[asterisk-users] Best place to issue tickets for Digium phones ?

2016-02-10 Thread Olivier 
Hello,

I've recently given a try to a Digium D70 phone.

At the moment, I'm configuring them though config files with a DHCP server
and not using DPMA.
Of course, I'm connecting them to Asteris (PJSIP stack on 13.7.0).


Which is the best place to:
- read about past issues
- open new tickets for remaining issues.

Best regards
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[asterisk-users] Authenticate() 11.21.0

2016-02-10 Thread Jerry Geis 
I am trying to use Authenticate() in the dialplan
for something other than "my password".
The message says "Please enter YOUR password followed by the pound key".

I'm not using this for my password.
Is there any way to change the message to "please enter the password
followed by the pound key"?

or is there another version of Authenticate() that I'm not aware of or
another way to prompt for a password?

Thanks,

Jerry
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Re: [asterisk-users] Authenticate() 11.21.0

2016-02-10 Thread Doug Lytle 
>>>  On Feb 10, 2016, at 9:20 AM, Jerry Geis ge...@pagestation.com wrote:

>>> or is there another version of Authenticate() that I'm not aware of or 
>>> another way to prompt for a password? 

As Steve said, use read.  This is a snippet from my dial plan

exten => s,1,Read(get-admin-password,enter-password,,,3,)
exten => s,n,Gotoif($["${LEN(${get-admin-password})}" < "1"]?5:3) 

Doug

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Re: [asterisk-users] Authenticate() 11.21.0

2016-02-10 Thread Steve Howes 

On 10/02/16 14:20, Jerry Geis wrote:

I am trying to use Authenticate() in the dialplan
for something other than "my password".
The message says "Please enter YOUR password followed by the pound key".

I'm not using this for my password.
Is there any way to change the message to "please enter the password 
followed by the pound key"?


or is there another version of Authenticate() that I'm not aware of or 
another way to prompt for a password?


READ() ?

Steve

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[asterisk-users] Unexpected termination of the call when pick up (res_pjsip)

2016-02-10 Thread Dmitriy Serov 

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 why asterisk suddenly decided to hangup i don't 
found :(


There are suggestions or strong belief about the reasons of such behavior?

Thanks.

Dmitriy.
[2016-02-10 22:58:17] VERBOSE[25442] res_pjsip_logger.c: <--- Received SIP 
response ( bytes) from UDP:83.143.192.141:5060 --->
SIP/2.0 183 Call progress
Via: SIP/2.0/UDP 
85.142.148.80:5060;rport=5060;branch=z9hG4bKPj1be0328e-fb97-426c-93fe-cf12e32f501a
Record-Route: 
Record-Route: 
From: "admin" 
;tag=0ea59f7e-817c-48a1-8e44-6e896322609a
To: ;tag=B955C4E4-606476-16E1127B
Call-ID: 5ac4642d-007b-4e29-908a-1b06417148c7
CSeq: 3711 INVITE
Contact: 
Supported: 100rel,timer,replaces,histinfo,precondition
User-Agent: CommuniGatePro-callLeg/5.4.10
Allow: 
INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Content-Type: application/sdp
Content-Length: 376

v=0
o=CGPLeg606476 1366433634 683216818 IN IP4 83.143.192.141
s=SIP Call
c=IN IP4 83.143.192.141
t=0 0
m=audio 60132 RTP/AVP 8 101
c=IN IP4 83.143.192.141
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcpping:F:1253794:125379478
m=video 60136 RTP/AVP 100
c=IN IP4 83.143.192.141
a=inactive
a=rtcpping:F:1253795:125379578

