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

[Marie Fischer]

Hi Stefan,

glad you got it solved.
Just to clarify, those are not global, but channel variables you are using - so 
they should be visible only to their respective channel (and child channels 
with inheritance).

Global variables are defined in a [globals] section in extensions.conf. 
(https://wiki.asterisk.org/wiki/display/AST/Global+Variables+Basics)

-- 
BR,

marie

On 11.05.2018, at 9:01, Stefan Viljoen <viljo...@verishare.co.za> wrote:

> Hi Marie
> 
> Thanks!
> 
> I was just worried about thread safety if I had to use a global variable, e.
> g. it might be set to a value by one call (since I'm using the same global
> for every incoming call to transfer the accountcode gotten from my HTTP
> endpoint to the same macro, and there can be several calls simultaneously
> all inserting HTTP-sourced values at more or less the same instant) and then
> another call is in such a state that it then reads this call's data - and
> never reads its logical "own" data. The classic concurrently accessed single
> variable issue.
> 
> Anyway, I've managed to solve this by declaring a variable in the main
> dialplan as inheritable and storing my back-office relevant GUID in there,
> then referencing that variable without the pre-prended _ in the macro:
> 
> E. g.
> 
> [verdianswer]
> exten=>s,n,NoOp(Lodging CDR accountcode: ${curIncAccCode} as an incoming
> call from ${numbersource} with VerDi and answered by ${MEMBERINTERFACE}...)
> exten=>s,n,MacroExit
> 
> [telkomin]
> .
> .
> .
> same=>n,Set(curlResult=${SHELL(/usr/src/verdi/bash/verdiIncGetUUID.sh)})
> same=>n,Set(_curIncAccCode=${curlResult})
> same=>n,Queue(stefantest,trhc,,,60,,verdianswer)
> 
> The above works just fine for doing what I want to do, e. g. pass a
> parameter from an Asterisk dialplan context into a queue-triggered "agent
> just answered in the queue" Asterisk macro.
> 
> Thanks for the reply!
> 
> Kind regards
> 
> Stefan
> -Original Message-
> From: Marie Fischer <ma...@vtl.ee> 
> Sent: Thursday, 10 May 2018 15:08
> To: viljo...@verishare.co.za; Asterisk Users Mailing List - Non-Commercial
> Discussion <asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] Passing parameter to Queue-called macro
> 
> Hi,
> 
> maybe I am overlooking something, but channel variables should be thread
> safe, shouldn't they?
> 
> I am using the following (sorry, in ael):
> 
> macro dial-queue (number) {
>   Set(_ORIG_UNIQUEID=${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 <viljo...@verishare.co.za> 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${CDR(accountco
>> de)},$
>> {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(2018050815141huzz

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

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

[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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[asterisk-users] dnsmgr - lots of entries for the same host

[Marie Fischer]

Hello everybody,

I am seeing a strange problem on Asterisk 1.8 with dnsmgr.

The number of entries in DNS Manager seems to be growing steadily and all are 
pointing to the sama host - a SIP trunk to a local provider, which uses SRV 
lookup.

So, when DNS manager refreshes, there are 6000+ messages like that:

ast_get_srv: SRV lookup for '_sip._udp.pbx..ee' mapped to host 
proxy1a.pbx..ee, port 5060
ast_get_srv: SRV lookup for '_sip._udp.pbx..ee' mapped to host 
proxy1a.pbx..ee, port 5060
ast_get_srv: SRV lookup for '_sip._udp.pbx..ee' mapped to host 
proxy1a.pbx..ee, port 5060
ast_get_srv: SRV lookup for '_sip._udp.pbx..ee' mapped to host 
proxy1a.pbx..ee, port 5060

'dnsmgr status' shows:

DNS Manager: enabled
Refresh Interval: 300 seconds
Number of entries: 6718

dnsmgr.conf is just 

[general]
enable=yes

Any ideas what could be wrong?

