Re: [asterisk-users] set framing on dynamic interface DAHDI

2016-01-25 Thread Vinicius Fontes 
You can do it like this:

dynamic=eth,eth3/04:74:a1:00:05:8e/1,31,0
bchan=32-46,48-62
dchan=47

2016-01-22 23:22 GMT-02:00 Rafael dos Santos Saraiva :

> Hi
>
> I working with DAHDI Dynamic Interfaces using ethernet boards. I need set
> the framing to CCS, but the documentation of DAHDI not refer to it. My
> question is: there is a way to do this?
>
> *system.conf*
> dynamic=eth,enp0s8/00:00:00:00:00:01/0,31,0
> echocanceller=mg2,1-15,17-31
> bchan=1-15,17-31
> dchan=16
> alaw=1-15,17-31
>
> *dahdi_scan*
> [1]
> active=yes
> alarms=RED
> description=Dynamic 'eth' span at 'enp0s8/00:00:00:
> name=DYN/eth/enp0s8/00:0
> manufacturer=
> devicetype=DYN/eth/enp0s8/00:00:00:00:00:01/0
> location=
> basechan=1
> totchans=31
> irq=0
> type=digital-DYNAM
> syncsrc=0
> lbo=0 db (CSU)/0-133 feet (DSX-1)
> coding_opts=B8ZS,AMI,HDB3
> framing_opts=ESF,D4,CCS,CRC4
> coding=
> framing=CAS
>
>
> Thank's in advance.
>
>
>
> [image: Sua Foto]  Rafael S. Saraiva
> Porto Alegre - RS | Mobile:  (51) 8174-7956
> 
> 
>
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Re: [asterisk-users] CEL entries over ODBC several hours late (Vinicius Fontes)

2015-12-11 Thread Vinicius Fontes 
Since the issue seems to be table locking, why not take a shot with
PostgreSQL? It's way better and more robust than MySQL/MariaDB.

You should be able to create an additional DSN and output CEL to both
PostgreSQL and MariaDB.

2015-12-11 8:59 GMT-02:00 Stefan Viljoen :

> Hi Vinicius
>
> Thanks for replying.
>
> >Sorry for the probably obvious question, but it's better to cover all
> bases.
>
> >The DBMS is running on the same box as Asterisk is? If that's the case
> then
> maybe the DBMS is using too much CPU and starving Asterisk?
>
> I don't think so - I think I have a locking issue in the DB specifically
> with my stored procedure -and- Asterisk trying to access the CEL table
> simultaneously.
>
> Load averages appear normal even if the DB is running in the background.
> The
> autodestruct problem occurs whenever the CEL table (InnoDB, so should NOT
> be
> locked for inserts while being queried in MySQL "nolock" equivalent, but it
> IS) is busy being read by the SP and Asterisk tries to write into it.
>
> This then back-flows to the Asterisk and delays the H extension execution
> to
> such a degree that handsets get locked tight until the CEL table lock held
> by the SP is released and Asterisk can write to it again over ODBC.
>
> I understand that CEL logging and call hangup should be an atomic operation
> (billing related) from there though the question about why it apparently
> (from observed effects) is a synchronous operation - but considering it
> must
> be atomic, it does make sense I suppose to write CELs synchronously with
> call teardown.
>
> It just requires that you keep the CEL table in MariaDB (or whatever) in an
> "un-locked" state at all times - which makes it pretty hard since to have
> any practical use out of the CEL table you should be able to query it
> without locking it - which InnoDB in MariaDB is supposd to provide, but
> clearly does not in my type of setup here.
>
> Anyway, thanks for the reply.
>
> Regards
>
> Stefan
>
>
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Re: [asterisk-users] CEL entries over ODBC several hours late (Matthew Jordan)

2015-12-10 Thread Vinicius Fontes 
Sorry for the probably obvious question, but it's better to cover all bases.

The DBMS is running on the same box as Asterisk is? If that's the case then
maybe the DBMS is using too much CPU and starving Asterisk?

