,
> thats the problem
ULT NULL,
UNIQUE KEY `id` (`id`),
KEY `ps_endpoint_id_ips_id` (`id`)
) ENGINE=InnoDB DEFAULT CHARSET=latin1;
with entry
10001 | 10001 | 85.195.98.178
but thats just didn’t works(
is this a bug and should I open ticket ?
On 16 Jul 2014, at 21:13, Nick Awesome wrote:
> Ok there is my test a
8:53 PM, Joshua Colp wrote:
> Nick Awesome wrote:
>> I thought that
>>>> type=identify
>> will match an IP address and accept it,
>>
>> well, in my example I can control both sides and able to configure it
>> without registration. in real life I have a p
07700d65055c9214@85.195.98.178) - No matching endpoint found
85.195.98.178 is an operator,
so what I should add to my config to be able accept calls from Registered peer ?
On Jul 16, 2014, at 7:55 PM, Joshua Colp wrote:
> Nick Awesome wrote:
>> Hi all, In my case I using realtime, here
Hi all,
In my case I using realtime,
here is how it looks in plant
[10001]
type=registration
transport=upd_static
outbound_auth=10001
server_uri=sip:600@192.168.1.1:5060
client_uri=sip:600@192.168.1.4:5060
[10001]
type=auth
auth_type=userpass
password=600
username=600
[10001]
type=aor
contact=sip
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