-- 
Greetings,


marie


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Re: [asterisk-users] Update peer IP address

[Marie Fischer]

On 14.09.2015, at 21:58, Sebastian Kemper  wrote:

> So I got rid of the firewall rule that opened the RTP ports. And then it
> dawned on me that I don't even need to open the 5060 port. The REGISTER
> requests established a UDP connection that the kernel's conntrack module
> was tracking anyway. The only issue was that the REGISTERs occurred only
> every 480s and the UDP connections were removed after 180s already.
> 
> So at first I raised net.netfilter.nf_conntrack_udp_timeout_stream to
> 500. That worked. But I didn't really want to raise the default. So
> instead I added "qualify=yes" to the dtag_inbound peer. Now asterisk is
> sending an OPTIONS request to Telekom every 120s (I raised the frequency
> from 60 to 120 by setting "qualifyfreq=120" under [general]), which
> keeps the connection open.

As far as I understand, raising the UDP session timeout (or lowering the 
REGISTER timeout, if possible) is actually the better solution. Most Telcos I 
know don't answer the OPTIONS request anyway and some might object to the 
traffic overhead.

-- 
marie
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Re: [asterisk-users] Strange Issue: asterisk deleted

[Marie Fischer]

On 26.11.2014, at 22:08, Antoine Megalla aa...@rocketmail.com wrote:
 The asterisk installation went fine but as soon as I start asterisk 
 executable it loads everything and then after the Ready line the process 
 gets killed and when I try to run it again i get: /usr/sbin/asterisk : 
 command not found
 
 I looked for asterisk in /usr/sbin using the commands ls and find and whereis 
 and it was not there.
 
 I know that the process is killed because when I start asterisk using the 
 command asterisk -c it starts and then it exits and the word killed is 
 wrote on the console.
 
 Ever time I copy a new executable to /usr/sbin either using cp command or 
 make install it gets deleted too.

Interesting problem, I'm quite curious what the cause is.

Are you 100% sure that the asterisk your are running is in /usr/sbin? Try 
'which asterisk' to see what your shell is running and/or start asterisk with a 
full path as /usr/sbin/asterisk -c.

You could also try renaming the binary to find out if indeed something kills 
Asterisk by name.

There's a tool called SystemTap which could give you information which process 
sent the SIGKILL:
https://sourceware.org/systemtap/
http://www.percona.com/blog/2014/07/18/systemtap-solves-phantom-mysqld-sigterm-sigkill-issue/

-- 

marie


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Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

[Marie Fischer]

On 22.11.2014, at 13:40, Yves A. yves...@gmx.de wrote:
 I have a really strange problem which is driving me crazy for days now.
 
 If I register my asterisk (tried all versions from 1.6 up to 13.x) with one 
 sip registrar,
 everything works... calls go out and call come in... no 32 seconds limit.
 
 but as soon as I configure another sip registration on another server, 
 outgoing
 calls  drop after 32 seconds.

Do a 'sip set debug on' and see what they (Asterisk and the registrar) are 
talking about just before the call drops.

-- 

marie


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Re: [asterisk-users] Voicemail message number off by one when using ODBC storage

[Marie Fischer]

... 'cause message file names start with 0 (msg.wav).

-- 

marie

On 05.10.2014, at 18:45, Leandro Dardini ldard...@gmail.com wrote:

 Hello,
 have you noticed the message num (VM_MSGNUM) is off by one?
 
 For example, I receive the following message:
 
 Just wanted to let you know you were just left a 0:03 long message (number 
 7)
 
 but in attach there is the msg0006.wav
 
 Leandro
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Re: [asterisk-users] how can queue agents choose which call to answer?

[Marie Fischer]

Thanks for your ideas. I set up a solution via AMI Redirect and it works nicely.

The only question now is queue metrics, as you also mentioned - the redirected 
calls get logged as ABANDON in the queue log. I could of course add a custom 
entry to the log via QueueLog function to show the call was actually 
redirected, but is there a way to disable/change the ABANDON log itself? It 
seems from this discussion FOP has the same problem: 
http://forum.fop2.com/1746-call-pickup-function-causes-dummy-entry-in-cdr-database/0

-- 

marie

On 23.09.2014, at 22:02, Scott Griepentrog sgriepent...@digium.com wrote:

 You can use any number of methods for redirecting a call from the queue to a 
 specific agent.  These include off the shelf products such as FOP or 
 iSymphony, or even something custom built that can display calls and direct 
 Asterisk (usually through AMI) to transfer the call to a new destination.
 
 However, you will need to be aware that your queue metrics may not count it 
 as a normally handled call, since the call is yanked out of the queue to 
 transfer directly to an agent via a separate tool.
 
