[asterisk-users] asterisk 13.9 with PJSIP -rejects with 488 Not Acceptable Here on invite with SRTP

2016-06-09 Thread Yaron Nachum 
Hi Everyone,
I am trying to setup an Audio Call from firefox WebRTC to Asterisk. The
Flow is:
PC -> SIPoWS -> KAMAILIO -> SIPoUDP -> ASTERISK
Regular call (no srtp)  works fine. However when I setup SRTP the asterisk
replies with 488 Not Acceptable Here.
I followed the Secure Calling Tutorial, but nothing seems to solve the
issue.

Below you can see the endpoint configuration + debug output.
Any help would be appreciated.

[acme]
type=endpoint
transport=transport-udp
context=app-router
disallow=all
allow=alaw
allow=ulaw
aors=acme
direct_media=no
media_encryption=dtls
dtls_verify=no
dtls_cert_file = /etc/asterisk/keys/asterisk.pem
dtls_private_key = /etc/asterisk/keys/asterisk.pem
dtls_setup=actpass
use_avpf=yes
ice_support=yes
media_use_received_transport=yes



<--- Received SIP request (2019 bytes) from UDP:10.25.133.209:5064 --->
INVITE sip:001...@ipcentrex.bezeq.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.25.133.209:5064;branch=z9hG4bK258keq2010n19mk406d1.1
Via: SIP/2.0/TCP 147.235.160.240:443
;branch=z9hG4bKd954.785f69fdfe198df7cabc66c0132a6b4c.0
Max-Forwards: 68
To: 
From: "1000" ;tag=1800vui39b
Call-ID: abars16ecm7s7icq4asq
CSeq: 3488 INVITE
Contact: 
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY
Content-Type: application/sdp
Supported: outbound
User-Agent: SIP.js/0.7.3
Content-Length: 1366

v=0
o=mozilla...THIS_IS_SDPARTA-46.0.1 4701336699161149943 0 IN IP4
10.25.133.241
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256
A8:CD:3D:44:3E:98:38:4F:3C:92:B7:05:B0:2B:91:0F:0F:39:7A:49:1F:8B:FB:26:18:1B:26:16:6B:2A:9C:03
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 20582 UDP/TLS/RTP/SAVPF 109 9 0 8
c=IN IP4 10.25.133.241
a=candidate:0 1 UDP 2122187007 147.235.159.2 58553 typ host
a=candidate:2 1 UDP 2122121471 2002:93eb:9f02::93eb:9f02 58554 typ host
a=candidate:4 1 UDP 2122252543 10.2.0.15 58555 typ host
a=candidate:0 2 UDP 2122187006 147.235.159.2 58556 typ host
a=candidate:2 2 UDP 2122121470 2002:93eb:9f02::93eb:9f02 58557 typ host
a=candidate:4 2 UDP 2122252542 10.2.0.15 58558 typ host
a=candidate:5 1 UDP 1686052863 62.219.92.9 58555 typ srflx raddr 10.2.0.15
rport 58555
a=candidate:5 2 UDP 1686052862 62.219.92.9 58558 typ srflx raddr 10.2.0.15
rport 58558
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=fmtp:109 maxplaybackrate=48000;stereo=1
a=ice-pwd:446b46366a44a35b757dbec6a85a06a7
a=ice-ufrag:8d8969e7
a=mid:sdparta_0
a=msid:{60af682c-6e6d-4ad9-a165-02ce89d3ca8a}
{37d6a8a7-78c0-42b4-b52e-cdec5b076b1f}
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=setup:actpass
a=ssrc:2150808732 cname:{4c9490b5-d183-4f6f-954a-56b87f361da2}

[Jun  9 17:04:18] DEBUG[13281]: pjproject:0 :sip_endpoint.c
Distributing rdata to modules: Request msg INVITE/cseq=3488
(rdata0x7f0908008b28)
[Jun  9 17:04:18] DEBUG[13281]: netsock2.c:172 ast_sockaddr_split_hostport:
Splitting '10.25.133.209' into...
[Jun  9 17:04:18] DEBUG[13281]: netsock2.c:226 ast_sockaddr_split_hostport:
...host '10.25.133.209' and port ''.
[Jun  9 17:04:18] DEBUG[13281]: res_pjsip_endpoint_identifier_ip.c:113
ip_identify_match_check: Source address 10.25.133.209:5064 does not match
identify 'KAMnet_TST'
[Jun  9 17:04:18] DEBUG[13281]: res_pjsip_endpoint_identifier_ip.c:113
ip_identify_match_check: Source address 10.25.133.209:5064 does not match
identify 'KAMnet_CBS'
[Jun  9 17:04:18] DEBUG[13281]: res_pjsip_endpoint_identifier_ip.c:113
ip_identify_match_check: Source address 10.25.133.209:5064 does not match
identify 'clacli_j5'
[Jun  9 17:04:18] DEBUG[13281]: res_pjsip_endpoint_identifier_ip.c:113
ip_identify_match_check: Source address 10.25.133.209:5064 does not match
identify 'clacli_j6'
[Jun  9 17:04:18] DEBUG[13281]: res_pjsip_endpoint_identifier_ip.c:113
ip_identify_match_check: Source address 10.25.133.209:5064 does not match
identify 'clacli_j7'
[Jun  9 17:04:18] DEBUG[13281]: res_pjsip_endpoint_identifier_ip.c:108
ip_identify_match_check: Source address 10.25.133.209:5064 matches identify
'acme'
[Jun  9 17:04:18] DEBUG[13281]: res_pjsip_endpoint_identifier_ip.c:143
ip_identify: Retrieved endpoint acme
[Jun  9 17:04:18] DEBUG[13281]: pjproject:0 :tsx0x7f090001f
..Transaction created for Request msg INVITE/cseq=3488 (rdata0x7f0908008b28)
[Jun  9 17:04:18] DEBUG[13281]: pjproject:0 :tsx0x7f090001f
.Incoming Request msg INVITE/cseq=3488 (rdata0x7f0908008b28) in state Null
[Jun  9 17:04:18] DEBUG[13281]: pjproject:0 :tsx0x7f090001f
..State changed from Null to Trying, event=RX_MSG
[Jun  9 17:04:18] DEBUG[13281]: pjproject:0 :dlg0x7f0900010
...Transaction tsx0x7f090001f8e8 state changed to Trying
[Jun  9 17:04:18] DEBUG[13281]: pjproject:0 :dlg0x7f0900010 .UAS
dialog created
[Jun  9 17:04:18] DEBUG[13281]: pjproject:0 :dlg0x7f0900010
.Module mod-invite added

Re: [asterisk-users] RES: Can I use PJSIP_HEADER to read the SIP 183 message header?

2015-07-19 Thread Yaron Nachum 
One way to do it is to use Transfer. This will cause the callee to send 302
redirect to the caller. The caller then will jump to the extension
specified in the contact. You will have to dial again to the callee in the
new extension.

This solution will increase the traffic on your asterisk and you have to be
careful from loops.



‫בתאריך יום ו׳, 10 ביולי 2015 ב-21:37 מאת ‪Rodrigo Pimenta Carvalho‬‏ ‪
pime...@inatel.br‬‏:‬

 Ok Mark Michelson.

 Thank you very much! You answer tells me that I was in the wrong path
 trying to access information from SIP 183 message.

 I need to find a way to let the callee pass information/data to the
 caller, even before accepting the call. That is, send data during the
 ringing time. And in my case, there will be more than one callee ringing at
 same time. As ASTERISK will not forward each SIP 183 message to the caller,
 I intend to get data from callees in dialplan by some another way before
 the call being accepted.

