Re: [asterisk-users] Meet me recording

[DHAVAL INDRODIYA]

Hi Satish,

You can Pass 'r' flag to meetme Application and file will be recorded nothin
to load mixmonitor and other Application on Channel, i think 'r' is better
than all options

Cheers
Dhaval

On Sat, Feb 19, 2011 at 1:37 AM, satish patel  wrote:

>  Thanks,
>
> look like monitor application resolved my issue.
>
> --
> From: da...@debsinc.com
> To: asterisk-users@lists.digium.com
> Date: Fri, 18 Feb 2011 09:16:36 -0600
> Subject: Re: [asterisk-users] Meet me recording
>
>
>   --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *satish patel
> *Sent:* Friday, February 18, 2011 9:12 AM
> *To:* asterisk-users
> *Subject:* [asterisk-users] Meet me recording
>
>
>
> Hey Users,
>
> I am using record application to record MeetMe conf. but look like its
> creating individual files for every channel. What applucation is best to
> record MeetMe conf ?
>
>
> ~ # ls -l /var/spool/asterisk/monitor/
> total 489220
> -rw-r--r-- 1 asterisk asterisk   44 Feb 16 08:42
> 8881-conf-20110216-084224.wav
> -rw-r--r-- 1 asterisk asterisk  1858284 Feb 16 13:05
> 8881-conf-20110216-130321.wav
> -rw-r--r-- 1 asterisk asterisk  1604204 Feb 16 13:05
> 8881-conf-20110216-130337.wav
> -rw-r--r-- 1 asterisk asterisk   241964 Feb 17 08:20
> 8881-conf-20110217-081957.wav
> -rw-r--r-- 1 asterisk asterisk 78678124 Feb 17 11:12
> 8881-conf-20110217-095056.wav
> -rw-r--r-- 1 asterisk asterisk   612204 Feb 17 09:53
> 8881-conf-20110217-095310.wav
> -rw-r--r-- 1 asterisk asterisk 81183084 Feb 17 11:13
> 8881-conf-20110217-095414.wav
> -rw-r--r-- 1 asterisk asterisk 69488044 Feb 17 11:12
> 8881-conf-20110217-100012.wav
> -rw-r--r-- 1 asterisk asterisk 68917164 Feb 17 11:12
> 8881-conf-20110217-100052.wav
> -rw-r--r-- 1 asterisk asterisk 66971884 Feb 17 11:11
> 8881-conf-20110217-100117.wav
> -rw-r--r-- 1 asterisk asterisk 66648684 Feb 17 11:12
> 8881-conf-20110217-100327.wav
> -rw-r--r-- 1 asterisk asterisk 45056044 Feb 17 11:06
> 8881-conf-20110217-102007.wav
>
>
> Thanks,
> S
>
>
>
> From what I read, mixmonitor.
>
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Re: [asterisk-users] AGI script dies after receivefax

[Mike Diehl]

--

Take care and have fun,
Mike Diehl.



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Re: [asterisk-users] AGI script dies after receivefax

[Mike Diehl]

--

Take care and have fun,
Mike Diehl.



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Re: [asterisk-users] AGI script dies after receivefax

[Steve Edwards]

On Fri, 18 Feb 2011, Mike Diehl wrote:


I've got a perl agi script that exec()'s the FFA version of receivefax to... 
receive a fax.

However, after the fax is received, the script seems to die.

This is what I have:

$main::agi->exec("receivefax","/tmp/${$}.tiff|fs");
$main::agi->verbose("FAX COMPLETE",1);

I never see the "FAX COMPLETE" message on the console, I've set verbose to 25.  
Any ideas?  I'd like to take the next few instructions to log
success/failure.


What version of Asterisk?

Would enabling AGI debugging on the console shed any light?

Any chance receivefax is generating a signal you're not trapping?

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000--
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[asterisk-users] AGI script dies after receivefax

[Mike Diehl]

Hi all,

I've got a perl agi script that exec()'s the FFA version of receivefax to...
receive a fax.

However, after the fax is received, the script seems to die.

This is what I have:

$main::agi->exec("receivefax","/tmp/${$}.tiff|fs");
$main::agi->verbose("FAX COMPLETE",1);

I never see the "FAX COMPLETE" message on the console, I've set verbose to
25.  Any ideas?  I'd like to take the next few instructions to log
success/failure.
--

Take care and have fun,
Mike Diehl.

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[asterisk-users] Problem in dialing out

[asterisk asterisk]

I have a sip trunk connecting to a huawei softx3000. At the moment, I can
register and dial in.

However, peer status shows not reachable

sip show peer as follow

  * Name   : cmphone
  Secret   : 
  MD5Secret: 
  Remote Secret: 
  Context  : from-cmphone
  Subscr.Cont. : device-hints
  Language :
  AMA flags: Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  MOH Suggest  :
  Mailbox  :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Max forwards : 0
  Dynamic  : No
  Callerid : "" <>
  MaxCallBR: 384 kbps
  Expire   : -1
  Insecure : port,invite
  Force rport  : Yes
  ACL  : No
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID: No
  Subscriptions: Yes
  Overlap dial : Yes
  Outb. proxy  : 202.0.179.3
  DTMFmode : rfc2833
  Timer T1 : 500
  Timer B  : 32000
  ToHost   : 202.0.179.3
  Addr->IP : 202.0.179.3:5060
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 852350xx
  SIP Options  : 100rel
  Codecs   : 0xe (gsm|ulaw|alaw)
  Codec Order  : (alaw:20,ulaw:20,gsm:20)
  Auto-Framing :  No
  100 on REG   : No
  Status   : UNREACHABLE
  Useragent:
  Reg. Contact :
  Qualify Freq : 6 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess : 90 secs
  RTP Engine   : asterisk
  Parkinglot   :
  Use Reason   : No
  Encryption   : No

In sip.conf

I have

register = 852350x:secret@202.0.179.3

[cmphone]
type = friend
host = 202.0.179.3
secret = secret
username = 852350x
context = from-cmphone
dtmfmode = rfc2833
outboundproxy = 202.0.179.3
caninvite=no
insecure = port,invite
nat = yes

When debug is on, the error message is


<--- SIP read from UDP:202.0.179.3:5060 --->
SIP/2.0 504 Server Time-out
From: "asterisk" ;tag=as2d14b9ec
To: ;tag=6b0704d0
CSeq: 102 OPTIONS
Call-ID: 17e0315c21d7dbc10e8c185740e21...@sip.x.xxx
Via: SIP/2.0/UDP
14.xxx.xxx.xxx:5060;branch=z9hG4bK3646eaf2;received=14.xxx.xxx.xxx;rport=5060
Content-Length: 0

Any help is appreciate.

CK
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Re: [asterisk-users] Newbie´s question about Asterisk...

[Francisco Javier Cintrón Olguín]

First of all, thank you for your help.

I was seing Cisco and Linsys web sites and I just came across this 2
devices:

Linksys SPA8000 8 phone ports, 1 port ethernet.
Cisco SPA8800 4 phone ports, 4 lines, 1 port ethernet.

I think they could work for us, because I need maximum 10 normal phones and
4 PSTN lines. Besides with these devices I could use my normal phones, so I
would not need additional wiring.

So far, this is the list  to evaluate the costs of this Linux PBX solution:

1 HP Proliant Micro Server
1 Linksys SPA8000
1 Cisco SPA8800
1 UPS
1 switch 16 ports( we use a 8 port switch)
1 shelf rack
3 patch cords.