 You may also want to look into building a custom queue-like solution through 
 ARI, using a Stasis application to manage callers on hold in waiting bridges, 
 and then delivering them to agents completely under control of your 
 application.  In this case you would need to create your own queue logging 
 data to your metrics solution, which would allow you to record calls 
 correctly even when transferred early.
 
 
 On Tue, Sep 23, 2014 at 1:41 PM, Michael Keuter li...@mksolutions.info 
 wrote:
 
 Am 23.09.2014 um 19:49 schrieb Marie Fischer ma...@vtl.ee:
 
  Hi everybody,
 
  I'm looking for a solution for the following scenario:
 
  • Asterisk queue
  • At peak hours, there will be more callers then queue members/agents, so 
  some callers will spend some time on hold
  • Agents should be able to choose which of the on hold calls to answer 
  instead of answering the next one in queue
 
  We already have a web interface where agents can see the callers on hold, 
  so the best solution would be if they could just click a callers number to 
  get his call. But I have not found a way to tell Asterisk to do something 
  to a call on hold in a queue.
 
  Priority queues are not really an option, as the agents will be deciding on 
  the fly which caller is more important.
 
  I am not really sure if queues are the correct solution for this problem. 
  However, we have existing statistics built for queue logs, so it would be 
  really nice if the solution was queue-based.
 
  Thanks for any thoughts,
 
  --
 
  marie
 
 
 Hello Marie,
 
 maybe FOP2  [1] is an option for you. There you can visually pick up a call 
 from a queue.
 It's not open source though.
 
 [1] http://www.fop2.com
 
 Michael
 
 http://www.mksolutions.info
 
 
 
 
 
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 Digium, Inc · Software Developer
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 direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029
 Check us out at: http://digium.com · http://asterisk.org
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Re: [asterisk-users] how can queue agents choose which call to answer?

[Marie Fischer]

... and to continue my thought, if nothing else is possible, would it be a Very 
Bad Idea to just delete the ABANDON log (queue_log goes to mysql via odbc) 
automatically after it's created? In h extension?

-- 

marie

On 05.10.2014, at 20:42, Marie Fischer ma...@vtl.ee wrote:

 Thanks for your ideas. I set up a solution via AMI Redirect and it works 
 nicely.
 
 The only question now is queue metrics, as you also mentioned - the 
 redirected calls get logged as ABANDON in the queue log. I could of course 
 add a custom entry to the log via QueueLog function to show the call was 
 actually redirected, but is there a way to disable/change the ABANDON log 
 itself? It seems from this discussion FOP has the same problem: 
 http://forum.fop2.com/1746-call-pickup-function-causes-dummy-entry-in-cdr-database/0
 
 -- 
 
 marie
 
 On 23.09.2014, at 22:02, Scott Griepentrog sgriepent...@digium.com wrote:
 
 You can use any number of methods for redirecting a call from the queue to a 
 specific agent.  These include off the shelf products such as FOP or 
 iSymphony, or even something custom built that can display calls and direct 
 Asterisk (usually through AMI) to transfer the call to a new destination.
 
 However, you will need to be aware that your queue metrics may not count it 
 as a normally handled call, since the call is yanked out of the queue to 
 transfer directly to an agent via a separate tool.
 
 You may also want to look into building a custom queue-like solution through 
 ARI, using a Stasis application to manage callers on hold in waiting 
 bridges, and then delivering them to agents completely under control of your 
 application.  In this case you would need to create your own queue logging 
 data to your metrics solution, which would allow you to record calls 
 correctly even when transferred early.
 
 
 On Tue, Sep 23, 2014 at 1:41 PM, Michael Keuter li...@mksolutions.info 
 wrote:
 
 Am 23.09.2014 um 19:49 schrieb Marie Fischer ma...@vtl.ee:
 
 Hi everybody,
 
 I'm looking for a solution for the following scenario:
 
 • Asterisk queue
 • At peak hours, there will be more callers then queue members/agents, so 
 some callers will spend some time on hold
 • Agents should be able to choose which of the on hold calls to answer 
 instead of answering the next one in queue
 
 We already have a web interface where agents can see the callers on hold, 
 so the best solution would be if they could just click a callers number to 
 get his call. But I have not found a way to tell Asterisk to do something 
 to a call on hold in a queue.
 
 Priority queues are not really an option, as the agents will be deciding on 
 the fly which caller is more important.
 