 1- Is there any way to do that?

 2 - SIP MESSAGE, if sent by the calle, enters the dialplan?

  Any hint will be very helpful!

  Best regards.



 RODRIGO PIMENTA CARVALHO
 Inatel Competence Center
 Software
 Ph: +55 35 3471 9300 (Brasil)
 
 De: asterisk-users-boun...@lists.digium.com [
 asterisk-users-boun...@lists.digium.com] em Nome de Mark Michelson [
 mmichel...@digium.com]
 Enviado: sexta-feira, 10 de julho de 2015 15:14
 Para: Asterisk Users Mailing List - Non-Commercial Discussion
 Assunto: Re: [asterisk-users] Can I use PJSIP_HEADER to read the SIP 183
 message header?

 On 07/10/2015 11:53 AM, Rodrigo Pimenta Carvalho wrote:
  Hi.
 
  The ASTERISK wiki has a page showing the function PJSIP_HEADER().
 However, it doesn't explain if such function works only over SIP INVITE
 messages or if it can be use, for example, to read headers from others
 types of SIP messages too.
 
  So, can I use PJSIP_HEADER to read the SIP 183 message header?
 
  Any hint will be very helpful!
 
  Best regards.
 
 
  RODRIGO PIMENTA CARVALHO
  Inatel Competence Center
  Software
  Ph: +55 35 3471 9200 RAMAL 979 (Brasil)
 Unfortunately, PJSIP_HEADER() cannot be used on responses because SIP
 responses do not enter the dialplan.

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Re: [asterisk-users] Dynamic Music on Hold

2015-02-24 Thread Yaron Nachum 
Thanks Carlos,
I have created the table and changed the extconfig to :
musiconhold = mysql,asterisk,musiconhold

It works fine.

Yaron

On Mon, Feb 23, 2015 at 6:57 PM, Carlos Chavez cur...@telecomabmex.com
wrote:

 On 2/23/15 3:03 AM, Yaron Nachum wrote:

 Hello everyone,
 I am trying to activate Music On Hold using DB on Asterisk 13.
 It works fine but in order to use new Music On hold definitions I have to
 reload the moh module.

 - The following is my configuration in extconfig.conf - I added the
 following line:
  musiconhold.conf = mysql,asterisk,bit_ast_config

 - The following is the table in the database:
 mysql select * from bit_ast_config;
 +++-++--
 -+---+---+--+
 | id   | cat_metric | var_metric | commented | filename  |
 category | var_name  | var_val |
 +++-++--
 -+---+---+--+
 |  2   | 0 | 0   | 0  |
 musiconhold.conf | yaron  | directory | moh   |
 |  3   | 0 | 0   | 0  |
 musiconhold.conf | yaron  | mode  | files  |
 | 10  | 0 |   0 | 0|
 musiconhold.conf | yaron1| directory | moh |
 | 11  | 0 |   0 | 0|
 musiconhold.conf | yaron1| mode  | files|
 +++-++--
 -+---+---+--+


 Is there a way to do automatically add new moh definitions without
 reloading the moh module?
 Thanks,
 Yaron.

 You actually want to use the realtime database and not the
 static.  With the realtime database all changes will take effect
 immediately.  The following link explains the difference between realtime
 and static:

 https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration

 Here is the structure I use:

 CREATE TABLE `musiconhold` (
   `name` varchar(80) COLLATE utf8_unicode_ci NOT NULL,
   `directory` varchar(255) COLLATE utf8_unicode_ci NOT NULL DEFAULT '',
   `application` varchar(255) COLLATE utf8_unicode_ci NOT NULL DEFAULT '',
   `mode` varchar(80) COLLATE utf8_unicode_ci NOT NULL DEFAULT '',
   `digit` char(1) COLLATE utf8_unicode_ci NOT NULL DEFAULT '',
   `sort` varchar(16) COLLATE utf8_unicode_ci NOT NULL DEFAULT '',
   `format` varchar(16) COLLATE utf8_unicode_ci NOT NULL DEFAULT '',
   PRIMARY KEY (`name`)
 ) ENGINE=MyISAM DEFAULT CHARSET=utf8 COLLATE=utf8_unicode_ci;

  -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52
 (55)9116-91161


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[asterisk-users] Dynamic Music on Hold

2015-02-23 Thread Yaron Nachum 
Hello everyone,
I am trying to activate Music On Hold using DB on Asterisk 13.
It works fine but in order to use new Music On hold definitions I have to
reload the moh module.

- The following is my configuration in extconfig.conf - I added the
following line:
 musiconhold.conf = mysql,asterisk,bit_ast_config

- The following is the table in the database:
mysql select * from bit_ast_config;
+++-++---+---+---+--+
| id   | cat_metric | var_metric | commented | filename|
category | var_name  | var_val |
+++-++---+---+---+--+
|  2   | 0 | 0   | 0  |
musiconhold.conf | yaron  | directory | moh |
|  3   | 0 | 0   | 0  |
musiconhold.conf | yaron  | mode  | files  |
| 10  | 0 |   0 | 0  |
musiconhold.conf | yaron1| directory | moh |
| 11  | 0 |   0 | 0  |
musiconhold.conf | yaron1| mode  | files  |
+++-++---+---+---+--+


Is there a way to do automatically add new moh definitions without
reloading the moh module?
Thanks,
Yaron.
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[asterisk-users] Fwd: Asterisk pjsip auto dtmf mode

2015-01-16 Thread Yaron Nachum 
Hello Asterisk Users,
I have been looking for similar auto dtmf mode implementation on pjsip, but
didn't see it coming, so I decided to give it a try.
My basic plan was to do it as simple as possible with minimum changes
because I am not familiar with all Asterisk code. My idea is to use rfc4733
settings, but when going over the codecs check if telephone-event appear
and if not set the dtmf mode to inband on rtp instance.
I would appreciate if someone would look at what I did and see if I didn't
do stupid things. If you think this is something you would like to add to
one of the next releases I am willing to help - add the additional dtmf
mode ...
I based my development on 13.1.0. The following are my changes:

In res/res_pjsip_sdp_rtp.c (I added session_media to get_codecs and used it
in order to update dtmf settings on rtp instance when telephone-event is
not included in the sdp):
150:
static void get_codecs(struct ast_sip_session *session, const struct
pjmedia_sdp_media *stream, struct ast_rtp_codecs *codecs, struct
ast_sip_session_media *session_media)
159:
char fmt_param[256];
int tel_event = 0;
177:
ast_copy_pj_str(name, rtpmap-enc_name, sizeof(name));
if (strcmp(name,telephone-event) == 0) {
tel_event++;
}
202:
}
if (tel_event==0) {
ast_rtp_instance_dtmf_mode_set(session_media-rtp,
AST_RTP_DTMF_MODE_INBAND);
}
/* Get the packetization, if it exists */
241:
get_codecs(session, stream, codecs, session_media);

In res/res_pjsip_session.c (Just activated DSP also on RFC dtmf mode - I
didn't find a way to test the rtp instance dtmf settiings because
session_media pointer is not there. Any advice for doing so would be
appreciated):
1062:
if (endpoint-dtmf == AST_SIP_DTMF_INBAND || endpoint-dtmf ==
AST_SIP_DTMF_RFC_4733) {
dsp_features |= DSP_FEATURE_DIGIT_DETECT;
}

In channels/chan_pjsip.c (1 change similar to the above, and 2 more changes
to send inband dtmf when rtp instance dtmf settings is inband)
543:
   if (session-endpoint-dtmf == AST_SIP_DTMF_INBAND ||
session-endpoint-dtmf == AST_SIP_DTMF_RFC_4733) {
ast_dsp_set_features(session-dsp,
DSP_FEATURE_DIGIT_DETECT);
1420:
   if (!media || !media-rtp ||
(ast_rtp_instance_dtmf_mode_get(media-rtp) == AST_RTP_DTMF_MODE_INBAND)) {
return -1;
1523:
   if (!media || !media-rtp ||
(ast_rtp_instance_dtmf_mode_get(media-rtp) == AST_RTP_DTMF_MODE_INBAND)) {
return -1;

That's it!!! It works fine for me. Any remarks / advice would be
appreciated.