2015-12-10 12:57 GMT-02:00 Stefan Viljoen :

> Hi Matthew
>
> Thank you very much for the reply.
>
> I must have something seriously wrong somewhere else then - I retested now
> and the "apparent" effect is as  I describe but your info definitely
> contradicts that. But you're obviously correct.
>
> One more question - I've noted that if I run a combination of queries in
> the
> CEL backing DB (MariaDB) and the CEL table is locked, this severely affects
> the Asterisk instance - thousands of occurrences of
>
> chan_sip.c:4057 __sip_autodestruct: Autodestruct on dialog
> '6a9f5d3543b619655e07c81437373a32@172.17.12.3:5060' with owner
> SIP/3034-000207c8 in place (Method: BYE). Rescheduling destruction for
> 1
> ms
>
> appear in the CLI and users complain that if they hang up then they cannot
> make another call on the same SIP handset for several minutes.
>
> This is obviously because the dialplan gets delayed in the H extension, and
> cannot write to the CEL table, waiting for the MariaDB instance to clear
> the
> locks so it can write again. The above apparently comes from a watchdog
> process that watches how "fast" the H extension is and if it takes "too
> long" it forces the channels closed.
>
> Is this assumption correct?
>
> Addtionally, it seems that the writing of CEL in Asterisk is NOT async?
> E.g.
> it appears the thread that was running the conversation ALSO does the CEL
> writing / pushing to the CEL core as you describe in a synchronous manner.
>
> For 1.8, is this correct that CELs are synchronous, and do newer Asterisk
> versions do it async?
>
> E. g. my point being if there are major DB issues, it is quite a bit
> kryptonite to have that "spill back" into Asterisk and start blocking off
> users from calling out - wouldn't it be much better to simply have failed
> CEL writes just die in a distal thread instead of the main call thread for
> the channels running on that handset.
>
> Or am I completely misunderstanding things?
>
> Anyway, thanks for the reply. :)
>
> Kind regards,
>
> > Hi guys
> >
> > I'm running 1.8.32.3 with CEL logging over ODBC to MariaDB 5.5.41 on the
> > same Centos 7 machine.
> >
> > I've noticed that the CDR entries made are all in-time, e. g. the call
> will
> > take place and the CDR entry is immediately written into the CDR table in
> > the MariaDB database.
> >
> > However, CEL events for that CDR (which I need to process for a realtime
> > display feature in my dialer software) are always several hours after the
> > fact. E. g. I will make a call at 09:30, see the call immediate pop up in
> > the MariaDB CDR table, but only at about 15:15 that afternoon will I see
> > that call's CEL events come into the CEL table, from Asterisk I have
> > examined the `show processlist` in MariaDB exhaustively to establish this
> > fact.
> >
> > The system doesn't appear loaded, load average is about 1.1 - it's a
> > quad-coare HT Intel Xeon E3-1225 with 8GB of DRAM running on an SSD for
> > main
> > storage.
> >
> > The system processes about 30 000 calls every 8 hour day, and services 90
> > SIP phones.
> >
> > I can stop and restart the MariaDB instance for several minutes, when I
> > restart it it immediately picks up on the "slow" CELs from where it was
> > interrupted - more evidence that Asterisk is very slowly streaming the
> CELs
> > through. I thought MariaDB was the bottleneck, but apparently not?
> >
> > If I make test inserts from a script into the CEL table, all of them
> > complete so quickly a time indication does not even register for the
> query
> > in MariaDB. Simple test queries on the CEL table are also instant, not
> even
> > counting in the internal MariaDB query duration timer.
> >
> > Can anybody explain why this is that the CELs asterisk emits over ODBC
> are
> > so delayed? Are CELs intended NOT to be realtime?
> >
> > So, logically, Asterisk appears to be caching CELs to the tune of
> hundreds
> > of thousands of them at any given time - meaning if it is stopped (either
> > killed, or core stop gracefully'ed, or just "core stop now")  potentially
> > hundreds of thousands of CELs will just evaporate irretrivably.
> >
> > What can I do to mitigate this extremely slow populating of CELs over
> ODBC?
> >
> >
> Asterisk does not buffer CEL entries. If anything, it pushes the entries
> out to ODBC much more aggressively than what you would get with CDRs.
>
> An event is generated in Asterisk that corresponds to the CEL entry. That
> entry is pushed over a message bus (the 'event' message bus in 1.8 - 11;
> 'stasis' in 12+) and is picked up by the CEL core. The events are
> immediately sent to the registered backends, who also immediately write it
> out to the backend they support. In the case of ODBC, this