 I am not really sure if queues are the correct solution for this problem. 
 However, we have existing statistics built for queue logs, so it would be 
 really nice if the solution was queue-based.
 
 Thanks for any thoughts,
 
 --
 
 marie
 
 
 Hello Marie,
 
 maybe FOP2  [1] is an option for you. There you can visually pick up a 
 call from a queue.
 It's not open source though.
 
 [1] http://www.fop2.com
 
 Michael
 
 http://www.mksolutions.info
 
 
 
 
 
 --
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 -- 
 
 Scott Griepentrog
 Digium, Inc · Software Developer
 445 Jan Davis Drive NW · Huntsville, AL 35806 · US
 direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029
 Check us out at: http://digium.com · http://asterisk.org
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[asterisk-users] how can queue agents choose which call to answer?

[Marie Fischer]

Hi everybody,

I'm looking for a solution for the following scenario:

• Asterisk queue
• At peak hours, there will be more callers then queue members/agents, so some 
callers will spend some time on hold
• Agents should be able to choose which of the on hold calls to answer instead 
of answering the next one in queue

We already have a web interface where agents can see the callers on hold, so 
the best solution would be if they could just click a callers number to get his 
call. But I have not found a way to tell Asterisk to do something to a call on 
hold in a queue.

Priority queues are not really an option, as the agents will be deciding on the 
fly which caller is more important.

I am not really sure if queues are the correct solution for this problem. 
However, we have existing statistics built for queue logs, so it would be 
really nice if the solution was queue-based.

Thanks for any thoughts,

-- 

marie


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Re: [asterisk-users] POSSA CID Superfecta to query a CardDAV server ?

[Marie Fischer]

On 01.09.2014, at 11:42, Lukasz Sokol el.es...@gmail.com wrote:

 On 31/08/14 17:40, Marie Fischer wrote:
 
 Le 29 août 2014 14:30, Marie Fischer ma...@vtl.ee a écrit :
 
 we use OSX CardDAV server and its response is very slow, so we
 ended up syncing all the CardDAV contacts to MySQL via cron.
 Asterisk dialplan then runs a query defined in func_odbc.conf.
 
 
 I have an EGroupware server (egroupware.org), CardDAV is 'built in' there.
 What timing is the CID Superfecta / Asterisk expecting ?
 
 (I do not need to run CardDAV sync between servers, supposedly this should be
 much faster)

Well, you made me curious - wrote up a little perl script to do a filtered 
report by phone number. It takes 2-3 seconds to get a response from OSX server 
(Mavericks). Which sure is shorter then doing a full sync, but still longish. 
Would be interesting to know how long other servers take.

Now, for CID you would want this to run in your dial plan after the call comes 
in and before you Dial() your local extension. One ringtone is 5 seconds (1 sec 
tone, 4 sec silence), so it's actually not too bad (remember those analog 
caller ID boxes which got the caller ID between first and second ringtone?). 
Maybe you'd need to send Progress() or Ringing() back to the calling party.

-- 

marie



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Re: [asterisk-users] POSSA CID Superfecta to query a CardDAV server ?

[Marie Fischer]

On 29.08.2014, at 22:44, Olivier oza.4...@gmail.com wrote:
 Le 29 août 2014 14:30, Marie Fischer ma...@vtl.ee a écrit :
 
  Hi,
 
  we use OSX CardDAV server and its response is very slow, so we ended up 
  syncing all the CardDAV contacts to MySQL via cron. Asterisk dialplan then 
  runs a query defined in func_odbc.conf.
 
 
 What kind of carddav query did you send to your server ? Finding caller's 
 name from phone number, I presume ?
 
 Which client side tools did you then use ?
 
 I may be wrong but googling a bit, most examples of carddav  I found where 
 for syncing directories, not querying so your experience is very interesting.


Sorry to disappoint you, but we also do just sync, not lookup by number. I just 
spent an evening or two on this years ago, didn't go very deeply into CardDAV 
protocol. I suppose you'd want to look at REPORT request with filter 
(http://tools.ietf.org/html/rfc6352#section-10.5) and search for numbers that 
end with the same string as the phone number you have (remove country prefix if 
needed).

As for tools, I sent raw requests via Perl's LWP::UserAgent and HTTP::Request 
and parsed the returned *.vcf's with Text::vCard::Addressbook. Didn't find a 
nice CardDAV module/library back then.