Yaron.
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Re: [asterisk-users] Six seconds hangup

2014-12-16 Thread Yaron Nachum 
Hello Binni,
It is hard to say anything without more information.
You need to understand what happens in those dropped calls.

Logs would help. Traces might help also. Try mirror the traffic to another
server and capture it using tcpdump, or even run tcpdump on the server
itself.



On Tue, Dec 16, 2014 at 3:05 PM, Brynjólfur Þorvarðsson bi...@binni.eu
wrote:



 Hello all



 Over the last couple of months we’ve been experiencing a strange problem,
 which I’ve been unable to solve.



 We have an Asterisk 1.4.19 that’s been running happily for the last
 several years. All calls go through an AGI server from dialplan.



 On average we have appr. 3000 incoming calls/day. All calls go in via the
 AGI server, various sound files and menus are played, and every single call
 ends up in a queue.



 Every now and then, when one of our SIP customer answers a call from
 his/her queue, the call is connected but hangs up in 6 seconds (seems
 surprisingly constant). Both ends of the call can hear each other for a
 second or two (not as many as 6 seconds) before the call hangs up.



 The frequency of this happening is difficult establish precisely, we have
 some 40 customers, and they don’t always tell me when this happens. The
 worst I have heard of is this morning, where one of our customers
 experienced 1 in 4 calls having this problem. Last week the frequency for
 this customer was more in the region of 1 in 50.



 In all cases the second attempt seems to succeed, i.e. the originating
 caller tries again and gets through “properly” the second time.



 I have not been able to find anything in the log files for Asterisk or the
 AGI server. I’ve not run a SIP trace, as this would be a major undertaking
 with our traffic and the sporadic nature of the problem – but if all else
 fails, I’ll try that!



 At one point I thought it might be a problem with RTP channels and tried
 setting them to default values, but that has not had any effect. The sound
 goes through fine in both directions, but only for a couple of seconds.



 I hope someone here will be able to help me!



 Thanks in advance.



 Binni

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[asterisk-users] Asterisk 12 crashes on CANCEL received during ANSWER handlingl

2014-11-12 Thread Yaron Nachum 
Hello Asterisk users and developers,
The last few weeks we had several crashes on live asterisks running
versions  12.2.0rc1 / 12.6.1 with PJPROJECT versions 2.1.0 / 2.2.1. We
opened a ticket - ASTERISK-24471.

After investigating the issue I can say that the scenario is a CANCEL being
received while handling ANSWER and before generating the 200OK response.

Looking at the core file we see that the problem is in
- pjsip/src/pjsip/sip_transaction.c line 3158 :
PJ_ASSERT_RETURN(event-type == PJSIP_EVENT_TX_MSG 
 event-body.tx_msg.tdata == tsx-last_tx,
 PJ_EINVALIDOP);

After investigating further I came to a conclusion that the second
expression fails (marked with yellow), and that causes the Asterisk to
crash.

I have already removed the expression and logged whenever this expression
fails. It seems to work fine. Since the change it the happened several
times, the application didn't crash and going over the debug it seems that
the call was handled fine.

Can anyone tell what is the purpose of this expression? Any explanation why
this expression fails in the above scenario?

Thanks,
Yaron.
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Re: [asterisk-users] Asterisk 12 crashes on CANCEL received during ANSWER handlingl

2014-11-12 Thread Yaron Nachum 
Hello Joshua,
Thank you for your response. It was important for us to make sure this fix
is harmless.

We will wait for someone to fix it. If you need any assistance we are here.
This issue occurs quite often in our environment.

Yaron.

On Wed, Nov 12, 2014 at 5:16 PM, Joshua Colp jc...@digium.com wrote:

 Yaron Nachum wrote:

 Hello Asterisk users and developers,
 The last few weeks we had several crashes on live asterisks running
 versions  12.2.0rc1 / 12.6.1 with PJPROJECT versions 2.1.0 / 2.2.1. We
 opened a ticket - ASTERISK-24471.

 After investigating the issue I can say that the scenario is a CANCEL
 being received while handling ANSWER and before generating the 200OK
 response.

 Looking at the core file we see that the problem is in
 - pjsip/src/pjsip/sip_transaction.c line 3158 :
  PJ_ASSERT_RETURN(event-type == PJSIP_EVENT_TX_MSG 
 event-body.tx_msg.tdata == tsx-last_tx,
   PJ_EINVALIDOP);

 After investigating further I came to a conclusion that the second
 expression fails (marked with yellow), and that causes the Asterisk to
 crash.

 I have already removed the expression and logged whenever this
 expression fails. It seems to work fine. Since the change it the
 happened several times, the application didn't crash and going over the
 debug it seems that the call was handled fine.

 Can anyone tell what is the purpose of this expression? Any explanation
 why this expression fails in the above scenario?


 PJSIP is written with assertions in various places which check various
 conditions to make sure that things are as expected. In the above scenario
 what's happening is that the CANCEL is coming in and changing the
 transaction so when the answer is sent (which occurs after handling the
 CANCEL) it is not in the expected state and it asserts. It's safe to do as
 you've done since the operation will do nothing, but that's not a viable
 fix to push upstream as it's a result of an implementation detail with
 chan_pjsip. The fix will occur in chan_pjsip once the issue is handled.

 --
 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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Re: [asterisk-users] Asterisk 12 - zombie processes

2014-11-05 Thread Yaron Nachum 
Hello Mathew and everyone,
We had another crash on the 12.6.1 machine. This time it was Sigmentation
Fault. I opened another issue - ASTERISK-24493
https://issues.asterisk.org/jira/browse/ASTERISK-24493.

Yaron.

On Tue, Nov 4, 2014 at 6:16 PM, Yaron Nachum nachum.ya...@gmail.com wrote:

 Mathew,
 We are aware that this is an open source product, and our expectations are
 clear.
 All we are asking is that once there is someone assigned to the issue, he
 will guide us in what other data or tests should be performed in order to
 diagnose and fix the issue in the shortest time.

 Sorry if the message is not understood.
 Yaron

 On Tue, Nov 4, 2014 at 3:59 PM, Matthew Jordan mjor...@digium.com wrote:

 On Tue, Nov 4, 2014 at 12:59 AM, Yaron Nachum nachum.ya...@gmail.com
 wrote:
  Hello Asterisk users  developers,
  I have opened an issue few days ago regarding the crash and the zombie
  processes. I haven't received any response and it has't been assigned.
  If something is wrong or missing with the issue please get back to me
 and I
  will handle it.
 
  Please look into it because we still have crashes every day or two, and
 we
  can't reproduce the issue in our lab with a simulator.
 

 To set expectations here:

 This is an open source project. No one is under any obligation to look
 at your issue.

 There are currently around 20 issues or so in the issue tracker in
 Triage. Bug marshals are working through those issues as fast as they
 can, but generally they work from oldest to newest. If an older issue
 takes a lot of investigation... well, there's only so many hours in a
 day.