I just have 3 doubts:


   1. What do you think about Linksys SPA8000 and Linksys SPA8800, are they
   good solutions in my case?
   2. What do you think about this list, am I missing something?
   3. I am thinking to buy a switch with VLANS to have one VLAN for my PBX,
   what do you think about this, is it necessary?



Thank you for your kind help.





On Fri, Feb 18, 2011 at 11:22 AM, Steve Edwards
wrote:

> (Please don't top-post and please trim posts that are no longer relevant.)
>
> There re USB and Ethernet devices (Xorcom, Sangoma, Sipura/Linksys/Cisco,
> and others) that can interface analog phones to your Asterisk server.
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
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Re: [asterisk-users] no progress indication

[Paul Belanger]

On 11-02-18 03:59 PM, Cassius Smith wrote:
> I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP
> only trunks, and this server only has soft phones.
> When I dial an extension and the phone is not registered, I don't get any
> ring or progress indications, and eventually the Dial() times out and
> drops into voicemail (as expected).
> 
*CLI> core show application Progress()

> CLI output:
> -- Executing [s@macro-StdExten:6] Dial("IAX2/barneveld-2036",
> "SIP/RickEndpoint&SIP/xlite-RickEndpoint,20") in new stack
>   == Using SIP RTP CoS mark 5
> [Feb 18 20:43:04] WARNING[6160]: acl.c:698 ast_ouraddrfor: Cannot connect
> [Feb 18 20:43:04] WARNING[6160]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
> -- Called RickEndpoint
> [Feb 18 20:43:04] WARNING[6160]: app_dial.c:2039 dial_exec_full: Unable to
> create channel of type 'SIP' (cause 20 - Unknown)
> [Feb 18 20:43:04] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
> [Feb 18 20:43:05] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
> [Feb 18 20:43:07] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
>   == Spawn extension (macro-StdExten, s, 6) exited non-zero on
> 'IAX2/barneveld-2036' in macro 'StdExten'
>   == Spawn extension (no911, RickEndpoint, 1) exited non-zero on
> 'IAX2/barneveld-2036'
> -- Hungup 'IAX2/barneveld-2036'
> [Feb 18 20:43:11] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
> [Feb 18 20:43:19] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
> [Feb 18 20:43:35] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
> [Feb 18 20:43:36] WARNING[3550]: chan_sip.c:3386 retrans_pkt:
> Retransmission timeout reached on transmission
> 367fd44f3a944b134765a4dc4c95b88d@127.0.0.1:5060 for seqno 102 (Critical
> Request) -- See doc/sip-retransmit.txt.
> 
There is something going wrong here, netsock2 is not parsing the IP
address correctly.  Are you using realtime?  It would be good to see a
full debug[1] log of your call.

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

-- 
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] no progress indication

[Satish Patel]

Try to use Answer() in your dial plan. Not sure though but it had been  
resoved my issue years ago.


--
Sent from my iPhone

On Feb 18, 2011, at 3:59 PM, Cassius Smith  wrote:

I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with  
VOIP

only trunks, and this server only has soft phones.
When I dial an extension and the phone is not registered, I don't  
get any

ring or progress indications, and eventually the Dial() times out and
drops into voicemail (as expected).

CLI output:
   -- Executing [s@macro-StdExten:6] Dial("IAX2/barneveld-2036",
"SIP/RickEndpoint&SIP/xlite-RickEndpoint,20") in new stack
 == Using SIP RTP CoS mark 5
[Feb 18 20:43:04] WARNING[6160]: acl.c:698 ast_ouraddrfor: Cannot  
connect
[Feb 18 20:43:04] WARNING[6160]: chan_sip.c:3115 __sip_xmit:  
sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid  
argument

   -- Called RickEndpoint
[Feb 18 20:43:04] WARNING[6160]: app_dial.c:2039 dial_exec_full:  
Unable to

create channel of type 'SIP' (cause 20 - Unknown)
[Feb 18 20:43:04] WARNING[3550]: chan_sip.c:3115 __sip_xmit:  
sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid  
argument
[Feb 18 20:43:05] WARNING[3550]: chan_sip.c:3115 __sip_xmit:  
sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid  
argument
[Feb 18 20:43:07] WARNING[3550]: chan_sip.c:3115 __sip_xmit:  
sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid  
argument

 == Spawn extension (macro-StdExten, s, 6) exited non-zero on
'IAX2/barneveld-2036' in macro 'StdExten'
 == Spawn extension (no911, RickEndpoint, 1) exited non-zero on
'IAX2/barneveld-2036'
   -- Hungup 'IAX2/barneveld-2036'
[Feb 18 20:43:11] WARNING[3550]: chan_sip.c:3115 __sip_xmit:  
sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid  
argument
[Feb 18 20:43:19] WARNING[3550]: chan_sip.c:3115 __sip_xmit:  
sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid  
argument
[Feb 18 20:43:35] WARNING[3550]: chan_sip.c:3115 __sip_xmit:  
sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid  
argument

[Feb 18 20:43:36] WARNING[3550]: chan_sip.c:3386 retrans_pkt:
Retransmission timeout reached on transmission
367fd44f3a944b134765a4dc4c95b88d@127.0.0.1:5060 for seqno 102  
(Critical

Request) -- See doc/sip-retransmit.txt.



Here is my StdExten macro:

[macro-StdExten]
exten => s,1,Verbose(2,>>>Processing StdExten call for
${MACRO_EXTEN})
exten => s,n,Verbose(2,CallerID => ${CALLERID(all)})
exten => s,n,Set(Device=${ARG1})
exten => s,n,Set(UserID=${MACRO_EXTEN})
exten => s,n,Dial(${ARG1},${ARG2})
exten => s,n,Verbose(2,==> Voicemail ${MACRO_EXTEN} -- unavail)
exten => s,n,Voicemail(${MACRO_EXTEN}@default,u)
exten => s,n,Hangup()


I was expecting the system to indicate that ringing was ?
I know I can check channel availability to bypass this behavior; just
curious why it's happening or whether it's expected.

Cassius

--





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[asterisk-users] no progress indication

[Cassius Smith]

I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP
only trunks, and this server only has soft phones.
When I dial an extension and the phone is not registered, I don't get any
ring or progress indications, and eventually the Dial() times out and
drops into voicemail (as expected).

CLI output:
-- Executing [s@macro-StdExten:6] Dial("IAX2/barneveld-2036",
"SIP/RickEndpoint&SIP/xlite-RickEndpoint,20") in new stack
  == Using SIP RTP CoS mark 5
[Feb 18 20:43:04] WARNING[6160]: acl.c:698 ast_ouraddrfor: Cannot connect
[Feb 18 20:43:04] WARNING[6160]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
-- Called RickEndpoint
[Feb 18 20:43:04] WARNING[6160]: app_dial.c:2039 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
[Feb 18 20:43:04] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
[Feb 18 20:43:05] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
[Feb 18 20:43:07] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
  == Spawn extension (macro-StdExten, s, 6) exited non-zero on
'IAX2/barneveld-2036' in macro 'StdExten'
  == Spawn extension (no911, RickEndpoint, 1) exited non-zero on
'IAX2/barneveld-2036'
-- Hungup 'IAX2/barneveld-2036'
[Feb 18 20:43:11] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
[Feb 18 20:43:19] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
[Feb 18 20:43:35] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
[Feb 18 20:43:36] WARNING[3550]: chan_sip.c:3386 retrans_pkt:
Retransmission timeout reached on transmission
367fd44f3a944b134765a4dc4c95b88d@127.0.0.1:5060 for seqno 102 (Critical
Request) -- See doc/sip-retransmit.txt.