-- 

marie


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Re: [asterisk-users] POSSA CID Superfecta to query a CardDAV server ?

[Marie Fischer]

Hi,

we use OSX CardDAV server and its response is very slow, so we ended up syncing 
all the CardDAV contacts to MySQL via cron. Asterisk dialplan then runs a query 
defined in func_odbc.conf.

-- 

marie

On 29.08.2014, at 15:03, Lukasz Sokol el.es...@gmail.com wrote:

 Hi,
 
 Would it be hard / anybody tried / any hints how / to add CardDAV server 
 query support to CID Superfecta ?
 
 Kind Regards,
 Lukasz
 
 
 
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[asterisk-users] queue stats last reset date/time

[Marie Fischer]

Hello everybody,

is there any way to find out when the queue stats ('queue show' / AMI action 
'QueueStatus') was last reset (by 'queue reset stats')? These counters would 
make much more sense if I knew what timeframe they cover. ;)

-- 

marie




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Re: [asterisk-users] Hangup cause 111 after call pickup

[Marie Fischer]

On 06.06.2013, at 15:05, Jonas Kellens jonas.kell...@telenet.be wrote:

 Hello,
 
 when picking up an incoming call from one ip phone on another ip phone, the 
 call terminates after about 5 to 10 seconds.
 
 When reading out the hangup cause variable in the h-extention of the 
 dialplan, the hangup cause seems to be 111.
 
 
 In the dialplan output, you can see that SIP-peer sipacc3 picks up the 
 incoming channel SipAgenT01-1454, and the call is answered. After 7 
 seconds, the conversation is terminated.
 
 [Jun  6 10:13:15] VERBOSE[21118] pbx.c: [Jun  6 10:13:15] -- Executing 
 [120@sub-pickup:25] Pickup(SIP/sipacc3-147c, 
 SIP/SipAgenT01-1454@PICKUPMARK) in new stack
 [Jun  6 10:13:15] VERBOSE[20788] app_queue.c: [Jun  6 10:13:15] -- 
 SIP/sipacc3-147c answered SIP/SipAgenT01-1454
 
 [Jun  6 10:13:22] VERBOSE[20788] pbx.c: [Jun  6 10:13:22] -- Executing 
 [h@pbx-routing:3] NoOp(SIP/SipAgenT01-1454, hangup cause = 111) in 
 new stack
 
 
 
 Questions :
 
 1. what can cause a hangup cause 111 ? What is the meaning of hangup cause 
 111 ?
 
 2. on voip-info.org I read 111 protocol error 500 Server internal error. 
 How can I fix this ?? Using Asterisk 1.8.12.2 on CentOS.

Hi Jonas,

when the calls is answered, do you have correct both-way audio as well?

Please enter sip set debug on on the Asterisk console and paste the output. 
It could also be helpful if you could paste your dialplan.

-- 
marie



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Re: [asterisk-users] Integration with skype

[Marie Fischer]

On 23.05.2013, at 12:57, bilal ghayyad bilmar...@yahoo.com wrote:

 There is no free channel to be used to have integration between asterisk and 
 skype? What is the software that I can use to send and receive chat messages 
 on skype network?

For voice calls, you could try Skype Connect, which is SIP - but needs a 
business account, so not free. http://www.skype.com/en/features/skype-connect/

Don't know about chat - but Skype chat is evil anyway.

-- 

marie




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Re: [asterisk-users] Stress testing Asterisk

[Marie Fischer]

On 21.05.2013, at 0:05, Tommy Cooper tomcoope...@yahoo.com wrote:

 Hi,
 I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is 
 generating are failing. I am trying to run Sipp on the same machine as 
 Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command.

Do you have a peer and extension configured for SIPP in your Asterisk 
configuration? You also needat least the -s extension_to_dial option on your 
sipp command line.
http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/
 has some simple instructions which should get you started.
If the calls still fail, Asterisk console output would be helpful.



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Re: [asterisk-users] Stress testing Asterisk

[Marie Fischer]

On 22.05.2013, at 16:18, Tommy Cooper tomcoope...@yahoo.com wrote:

 Thank you for your help I finally solved this issue. Is it possible that my 
 setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core 
 using 3.5 GHz, and 1Gb of RAM?