 Even after an issue is triaged, that is not a guarantee that someone
 will fix your issue. It is a crash, and that generally means it is
 higher priority - however, if you can't reproduce it in a lab
 environment or provide instructions on how it is reproduced, then you
 have to hope that a developer who does look at it can infer the cause
 of the crash from the information available. Any information you can
 provide beyond the backtrace on how to reproduce the issue will help a
 developer who looks at it.

 Again, however, no one is under any obligation to fix the issue. If
 you need more assurance that your issue is resolved, I'd highly
 recommend looking at issuing a bug bounty [1], or contacting a
 developer in the Asterisk Developer Community for assistance.

 [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties

 --
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

 --
 _
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Re: [asterisk-users] Asterisk 12 - zombie processes

2014-11-04 Thread Yaron Nachum 
Mathew,
We are aware that this is an open source product, and our expectations are
clear.
All we are asking is that once there is someone assigned to the issue, he
will guide us in what other data or tests should be performed in order to
diagnose and fix the issue in the shortest time.

Sorry if the message is not understood.
Yaron

On Tue, Nov 4, 2014 at 3:59 PM, Matthew Jordan mjor...@digium.com wrote:

 On Tue, Nov 4, 2014 at 12:59 AM, Yaron Nachum nachum.ya...@gmail.com
 wrote:
  Hello Asterisk users  developers,
  I have opened an issue few days ago regarding the crash and the zombie
  processes. I haven't received any response and it has't been assigned.
  If something is wrong or missing with the issue please get back to me
 and I
  will handle it.
 
  Please look into it because we still have crashes every day or two, and
 we
  can't reproduce the issue in our lab with a simulator.
 

 To set expectations here:

 This is an open source project. No one is under any obligation to look
 at your issue.

 There are currently around 20 issues or so in the issue tracker in
 Triage. Bug marshals are working through those issues as fast as they
 can, but generally they work from oldest to newest. If an older issue
 takes a lot of investigation... well, there's only so many hours in a
 day.

 Even after an issue is triaged, that is not a guarantee that someone
 will fix your issue. It is a crash, and that generally means it is
 higher priority - however, if you can't reproduce it in a lab
 environment or provide instructions on how it is reproduced, then you
 have to hope that a developer who does look at it can infer the cause
 of the crash from the information available. Any information you can
 provide beyond the backtrace on how to reproduce the issue will help a
 developer who looks at it.

 Again, however, no one is under any obligation to fix the issue. If
 you need more assurance that your issue is resolved, I'd highly
 recommend looking at issuing a bug bounty [1], or contacting a
 developer in the Asterisk Developer Community for assistance.

 [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties

 --
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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 asterisk-users mailing list
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Re: [asterisk-users] Asterisk 12 - zombie processes

2014-11-03 Thread Yaron Nachum 
Hello Asterisk users  developers,
I have opened an issue few days ago regarding the crash and the zombie
processes. I haven't received any response and it has't been assigned.
If something is wrong or missing with the issue please get back to me and I
will handle it.

Please look into it because we still have crashes every day or two, and we
can't reproduce the issue in our lab with a simulator.

Thank you,
Yaron.


On Thu, Oct 30, 2014 at 9:01 AM, Yaron Nachum nachum.ya...@gmail.com
wrote:

 Hello everyone,
 I have opened a ticket number - ASTERISK-24471
 https://issues.asterisk.org/jira/browse/ASTERISK-24471.

 I have attached the backtrace of the core file. The backtrace was taken on
 the server running 12.6.1.

 If you need any information please get back to me.

 Thank you.
 Yaron.

 On Thu, Oct 30, 2014 at 8:43 AM, Yaron Nachum nachum.ya...@gmail.com
 wrote:

 No,
 I went over all my scripts.

 Thanks for the help.

 Yaron

 On Wed, Oct 29, 2014 at 6:11 PM, Paul Belanger 
 paul.belan...@polybeacon.com wrote:

 On Tue, Oct 28, 2014 at 11:10 AM, Yaron Nachum nachum.ya...@gmail.com
 wrote:
  Mathew,
  When I run 'ps -ef|grep asterisk' the following processes are
 displayed:
  root  6861 1  0 Aug27 ?00:00:00 /bin/sh
  /ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C
  /ivr/app/asterisk/etc/asterisk/asterisk.conf
  asterisk  8062  6861  3 Oct27 ?00:44:56
  /ivr/app/asterisk/sbin/asterisk -f -U asterisk -G asterisk -C
  /ivr/app/asterisk/etc/asterisk/asterisk.conf -vvvg -c
  root 20776  2200  0 11:20 pts/200:00:33 tail -f asterisk.log
  asterisk 23076  8062  0 17:01 ?00:00:00 [asterisk] defunct
  asterisk 23897  8062  0 17:03 ?00:00:00 [asterisk] defunct
 
  also when I run top the same amount of zombie processes are displayed:
  Tasks: 185 total,   1 running, 182 sleeping,   0 stopped,   2 zombie
 
  Regarding the AGI - we are using AGI in order to run php scripts for
  external logic. I have printed the PIDs of the php scripts and none of
 them
  are related to the PID's of those zombie processes.
  Do you have any idea how to find out what are these processes?
  Yaron.
 
 Are you doing anything like:

 # asterisk -rx 'core show channels'

 via an external process?

 --
 Paul Belanger | PolyBeacon, Inc.
 Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
 Github: https://github.com/pabelanger | Twitter:
 https://twitter.com/pabelanger

 --
 _
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 asterisk-users mailing list
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Re: [asterisk-users] Asterisk 12 - zombie processes

2014-10-30 Thread Yaron Nachum 
No,
I went over all my scripts.

Thanks for the help.

Yaron

On Wed, Oct 29, 2014 at 6:11 PM, Paul Belanger paul.belan...@polybeacon.com
 wrote:

 On Tue, Oct 28, 2014 at 11:10 AM, Yaron Nachum nachum.ya...@gmail.com
 wrote:
  Mathew,
  When I run 'ps -ef|grep asterisk' the following processes are displayed:
  root  6861 1  0 Aug27 ?00:00:00 /bin/sh
  /ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C
  /ivr/app/asterisk/etc/asterisk/asterisk.conf
  asterisk  8062  6861  3 Oct27 ?00:44:56
  /ivr/app/asterisk/sbin/asterisk -f -U asterisk -G asterisk -C
  /ivr/app/asterisk/etc/asterisk/asterisk.conf -vvvg -c
  root 20776  2200  0 11:20 pts/200:00:33 tail -f asterisk.log
  asterisk 23076  8062  0 17:01 ?00:00:00 [asterisk] defunct
  asterisk 23897  8062  0 17:03 ?00:00:00 [asterisk] defunct
 
  also when I run top the same amount of zombie processes are displayed:
  Tasks: 185 total,   1 running, 182 sleeping,   0 stopped,   2 zombie
 
  Regarding the AGI - we are using AGI in order to run php scripts for
  external logic. I have printed the PIDs of the php scripts and none of
 them
  are related to the PID's of those zombie processes.
  Do you have any idea how to find out what are these processes?
  Yaron.
 
 Are you doing anything like:

 # asterisk -rx 'core show channels'

 via an external process?

 --
 Paul Belanger | PolyBeacon, Inc.
 Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
 Github: https://github.com/pabelanger | Twitter:
 https://twitter.com/pabelanger

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Asterisk 12 - zombie processes

2014-10-30 Thread Yaron Nachum 
Hello everyone,
I have opened a ticket number - ASTERISK-24471
https://issues.asterisk.org/jira/browse/ASTERISK-24471.

I have attached the backtrace of the core file. The backtrace was taken on
the server running 12.6.1.