Re: [asterisk-users] Call forwarding in Asterisk

2015-09-03 Thread Vinicius Fontes 
You might want to use the Originate() application instead. Check its usage
by issuing the command 'core show application originate' on Asterisk CLI.

2015-09-03 9:09 GMT-03:00 Kantharuban Ruban :

> Hello Group,
>
> I have a requirement to dialout some external number, once the
> call is answered the same has to be forwarded to an Internal Queue.
>
> Please help me.
>
> I have tried calling with two SIP end point forwarding , even that is not
> working,
>
> My dial plan line is , Dial(SIP/19201/19202,300)
>
>
> --
> *Best regards,*
> *Ruban.S*
>
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Re: [asterisk-users] webrtc no audio

2015-08-28 Thread Vinicius Fontes 
I tested and it seems like I do have
https://issues.asterisk.org/jira/browse/ASTERISK-24146 but in a different
way. If I take more than 7s to answer the call, I don't get audio for a few
seconds (about 3), after that it works okay.



2015-08-28 10:43 GMT-03:00 Marek Červenka cerv...@fpf.slu.cz:

 are you sure you dont have this problem?
 https://issues.asterisk.org/jira/browse/ASTERISK-24146

 i'm now fighting with
 https://issues.asterisk.org/jira/browse/ASTERISK-24602

 Dne 27.8.2015 v 20:07 Vinicius Fontes napsal(a):

 I have it working now!

 *I had to install Asterisk 13 with PJSIP support.That's important, even if
 you won't use PJSIP.* Then I did this configuration, which is working
 fine under NAT:

 *sip.conf:*
 [6001]
 type=friend
 secret=REDACTED
 host=dynamic
 context=interno
 disallow=all
 ;allow=alaw,h263,h264,vp8
 allow=g722
 dtmf=auto
 videosupport=yes
 transport=ws,udp
 avpf=yes
 callerid=WebRTC 6001
 encryption=yes
 qualify=yes
 directmedia=no
 nat=force_rport,comedia
 icesupport=yes
 dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
 dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
 dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your
 DTLS cert file is
 dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your
 DTLS private key is
 dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when
 setting up DTLS

 *rtp.conf:*
 icesupport=true
 stunaddr=stun.l.google.com:19302

 *res_stun_monitor.conf:*
 stunaddr = stun.l.google.com:19302; Address of the STUN server to
 query.
 stunrefresh = 30

 2015-08-12 5:23 GMT-03:00 Marek Červenka cerv...@fpf.slu.cz:

 Dne 11.8.2015 v 12:18 Joshua Colp napsal(a):

 Vinicius Fontes wrote:

 I'm having the same issue! The difference in my case is Asterisk server
 has a public IPv4 and the browser is behind a single NAT.

 I'm forwarding my configuration below (which I posted previously on
 asterisk-users).

 How can we debug ICE negotiation?


 You have to do a packet capture, look at the exchange in Wireshark, and
 see how the negotiation flows. It requires a basic understanding of ICE.


 it looks like we are facing this problem
 https://issues.asterisk.org/jira/browse/ASTERISK-24146
 https://issues.asterisk.org/jira/browse/ASTERISK-24146 too
 if we use [] in sipml5 expert config To disable TURN/STUN to speedup
 ICE candidates gathering you can use an empty array. e.g. [].
 it works better




 --
 ---
 Marek Cervenka
 ===


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 ---
 Marek Cervenka
 ===


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Re: [asterisk-users] webrtc no audio

2015-08-27 Thread Vinicius Fontes 
I have it working now!

