[asterisk-users] cdr_odbc: Error in ExecDirect: -1

[Vitor Mazuco]

Hi everybody!

I'm trying to install a CDR Viewer https://github.com/g613/asterisk-cdr-viewer

I'ts work well, but my problem is to make Asterisk store to MySQL using ODBC

When I make a call the CLI returns for me

See the log:
== Using SIP RTP CoS mark 5
-- Executing [2021@ramais:1] Dial("SIP/2020-",
"SIP/2021,60,tT") in new stack
  == Using SIP RTP CoS mark 5
-- Called SIP/2021
-- SIP/2021-0001 is ringing
   > 0x7fd3f8014240 -- Probation passed - setting RTP source
address to 192.168.25.49:35528
[Jan 13 11:31:07] NOTICE[2279][C-]: res_rtp_asterisk.c:4441
ast_rtp_read: Unknown RTP codec 95 received from '(null)'
-- SIP/2021-0001 answered SIP/2020-
   > 0x7fd3b4004eb0 -- Probation passed - setting RTP source
address to 192.168.25.100:8000
   > 0x7fd3f8014240 -- Probation passed - setting RTP source
address to 192.168.25.49:35528
   > cdr_odbc: Error in ExecDirect: -1
[Jan 13 11:31:08] WARNING[2279][C-]: res_odbc.c:604
ast_odbc_direct_execute: SQL Execute error! Verifying connection to
asterisk [asterisk-connector]...
   > cdr_odbc: Error in ExecDirect: -1
[Jan 13 11:31:08] ERROR[2279][C-]: cdr_odbc.c:177 odbc_log:
CDR direct execute failed


See my res_odbc.conf

[asterisk]
enabled = yes
dsn = asterisk-connector
username = root
password = 100567
pooling = no
limit = 1
pre-connect = yes

What can be happened?

Thank in advanced.

-- 
_
-- 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:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk-users Digest, Vol 138, Issue 8

[waqas.mehmood90]

How to get user extention no in agi php scrip from which he's calling on ivr i 
am using cid and able to get his name but not his extention no please help me



Sent from my Samsung Galaxy smartphone.

 Original message 
From: asterisk-users-requ...@lists.digium.com 
Date:13/01/2016  11:00 PM  (GMT+05:00) 
To: asterisk-users@lists.digium.com 
Cc:  
Subject: asterisk-users Digest, Vol 138, Issue 8 

Send asterisk-users mailing list submissions to
asterisk-users@lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
asterisk-users-requ...@lists.digium.com

You can reach the person managing the list at
asterisk-users-ow...@lists.digium.com

When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-users digest..."


Today's Topics:

   1. cdr_odbc: Error in ExecDirect: -1 (Vitor Mazuco)
   2. "pjsip show endpoints" returns "No Objects Found" in
  13.7.0-rc2 (Matthew Murphy)
   3. Re: "pjsip show endpoints" returns "No Objects Found" in
  13.7.0-rc2 (Joshua Colp)
   4. Re: "pjsip show endpoints" returns "No Objects Found" in
  13.7.0-rc2 (Matthew Murphy)
   5. Re: cdr_odbc: Error in ExecDirect: -1 (Patrick Laimbock)


--

Message: 1
Date: Wed, 13 Jan 2016 11:48:51 -0200
From: Vitor Mazuco <vitor.maz...@gmail.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: [asterisk-users] cdr_odbc: Error in ExecDirect: -1
Message-ID:
<caos3alzyi5wqeh89ueztwnhelj+ttlqi8r1+jxpis-sbfp-...@mail.gmail.com>
Content-Type: text/plain; charset=UTF-8

Hi everybody!