If you need any information please get back to me.

Thank you.
Yaron.

On Thu, Oct 30, 2014 at 8:43 AM, Yaron Nachum nachum.ya...@gmail.com
wrote:

 No,
 I went over all my scripts.

 Thanks for the help.

 Yaron

 On Wed, Oct 29, 2014 at 6:11 PM, Paul Belanger 
 paul.belan...@polybeacon.com wrote:

 On Tue, Oct 28, 2014 at 11:10 AM, Yaron Nachum nachum.ya...@gmail.com
 wrote:
  Mathew,
  When I run 'ps -ef|grep asterisk' the following processes are displayed:
  root  6861 1  0 Aug27 ?00:00:00 /bin/sh
  /ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C
  /ivr/app/asterisk/etc/asterisk/asterisk.conf
  asterisk  8062  6861  3 Oct27 ?00:44:56
  /ivr/app/asterisk/sbin/asterisk -f -U asterisk -G asterisk -C
  /ivr/app/asterisk/etc/asterisk/asterisk.conf -vvvg -c
  root 20776  2200  0 11:20 pts/200:00:33 tail -f asterisk.log
  asterisk 23076  8062  0 17:01 ?00:00:00 [asterisk] defunct
  asterisk 23897  8062  0 17:03 ?00:00:00 [asterisk] defunct
 
  also when I run top the same amount of zombie processes are displayed:
  Tasks: 185 total,   1 running, 182 sleeping,   0 stopped,   2 zombie
 
  Regarding the AGI - we are using AGI in order to run php scripts for
  external logic. I have printed the PIDs of the php scripts and none of
 them
  are related to the PID's of those zombie processes.
  Do you have any idea how to find out what are these processes?
  Yaron.
 
 Are you doing anything like:

 # asterisk -rx 'core show channels'

 via an external process?

 --
 Paul Belanger | PolyBeacon, Inc.
 Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
 Github: https://github.com/pabelanger | Twitter:
 https://twitter.com/pabelanger

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Asterisk 12 - zombie processes

2014-10-29 Thread Yaron Nachum 
Hello Mathew and everyone,
We are still having reboots on our asterisk servers. The latest 12.6.1
release doesn't fix the issue.

We have the core files of the latest reboots and also debug taken during
the reboot.

We would like to open an issue. What kind of information you need for the
issue?

Thanks,
Yaron.


On Tue, Oct 28, 2014 at 5:10 PM, Yaron Nachum nachum.ya...@gmail.com
wrote:

 Mathew,
 When I run 'ps -ef|grep asterisk' the following processes are displayed:
 root  6861 1  0 Aug27 ?00:00:00 /bin/sh
 /ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C
 /ivr/app/asterisk/etc/asterisk/asterisk.conf
 asterisk  8062  6861  3 Oct27 ?00:44:56
 /ivr/app/asterisk/sbin/asterisk -f -U asterisk -G asterisk -C
 /ivr/app/asterisk/etc/asterisk/asterisk.conf -vvvg -c
 root 20776  2200  0 11:20 pts/200:00:33 tail -f asterisk.log
 asterisk 23076  8062  0 17:01 ?00:00:00 [asterisk] defunct
 asterisk 23897  8062  0 17:03 ?00:00:00 [asterisk] defunct

 also when I run top the same amount of zombie processes are displayed:
 Tasks: 185 total,   1 running, 182 sleeping,   0 stopped,   2 zombie

 Regarding the AGI - we are using AGI in order to run php scripts for
 external logic. I have printed the PIDs of the php scripts and none of them
 are related to the PID's of those zombie processes.
 Do you have any idea how to find out what are these processes?
 Yaron.

 On Tue, Oct 28, 2014 at 4:53 PM, Matthew Jordan mjor...@digium.com
 wrote:

 On Tue, Oct 28, 2014 at 9:44 AM, Yaron Nachum nachum.ya...@gmail.com
 wrote:
  Hello Mathew,
  In the following tutorial it says that channel are marked with ZOMBIE
 flag.
  From your response I assume it has no connection to my problem.
  https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Bridging+Project
 
  Regrading the zombie processes issue we are having. We have debug taken
 from
  the server during such process is invoked. If you want I can attache it.

 A zombie channel has nothing to do with a process. It was an artefact
 of an internal process known as a masquerade. While masquerades do
 sometimes still occur in Asterisk 12+, they are far less frequent and
 are no longer externally visible.

 Why do you think you have zombie processes? Asterisk does use a large
 number of threads, but generally rarely forks processes unless you are
 using something like original AGI.

 --
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

 asterisk-users mailing list
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Re: [asterisk-users] DTMF behavior in asterisk 12 with PJSIP

2014-10-28 Thread Yaron Nachum 
Thank you Mathew,
We tested the feature flag workaround and it worked.

We opened a ticket - Asterisk-24459.

If you need any information please get back to us and we will do our best.

Thanks again,
Yaron.

On Mon, Oct 27, 2014 at 3:48 PM, Matthew Jordan mjor...@digium.com wrote:



 On Mon, Oct 27, 2014 at 1:20 AM, Yaron Nachum nachum.ya...@gmail.com
 wrote:

 Hello Mathew,
 Thank you for the reply.

 I will open an issue and send debug information.

 Can you explain more about the workaround? A reference to the
 documentation would be fine.



 Sure - really, what you are running into is a difference in how Asterisk
 bridges channels:

 https://wiki.asterisk.org/wiki/display/AST/Bridges

 I suspect the reason DTMF is not decoded is because you are in a native
 bridge (local or remote). You can force a core two-party bridge by
 requiring that Asterisk decode the media and detect DTMF. Those
 requirements are done by setting the various 'feature' flags on whatever
 dialplan application is causing the channels to be bridged. For an example,
 see the 't' or 'T' options in Dial:

 https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial

 --
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Asterisk 12 - zombie processes

2014-10-28 Thread Yaron Nachum 
Hello Asterisk users,
We noticed that on Asterisk 12 zombie processes are being generated - They
are released after a while, but we have around 10-20 zombie processes
running.

We are not sure if this is a normal behavior or an issue.

We saw in the documentation that the bridging module creates zombie
processes - is it related?

Thank you,
Yaron.
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Re: [asterisk-users] Asterisk 12 - zombie processes

2014-10-28 Thread Yaron Nachum 
Hello Mathew,
In the following tutorial it says that channel are marked with ZOMBIE flag.
From your response I assume it has no connection to my problem.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Bridging+Project

Regrading the zombie processes issue we are having. We have debug taken
from the server during such process is invoked. If you want I can attache
it.

Furthermore,  in recent days we have had a number of reboots. Most of our
servers are on release 12.2.0.RC1, and we have just upgraded to 12.6.1 on
one of the servers. since the upgrade we have't had a reboot, but it is too
early to be sure.

Somehow the core files are not written - in the log file it says -
Executable '/ivr/app/asterisk/sbin/asterisk' doesn't belong to any package.
Do you have any idea?

Thank you,
Yaron.




On Tue, Oct 28, 2014 at 2:34 PM, Matthew Jordan mjor...@digium.com wrote:


 On Tue, Oct 28, 2014 at 4:58 AM, Yaron Nachum nachum.ya...@gmail.com
 wrote:

 Hello Asterisk users,
 We noticed that on Asterisk 12 zombie processes are being generated -
 They are released after a while, but we have around 10-20 zombie processes
 running.

 We are not sure if this is a normal behavior or an issue.

 We saw in the documentation that the bridging module creates zombie
 processes - is it related?



 Where in the documentation (or in what documentation) does it say that?