*I had to install Asterisk 13 with PJSIP support.That's important, even if
you won't use PJSIP.* Then I did this configuration, which is working fine
under NAT:

*sip.conf:*
[6001]
type=friend
secret=REDACTED
host=dynamic
context=interno
disallow=all
;allow=alaw,h263,h264,vp8
allow=g722
dtmf=auto
videosupport=yes
transport=ws,udp
avpf=yes
callerid=WebRTC 6001
encryption=yes
qualify=yes
directmedia=no
nat=force_rport,comedia
icesupport=yes
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your
DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your
DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting
up DTLS

*rtp.conf:*
icesupport=true
stunaddr=stun.l.google.com:19302

*res_stun_monitor.conf:*
stunaddr = stun.l.google.com:19302; Address of the STUN server to query.
stunrefresh = 30

2015-08-12 5:23 GMT-03:00 Marek Červenka cerv...@fpf.slu.cz:

 Dne 11.8.2015 v 12:18 Joshua Colp napsal(a):

 Vinicius Fontes wrote:

 I'm having the same issue! The difference in my case is Asterisk server
 has a public IPv4 and the browser is behind a single NAT.

 I'm forwarding my configuration below (which I posted previously on
 asterisk-users).

 How can we debug ICE negotiation?


 You have to do a packet capture, look at the exchange in Wireshark, and
 see how the negotiation flows. It requires a basic understanding of ICE.


 it looks like we are facing this problem
 https://issues.asterisk.org/jira/browse/ASTERISK-24146 too
 if we use [] in sipml5 expert config To disable TURN/STUN to speedup
 ICE candidates gathering you can use an empty array. e.g. [].
 it works better




 --
 ---
 Marek Cervenka
 ===


 --
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Re: [asterisk-users] webrtc no audio

2015-08-10 Thread Vinicius Fontes 
I'm having the same issue! The difference in my case is Asterisk server has
a public IPv4 and the browser is behind a single NAT.

I'm forwarding my configuration below (which I posted previously on
asterisk-users).

How can we debug ICE negotiation?


-- Forwarded message --
From: Vinicius Fontes vinic...@aittelecom.com.br
Date: 2015-07-27 13:54 GMT-03:00
Subject: No audio on SIP over WebRTC
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com


I'm following this tutorial (
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5) to
deploy WebRTC support but I'm having an issue with RTP when the WebRTC
softphone is behind NAT.

In my scenario, the Asterisk server is running a public IPv4, and the
softphone is behind NAT. I can register and make a call normally, but I
don't get any audio in neither way (Asterisk/softphone and
softphone/Asterisk). Using the very same config files but having the
softphone and Asterisk on the same network it works fine.

Any tips on how to solve this? Here's my relevant files.

*;sip.conf:*
[general]
udpbindaddr=0.0.0.0:5060
realm=10.201.0.106 ;replace with your Asterisk server public IP address or
host
transport=udp,ws,wss
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1

[6000]
host=dynamic
secret=mysecret
context=default
type=friend
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
qualify=yes

[6001]
host=dynamic
secret=mysecret
context=default
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass


*extensions.conf:*
[default]
exten = _6XXX,1,Dial(SIP/${EXTEN})


*rtp.conf:*
[general]
rtpstart=1
rtpend=2
icesupport=yes
stunaddr=stun.l.google.com:19302



2015-08-10 12:35 GMT-03:00 Joshua Colp jc...@digium.com:

 Marek Cervenka wrote:

 hello,

 i'm facing strange problem

 asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230
 person1 to person3 are behind different NATs
 audio devices double checked

 call from person1(chrome) to person2(chrome) works
 call from person1(chrome) to person 3(chrome) - no audio on both side
 (RTP flowing only in one direction)
 call from person2(chrome) to person 3(chrome) - no audio on both side
 (RTP flowing only in one direction)
 BUT
 call from person2(chrome) to person 3(Jitsi sip client) - works!

 any tips howto find the problem?