Here is my StdExten macro:

[macro-StdExten]
exten => s,1,Verbose(2,>>>Processing StdExten call for
${MACRO_EXTEN})
exten => s,n,Verbose(2,CallerID => ${CALLERID(all)})
exten => s,n,Set(Device=${ARG1})
exten => s,n,Set(UserID=${MACRO_EXTEN})
exten => s,n,Dial(${ARG1},${ARG2})
exten => s,n,Verbose(2,==> Voicemail ${MACRO_EXTEN} -- unavail)
exten => s,n,Voicemail(${MACRO_EXTEN}@default,u)
exten => s,n,Hangup()


I was expecting the system to indicate that ringing was ?
I know I can check channel availability to bypass this behavior; just
curious why it's happening or whether it's expected.

Cassius

-- 





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Re: [asterisk-users] Meet me recording

[satish patel]

Thanks, 

look like monitor application resolved my issue. 

From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Fri, 18 Feb 2011 09:16:36 -0600
Subject: Re: [asterisk-users] Meet me recording



























From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel

Sent: Friday, February 18, 2011
9:12 AM

To: asterisk-users

Subject: [asterisk-users] Meet me
recording



 

Hey Users,



I am using record application to record MeetMe conf. but look like its creating
individual files for every channel. What applucation is best to record MeetMe
conf ?





~ # ls -l /var/spool/asterisk/monitor/

total 489220

-rw-r--r-- 1 asterisk asterisk   44 Feb 16
08:42 8881-conf-20110216-084224.wav

-rw-r--r-- 1 asterisk asterisk  1858284 Feb 16 13:05
8881-conf-20110216-130321.wav

-rw-r--r-- 1 asterisk asterisk  1604204 Feb 16 13:05
8881-conf-20110216-130337.wav

-rw-r--r-- 1 asterisk asterisk   241964 Feb 17 08:20
8881-conf-20110217-081957.wav

-rw-r--r-- 1 asterisk asterisk 78678124 Feb 17 11:12
8881-conf-20110217-095056.wav

-rw-r--r-- 1 asterisk asterisk   612204 Feb 17 09:53
8881-conf-20110217-095310.wav

-rw-r--r-- 1 asterisk asterisk 81183084 Feb 17 11:13
8881-conf-20110217-095414.wav

-rw-r--r-- 1 asterisk asterisk 69488044 Feb 17 11:12
8881-conf-20110217-100012.wav

-rw-r--r-- 1 asterisk asterisk 68917164 Feb 17 11:12
8881-conf-20110217-100052.wav

-rw-r--r-- 1 asterisk asterisk 66971884 Feb 17 11:11
8881-conf-20110217-100117.wav

-rw-r--r-- 1 asterisk asterisk 66648684 Feb 17 11:12
8881-conf-20110217-100327.wav

-rw-r--r-- 1 asterisk asterisk 45056044 Feb 17 11:06
8881-conf-20110217-102007.wav





Thanks,

S

 

>From what I read, mixmonitor.







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Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2

[Tilghman Lesher]

On Friday 18 February 2011 05:29:56 Borin wrote:
> Hello,
> trying to load ael module in asterisk ver 1.6.2 got the following:
> 
> asterisk*CLI> module load pbx_ael.so
> Unable to load module pbx_ael.so
> Command 'module load pbx_ael.so' failed.
> [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error
> loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so:
> undefined symbol: ast_compile_ael2
> [Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module
> 'pbx_ael.so' could not be loaded.
> 
> I did not find in google what it could be and what should be done to
> solve this. I also tried the same on ast ver 1.8.2.3, got the same. I
> am usind debian as OS and install asterisk from sources that I took on
> digium site. Did anyone have the same issue?

Make sure res_ael_share.so is loaded first.

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Re: [asterisk-users] Fwd: cmd MySQL

[Felipe Figueiredo]

Faisal,

yes, that's it!! I'm using 1.8.x , I didn't know about it!!!
Thank you so much, guy!!!

On Fri, Feb 18, 2011 at 4:26 PM, Faisal Hanif  wrote:

> If you are using asterisk 1.8.x you don’t need to type \ for spaces you can
> write simple query and use spaces as normal it will work fine.
>
>
>
> Faisal
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felipe Figueiredo
> *Sent:* Friday, February 18, 2011 11:04 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Fwd: cmd MySQL
>
>
>
>
>
> -- Forwarded message --
> From: *Felipe Figueiredo* 
> Date: Fri, Feb 18, 2011 at 4:03 PM
> Subject: Re: [asterisk-users] cmd MySQL
> To: Gerald A 
>
> - Executing [200@teste:2] MYSQL("Console/dsp", "Query resultid 1 SELECT\
> ramal\ FROM\ colaboradores\ WHERE\ ramal=200") in new stack
>
> [Feb 18 16:01:42] WARNING[7749]: app_mysql.c:393 aMYSQL_query:
> aMYSQL_query: mysql_query failed. Error: You have an error in your SQL
> syntax; check the manual that corresponds to your MySQL server version for
> the right syntax to use near '\ ramal\ FROM\ colaboradores\ WHERE\
> ramal=200' at line 1
>
>
>
> hi Gerald,
>
>
>
> look, the error is the same. Eveng changing the "/" for "\" ...
>
>
>
> On Fri, Feb 18, 2011 at 4:00 PM, Gerald A  wrote:
>
> Hi Felipe,
>
> On Fri, Feb 18, 2011 at 12:56 PM, Felipe Figueiredo <
> felipe.figueired...@gmail.com> wrote:
>
>
>
> -- Executing [200@teste:2] MYSQL("Console/dsp", "Query resultid 1 SELECT/
> ramal/ FROM/ colaboradores/ WHERE/ ramal=200") in new stack
>
> [Feb 18 15:55:13] WARNING[7696]: app_mysql.c:393 aMYSQL_query:
> aMYSQL_query: mysql_query failed. Error: You have an error in your SQL
> syntax; check the manual that corresponds to your MySQL server version for
> the right syntax to use near '/ ramal/ FROM/ colaboradores/ WHERE/
> ramal=200' at line 1
>
>
> I'm not Asterisk-MySQL guru, but shouldn't the "/" be "\"?
>
> I'm guessing you are trying to keep a string together here, but maybe I'm
> mistaken.
>
> Thanks,
> Gerald
>
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Fwd: cmd MySQL

[Faisal Hanif]

If you are using asterisk 1.8.x you don't need to type \ for spaces you can
write simple query and use spaces as normal it will work fine.

 

Faisal

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felipe
Figueiredo
Sent: Friday, February 18, 2011 11:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Fwd: cmd MySQL

 

 

-- Forwarded message --
From: Felipe Figueiredo 
Date: Fri, Feb 18, 2011 at 4:03 PM
Subject: Re: [asterisk-users] cmd MySQL
To: Gerald A 



- Executing [200@teste:2] MYSQL("Console/dsp", "Query resultid 1 SELECT\
ramal\ FROM\ colaboradores\ WHERE\ ramal=200") in new stack

[Feb 18 16:01:42] WARNING[7749]: app_mysql.c:393 aMYSQL_query: aMYSQL_query:
mysql_query failed. Error: You have an error in your SQL syntax; check the
manual that corresponds to your MySQL server version for the right syntax to
use near '\ ramal\ FROM\ colaboradores\ WHERE\ ramal=200' at line 1

 

hi Gerald, 

 

look, the error is the same. Eveng changing the "/" for "\" ... 