 --
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Asterisk 12 - zombie processes

2014-10-28 Thread Yaron Nachum 
Mathew,
When I run 'ps -ef|grep asterisk' the following processes are displayed:
root  6861 1  0 Aug27 ?00:00:00 /bin/sh
/ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C
/ivr/app/asterisk/etc/asterisk/asterisk.conf
asterisk  8062  6861  3 Oct27 ?00:44:56
/ivr/app/asterisk/sbin/asterisk -f -U asterisk -G asterisk -C
/ivr/app/asterisk/etc/asterisk/asterisk.conf -vvvg -c
root 20776  2200  0 11:20 pts/200:00:33 tail -f asterisk.log
asterisk 23076  8062  0 17:01 ?00:00:00 [asterisk] defunct
asterisk 23897  8062  0 17:03 ?00:00:00 [asterisk] defunct

also when I run top the same amount of zombie processes are displayed:
Tasks: 185 total,   1 running, 182 sleeping,   0 stopped,   2 zombie

Regarding the AGI - we are using AGI in order to run php scripts for
external logic. I have printed the PIDs of the php scripts and none of them
are related to the PID's of those zombie processes.
Do you have any idea how to find out what are these processes?
Yaron.

On Tue, Oct 28, 2014 at 4:53 PM, Matthew Jordan mjor...@digium.com wrote:

 On Tue, Oct 28, 2014 at 9:44 AM, Yaron Nachum nachum.ya...@gmail.com
 wrote:
  Hello Mathew,
  In the following tutorial it says that channel are marked with ZOMBIE
 flag.
  From your response I assume it has no connection to my problem.
  https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Bridging+Project
 
  Regrading the zombie processes issue we are having. We have debug taken
 from
  the server during such process is invoked. If you want I can attache it.

 A zombie channel has nothing to do with a process. It was an artefact
 of an internal process known as a masquerade. While masquerades do
 sometimes still occur in Asterisk 12+, they are far less frequent and
 are no longer externally visible.

 Why do you think you have zombie processes? Asterisk does use a large
 number of threads, but generally rarely forks processes unless you are
 using something like original AGI.

 --
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] DTMF behavior in asterisk 12 with PJSIP

2014-10-27 Thread Yaron Nachum 
Hello Mathew,
Thank you for the reply.

I will open an issue and send debug information.

Can you explain more about the workaround? A reference to the documentation
would be fine.


Thanks again,
Yaron.

On Sun, Oct 26, 2014 at 10:46 PM, Matthew Jordan mjor...@digium.com wrote:



 On Sun, Oct 26, 2014 at 3:22 AM, Yaron Nachum nachum.ya...@gmail.com
 wrote:

 Hello all,
 We have recently upgraded some of our services to Asterisk 12 with PJSIP.
 We have 2 issues related to DTMF:
 1. in the regular SIP channel we had DTMF auto mode, which adapted the
 DTMF settings according to the incoming INVITE - RFC2833 or inband. The is
 no such settings in PJSIP. Do you know is there is a plan to develop it?


 No one that I'm aware of is currently working on that.

 As Asterisk is an open source project, if having the 'auto' feature added
 to the PJSIP stack is something you're interested in, you should consider
 writing a patch for the project [1].

 [1] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process


 2. When we setup 2 peers, one RFC4733 and the other inband, the asterisk
 does not transcode the DTMF signals, therefore DTMF is not working. It used
 to work on release 11. This is really bad. Do you know of a solution to
 this issue? Maybe some settings?


 That actually is a bug. You are most likely ending up in a native packet
 to packet bridge (or a native remote bridge), which does not decode the RTP
 stream. Hence, the inband DTMF or RFC 2833 DTMF is not being decoded and is
 being passed to the other side. Please do open an issue for that [2]. Make
 sure you provide a full DEBUG log, as that will illustrate what is actually
 occurring.

 Note that you can work around that issue by adding a feature flag to
 whatever application caused the bridging to occur.

 [2] https://issues.asterisk.org/jira

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[asterisk-users] DTMF behavior in asterisk 12 with PJSIP

2014-10-26 Thread Yaron Nachum 
Hello all,
We have recently upgraded some of our services to Asterisk 12 with PJSIP.
We have 2 issues related to DTMF:
1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMF
settings according to the incoming INVITE - RFC2833 or inband. The is no
such settings in PJSIP. Do you know is there is a plan to develop it?
2. When we setup 2 peers, one RFC4733 and the other inband, the asterisk
does not transcode the DTMF signals, therefore DTMF is not working. It used
to work on release 11. This is really bad. Do you know of a solution to
this issue? Maybe some settings?

Thanks,
Yaron.
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[asterisk-users] Port number in From URI on Asterisk 12 PJSIP

2014-10-26 Thread Yaron Nachum 
Hello,
I have an issue with Asterisk 12 PJSIP. When receving an INVITE with FROM
URI that has a port number, the Asterisk removes the port from URI on
consecutive Responses / Requests. This causes an issue with one of our SIP
servers (it doesn't recognize the response / request).
Below you can see an incoming INVITE and the outgoing 200OK response. I
have highlighted the issue in Yellow.
Does anyone know of a solution / workaround for this issue?

--- Received SIP request (648 bytes) from UDP:172.16.60.160:5061 ---
INVITE sip:039988120F@172.16.60.160:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK-29450-3-0
Max-Forwards: 60
From: sip:39937841@192.168.225.2:5061;user=phone;tag=3
To: sip:039988120F@172.16.60.160:5060;user=phone
Call-ID: 3-29450@172.16.60.160
CSeq: 1 INVITE
Contact: sip:10.1.1.1:5060
User-Agent: Simulator
Supported: 100rel
Privacy: id
Min-SE: 90
Content-Type: application/sdp
Content-Length:   201

v=0
o=172.16.60.160 10864 2 IN IP4 172.16.60.160
s=SIP Call
c=IN IP4 172.16.60.160
t=0 0
a=sendrecv
m=audio 6 RTP/AVP 8 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- Transmitting SIP response (730 bytes) to UDP:172.16.60.160:5061 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.1:5060
;rport;received=172.16.60.160;branch=z9hG4bK-29450-3-0
Call-ID: 3-29450@172.16.60.160
From: sip:39937841@192.168.225.2;user=phone;tag=3
To: sip:039988120F@172.16.60.160
;user=phone;tag=4f7ef94f-fb15-4bf5-94bd-4e43fe-299655
CSeq: 1 INVITE
Contact: sip:172.16.60.160:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   189

v=0
o=- 10864 4 IN IP4 10.2.0.67
s=Asterisk
c=IN IP4 172.16.60.160
t=0 0
m=audio 19404 RTP/AVP 8
c=IN IP4 172.16.60.160
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:150
a=sendrecv
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Re: [asterisk-users] PJSIP usereqphone setting in config file

2014-04-10 Thread Yaron Nachum 
Thanks a lot Joshua. That's a great idea :-)
I will try it and get back to you.


On Thu, Apr 10, 2014 at 5:50 PM, Joshua Colp jc...@digium.com wrote:

 Yaron Nachum wrote:

 Hi everyone,


 Kia ora,


  I am starting to work with PJSIP on release 12.1.0.rc3.

 I used to have Asterisk 1.8 with the regular sip channel. I was using
 the usereqphone settings in order to set user=phone on from and to URIs.

 Is there a similar config in PJSIP?


 There is currently no option which explicitly does this. A core difference
 between chan_sip and chan_pjsip, though, is that we treat stuff as SIP URIs
 in the first place. To that end it may be possible to simply add
 ;user=phone to the SIP URI yourself and have it work. Since ';' is used for
 comments though you will need to escape it using \.