 You would need to look at the ICE negotiation to see if it tried and
 failed. After that would be looking at the DTLS negotiation. Asterisk
 console output could provide some information.

 --
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com  www.asterisk.org


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[asterisk-users] No audio on SIP over WebRTC

2015-07-27 Thread Vinicius Fontes 
I'm following this tutorial (
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5) to
deploy WebRTC support but I'm having an issue with RTP when the WebRTC
softphone is behind NAT.

In my scenario, the Asterisk server is running a public IPv4, and the
softphone is behind NAT. I can register and make a call normally, but I
don't get any audio in neither way (Asterisk/softphone and
softphone/Asterisk). Using the very same config files but having the
softphone and Asterisk on the same network it works fine.

Any tips on how to solve this? Here's my relevant files.

*;sip.conf:*
[general]
udpbindaddr=0.0.0.0:5060
realm=10.201.0.106 ;replace with your Asterisk server public IP address or
host
transport=udp,ws,wss
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1

[6000]
host=dynamic
secret=mysecret
context=default
type=friend
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
qualify=yes

[6001]
host=dynamic
secret=mysecret
context=default
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass


*extensions.conf:*
[default]
exten = _6XXX,1,Dial(SIP/${EXTEN})


*rtp.conf:*
[general]
rtpstart=1
rtpend=2
icesupport=yes
stunaddr=stun.l.google.com:19302
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Re: [asterisk-users] How to make asterisk work with remote mysql database?

2015-05-07 Thread Vinicius Fontes 
To install start script: make config
To install samples (this will overwrite all files in /etc/asterisk!): make
samples

I usually do this when I need to compile and install Asterisk:

./configure  make menuselect  make  make install  make config 
make samples

2015-05-07 14:44 GMT-03:00 Manish Jain manish.j...@roundabouttech.com:

 Hi,

 I am trying to make asterisk 11.7 work on Ubuntu 12 (amd64). I would like
 asterisk to not use the inbuilt sqlite database. Instead, I want it use a
 remote mysql database. Is this possible ? If yes, is there any good HowTo
 on this ?

 Running 'make install' installed my asterisk successfully, but no conf
 files were copied to /etc/asterisk. Nor was a service start script placed
 in /etc/init.d. Is there normal ? fixable?

 One additional problem is that, after installation, the CLI has no odbc
 command. Why could this be ? I suspect that for asterisk to work with
 remote mysql database, odbc functionality needs to be fully workable.

 Thanks for any help.

 Manish Jain

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Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

2015-04-08 Thread Vinicius Fontes 
Have you tried Asterisk 13? The bridging mechanism has been completely
rewritten on Asterisk 12, so there's no longer channel masquerading and
zombie channels. Might be worth a try.

2015-04-07 20:33 GMT-03:00 Alex Villací­s Lasso a_villa...@palosanto.com:

 El 07/04/15 a las 17:38, Alex Villací­s Lasso escribió:

  I am trying to collect enough information about an problem a client is
 having with its asterisk 11.17.0  x86_64. This issue was observed before in
 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution.

 Background: this client is a telemarketing call-center that generates
 outgoing calls with close to a hundred agents operating simultaneously in
 peak hours. The system uses asterisk with FreePBX 2.8. In order to generate
 the calls, I wrote a program that connects to Asterisk using the AMI
 protocol. This program expects the SIP agent extensions to be assigned as
 members of queues, of which there are about 20, as shown below:

 9007 has 0 calls (max unlimited) in 'random' strategy (5s holdtime, 68s
 talktime), W:0, C:581, A:260, SL:82.6% within 60s
Members:
   SIP/147 (ringinuse disabled) (dynamic) (On Hold) has taken 21 calls
 (last was 800 secs ago)
   SIP/417 (ringinuse disabled) (dynamic) (In use) has taken 77 calls
 (last was 708 secs ago)
   SIP/416 (ringinuse disabled) (dynamic) (In use) has taken 41 calls
 (last was 656 secs ago)
   SIP/408 (ringinuse disabled) (dynamic) (In use) has taken 50 calls
 (last was 789 secs ago)
No Callers