I'm trying to install a CDR Viewer https://github.com/g613/asterisk-cdr-viewer

I'ts work well, but my problem is to make Asterisk store to MySQL using ODBC

When I make a call the CLI returns for me

See the log:
== Using SIP RTP CoS mark 5
-- Executing [2021@ramais:1] Dial("SIP/2020-",
"SIP/2021,60,tT") in new stack
  == Using SIP RTP CoS mark 5
-- Called SIP/2021
-- SIP/2021-0001 is ringing
   > 0x7fd3f8014240 -- Probation passed - setting RTP source
address to 192.168.25.49:35528
[Jan 13 11:31:07] NOTICE[2279][C-]: res_rtp_asterisk.c:4441
ast_rtp_read: Unknown RTP codec 95 received from '(null)'
-- SIP/2021-0001 answered SIP/2020-
   > 0x7fd3b4004eb0 -- Probation passed - setting RTP source
address to 192.168.25.100:8000
   > 0x7fd3f8014240 -- Probation passed - setting RTP source
address to 192.168.25.49:35528
   > cdr_odbc: Error in ExecDirect: -1
[Jan 13 11:31:08] WARNING[2279][C-]: res_odbc.c:604
ast_odbc_direct_execute: SQL Execute error! Verifying connection to
asterisk [asterisk-connector]...
   > cdr_odbc: Error in ExecDirect: -1
[Jan 13 11:31:08] ERROR[2279][C-]: cdr_odbc.c:177 odbc_log:
CDR direct execute failed


See my res_odbc.conf

[asterisk]
enabled = yes
dsn = asterisk-connector
username = root
password = 100567
pooling = no
limit = 1
pre-connect = yes

What can be happened?

Thank in advanced.



--

Message: 2
Date: Wed, 13 Jan 2016 14:26:08 +
From: Matthew Murphy <mrm...@outlook.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: [asterisk-users] "pjsip show endpoints" returns "No Objects
Found"  in 13.7.0-rc2
Message-ID:

<cy1pr15mb0314208ec2c6d60fa4a5f6afd9...@cy1pr15mb0314.namprd15.prod.outlook.com>

Content-Type: text/plain; charset="iso-8859-1"

Hi everyone,


I have just upgraded to Asterisk 13.7.0-rc2 and noticed that when I type "pjsip 
show endpoints" at the CLI, I get "No Objects Found".


However, if I request information on a specific endpoint, (for example: "pjsip 
show endpoint 101") then I get all of the information for that endpoint as 
expected.


This seems to have started as soon as I upgraded to 13.7.0-rc2. I tried with 
pjproject 2.4 and now pjproject 2.4.5 and get the same result.


Has anyone else seen this or is it something that is unique to my situation?


Thanks,


--Matt
-- next part --
An HTML attachment was scrubbed...
URL: 
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160113/d8804cd5/attachment-0001.html>

--

Message: 3
Date: Wed, 13 Jan 2016 10:34:49 -0400
From: Joshua Colp <jc...@digium.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] "pjsip show endpoints"

Re: [asterisk-users] asterisk-users Digest, Vol 138, Issue 8

[A J Stiles]

On Wednesday 13 Jan 2016, waqas.mehmood90 wrote:
> How to get user extention no in agi php scrip from which he's calling on
> ivr i am using cid and able to get his name but not his extention no
> please help me

Within the dialplan, what you are looking for would be ${CALLERID(num)} .  So 
you could just pass this as a parameter to your AGI script.  BUT, it may well 
be already there as an environment variable within the script.

Try writing a "dummy" AGI script that simply dumps the environment array to a 
temporary file and then returns to the dialplan.

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

-- 
_
-- 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:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] warble or clicking sound with 11.20.0 with Console/dsp

[Jerry Geis]

I am running 11.20.0 (64 bit) as a user other than root
and using the Console/dsp port (soundcard) output and HDMI.

I am getting a warble or clicking noise on the audio.
I'm connected into the pulseaudio for the logged in user.
Pulseaudio works fine for everything else. Aplay is fine, totem is fine
with audio for a movie clip etc...

What can I do on asterisk to work better with pulse audio so no clicking ?

Thanks,

Jerry
-- 
_
-- 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:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] PJSIP Returning 421 Extension Required

[Trey Hilyard]

I am turning up a PJSIP Endpoint and am having problems when they send an
INVITE to my server. Asterisk is returning a 421 Extenstion Required. Since
"extension" means different things in the SIP stack versus Asterisk, I
don't know what it is complaining about.

I have attached the trace below. Nothing else shows up with core verbose or
core debug enabled, so I am assuming it has to be dying at the PJSIP
module. The INVITE does come from an abnormal UDP Port, which is also shown
in the Via header, but the fact that the PBX is responding makes me think
that isn't the culprit.