Easily, as long as you have no media :)

Use -sn uac_pcap instead of -sn uac to test with RTP (and watch your call count 
drop). Add recording (MixMonitor()) to your dialplan and watch the call count 
go down even more. ;)

A rough way to see if call quality is deteriorating would be to call your 
Asterisk box while the SIPP test is running and listen to some message played 
via Background().

 
 - Forwarded Message -
 From: Marie Fischer ma...@vtl.ee
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com 
 Sent: Wednesday, May 22, 2013 1:16 PM
 Subject: Re: [asterisk-users] Stress testing Asterisk
 
 
 On 21.05.2013, at 0:05, Tommy Cooper tomcoope...@yahoo.com wrote:
 
  Hi,
  I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is 
  generating are failing. I am trying to run Sipp on the same machine as 
  Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command.
 
 Do you have a peer and extension configured for SIPP in your Asterisk 
 configuration? You also needat least the -s extension_to_dial option on 
 your sipp command line.
 http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has
  some simple instructions which should get you started.
 If the calls still fail, Asterisk console output would be helpful.

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Re: [asterisk-users] debug strategy for one-way audio calls

[Marie Fischer]

On 04.05.2013, at 20:20, Olivier oza_4...@yahoo.fr wrote:
 Le 2 mai 2013 13:23, Marie Fischer ma...@vtl.ee a écrit :
 
 from time to time, we get so-called simplex / one-way audio calls, where one 
 party cannot hear the other. The only thing in common is that is does happen 
 with calls via SIP trunk, not ISDN and not internal calls. Nothing strange 
 in verbose and SIP logs. Could even be some weird intermittent firewall 
 issue I guess.
 
 Which audio flow is missing ? Inbound ?
 
 I suppose it should be easier to automatically detect missing inbound audio.

Not sure about older calls, but outbound was missing the last few times. We use 
call recording via MixMonitor and the recording has both flows, so I guess rtp 
debug would have shown both as well.

 Apart from logging all traffic 24/7 via tcpdump (not really convenient), can 
 you give me some ideas how to debug this kind of issue?
 Asterisk 1.8.11-cert10 on CentOS 6.3, if that matters.

-- 

marie




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[asterisk-users] debug strategy for one-way audio calls

[Marie Fischer]

Hello everybody,

from time to time, we get so-called simplex / one-way audio calls, where one 
party cannot hear the other. The only thing in common is that is does happen 
with calls via SIP trunk, not ISDN and not internal calls. Nothing strange in 
verbose and SIP logs. Could even be some weird intermittent firewall issue I 
guess.

Apart from logging all traffic 24/7 via tcpdump (not really convenient), can 
you give me some ideas how to debug this kind of issue?

Asterisk 1.8.11-cert10 on CentOS 6.3, if that matters.

-- 
Thanks,

marie




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Re: [asterisk-users] CDR unanswered setting

[Marie Fischer]

On 09.04.2013, at 10:33, Shanavaz E A shanava...@yahoo.com wrote:

  
 Hi,
  
 From asterisk 1.8, the CDR table is not logging the unanswered or extn busy 
 calls which hit while in the queue. I am talking about this setting in the 
 cdr.conf :
  
 ; In brief, this option controls the reporting of unanswered calls which only 
 have an A 
 ; party. Calls which get offered to an outgoing line, but are unanswered, are 
 still 
 ; logged, and that is the intended behaviour. (It also results in some B side 
 CDRs being
 ; output, as they have the B side channel as their source channel, and no 
 destination 
 ; channel.)
 ;unanswered = no
  
 I require this to find out which all extensions didnt respond to a call when 
 the call hit them while in the queue. This is working fine until 1.6. But I 
 tried in 1.8 and 11, but I am getting only one record in the CDR table even 
 after setting the value as
  
 unanswered = yes
  
 Is there anything more I have to do, to make it working? Kindly help me.

I don't know about 1.6, but 1.8 and 11 both behave this way for me, too, so 
it's probably by design.
You can (and probably should) use queue_log for information about unanswered 
calls, more specifically, the RINGNOANSWER and EXIT* events. 
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289009.html

queue_log logs the call's uniqueid, so you can use that to link it to CDR. And 
of course, life is much easier when both CDR and queue_log go to database.

-- 
marie
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Re: [asterisk-users] Connect to an outbound channel and dial a phone number??