 The program runs queue show through AMI every few seconds. For each
 queue to be used in telemarketing, the program counts the number of members
 that are Not In Use. If at least one is found, it reads that many phone
 numbers from the database and uses the AMI Originate command on each one,
 as follows:

 Action: Originate
 Channel: Local/NN@from-internal
 Exten: 
 Context: from-internal
 Priority: 1
 Async: true
 ActionID: xxx

 Here, NN is the number read from the database and  is the
 queue extension in the FreePBX-created context that eventually runs the
 Queue() dialplan application for the corresponding queue. This causes the
 call to be connected between the outgoing number and the queue, and is then
 assigned to a queue member by Asterisk. The dialplan is configured to route
 NN through one of a series of SIP trunks using the outbound routes
 as configured by FreePBX.

 The issue is that although this strategy works correctly on the user's
 machine for a few days, we have been observing that eventually the
 application stops placing calls. The agents are all idle (all 90 to 100 of
 them), but the queue show command shows them to be In Use on all
 queues. Furthermore, in normal operation, the core show channels command
 shows at most one channel for each configured SIP client in the Up state,
 but when calls stop being placed, the same command reports multiple
 channels in the Up state, as follows (after sorting):

 Local/9757007441@from-internal-a447;2!macro-
 dialout-trunk!s!19!Up!Dial!SIP/5547740412/5557007441,300,
 !47740412!!!3!572!(None)!1428426012.169192
 Local/9759315789@from-internal-a456;1ZOMBIE!
 from-trunk-sip-5547740412!!1!Up!AppDial!(Outgoing
 Line)!9759315789!!!3!500!(None)!1428426084.169326
 Local/9759315789@from-internal-a456;2!macro-
 dialout-trunk!s!19!Up!Dial!SIP/5547740412/5559315789,300,
 !47740412!!!3!500!(None)!1428426084.169323
 SIP/104-00014c61!macro-dialout-trunk!s!19!Up!Dial!
 SIP/5547740413/0453511309468,300,!47740413!!!3!562!SIP/
 5547740413-00014c62!1428426022.169224
 SIP/110-00014c2b!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing
 Line)!110!!!3!590!(None)!1428425994.169124
 SIP/110-00014e4e!macro-dialout-trunk!s!19!Up!Dial!
 SIP/5547740413/5552114757,300,!47740413!!!3!92!(None)!1428426491.169760
 SIP/113-00014c8c!macro-dialout-trunk!s!19!Up!Dial!
 SIP/5547740420/0016499465293,300,!47740420!!!3!532!(None)!
 1428426052.169273
 SIP/114-00014ce6!macro-dialout-trunk!s!19!Up!Dial!
 SIP/5547740410/040,300,!47740410!!!3!430!(None)!1428426154.169384
 SIP/115-00014ea4!macro-dialout-trunk!s!19!Ring!Dial!
 SIP/5547740400/0059144681666,300,!47740400!!!3!10!(None)!
 1428426574.169850
 SIP/119-00014c26!macro-dialout-trunk!s!19!Up!Dial!
 SIP/5547740413/016255863252,300,!47740413!!!3!593!(None)!
 1428425991.169113
 SIP/119-00014d1a!macro-dialout-trunk!s!19!Up!Dial!
 SIP/5547740413/5552716011,300,!47740413!!!3!383!(None)!1428426201.169436
 SIP/119-00014d4d!macro-dialout-trunk!s!19!Up!Dial!
 SIP/5547740413/5556802622,300,!47740413!!!3!327!(None)!1428426257.169493
 SIP/120-00014d5e!macro-dial-one!s!37!Up!Dial!SIP/230,,
 trT!120!!!3!314!SIP/230-00014d5f!1428426270.169510
 SIP/121-00014c24!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing
 Line)!121!!!3!596!(None)!1428425988.169111
 SIP/122-00014b56!EjecutivoQUADS!93000!1!Up!AppQueue!(Outgoing
 Line)!122!!!3!677!(None)!1428425906.168693
 SIP/123-00014d53!macro-dialout-trunk!s!19!Up!Dial!