 

On Fri, Feb 18, 2011 at 4:00 PM, Gerald A  wrote:

Hi Felipe,

On Fri, Feb 18, 2011 at 12:56 PM, Felipe Figueiredo
 wrote:

 

-- Executing [200@teste:2] MYSQL("Console/dsp", "Query resultid 1 SELECT/
ramal/ FROM/ colaboradores/ WHERE/ ramal=200") in new stack

[Feb 18 15:55:13] WARNING[7696]: app_mysql.c:393 aMYSQL_query: aMYSQL_query:
mysql_query failed. Error: You have an error in your SQL syntax; check the
manual that corresponds to your MySQL server version for the right syntax to
use near '/ ramal/ FROM/ colaboradores/ WHERE/ ramal=200' at line 1


I'm not Asterisk-MySQL guru, but shouldn't the "/" be "\"?

I'm guessing you are trying to keep a string together here, but maybe I'm
mistaken.

Thanks,
Gerald

 

 

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[asterisk-users] Fwd: cmd MySQL

[Felipe Figueiredo]

-- Forwarded message --
From: Felipe Figueiredo 
Date: Fri, Feb 18, 2011 at 4:03 PM
Subject: Re: [asterisk-users] cmd MySQL
To: Gerald A 


- Executing [200@teste:2] MYSQL("Console/dsp", "Query resultid 1 SELECT\
ramal\ FROM\ colaboradores\ WHERE\ ramal=200") in new stack
[Feb 18 16:01:42] WARNING[7749]: app_mysql.c:393 aMYSQL_query: aMYSQL_query:
mysql_query failed. Error: You have an error in your SQL syntax; check the
manual that corresponds to your MySQL server version for the right syntax to
use near '\ ramal\ FROM\ colaboradores\ WHERE\ ramal=200' at line 1

hi Gerald,

look, the error is the same. Eveng changing the "/" for "\" ...

On Fri, Feb 18, 2011 at 4:00 PM, Gerald A  wrote:

> Hi Felipe,
>
> On Fri, Feb 18, 2011 at 12:56 PM, Felipe Figueiredo <
> felipe.figueired...@gmail.com> wrote:
>
>>
>> -- Executing [200@teste:2] MYSQL("Console/dsp", "Query resultid 1 SELECT/
>> ramal/ FROM/ colaboradores/ WHERE/ ramal=200") in new stack
>> [Feb 18 15:55:13] WARNING[7696]: app_mysql.c:393 aMYSQL_query:
>> aMYSQL_query: mysql_query failed. Error: You have an error in your SQL
>> syntax; check the manual that corresponds to your MySQL server version for
>> the right syntax to use near '/ ramal/ FROM/ colaboradores/ WHERE/
>> ramal=200' at line 1
>>
>
> I'm not Asterisk-MySQL guru, but shouldn't the "/" be "\"?
>
> I'm guessing you are trying to keep a string together here, but maybe I'm
> mistaken.
>
> Thanks,
> Gerald
>
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Re: [asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?

[Steve Edwards]

On Fri, 18 Feb 2011 09:19:01 -0800 (PST), Steve Edwards



The action and username lines were followed by pressing .
The secret line was followed by pressing .


On Fri, 18 Feb 2011, Gilles wrote:


Thanks for the tip. I figured this out after a while ;-)

I can now successfully log on, although I have to type the whole set
twice:


Odd. I don't even using telnet which is not a 'transparent' connection. 
Netcat (nc) would probably make a better took for testing your 
understanding of the protocol.


I'd resolve the 'twice' bit before going further.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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[asterisk-users] cmd MySQL

[Felipe Figueiredo]

Hi guys,

I'm trying to connect Asterisk to the MySQL, but I can't execute it. It
returns an error, as below:

-- Executing [200@teste:2] MYSQL("Console/dsp", "Query resultid 1 SELECT/
ramal/ FROM/ colaboradores/ WHERE/ ramal=200") in new stack
[Feb 18 15:55:13] WARNING[7696]: app_mysql.c:393 aMYSQL_query: aMYSQL_query:
mysql_query failed. Error: You have an error in your SQL syntax; check the
manual that corresponds to your MySQL server version for the right syntax to
use near '/ ramal/ FROM/ colaboradores/ WHERE/ ramal=200' at line 1

Its seems it can connect to mysql


My extension (AEL) is:

MySQL(Connect conn_id localhost root 123456 crm);
MySQL(Query resultid ${conn_id} SELECT/ ramal/ FROM/
colaboradores/ WHERE/ ramal=${EXTEN});
MySQL(Fetch fetchid ${resultid} RAMAL);
MySQL(Clear ${fetchid});
MySQL(Disconnect ${connid});
MySQL(Clear ${connid});
NoOp(${RAMAL});



Where is the error? Thanks!!



The MySQL server is in the same server where Asterisk is running.

Thanks!!!
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Re: [asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?

[Gilles]

On Fri, 18 Feb 2011 09:19:01 -0800 (PST), Steve Edwards
 wrote:
>The action and username lines were followed by pressing .
>The secret line was followed by pressing .

Thanks for the tip. I figured this out after a while ;-)

I can now successfully log on, although I have to type the whole set
twice:

== telnet
Asterisk Call Manager/1.0
action: Login
username: admin
secret: test

Response: Error
Message: Missing action in request

action: Login
username: admin
secret: test

Response: Success
Message: Authentication accepted
==

Could it be that AMI uses Asterisk's web server and relies on cookies?

Thank you.


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Re: [asterisk-users] Assigning an extension to a roaming phone

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle
Sent: Friday, February 18, 2011 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Assigning an extension to a roaming phone

Hi,

>I'm thinking that _4XXX is an "over-complication".  _4XXX means you could
>dial any number from 4000 through 4999 inclusive and get the extension at
>SIP/${ROAM}.

Well it's kind of what I want.
I have a roaming phone that comes in. He dials 3001, sets his
extension to 123, so that he is assigned 4123.
I have another roamding phone that comes in. Dials 3001, sets his
extension to 124. He is assigned 4124.
Or at least that's how I understand it.

In reality, what I only need is all roaming phones to get assigned an
extension (within a given range) and to have a way to find their
extension number.
Roaming phone 1 comes in. Get assigned (automatically) 4001.
Roaming phone 2 comes in. Get assigned (automatically) 4002.
Roaming phone 3 comes in. Get assigned (automatically) 4003.
etc

>I'd change the line 2
>- exten => _4XXX,n,Dial(SIP/${ROAM},30,,mKkTt)

looks pretty similar to the previous line - apart from that mKkTt.
What is that for?
What's wrong with the previous line.

>Or
>- exten => 4123,1,Dial(SIP/${ROAM},30,,mKkTt)

That would only match case where roaming user wants to be assigned
4123, but it would not work for - say - 4124.

Thanks.