[Marie Fischer]

On 09.04.2013, at 23:12, Thomas Perron thomas.per...@gmail.com wrote:

 This seems basic but something is missing.
  
  
 I dial from my cell phone to my DID and enter the context in extensions.conf
 I am hoping to cascade through the plan and successfully automatically dial 
 the 1444 number listed.
 But it fails.
   
 And, I dpon't know why?   Should I removed the Hangup application?
 Syntax issue somewhere?
  
 I have a good SIP registration with the vendor, voipvoip.
  
 Thanks in advance for any feedback...
  
  
  
 [incoming]
 exten = 5552530146,1,Answer()
 exten = 5552530146,n,Wait(1)
 exten = 5552530146,n,Playback(beep)
 exten = 5552530146,n,Goto(105,105,1)
 ;
 ;
 [105]
 exten = 105,1,Wait(2)
 exten = 105,n,Playback(hello-world)
 exten = 105,n,Dial(SIP/voipvoip/1444514)
 exten = 105,n,Hangup()
  
 console output ...
  
 -- Executing [5552530146@incoming:1] Answer(SIP/voipvoip.com-000f, 
 ) in new stack
 -- Executing [5552530146@incoming:2] Wait(SIP/voipvoip.com-000f, 
 1) in new stack
 -- Executing [5552530146@incoming:3] 
 Playback(SIP/voipvoip.com-000f, beep) in new stack
 -- SIP/voipvoip.com-000f Playing 'beep.alaw' (language 'en')
 -- Executing [5552530146@incoming:4] Goto(SIP/voipvoip.com-000f, 
 105,105,1) in new stack
 -- Goto (105,105,1)
 -- Executing [105@105:1] Wait(SIP/voipvoip.com-000f, 2) in new 
 stack
 -- Executing [105@105:2] Playback(SIP/voipvoip.com-000f, 
 hello-world) in new stack
 -- SIP/voipvoip.com-000f Playing 'hello-world.alaw' (language 'en')
 -- Executing [105@105:3] Dial(SIP/voipvoip.com-000f, 
 SIP/sip3.voipvoip.com/17037171624) in new stack
   == Using SIP RTP CoS mark 5
 -- Called SIP/sip3.voipvoip.com/1444514
 [Apr  9 16:07:11] WARNING[994]: chan_sip.c:4169 retrans_pkt: Retransmission 
 timeout reached on transmission 
 4dd167154ea52bd26d63a95a56aa9526@192.168.1.10:5060 for seqno 102 (Critical 
 Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
 Packet timed out after 32000ms with no response
 [Apr  9 16:07:11] WARNING[994]: chan_sip.c:4198 retrans_pkt: Hanging up call 
 4dd167154ea52bd26d63a95a56aa9526@192.168.1.10:5060 - no reply to our critical 
 packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
 -- SIP/sip3.voipvoip.com-0010 is circuit-busy
   == Everyone is busy/congested at this time (1:0/1/0)
 -- Executing [105@105:4] Hangup(SIP/voipvoip.com-000f, ) in new 
 stack
   == Spawn extension (105, 105, 4) exited non-zero on 
 'SIP/voipvoip.com-000f'
 Asterisk*CLI


Enter sip set debug on at the console and show us the output from the call 
attempt (you should get a log of your SIP traffic together with the normal 
console output).

-- 
marie


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Re: [asterisk-users] Asterisk Peaking and 91 Calls And not a Dime More!

[Marie Fischer]

On 09.04.2013, at 23:43, Nick Khamis sym...@gmail.com wrote:

 That's just it! Nothing! It just does not pass the 91 mark. There are
 no failed calls during the test:
 
  Successful call|0  |20802
  Failed call|0  |0
 
 It's locked on 91 calls. I think I have a channel limit or call limit
 thing set somewhere by accident?
 
 
 The SIPP Results
 
 
 -- Scenario Screen  [1-9]: Change
 Screen --
   Call-rate(length)   Port   Total-time  Total-calls  Remote-host
   10.0(0 ms)/1.000s   50602089.21 s20802 192.168.2.10:5060(UDP)

So you have total calls = 20802. Does this number grow over time?