Any thoughts?

SIP Logger:
INVITE sip:+18165116504@12.4.240.200:5060;user=phone SIP/2.0
v: SIP/2.0/UDP 10.77.27.103:20065
;branch=z9hG4bK0020C575A392E895C39051;oc-accept
Max-Forwards: 70
t: 
f: ;tag=10847511385389740959
i: 117620342110831512016142@10.77.27.103
CSeq: 1 INVITE
d: no-fork
Privacy: none
P-Asserted-Identity: 
Require: 100rel
Accept: application/sdp
k: histinfo,resource-priority
c: application/sdp
m: 
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,UPDATE
l:   228

v=0
o=PVG 1452710812870 1452710812870 IN IP4 10.77.160.55
s=-
c=IN IP4 10.77.160.55
t=0 0
m=audio 37700 RTP/AVP 0 101
b=AS:80
b=RR:0
b=RS:0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20

<--- Transmitting SIP response (495 bytes) to UDP:10.77.27.103:20065 --->
SIP/2.0 421 Extension Required
Via: SIP/2.0/UDP 10.77.27.103:20065
;rport=20065;received=10.77.27.103;branch=z9hG4bK0020C575A392E895C39051;oc-accept
Call-ID: 117620342110831512016142@10.77.27.103
From: ;tag=10847511385389740959
To: ;tag=z9hG4bK0020C575A392E895C39051
CSeq: 1 INVITE
Require: 100rel
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 13.3.0-rc1
Content-Length:  0

PJSIP Endpoint:
zeus*CLI> pjsip show endpoint erc905

 Endpoint:  
   
I/OAuth:
 
Aor:  
 
  Contact:  
   
  Transport:
 
   Identify:
 
Match:  
Channel:  
   
Exten:   CLCID: 
 
=

 Endpoint:  erc905   Invalid
0 of inf
Aor:  erc905 0
  Contact:  erc905/sip:10.77.27.103:5060 Avail
 32.887
  Transport:  ngvn  udp  0 40  12.4.240.200:5060
   Identify:  erc905_1/erc905
Match: 10.77.27.103/32


 ParameterName : ParameterValue
 
 100rel: required
 accountcode   :
 aggregate_mwi : true
 allow : (ulaw)
 allow_subscribe   : true
 allow_transfer: true
 aors  : erc905
 auth  :
 call_group:
 callerid  : 
 callerid_privacy  : allowed_not_screened
 callerid_tag  :
 connected_line_method : invite
 context   : from_pstn
 cos_audio : 0
 cos_video : 0
 device_state_busy_at  : 0
 direct_media  : true
 direct_media_glare_mitigation : none
 direct_media_method   : invite
 disable_direct_media_on_nat   : false
 dtls_ca_file  :
 dtls_ca_path  :
 dtls_cert_file:
 dtls_cipher   :
 dtls_fingerprint  : SHA-256
 dtls_private_key  :
 dtls_rekey: 0
 dtls_setup: active
 dtls_verify   : No
 dtmf_mode : rfc4733
 fax_detect: false
 force_avp : false
 force_rport   : true
 from_domain   :
 from_user :
 ice_support   : false
 identify_by   : username
 inband_progress   : false
 language  :
 mailboxes :
 media_address :
 media_encryption  : none
 media_encryption_optimistic   : false
 media_use_received_transport  : false
 message_context   :
 moh_suggest   : default
 mwi_from_user :
 named_call_group  :
 named_pickup_group:
 one_touch_recording   : false
 outbound_auth :
 outbound_proxy:
 pickup_group  :
 record_off_feature: automixmon
 record_on_feature : automixmon
 rewrite_contact   : false
 rtp_engine: asterisk
 rtp_ipv6   

Re: [asterisk-users] "pjsip show endpoints" returns "No Objects Found" in 13.7.0-rc2

[Joshua Colp]

Matthew Murphy wrote:

Hi everyone,


I have just upgraded to *Asterisk 13.7.0-rc2* and noticed that when I
type "/pjsip show endpoints/" at the CLI, I get "/No Objects Found/".