[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: Function 
session_inv_on_state_changed called on event TSX_STATE
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The state change 
pertains to the endpoint 'srv_d15140(PJSIP/srv_d15140-0255)'
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The inv session still 
has an invite_tsx (0x7ffdf039dc58)
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: There is no transaction 
involved in this state change
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The current inv state 
is EARLY
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: Source of transaction 
state change is RX_MSG
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: Received response
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: Response is 183 Call 
progress
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: Function 
session_inv_on_tsx_state_changed called on event TSX_STATE
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The state change 
pertains to the endpoint 'srv_d15140(PJSIP/srv_d15140-0255)'
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The inv session still 
has an invite_tsx (0x7ffdf039dc58)
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The UAC INVITE 
transaction involved in this state change is 0x7ffdf039dc58
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The current transaction 
state is Proceeding
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The transaction state 
change event is RX_MSG
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The current inv state 
is EARLY
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: Received response
[2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: Response is 183 Call 
progress
[2016-02-10 22:58:17] VERBOSE[7404][C-018e] app_dial.c: 
PJSIP/srv_d15140-0255 is making progress passing it to PJSIP/admin-0254
[2016-02-10 22:58:17] DEBUG[31024] res_pjsip_session.c: Applying negotiated SDP 
media stream 'audio' using audio SDP handler
[2016-02-10 22:58:17] DEBUG[31024] netsock2.c: Splitting '109.60.222.253' 
into...
[2016-02-10 22:58:17] DEBUG[31024] netsock2.c: ...host '109.60.222.253' and 
port ''.
[2016-02-10 22:58:17] DEBUG[31024] res_rtp_asterisk.c: Setting RTCP address on 
RTP instance '0x7ffddc198f58'
[2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Setting tx payload type 8 
based on m type on 0x7ffdc23b7320
[2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Setting tx payload type 0 
based on m type on 0x7ffdc23b7320
[2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Don't have a default tx 
payload type 96 format for m type on 0x7ffdc23b7320
[2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Setting tx payload type 97 
based on m type on 0x7ffdc23b7320
[2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Don't have a default tx 
payload type 2 format for m type on 0x7ffdc23b7320
[2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Setting tx payload type 18 
based on m type on 0x7ffdc23b7320
[2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Setting tx payload type 101 
based on m type on 0x7ffdc23b7320
[2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Copying payload 0 
(0x7ffddd7ac7a0) from 0x7ffdc23b7320 to 0x7ffddc199120
[2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Copying

[asterisk-users] looking for soft phone can be manged like Snom phones

2016-02-10 Thread Thomas 
Hi,

its easy to control Snom phones from outside by an http request
http://username:password@192.168.0.1/command.htm?key=KEYEVENT

http://wiki.snom.com/FAQ/Can_I_control_my_snom_phone_remotely

also its easy to configure that an Snom phone send me an http request
for example for an incomming call
http://my.serverIP/pbx/snomcom.php?mac=$mac=1401


I would need an softphone for Linux and/or Mac with same or similar 
functionality.


best regads
Thomas

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Re: [asterisk-users] Unexpected termination of the call when pick up (res_pjsip)

2016-02-10 Thread Trey Hilyard 
How are you initiating the call out to that server? Are you dialing from an
internal phone or doing it from the CLI? It looks like it is from an
internal extension, if I were 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  wrote:

> 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 why asterisk suddenly decided to hangup i don't
> found :(
>
> There are suggestions or strong belief about the reasons of such behavior?
>
> Thanks.
>
> Dmitriy.
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Re: [asterisk-users] Unexpected termination of the call when pick up (res_pjsip)

2016-02-10 Thread Dmitriy Serov 

The call initiated from internal extension.

I have made two test call:
Successful call from device on res_pjsip via endpoint on chan_sip: 
http://pastebin.com/LWeDYstj
Unsuccessful call from device on res_pjsip via endpoint on res_pjsip: 
http://pastebin.com/hepVb6Nu


And ones again i don't see anything that would make asterisk send BYE.

I would be grateful for any ideas.

11.02.2016 1:47, Trey Hilyard пишет:


How are you initiating the call out to that server? Are you dialing 
from an internal phone or doing it from the CLI? It looks like it is 
from an internal extension, if I were 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 > wrote:


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 why asterisk suddenly decided to hangup i
don't found :(

There are suggestions or strong belief about the reasons of such
behavior?

Thanks.

Dmitriy.
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