Ok, so you need a "roaming magic dial" where 4XXX dials the assigned phone.
Your original command
- exten => _4XXX,n,Dial(SIP/${ROAM}) 
Technically should work, just has no timeouts or control on it.
The ,30 gives the Dial 30 seconds (about 6 rings) to complete, and mKkTt
gives the caller music-on-hold until answer/time then lets the dialee
re-transfer the call.

As I understand what you just wrote, folks come into your shop with a cell
phone (555-1212) and dial a number 3001 to tell your inhouse folks to reach
the cell at 4XXX.  Say "Joe" comes in and dials 3001.  Asterisk should say
"what is your name".  He says "Joe" and Asterisk says "callers can now reach
Joe at 4001".  "Jim" comes in and does the same thing and gets 4002 until
999 folks do it.  Is that correct?



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Re: [asterisk-users] Newbie´s question about Asterisk...

[Steve Edwards]

(Please don't top-post and please trim posts that are no longer relevant.)

On Fri, 18 Feb 2011, Francisco Javier Cintrón Olguín wrote:

I think I have 3 PSTN lines because I can connect a normal telephone to 
them all and make calls between each of them. We have 5 normal 
telephones and 1 panasonic.


From what I got I need a PC  and a of PCI card to interface to my 3 
external lines and my 6 internal lines.


For the PC I was planning to use the smallest PC posible like a HP 
Proliant Microserver  but it doesn´t have space for this PCI card. Is 
there another way to interface to 3 external and 6 internal lines??


There re USB and Ethernet devices (Xorcom, Sangoma, Sipura/Linksys/Cisco, 
and others) that can interface analog phones to your Asterisk server.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000--
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Re: [asterisk-users] Newbie´s question about Asterisk...

[Terry Brummell]

Yes, use a FXO device, like the AudioCodes MP-114.  It is an external gateway 
that will allow you to interface your PSTN lines to Asterisk via IP.  There are 
other brands out there but in my line of business we only use AudioCodes.



From: asterisk-users-boun...@lists.digium.com on behalf of Francisco Javier 
Cintrón Olguín
Sent: Fri 2/18/2011 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]Newbie´s question about Asterisk...


Is there another way to interface to 3 external and 6 internal lines??

Thank you for your kind help


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Re: [asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?

[Steve Edwards]

On Fri, 18 Feb 2011, Gilles wrote:


I'm not having much luck with AMI: After typing the right commands, it
just stays there, not replying to the Login action:



= Telnet to TCP5038
Action: Login
Username: admin
Secret: secret



It's waiting for another .

This is from a 1.2 box, but it should look like this:

-t2::sedwards:~$ telnet localhost 5038
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/1.0
action: login
username: example
secret: example

Response: Success
Message: Authentication accepted

The action and username lines were followed by pressing .
The secret line was followed by pressing .

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Newbie´s question about Asterisk...

[Francisco Javier Cintrón Olguín]

I think I have 3 PSTN lines because I can connect a normal telephone to them
all and make calls between each of them.
We have 5 normal telephones and 1 panasonic.

>From what I got I need a PC  and a of PCI card to interface to my 3 external
lines and my 6 internal lines.

For the PC I was planning to use the smallest PC posible like a HP Proliant
Microserver  but it doesn´t have space for this PCI card.
Is there another way to interface to 3 external and 6 internal lines??

Thank you for your kind help.


On Fri, Feb 18, 2011 at 6:52 AM, Gopalakrishnan A.N wrote:

> Asterisk is open source and you can install in a normal PC itself and you
> can avail all the features that proprietary system has.
>
> If you want to integrate with any VoIP service then a PC with Asterisk is
> enough or else if you want to integrate with PSTN lines then you need FXO
> card to be installed, its a PCI card. Vendors like Sangoma, Digium (from
> Asterisk) were selling these cards. And for internal for your agents if you
> need analog hard phones then you need to have FXS card you can avail these
> FXO and FXS cards in combination. These cards will fit in your PCI slot of
> machine. Configuring these cards are also very easy.
>
> If it is VoIP then you dont need these cards simply install Asterisk in a
> PC and you are done.
>
>
> On Fri, Feb 18, 2011 at 5:38 PM, Terry Brummell wrote:
>
>> Yes, I use Elastix myself too.  Funny that I didn’t mention that one!
>>
>>
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA
>> *Sent:* Friday, February 18, 2011 6:11 AM
>>
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...
>>
>>
>>
>> i prefer to go with Elastix very easy to setup and maintain and reach UI
>> rather than freePBX
>>
>> cheers
>> Dhaval
>>
>> On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell 
>> wrote:
>>
>> Dean’s link has references to Trixbox.  TB has a bad, bad, very bad
>> reputation for being very insecure.  Alternatives to TB are FreePBX & PBX in
>> a Flash.  All are Asterisk based and very easy to set up.
>>
>>
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dean Collins
>> *Sent:* Thursday, February 17, 2011 7:29 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...
>>
>>
>>
>> If you already have experience with linux asterisk will be easy for you.
>>
>>
>>
>> Other people will reply with official links but here is how I use Asterisk
>> in my small home office www.cognation.net/asterisk
>>
>>
>>
>>
>>
>> Cheers,
>>
>> Dean
>>
>>
>>
>>
>> --
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Francisco Javier
>> Cintrón Olguín
>> *Sent:* Thursday, February 17, 2011 7:26 PM
>> *To:* asterisk-users@lists.digium.com
>> *Subject:* [asterisk-users] Newbie´s question about Asterisk...
>>
>>
>>
>> Hi, My name is Francisco from México.
>>
>> Here, in my work we have a very very old panasonic PBX(12 years old). We
>> are growing and we need to increase our external lines(from 3 to 4) and our
>> internal lines(from 6 to 10). Besides we need voice mail and voice menu too.
>>
>>
>> We asked for a quote to our panasonic dealer. The whole thing cost about
>> 4,500 dollars.
>>
>> My boss just saw a thing called Asterisk this morning looking for options
>> in Google. He asked my to investigate what this thing called Asterisk is and
>> if we could save some money using it instead of the panasonic solution. So,
>> here I am.
>>
>> I have some experience as linux sysadmin(we have 1 oracle linux server and
>> 1 linux print server) nevertheless I don´t have any idea where and how to
>> start this evaluation?
>>
>>
>> Please
>> Would you give us a clue where to see If Asterisk could work for us?
>>
>> Thanks for your kind help.
>>
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> Thank you  with regards,
> Gopalakrishnan A.N.
> VoIP call -

Re: [asterisk-users] Newbie´s question about Asterisk...

[Francisco Javier Cintrón Olguín]

I think I have 3 PSTN lines because I can connect a normal telephone to them
all and make calls between each of them.
We have 5 normal telephones and 1 panasonic.

>From what I got I need a PC  and a of PCI card to interface to my 3 external
lines and my 6 internal lines.

For the PC I was planning to use the smallest PC posible like a HP Proliant
Microserver  but it doesn´t have space for this PCI card.
Is there another way to interface to 3 external and 6 internal lines??