   0 new calls during 0.000 s period  0 ms scheduler resolution
   0 calls (limit 100)Peak was 91 calls, after 9 s

IIRC, peak shows maximum concurrent calls.
What command line do you use to start SIPP? I see your call rate is 10 
calls/sec and maximum calls set to 100. Have you tried experimenting with 
increasing the call rate (-r command line parameter)? How long is the recording 
you are playing or have you set a call length for SIPP (-d command line option) 
- that is, how long are your calls?

SIPP generates just as many calls as specified - if you have 10 calls per sec, 
it's quite logical to have ~90 ongoing calls after 9 secs. If your recording is 
about 9 secs, then the first calls will end at that time and you will never 
have more than ~90 concurrent calls.

-- 
marie


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[asterisk-users] realtime peer w/ callbackextension does not register after 'sip reload'

[Marie Fischer]

Hello everybody,

I am having a problem with realtime SIP peers.
On Asterisk 1.8, I had SIP peers for external SIP providers configured in 
database and additional register lines in sip.conf so they would register.
Now I upgraded to Asterisk 11.3.0, partly because of the promised 
callbackextension feature for realtime peers 
(https://reviewboard.asterisk.org/r/1717/). Removed the 'register' lines from 
sip.conf. My peers register correctly when Asterisk is started or if I do 
'module unload chan_sip.so; module load chan_sip.so', but if I do 'sip reload', 
they stay in 'Unregistered' state forever.

*CLI sip show registry
Hostdnsmgr Username   Refresh State 
   Reg.Time 
xxx.xxx.xxx.xxx:5060  N    45 
Registered   Fri, 05 Apr 2013 05:37:02
1 SIP registrations.
*CLI 
*CLI sip reload
*CLI  Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found
  == Using SIP CoS mark 4
[Apr  5 05:37:59] NOTICE[16991]: chan_sip.c:5527 
register_realtime_peers_with_callbackextens: Created realtime peer 'peer' for 
registration
  == Parsing '/etc/asterisk/sip_notify.conf': Found
*CLI 
*CLI 
*CLI sip show registry
Hostdnsmgr Username   Refresh State 
   Reg.Time 
xxx.xxx.xxx.xxx:5060  N    60 
Unregistered  
1 SIP registrations.
*CLI 

Also, sip show peers shows the peer correctly after restart, but is empty 
after 'sip reload'.

If I add the register line back to sip.conf, I get 2 lines for the same peer 
(in 'sip show registry') and both show state = registered. Strange.

Tried to dig through the code in chan_sip.c and one difference seems to be in 
the register line created by build_peer() - it includes the peername 
(register = peer?user:secret@host/extension), whereas in my config file I had 
just register = user:secret@host/extension. I removed the peer part from the 
source and recompiled, and if I recall correctly the registration survived 
sip reload after that, but that's a hack, not a solution. :)

I found this bug: https://issues.asterisk.org/jira/browse/ASTERISK-20611, but I 
don't think that's my issue - anyway, it should be fixed by now, but I still 
had the same issue with 11.4.0-rc1.

Does anybody have experience with realtime peers registering using 
callbackextension? Does this problem seem like a configuration issue or should 
I file a bug report? 

Sorry if you read this twice, I am crossposting to http://forums.asterisk.org 
as I still haven't figured out the best place to get answers. ;)

-- 

marie




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[asterisk-users] sip set debug on output to file only (not to console)

[Marie Fischer]

Hello everybody,

I am trying to find an intermittent SIP error with one provider and thought the 
best first step would be to have sip set debug on for some days and check the 
logs.

Everything gets logged nicely, but the SIP log clutters up the console quite 
badly. Is it possible to have SIP debug log go only to the log file and not to 
the console?

My logger.conf:

console = notice,warning,error
messages = notice,warning,error
full = notice,warning,error,debug,verbose,dtmf,fax

On the console, I entered:

core set verbose 3
core set debug 0
sip set debug on

Thanks,

-- 

marie




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Re: [asterisk-users] sip set debug on output to file only (not to console)

[Marie Fischer]

On 29.03.2013, at 15:05, Doug Lytle supp...@drdos.info wrote:

 Marie Fischer wrote:
 full = notice,warning,error,debug,verbose,dtmf,fax
 
 You should have a log called full in:
 
 /var/log/asterisk

Sure I do and happy with that. :)

The point is, I also have my Asterisk console full of SIP messages and I asked 
if it was possible to switch those off.

-- 

marie


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