 An example for a static contact on an AOR:

 contact=sip:172.16.1.1:5060;user=phone

 Cheers,

 --
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 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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Re: [asterisk-users] PJSIP in dialog OPTIONS method handling

2014-04-09 Thread Yaron Nachum 
Hi,
Anyone has a workaround?


On Tue, Apr 8, 2014 at 9:17 PM, Yaron Nachum nachum.ya...@gmail.com wrote:

 Hi everyone,
 I am running asterisk with release 12.1.0.rc3 and PJSIP.
 I have a peer which sends OPTIONS method for session keep-alive, and the
 asterisk is not responding to it. That of course disconnects the call after
 a few minutes.

 Is there a settings in the PJSIP.conf to respond to in dialog OPTIONS
 method? Looking at the documentation I haven't seen it. Does anybody know a
 workaround?

 Thanks,
 Yaron.



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[asterisk-users] PJSIP usereqphone setting in config file

2014-04-09 Thread Yaron Nachum 
Hi everyone,
I am starting to work with PJSIP on release 12.1.0.rc3.

I used to have Asterisk 1.8 with the regular sip channel. I was using the
usereqphone settings in order to set user=phone on from and to URIs.

Is there a similar config in PJSIP?
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[asterisk-users] PJSIP in dialog OPTIONS method handling

2014-04-08 Thread Yaron Nachum 
Hi everyone,
I am running asterisk with release 12.1.0.rc3 and PJSIP.
I have a peer which sends OPTIONS method for session keep-alive, and the
asterisk is not responding to it. That of course disconnects the call after
a few minutes.

Is there a settings in the PJSIP.conf to respond to in dialog OPTIONS
method? Looking at the documentation I haven't seen it. Does anybody know a
workaround?

Thanks,
Yaron.
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[asterisk-users] Asterisk 12 - CDR changes

2014-03-19 Thread Yaron Nachum 
Hello everyone,
I am upgrading from release 1.8 and I have a strange behavior with CDR
generation.
We are using a Redirect server for Number portability, and I see that once
the call is going through the Redirect Server additional CDR records are
generated - we have 3 additional records.
This Behavior is different then what we had on Release 1.8.

Does anyone have a clue how to remove these CDR records?
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[asterisk-users] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833

2014-03-11 Thread Yaron Nachum 
Hello,
I have installed the latest version 12 that has been released (12.1.0.rc3).

I have setup default dtmf mode (rfc47..) but when I am calling to a
endpoint that doesn't support it (no telephony event in the rtpmap) the
asterisk responds OK in the signalling but DTMF is not working.

Is it a known issue?

Below you can see the output of the asterisk monitor.


--- Received SIP request (1182 bytes) from UDP:10.25.153.150:5060 ---
INVITE sip:039988120@172.16.60.160:5060;user=phone SIP/2.0
Record-Route: sip:10.25.153.150;lr;ftag=02e3a8c0-33807b-t-2
Via: SIP/2.0/UDP 10.25.153.150:5060;branch=z9hG4bK587.67258295.0
Via: SIP/2.0/UDP
10.1.1.10;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.qLOBF6zGZLMBj7MBvuMx3AMB1jmxuqC93X3heroEWvH9vsCFN43qdAMxyAMxyAMxyAMlMZMxpJ3lqwWxarW.gqWReJMEPA36juW6WBzR363RVA3Ejugx3*
Max-Forwards: 68
From: 39937841 39937841 sip:39937841;cpc=payphone@192.168.225.2:5060
;user=phone;tag=02e3a8c0-33807b-t-2
To: sip:D39539988120@192.168.225.2:5060;user=phone
Call-ID: 2915b6e4-02e3a8c0-be53@192.168.225.2
CSeq: 2 INVITE
Contact:
sip:10.1.1.10;line=sr-N6IAzBMsz.MwzxPfPxFsMJZfWBc7MBVuOBV-W.y6MxV*
User-Agent: NetCentrex CCS Softswitch/7.16.0
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, INFO, PRACK, UPDATE, NOTIFY
Supported: 100rel
P-Asserted-Identity: 39937841 39937841
sip:39937841;cpc=payphone@192.168.225.2:5060;user=phone
Min-SE: 90
Privacy: none
Content-Type: application/sdp
Content-Length: 167

v=0
o=10.206.22.171 62708 2 IN IP4 10.206.22.171
s=SIP Call
c=IN IP4 10.206.22.171
t=0 0
a=sendrecv
m=audio 41040 RTP/AVP 8
a=rtpmap:8 PCMA/8000/1
a=ptime:20

--- Transmitting SIP response (602 bytes) to UDP:10.25.153.150:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.25.153.150:5060
;rport;received=10.25.153.150;branch=z9hG4bK587.67258295.0
Via: SIP/2.0/UDP
10.1.1.10;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.qLOBF6zGZLMBj7MBvuMx3AMB1jmxuqC93X3heroEWvH9vsCFN43qdAMxyAMxyAMxyAMlMZMxpJ3lqwWxarW.gqWReJMEPA36juW6WBzR363RVA3Ejugx3*
Record-Route: sip:10.25.153.150;lr;ftag=02e3a8c0-33807b-t-2
Call-ID: 2915b6e4-02e3a8c0-be53@192.168.225.2
From: 39937841 39937841 sip:39937841;cpc=payphone@192.168.225.2
;user=phone;tag=02e3a8c0-33807b-t-2
To: sip:D39539988120@192.168.225.2;user=phone
CSeq: 2 INVITE
Content-Length:  0


-- Executing [039988120@from-external:1] NoOp(PJSIP/sipp-, 
H E L L O ! ! !) in new stack
-- Executing [039988120@from-external:2]
DumpChan(PJSIP/sipp-, ) in new stack

Dumping Info For Channel: PJSIP/sipp-:

Info:
Name=   PJSIP/sipp-
Type=   PJSIP
UniqueID=   172.16.60.160-1394542052.0
LinkedID=   172.16.60.160-1394542052.0
CallerIDNum=39937841;cpc=payphone
CallerIDName=   39937841 39937841
ConnectedLineIDNum= (N/A)
ConnectedLineIDName=(N/A)
DNIDDigits= (N/A)
RDNIS=  (N/A)
Parkinglot=
Language=   en
State=  Ring (4)
Rings=  1
NativeFormat=   (alaw)
WriteFormat=alaw
ReadFormat= alaw
RawWriteFormat= alaw
RawReadFormat=  alaw
WriteTranscode= No
ReadTranscode=  No
1stFileDescriptor=  -1
Framesin=   0
Framesout=  0
TimetoHangup=   0
ElapsedTime=0h0m0s
BridgeID=   (Not bridged)
Context=from-external
Extension=  039988120
Priority=   2
CallGroup=
PickupGroup=
Application=DumpChan
Data=   (Empty)
Blocking_in=(Not Blocking)

Variables:

-- Executing [039988120@from-external:3] Answer(PJSIP/sipp-,
) in new stack
--- Transmitting SIP response (1060 bytes) to UDP:10.25.153.150:5060 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.25.153.150:5060
;rport;received=10.25.153.150;branch=z9hG4bK587.67258295.0
Via: SIP/2.0/UDP
10.1.1.10;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.qLOBF6zGZLMBj7MBvuMx3AMB1jmxuqC93X3heroEWvH9vsCFN43qdAMxyAMxyAMxyAMlMZMxpJ3lqwWxarW.gqWReJMEPA36juW6WBzR363RVA3Ejugx3*
Record-Route: sip:10.25.153.150;lr;ftag=02e3a8c0-33807b-t-2
Call-ID: 2915b6e4-02e3a8c0-be53@192.168.225.2
From: 39937841 39937841 sip:39937841;cpc=payphone@192.168.225.2
;user=phone;tag=02e3a8c0-33807b-t-2
To: sip:D39539988120@192.168.225.2
;user=phone;tag=b23cda89-931c-4a95-85c5-0ec8b03f895c
CSeq: 2 INVITE
Contact: sip:172.16.60.160:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   193

v=0
o=- 62708 4 IN IP4 172.16.60.160
s=Asterisk
c=IN IP4 172.16.60.160
t=0 0
m=audio 13644 RTP/AVP 8
c=IN IP4 172.16.60.160
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:150
a=sendrecv

--- Received SIP request (703 bytes) from UDP:10.25.153.150:5060 ---
ACK sip:172.16.60.160:5060 SIP/2.0
Via: SIP/2.0/UDP 10.25.153.150:5060;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP

[asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint

2014-03-11 Thread Yaron Nachum 
Hello everyone,
I have started testing the PJSIP stack.