Thank you for your kind help

On Fri, Feb 18, 2011 at 6:52 AM, Gopalakrishnan A.N wrote:

> Asterisk is open source and you can install in a normal PC itself and you
> can avail all the features that proprietary system has.
>
> If you want to integrate with any VoIP service then a PC with Asterisk is
> enough or else if you want to integrate with PSTN lines then you need FXO
> card to be installed, its a PCI card. Vendors like Sangoma, Digium (from
> Asterisk) were selling these cards. And for internal for your agents if you
> need analog hard phones then you need to have FXS card you can avail these
> FXO and FXS cards in combination. These cards will fit in your PCI slot of
> machine. Configuring these cards are also very easy.
>
> If it is VoIP then you dont need these cards simply install Asterisk in a
> PC and you are done.
>
>
> On Fri, Feb 18, 2011 at 5:38 PM, Terry Brummell wrote:
>
>> Yes, I use Elastix myself too.  Funny that I didn’t mention that one!
>>
>>
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA
>> *Sent:* Friday, February 18, 2011 6:11 AM
>>
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...
>>
>>
>>
>> i prefer to go with Elastix very easy to setup and maintain and reach UI
>> rather than freePBX
>>
>> cheers
>> Dhaval
>>
>> On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell 
>> wrote:
>>
>> Dean’s link has references to Trixbox.  TB has a bad, bad, very bad
>> reputation for being very insecure.  Alternatives to TB are FreePBX & PBX in
>> a Flash.  All are Asterisk based and very easy to set up.
>>
>>
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dean Collins
>> *Sent:* Thursday, February 17, 2011 7:29 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...
>>
>>
>>
>> If you already have experience with linux asterisk will be easy for you.
>>
>>
>>
>> Other people will reply with official links but here is how I use Asterisk
>> in my small home office www.cognation.net/asterisk
>>
>>
>>
>>
>>
>> Cheers,
>>
>> Dean
>>
>>
>>
>>
>> --
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Francisco Javier
>> Cintrón Olguín
>> *Sent:* Thursday, February 17, 2011 7:26 PM
>> *To:* asterisk-users@lists.digium.com
>> *Subject:* [asterisk-users] Newbie´s question about Asterisk...
>>
>>
>>
>> Hi, My name is Francisco from México.
>>
>> Here, in my work we have a very very old panasonic PBX(12 years old). We
>> are growing and we need to increase our external lines(from 3 to 4) and our
>> internal lines(from 6 to 10). Besides we need voice mail and voice menu too.
>>
>>
>> We asked for a quote to our panasonic dealer. The whole thing cost about
>> 4,500 dollars.
>>
>> My boss just saw a thing called Asterisk this morning looking for options
>> in Google. He asked my to investigate what this thing called Asterisk is and
>> if we could save some money using it instead of the panasonic solution. So,
>> here I am.
>>
>> I have some experience as linux sysadmin(we have 1 oracle linux server and
>> 1 linux print server) nevertheless I don´t have any idea where and how to
>> start this evaluation?
>>
>>
>> Please
>> Would you give us a clue where to see If Asterisk could work for us?
>>
>> Thanks for your kind help.
>>
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> Thank you  with regards,
> Gopalakrishnan A.N.
> VoIP call - s

Re: [asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?

[Gilles]

On Fri, 18 Feb 2011 17:11:18 +0100, Gilles 
wrote:
>>I'm guessing you would have better luck kicking off an external process 
>>that checks the channel status via AMI.
>
>Yes, it looks like it's not possible to reuse the FXO from either
>extensions.conf or through an AGI script.

I'm not having much luck with AMI: After typing the right commands, it
just stays there, not replying to the Login action:

= /etc/asterisk> cat manager.conf
[general]
enabled = yes
webenabled = yes
port = 5038
bindaddr = 0.0.0.0

[admin]
secret=test
read = system,call,log,verbose,command,agent,user,config
write = system,call,log,verbose,command,agent,user,config
= Telnet to TCP5038
Action: Login
Username: admin
Secret: secret

= CLI
CLI> manager show commands
Action   PrivilegeSynopsis
--   -
...
Hangup   call,all Hangup Channel
ListCommandsList available manager commands
Logoff  Logoff Manager
MailboxCount call,all Check Mailbox Message Count
...
=

It's odd that the Login command is not listed. FWIW, it's Asterisk
1.4.20.

Has someone seen this behavior and knows how to solve this?

Thank you.


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Re: [asterisk-users] Assigning an extension to a roaming phone

[Axelle]

Hi,

>I'm thinking that _4XXX is an "over-complication".  _4XXX means you could
>dial any number from 4000 through 4999 inclusive and get the extension at
>SIP/${ROAM}.

Well it's kind of what I want.
I have a roaming phone that comes in. He dials 3001, sets his
extension to 123, so that he is assigned 4123.
I have another roamding phone that comes in. Dials 3001, sets his
extension to 124. He is assigned 4124.
Or at least that's how I understand it.

In reality, what I only need is all roaming phones to get assigned an
extension (within a given range) and to have a way to find their
extension number.
Roaming phone 1 comes in. Get assigned (automatically) 4001.
Roaming phone 2 comes in. Get assigned (automatically) 4002.
Roaming phone 3 comes in. Get assigned (automatically) 4003.
etc

>I'd change the line 2
>- exten => _4XXX,n,Dial(SIP/${ROAM},30,,mKkTt)

looks pretty similar to the previous line - apart from that mKkTt.
What is that for?
What's wrong with the previous line.

>Or
>- exten => 4123,1,Dial(SIP/${ROAM},30,,mKkTt)

That would only match case where roaming user wants to be assigned
4123, but it would not work for - say - 4124.

Thanks.

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Re: [asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?

[Gilles]

On Fri, 18 Feb 2011 07:52:40 -0800 (PST), Steve Edwards
 wrote:
>I've never written an AGI in lua, but don't you have to read the AGI 
>environment (from STDIN) before issuing requests?

Thanks for pointing it out. I forgot to prepend that part to that test
script.

>Also, you execute your AGI in the 'h' extension. I think once a channel is 
>hung up, it's state will not change until you reach the end of your 
>dialplan execution and the channel is destroyed.

Mmm... looks like it :-/

>I'm guessing you would have better luck kicking off an external process 
>that checks the channel status via AMI.

Yes, it looks like it's not possible to reuse the FXO from either
extensions.conf or through an AGI script.

Since the callback must be automated, I guess I'll have to add the
script to "at" to schedule the call.
Too bad Asterisk doesn't provide a way to reuse a port after calling
Hangup().

Thank you.


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Re: [asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?

[Steve Edwards]

On Fri, 18 Feb 2011, Gilles wrote:


I'm using an AGI script in Lua to make a callback through Zaptel.



=== AGI script
#!/var/tmp/lua
for i=1,10 do
   io.write("CHANNEL STATUS\n")
   response=io.read()

   _, _, key, value = string.find(response, "(%a+)=(%d+)")

   --Channel never "down and available"!
   if value=="0" then
   io.write("NOOP Channel idle\n")
   response=io.read()
   else
   io.write("NOOP Channel N.A.\n")
   response=io.read()
   end
   os.execute("/bin/sleep 2")
end


I'm just a 1.2 Luddite...

I've never written an AGI in lua, but don't you have to read the AGI 
environment (from STDIN) before issuing requests?


Also, you execute your AGI in the 'h' extension. I think once a channel is 
hung up, it's state will not change until you reach the end of your 
dialplan execution and the channel is destroyed.


I'm guessing you would have better luck kicking off an external process 
that checks the channel status via AMI.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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[asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?

[Gilles]

Hello

I'm using an AGI script in Lua to make a callback through Zaptel.

