Re: [asterisk-users] 7965G sporadically not able to make calls via chan_sip

2018-05-10 Thread Marie Fischer 
Hi,

the tcpdump starts with a pretty standard INVITE sequence:

10.0.0.121 -- INVITE --> 10.0.0.3
10.0.0.121 <-- 401 Unauthorized -- 10.0.0.3 // asterisk gives nonce in 
WWW-Authenticate: header
10.0.0.121 -- ACK --> 10.0.0.3

After that, normally you would see a new INVITE from the phone with 
Authorization: header, but in your case the phone does not send this - although 
it is clearly reachable as indicated by the SIP OPTIONS dialogue.

In the Asterisk SIP debug, I see only the packets sent by Asterisk to the 
phone, but not the phone's responses. Did you do just 'sip set debug on' or 
something different?

Can you provide the same logs for a successful call?

Do incoming calls to the phone work when this happens?

-- 
BR,

marie

On 02.05.2018, at 20:23, John Kinsner  wrote:

> sometime during the past few upgrades on asterisk 13, my Cisco 7965G phones 
> are sporadically not able to make calls.  after a few seconds, they just play 
> a fast-busy tone.  I tried upgrading the 7965G OS from their original 
> (working for years) 9.4.2SR1 to 9.4.2SR3 and the behavior did not change.
> 
> they are talking via chan_sip on asterisk 13.19.0.  I cannot determine the 
> sporadic part, sometimes the call goes through fine with no configuration 
> changes or restarts/reboots on either end.
> 
> sip debug from asterisk:
> https://pastebin.com/Mmz9JsAP
> 
> tcpdump from pbx:
> https://pastebin.com/jRT9QJwq
> 
> sip.conf:
> [121]
> type=friend
> ;qualify=yes
> ;qualifyfreq=300
> host=dynamic
> context=extensions
> secret=MySecret
> nat=no
> callerid="MBR" <121>
> 
> 
> can anyone give me clues to troubleshoot?
> 
> 
> 
> 
> 
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> 
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Re: [asterisk-users] SIP Codec negotiation

2018-05-10 Thread Daniel Tryba 
On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote:
> I receive an INVITE/SDP containing:
> 
>   m=audio 11310 RTP/AVP 3 0 101
> 
> which I interpret as gsm, ulaw, rfc2833.
> 
> and I reply with an OK/SDP containing:
> 
>   m=audio 15884 RTP/AVP 0 3 101
> 
> which I interpret as ulaw, gsm, rfc2833.
> 
> How can I tell which codec was actually used for the call?

AFAIK this is undetermined. The callee can send either ulaw or gsm,
unless the caller wants to narrow it down to 1 codec, see
https://tools.ietf.org/html/rfc4317#section-2.2

Most of the time the callee will pick the first (so in this case ulaw).
But there are media gateways out there that choose g711[au] above "more
complex" codecs regardless order in SDP. My prefer PSTN provider will
always prefer alaw if offered since that will prevent transcoding on
their side if the call goes to ISDN/POTS, but AMR if the call goes to
VoLTE.


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[asterisk-users] SIP Codec negotiation

2018-05-10 Thread Steve Edwards 

I receive an INVITE/SDP containing:

m=audio 11310 RTP/AVP 3 0 101

which I interpret as gsm, ulaw, rfc2833.

and I reply with an OK/SDP containing:

m=audio 15884 RTP/AVP 0 3 101

which I interpret as ulaw, gsm, rfc2833.

How can I tell which codec was actually used for the call?

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] Passing parameter to Queue-called macro

2018-05-10 Thread Marie Fischer 
Hi,

maybe I am overlooking something, but channel variables should be thread safe, 
shouldn't they?

I am using the following (sorry, in ael):

macro dial-queue (number) {
Set(_ORIG_UNIQUEID=${UNIQUEID});
Queue(${number},rCt,,,${timeout},,set-dst-agent);
..
}

// the "context macro-..." things is an ael-specific workaround to get transfer 
working (macro sets context to app_queue_gosub_virtual_context)
context macro-set-dst-agent {
s => {
Noop(${ORIG_UNIQUEID});
(${ORIG_UNIQUEID},${MEMBERNAME});
}
}

macro add-current-call-agent (id,num) {
Set(ODBC_ADD_CURRENT_AGENT(${id},${num})=1);
return;
}

-- 

marie

On 08.05.2018, at 16:16, Stefan Viljoen  wrote:

> Hi all
> 
> I need to pass a parameter in a thread-safe manner to the Queue pickup
> macro. This is to know when (and who) picked up an incoming call to a queue
> and log that to my back-office system with a CURL to a HTTP endpoint.
> 
> However, the Queue application does not appear to allow passing of
> parameters to the called queue pickup macro.
> 
> E. g. non-working code is:
> 
> [queuetest]
> timeout = 60
> retry = 2
> member=>SIP/testnum
> 
> [macro-verdianswer]
> exten=>s,1,NoOp(Entering Verdi answer macro)
> exten=>s,n,NoOp(Value: ${ARG1})
> exten=>s,n,MacroExit
> 
> [incomingcontext]
> 
> exten=>tstqueue,1,NoOp(Incoming call for VerDi)
> same=>n,Set(curlResult=${SHELL(/usr/src/verdi/bash/verdiIncGetUUID.sh)})
> same=>n,Set(curlResultLength=${LEN(${curlResult})})
> same=>n,NoOp(Curl result for incoming call UUID from VerDi: ${curlResult})
> same=>n,Set(CDR(accountcode)=${curlResult})
> same=>n,Set(curIncAccCode=${curlResult},g)
> same=>n,Macro(VCRECORD,stefantestEXT${CALLERID(num)}ACC${CDR(accountcode)},$
> {EXTEN})
> same=>n,Queue(queuetest,trhc,,,60,,verdianswer(${curIncAccCode}))
> same=>n,Hangup()
> 
> This results, when executed, in:
> 
> [May  8 15:14:50] WARNING[20921]: app_macro.c:309 _macro_exec: No such
> context 'macro-verdianswer(2018050815141huzzu4
> ' for macro 'verdianswer(2018050815141huzzu4
> 
> How can one pass a paramter into the macro called by the Asterisk queue
> application on queue pickup?
> 
> Alternatively, how can a global variable or ASTDB entry be made thread safe
> to do the same?
> 
> Thank you
> 
> Stefan
> 
> 
> 
> 
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> 
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
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