Are you using realtime? A regression was found[1] in rc2 when realtime 
was in use which would cause this to happen. Only contacts were 
mentioned but it would impact other things. There is now an rc3 which 
has a fix for this in it.


[1] https://issues.asterisk.org/jira/browse/ASTERISK-25689

--
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


--
_
-- 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:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] "pjsip show endpoints" returns "No Objects Found" in 13.7.0-rc2

[Matthew Murphy]

I am using realtime - you got it! 

Great to know it already has a fix. I'll pull in rc3 and go from there.

Thanks a lot for your help!

--Matt


From: asterisk-users-boun...@lists.digium.com 
 on behalf of Joshua Colp 

Sent: Wednesday, January 13, 2016 9:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] "pjsip show endpoints" returns "No Objects Found" 
in 13.7.0-rc2

Matthew Murphy wrote:
> Hi everyone,
>
>
> I have just upgraded to *Asterisk 13.7.0-rc2* and noticed that when I
> type "/pjsip show endpoints/" at the CLI, I get "/No Objects Found/".

Are you using realtime? A regression was found[1] in rc2 when realtime
was in use which would cause this to happen. Only contacts were
mentioned but it would impact other things. There is now an rc3 which
has a fix for this in it.

[1] https://issues.asterisk.org/jira/browse/ASTERISK-25689

--
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


--
_
-- 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:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- 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:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] "pjsip show endpoints" returns "No Objects Found" in 13.7.0-rc2

[Matthew Murphy]

Hi everyone,


I have just upgraded to Asterisk 13.7.0-rc2 and noticed that when I type "pjsip 
show endpoints" at the CLI, I get "No Objects Found".


However, if I request information on a specific endpoint, (for example: "pjsip 
show endpoint 101") then I get all of the information for that endpoint as 
expected.


This seems to have started as soon as I upgraded to 13.7.0-rc2. I tried with 
pjproject 2.4 and now pjproject 2.4.5 and get the same result.


Has anyone else seen this or is it something that is unique to my situation?


Thanks,


--Matt
-- 
_
-- 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:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] cdr_odbc: Error in ExecDirect: -1

[Patrick Laimbock]

On 01/13/16 14:48, Vitor Mazuco wrote:

Hi everybody!

I'm trying to install a CDR Viewer https://github.com/g613/asterisk-cdr-viewer

I'ts work well, but my problem is to make Asterisk store to MySQL using ODBC

When I make a call the CLI returns for me

See the log:
== Using SIP RTP CoS mark 5
 -- Executing [2021@ramais:1] Dial("SIP/2020-",
"SIP/2021,60,tT") in new stack
   == Using SIP RTP CoS mark 5
 -- Called SIP/2021
 -- SIP/2021-0001 is ringing
> 0x7fd3f8014240 -- Probation passed - setting RTP source
address to 192.168.25.49:35528
[Jan 13 11:31:07] NOTICE[2279][C-]: res_rtp_asterisk.c:4441
ast_rtp_read: Unknown RTP codec 95 received from '(null)'
 -- SIP/2021-0001 answered SIP/2020-
> 0x7fd3b4004eb0 -- Probation passed - setting RTP source
address to 192.168.25.100:8000
> 0x7fd3f8014240 -- Probation passed - setting RTP source
address to 192.168.25.49:35528
> cdr_odbc: Error in ExecDirect: -1
[Jan 13 11:31:08] WARNING[2279][C-]: res_odbc.c:604
ast_odbc_direct_execute: SQL Execute error! Verifying connection to
asterisk [asterisk-connector]...
> cdr_odbc: Error in ExecDirect: -1
[Jan 13 11:31:08] ERROR[2279][C-]: cdr_odbc.c:177 odbc_log:
CDR direct execute failed


See my res_odbc.conf

[asterisk]
enabled = yes
dsn = asterisk-connector
username = root
password = 100567
pooling = no
limit = 1
pre-connect = yes

What can be happened?

Thank in advanced.


Just a guess but try setting "pooling" to yes and "limit" to a higher value.

Best,
Patrick

--
_
-- 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:
  http://lists.digium.com/mailman/listinfo/asterisk-users