Re: [asterisk-users] OpenVZ with asterisk 13

2015-04-07 Thread Vinicius Fontes 
I have several large customers (200+ extensions) running on vSphere without
issue. Not sure about OpenVZ, thought.

2015-04-07 11:36 GMT-03:00 Mitul Limbani mi...@enterux.in:

 Show him this freaking thread, or else ask him to prove it otherwise.

 We all here have decades of exp dealing with asterisk.

 Mitul
 On 07-Apr-2015 7:27 PM, Ikka Tirtawidjaja ikka.ti...@gmail.com wrote:

 Dear Mitul,

 I already told my boss about it, I really want a single box, no virtual,
 but my boss insist.
 He said that openvz use less resource then KVM (or other virtual for
 cloud).
 I really need a solid analysis to argue with him.

 On the other hand, dahdi   cannot be installed in openvz virtual server.

 I dont have any experience with openvz at all.

 Thx,

 On Tue, Apr 7, 2015 at 8:47 PM, Ikka Tirtawidjaja ikka.ti...@gmail.com
 wrote:

 Dear all,

 Is anyone has experience making Asterisk server with virtual server
 OPEN-VZ (in proxmox 3.4 box) ?

 My boss want to build a production server with it, and it will have +/-
 300 sip user (concurrent call maybe  150 call)

 Is it good to go, or not ?

 I really hope someone who have experience with it  willing to share with
 me...

 Thanks in advance...


 Best Regards,


 Ikka - Jakarta, Indonesia





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[asterisk-users] CDR dst value null after attended transfer

2015-03-26 Thread Vinicius Fontes 
I'm having an issue with CDR. Basically, I expect to have all legs of a
call having the same linkedid and differing only by the sequence value.
That does happen, but I'm getting null dst values after doing an attended
transfer.

I'm not sure if this is a bug or I'm doing something wrong. I'm running
Asterisk 13.2.0.

Here's the console log, step by step:

First, I receive a call from 5491549116 on extension 7051 (DID 5421047051):

[Mar 26 12:11:04]   == Using SIP RTP TOS bits 184
[Mar 26 12:11:04]   == Using SIP RTP CoS mark 5
[Mar 26 12:11:04] -- Executing [5421047051@restrito:1]
Goto(SIP/pabx-e1-0252, interno,7051,1) in new stack
[Mar 26 12:11:04] -- Goto (interno,7051,1)
[Mar 26 12:11:04] -- Executing [7051@interno:1]
Macro(SIP/pabx-e1-0252, stdexten,7051,SIP/7051) in new stack
[Mar 26 12:11:04] -- Executing [s@macro-stdexten:1]
NoOp(SIP/pabx-e1-0252, STDEXTEN: Arg1 = 7051   Arg2 = SIP/7051
Arg3 = ) in new stack
[Mar 26 12:11:04] -- Executing [s@macro-stdexten:2]
Dial(SIP/pabx-e1-0252, SIP/7051,45,tT) in new stack
[Mar 26 12:11:04]   == Using SIP RTP TOS bits 184
[Mar 26 12:11:04]   == Using SIP RTP CoS mark 5
[Mar 26 12:11:04] -- Called SIP/7051
[Mar 26 12:11:05] -- SIP/7051-0253 is ringing
[Mar 26 12:11:11] -- SIP/7051-0253 answered SIP/pabx-e1-0252
[Mar 26 12:11:11] -- Channel SIP/pabx-e1-0252 joined
'simple_bridge' basic-bridge b1c97b75-bd5f-4762-96dd-7aa68c472827
[Mar 26 12:11:11] -- Channel SIP/7051-0253 joined 'simple_bridge'
basic-bridge b1c97b75-bd5f-4762-96dd-7aa68c472827