For this to work, I must wait until the channel is idle, or I get this
kind of error, even after waiting over 10 seconds after the remote end
rings once and hangs up:
==
channel.c:2863 __ast_request_and_dial: Unable to request channel
Zap/1/123456
pbx_spool.c:341 attempt_thread: Call failed to go through, reason (0)
Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or
down?)
==

According to this article, CHANNEL STATUS=0 means that the line is
available: www.voip-info.org/wiki/view/channel+status

Here are extensions.conf, the AGI script, and what the CLI says:
=== extensions.conf
[from_fxo]
exten => s,1,Wait(2)
exten => s,n,Set(CID=${CALLERID(num)})
exten => s,n,Wait(10)
exten => s,n,Hangup

exten => h,1,DeadAGI(/var/tmp/test.lua)
=== AGI script
#!/var/tmp/lua
for i=1,10 do
io.write("CHANNEL STATUS\n")
response=io.read()

_, _, key, value = string.find(response, "(%a+)=(%d+)")

--Channel never "down and available"!
if value=="0" then
io.write("NOOP Channel idle\n")
response=io.read()
else
io.write("NOOP Channel N.A.\n")
response=io.read()
end
os.execute("/bin/sleep 2")
end
=== CLI
...
AGI Rx << CHANNEL STATUS
AGI Tx >> 200 result=4
AGI Rx << NOOP Channel N.A.
AGI Tx >> 200 result=0
AGI Rx << CHANNEL STATUS
AGI Tx >> 200 result=4
AGI Rx << NOOP Channel N.A.
AGI Tx >> 200 result=0
...
=== 

IOW, although I use Hangup() in extensions.conf and wait several
seconds before calling "CHANNEL STATUS", the channel is never marked
as "down and available".

Do I need to perform some extra steps in extensions.conf or in the
script for the FXO port to finally be available for a callback?

Thank you for any help.


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Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2

[Borin]

The error appears only if I load module. There is no warning during
installation, so module pbx_ael.so is compiled and placed in modules dir of
asterisk

On Fri, Feb 18, 2011 at 4:15 PM, Borin  wrote:

> Linux version 2.6.26-2-686 (Debian 2.6.26-24) (da...@debian.org) (gcc
> version 4.1.3 20080704 (prerelease) (Debian 4.1.2-25))
>
>
>
> On Fri, Feb 18, 2011 at 4:09 PM, Faisal Hanif  wrote:
>
>> Are you on CentOS?
>>
>>
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Borin
>> *Sent:* Friday, February 18, 2011 7:46 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] pbx_ael.so: undefined symbol:
>> ast_compile_ael2
>>
>>
>>
>>
>>
>> On Fri, Feb 18, 2011 at 3:29 PM, Faisal Hanif  wrote:
>>
>> Did you checked if you extension.ael doesn’t have syntax error?
>>
>>
>> I think there is no error. I loaded the standard ael first (provided by
>> asterisk) then my test config, got the same result.
>>
>> Did you upgraded anything after last compile?
>>
>> No. I just took ver 1.6.2.16.1 , compiled with ael support got this error.
>> then decided to check with ver 1.8.2. Error remained the same.
>>
>> Or
>>
>>
>>
>> Try a clean recompile
>>
>>
>>
>> Faisal Hanif
>>
>>
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Borin
>> *Sent:* Friday, February 18, 2011 4:30 PM
>> *To:* asterisk-users@lists.digium.com
>> *Subject:* [asterisk-users] pbx_ael.so: undefined symbol:
>> ast_compile_ael2
>>
>>
>>
>> Hello,
>> trying to load ael module in asterisk ver 1.6.2 got the following:
>>
>> asterisk*CLI> module load pbx_ael.so
>> Unable to load module pbx_ael.so
>> Command 'module load pbx_ael.so' failed.
>> [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error
>> loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined
>> symbol: ast_compile_ael2
>> [Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module
>> 'pbx_ael.so' could not be loaded.
>>
>> I did not find in google what it could be and what should be done to solve
>> this. I also tried the same on ast ver 1.8.2.3, got the same. I am usind
>> debian as OS and install asterisk from sources that I took on digium site.
>> Did anyone have the same issue?
>>
>> Regards, Kate
>>
>>
>> --
>> _
>> -- 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
>>
>
>
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Re: [asterisk-users] Meet me recording

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Friday, February 18, 2011 9:12 AM
To: asterisk-users
Subject: [asterisk-users] Meet me recording

 

Hey Users,

I am using record application to record MeetMe conf. but look like its
creating individual files for every channel. What applucation is best to
record MeetMe conf ?


~ # ls -l /var/spool/asterisk/monitor/
total 489220
-rw-r--r-- 1 asterisk asterisk   44 Feb 16 08:42
8881-conf-20110216-084224.wav
-rw-r--r-- 1 asterisk asterisk  1858284 Feb 16 13:05
8881-conf-20110216-130321.wav
-rw-r--r-- 1 asterisk asterisk  1604204 Feb 16 13:05
8881-conf-20110216-130337.wav
-rw-r--r-- 1 asterisk asterisk   241964 Feb 17 08:20
8881-conf-20110217-081957.wav
-rw-r--r-- 1 asterisk asterisk 78678124 Feb 17 11:12
8881-conf-20110217-095056.wav
-rw-r--r-- 1 asterisk asterisk   612204 Feb 17 09:53
8881-conf-20110217-095310.wav
-rw-r--r-- 1 asterisk asterisk 81183084 Feb 17 11:13
8881-conf-20110217-095414.wav
-rw-r--r-- 1 asterisk asterisk 69488044 Feb 17 11:12
8881-conf-20110217-100012.wav
-rw-r--r-- 1 asterisk asterisk 68917164 Feb 17 11:12
8881-conf-20110217-100052.wav
-rw-r--r-- 1 asterisk asterisk 66971884 Feb 17 11:11
8881-conf-20110217-100117.wav
-rw-r--r-- 1 asterisk asterisk 66648684 Feb 17 11:12
8881-conf-20110217-100327.wav
-rw-r--r-- 1 asterisk asterisk 45056044 Feb 17 11:06
8881-conf-20110217-102007.wav


Thanks,
S

 

>From what I read, mixmonitor.

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Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2

[Borin]

Linux version 2.6.26-2-686 (Debian 2.6.26-24) (da...@debian.org) (gcc
version 4.1.3 20080704 (prerelease) (Debian 4.1.2-25))


On Fri, Feb 18, 2011 at 4:09 PM, Faisal Hanif  wrote:

> Are you on CentOS?
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Borin
> *Sent:* Friday, February 18, 2011 7:46 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] pbx_ael.so: undefined symbol:
> ast_compile_ael2
>
>
>
>
>
> On Fri, Feb 18, 2011 at 3:29 PM, Faisal Hanif  wrote:
>
> Did you checked if you extension.ael doesn’t have syntax error?
>
>
> I think there is no error. I loaded the standard ael first (provided by
> asterisk) then my test config, got the same result.
>
> Did you upgraded anything after last compile?
>
> No. I just took ver 1.6.2.16.1 , compiled with ael support got this error.
> then decided to check with ver 1.8.2. Error remained the same.
>
> Or
>
>
>
> Try a clean recompile
>
>
>
> Faisal Hanif
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Borin
> *Sent:* Friday, February 18, 2011 4:30 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2
>
>
>
> Hello,
> trying to load ael module in asterisk ver 1.6.2 got the following:
>
> asterisk*CLI> module load pbx_ael.so
> Unable to load module pbx_ael.so
> Command 'module load pbx_ael.so' failed.
> [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error
> loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined
> symbol: ast_compile_ael2
> [Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module
> 'pbx_ael.so' could not be loaded.
>
> I did not find in google what it could be and what should be done to solve
> this. I also tried the same on ast ver 1.8.2.3, got the same. I am usind
> debian as OS and install asterisk from sources that I took on digium site.
> Did anyone have the same issue?
>
> Regards, Kate
>
>
> --
> _
> -- 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
>
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[asterisk-users] Meet me recording

[satish patel]

Hey Users,

I am using record application to record MeetMe conf. but look like its creating 
individual files for every channel. What applucation is best to record MeetMe 
conf ?