I saw that it is possible to setup statically multiple AOR contacts, setup
qualify_timeout and attach it to an endpoint, and then dial using this
endpoint.

When I setup the configuration I used the cli in order to see the status of
the contacts, and it worked fine - whenever a contact is unreachable, the
status is updated to Unavailable.

However, when I dial through this endpoint the asterisk doesn't use other
contacts which are available in this endpoint.

Is it a known issue?  Are you planning to solve it?

Below is my pjsip.conf:

[transport-udp]
type=transport
protocol=udp
bind=172.16.60.160:5060

;SIPP
[sipp]
type=endpoint
transport=transport-udp
context=from-external
disallow=all
allow=alaw
100rel=required
aors=sipp

[sipp]
type=aor
contact=sip:172.16.60.160:5080
contact=sip:10.25.153.150:5060
qualify_frequency=10



[sipp]
type=identify
endpoint=sipp
match=10.25.153.150
match=172.16.60.160
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Re: [asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint

2014-03-11 Thread Yaron Nachum 
Thanks Joshua,
I tried it already. That would generate a call to both AORs which is not
what I was looking for.

Isn't there a way to retrieve the AOR status from the dialplan?


On Tue, Mar 11, 2014 at 3:27 PM, Joshua Colp jc...@digium.com wrote:

 Yaron Nachum wrote:

 Hello everyone,
 I have started testing the PJSIP stack.

 I saw that it is possible to setup statically multiple AOR contacts,
 setup qualify_timeout and attach it to an endpoint, and then dial using
 this endpoint.

 When I setup the configuration I used the cli in order to see the status
 of the contacts, and it worked fine - whenever a contact is unreachable,
 the status is updated to Unavailable.

 However, when I dial through this endpoint the asterisk doesn't use
 other contacts which are available in this endpoint.

 Is it a known issue?  Are you planning to solve it?


 Due to limitations within the Asterisk core you have to use the
 PJSIP_DIAL_CONTACTS dialplan function[1].

 [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+
 Function_PJSIP_DIAL_CONTACTS

 --
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Re: [asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint

2014-03-11 Thread Yaron Nachum 
Thanks for the response anyway.

I think that it would be great if someone would make it happen. It seems to
me trivial that once you enable to setup multiple AORs you would use them
:-)

Yaron.


On Tue, Mar 11, 2014 at 3:38 PM, Joshua Colp jc...@digium.com wrote:

 Yaron Nachum wrote:

 Thanks Joshua,
 I tried it already. That would generate a call to both AORs which is not
 what I was looking for.

 Isn't there a way to retrieve the AOR status from the dialplan?


 Not currently.


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Re: [asterisk-users] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833

2014-03-11 Thread Yaron Nachum 
Hi Mathew,
The regular sip stack has 'auto' dtmfmode which behaved as I said - if the
remote replied with telephony event it used RFC2833 otherwise it used
inband.




On Tue, Mar 11, 2014 at 5:43 PM, Matthew Jordan mjor...@digium.com wrote:

 On Tue, Mar 11, 2014 at 8:23 AM, Yaron Nachum nachum.ya...@gmail.com
 wrote:
  Hello,
  I have installed the latest version 12 that has been released
 (12.1.0.rc3).
 
  I have setup default dtmf mode (rfc47..) but when I am calling to a
 endpoint
  that doesn't support it (no telephony event in the rtpmap) the asterisk
  responds OK in the signalling but DTMF is not working.
 
  Is it a known issue?
 

 I don't think that's an issue at all.

 Your configured your endpoint to support RFC 4733 DTMF. However, the
 INVITE request that was received by Asterisk didn't offer support for
 DTMF, so Asterisk can't accept it. It has to accept only what is in
 the offer.

 Your configuration can't force the UA to offer what it wants - you can
 only configure Asterisk with what it should support with that UA.

 There's really only two possible outcomes here:
 (1) Reject the INVITE request with a 488 (you didn't offer me DTMF!)
 (2) Accept the INVITE request but not have DTMF over RFC 4733.

 What you're seeing is option (2), which I think is better than
 rejecting the entire call simply because the thing you are talking to
 doesn't support the DTMF mode you configured it to have.

 --
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 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833

2014-03-11 Thread Yaron Nachum 
Mathew,
Thanks Mathew. It's good to know the limitations :-)

Is there any plan to add it?


On Tue, Mar 11, 2014 at 6:38 PM, Matthew Jordan mjor...@digium.com wrote:

 On Tue, Mar 11, 2014 at 11:23 AM, Yaron Nachum nachum.ya...@gmail.com
 wrote:
  Hi Mathew,
  The regular sip stack has 'auto' dtmfmode which behaved as I said - if
 the
  remote replied with telephony event it used RFC2833 otherwise it used
  inband.
 

 Correct. There is no setting for dtmf_mode that is analogous to the
 chan_sip 'auto' setting - what you configure for you endpoint today is
 what it will use.

 That's not a bug, just something not existing yet.

 Matt

 --
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint

2014-03-11 Thread Yaron Nachum 
Thanks Mathew,
That would be great - just to validate the status of the AOR before you
send the INVITE.

Great mailing list.



On Tue, Mar 11, 2014 at 5:38 PM, Matthew Jordan mjor...@digium.com wrote:

 On Tue, Mar 11, 2014 at 8:45 AM, Yaron Nachum nachum.ya...@gmail.com
 wrote:
 
  Thanks for the response anyway.
 
  I think that it would be great if someone would make it happen. It seems
 to me trivial that once you enable to setup multiple AORs you would use
 them :-)
 
  Yaron.
 
 
  On Tue, Mar 11, 2014 at 3:38 PM, Joshua Colp jc...@digium.com wrote:
 
  Yaron Nachum wrote:
 
  Thanks Joshua,
  I tried it already. That would generate a call to both AORs which is
 not
  what I was looking for.
 
  Isn't there a way to retrieve the AOR status from the dialplan?
 
 
  Not currently.
 

 We're still adding dialplan functions and CLI commands to the PJSIP
 stack. Right now there's a way to drill down into endpoint
 configuration via the PJSIP_ENDPOINT function, but we haven't yet
 expanded that to AORs. Doing so is a pretty natural next step.

 There's some discussion of this on the following JIRA issue, where
 Josh mentions we could query down into the contacts for some of the
 information:

 https://issues.asterisk.org/jira/browse/ASTERISK-23173

 We'd probably have something similar to PJSIP_ENDPOINT, such as
 PJSIP_AOR or PJSIP_CONTACT (or something like that), that lets you get
 at the run-time information of an AOR.

 --
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

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