Now, extension 7051 places the call on hold and calls 7003, who answers:

[Mar 26 12:11:17] -- Started music on hold, class 'default', on channel
'SIP/pabx-e1-0252'
[Mar 26 12:11:20]   == Using SIP RTP TOS bits 184
[Mar 26 12:11:20]   == Using SIP RTP CoS mark 5
[Mar 26 12:11:20] -- Executing [7003@ddi:1] Macro(SIP/7051-0254,
stdexten,7003,SIP/7003) in new stack
[Mar 26 12:11:20] -- Executing [s@macro-stdexten:1]
NoOp(SIP/7051-0254, STDEXTEN: Arg1 = 7003   Arg2 = SIP/7003   Arg3 =
) in new stack
[Mar 26 12:11:20] -- Executing [s@macro-stdexten:2]
Dial(SIP/7051-0254, SIP/7003,45,tT) in new stack
[Mar 26 12:11:20]   == Using SIP RTP TOS bits 184
[Mar 26 12:11:20]   == Using SIP RTP CoS mark 5
[Mar 26 12:11:20] -- Called SIP/7003
[Mar 26 12:11:20] -- SIP/7003-0255 is ringing
[Mar 26 12:11:25] -- SIP/7003-0255 answered SIP/7051-0254
[Mar 26 12:11:25] -- Channel SIP/7051-0254 joined 'simple_bridge'
basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2
[Mar 26 12:11:25] -- Channel SIP/7003-0255 joined 'simple_bridge'
basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2


Then, extension 7051 transfers the call to 7003, who hangs up after a few
seconds:

[Mar 26 12:11:32] -- Channel SIP/pabx-e1-0252 left 'simple_bridge'
basic-bridge b1c97b75-bd5f-4762-96dd-7aa68c472827
[Mar 26 12:11:32] -- Channel SIP/7051-0254 left 'simple_bridge'
basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2
[Mar 26 12:11:32] -- Channel SIP/pabx-e1-0252 swapped with
SIP/7051-0254 into 'simple_bridge' basic-bridge
f4fb9d99-24b9-4d3c-9b63-41a1b84484b2
[Mar 26 12:11:32] -- Stopped music on hold on SIP/pabx-e1-0252
[Mar 26 12:11:32] -- Channel SIP/7051-0253 left 'simple_bridge'
basic-bridge b1c97b75-bd5f-4762-96dd-7aa68c472827
[Mar 26 12:11:32]   == Spawn extension (macro-stdexten, s, 2) exited
non-zero on 'SIP/7051-0254' in macro 'stdexten'
[Mar 26 12:11:32]   == Spawn extension (ddi, 7003, 1) exited non-zero on
'SIP/7051-0254'
[2015-03-26 12:11:32] WARNING[1561][C-015c]: channel.c:5070 ast_write:
Codec mismatch on channel SIP/pabx-e1-0252 setting write format to slin
from alaw native formats (alaw)
[Mar 26 12:11:40] -- Channel SIP/pabx-e1-0252 left 'simple_bridge'
basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2
[Mar 26 12:11:40]   == Spawn extension (macro-stdexten, s, 2) exited
non-zero on 'SIP/pabx-e1-0252' in macro 'stdexten'
[Mar 26 12:11:40]   == Spawn extension (interno, 7051, 1) exited non-zero
on 'SIP/pabx-e1-0252'
[Mar 26 12:11:40] -- Channel SIP/7003-0255 left 'simple_bridge'
basic-bridge f4fb9d99-24b9-4d3c-9b63-41a1b84484b2

So far so good, except that when I check the CDR lines generated, this is
what I get:

mysql select calldate, uniqueid, linkedid, sequence, src, dst, duration,
disposition, channel, dstchannel from cdr where uniqueid = '1427382664.963';
+-+++--++--+--+-+--+---+
| calldate| uniqueid   | linkedid   | sequence | src
 | dst  | duration | disposition | channel  | dstchannel
 |
+-+++--++--+--+-+--+---+
|