~ # ls -l /var/spool/asterisk/monitor/
total 489220
-rw-r--r-- 1 asterisk asterisk   44 Feb 16 08:42 
8881-conf-20110216-084224.wav
-rw-r--r-- 1 asterisk asterisk  1858284 Feb 16 13:05 
8881-conf-20110216-130321.wav
-rw-r--r-- 1 asterisk asterisk  1604204 Feb 16 13:05 
8881-conf-20110216-130337.wav
-rw-r--r-- 1 asterisk asterisk   241964 Feb 17 08:20 
8881-conf-20110217-081957.wav
-rw-r--r-- 1 asterisk asterisk 78678124 Feb 17 11:12 
8881-conf-20110217-095056.wav
-rw-r--r-- 1 asterisk asterisk   612204 Feb 17 09:53 
8881-conf-20110217-095310.wav
-rw-r--r-- 1 asterisk asterisk 81183084 Feb 17 11:13 
8881-conf-20110217-095414.wav
-rw-r--r-- 1 asterisk asterisk 69488044 Feb 17 11:12 
8881-conf-20110217-100012.wav
-rw-r--r-- 1 asterisk asterisk 68917164 Feb 17 11:12 
8881-conf-20110217-100052.wav
-rw-r--r-- 1 asterisk asterisk 66971884 Feb 17 11:11 
8881-conf-20110217-100117.wav
-rw-r--r-- 1 asterisk asterisk 66648684 Feb 17 11:12 
8881-conf-20110217-100327.wav
-rw-r--r-- 1 asterisk asterisk 45056044 Feb 17 11:06 
8881-conf-20110217-102007.wav


Thanks,
S
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Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2

[Faisal Hanif]

Are you on CentOS?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Borin
Sent: Friday, February 18, 2011 7:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2

 

 

On Fri, Feb 18, 2011 at 3:29 PM, Faisal Hanif  wrote:

Did you checked if you extension.ael doesn't have syntax error?


I think there is no error. I loaded the standard ael first (provided by
asterisk) then my test config, got the same result. 

Did you upgraded anything after last compile?

No. I just took ver 1.6.2.16.1 , compiled with ael support got this error.
then decided to check with ver 1.8.2. Error remained the same.  

Or

 

Try a clean recompile

 

Faisal Hanif

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Borin
Sent: Friday, February 18, 2011 4:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2

 

Hello, 
trying to load ael module in asterisk ver 1.6.2 got the following:

asterisk*CLI> module load pbx_ael.so
Unable to load module pbx_ael.so
Command 'module load pbx_ael.so' failed.
[Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error
loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined
symbol: ast_compile_ael2
[Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module
'pbx_ael.so' could not be loaded.

I did not find in google what it could be and what should be done to solve
this. I also tried the same on ast ver 1.8.2.3, got the same. I am usind
debian as OS and install asterisk from sources that I took on digium site.
Did anyone have the same issue? 

Regards, Kate


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Re: [asterisk-users] lua -asterisk manual

[Borin]

Pls could you share some lua config which contains mysql quires

On Fri, Feb 18, 2011 at 3:38 PM, Faisal Hanif  wrote:

> The only specific you need to do in extensions.lua is create a table to put
> your extensions in like,
>
>
>
> Extension{
>
>
>
>
>
> }
>
>
>
> Else all will be LUA code and all asterisk applications can be called as
> app.application_name.
>
>
>
> Regards,
>
>
>
> Faisal Hanif
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Borin
> *Sent:* Friday, February 18, 2011 4:33 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] lua -asterisk manual
>
>
>
> Please could someone advise good manual for using lua for asterisk
> dialplan. There is not much docu about it.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2

[Borin]

On Fri, Feb 18, 2011 at 3:29 PM, Faisal Hanif  wrote:

> Did you checked if you extension.ael doesn’t have syntax error?
>

I think there is no error. I loaded the standard ael first (provided by
asterisk) then my test config, got the same result.

> Did you upgraded anything after last compile?
>
No. I just took ver 1.6.2.16.1 , compiled with ael support got this error.
then decided to check with ver 1.8.2. Error remained the same.

> Or
>
>
>
> Try a clean recompile
>
>
>
> Faisal Hanif
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Borin
> *Sent:* Friday, February 18, 2011 4:30 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2
>
>
>
> Hello,
> trying to load ael module in asterisk ver 1.6.2 got the following:
>
> asterisk*CLI> module load pbx_ael.so
> Unable to load module pbx_ael.so
> Command 'module load pbx_ael.so' failed.
> [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error
> loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined
> symbol: ast_compile_ael2
> [Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module
> 'pbx_ael.so' could not be loaded.
>
> I did not find in google what it could be and what should be done to solve
> this. I also tried the same on ast ver 1.8.2.3, got the same. I am usind
> debian as OS and install asterisk from sources that I took on digium site.
> Did anyone have the same issue?
>
> Regards, Kate
>
> --
> _
> -- 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
>
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Re: [asterisk-users] lua -asterisk manual

[Faisal Hanif]

The only specific you need to do in extensions.lua is create a table to put
your extensions in like,

 

Extension{

 

 

}

 

Else all will be LUA code and all asterisk applications can be called as
app.application_name.

 

Regards,

 

Faisal Hanif

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Borin
Sent: Friday, February 18, 2011 4:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] lua -asterisk manual

 

Please could someone advise good manual for using lua for asterisk dialplan.
There is not much docu about it. 

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Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2

[Faisal Hanif]

Did you checked if you extension.ael doesn't have syntax error?

Did you upgraded anything after last compile?

Or

 

Try a clean recompile

 

Faisal Hanif

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Borin
Sent: Friday, February 18, 2011 4:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2

 

Hello, 
trying to load ael module in asterisk ver 1.6.2 got the following:

asterisk*CLI> module load pbx_ael.so
Unable to load module pbx_ael.so
Command 'module load pbx_ael.so' failed.
[Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error
loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined
symbol: ast_compile_ael2
[Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module
'pbx_ael.so' could not be loaded.

I did not find in google what it could be and what should be done to solve
this. I also tried the same on ast ver 1.8.2.3, got the same. I am usind
debian as OS and install asterisk from sources that I took on digium site.
Did anyone have the same issue? 

Regards, Kate

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Re: [asterisk-users] Assigning an extension to a roaming phone

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle
Sent: Friday, February 18, 2011 5:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Assigning an extension to a roaming phone

Hi,
I'm trying to automatically have the dialplan assign an extension to a
roaming phone on my network.
I tried the following without success:

exten => 3001,1(readop),BackGround(beep)
exten => 3001,n,Read(digito,vm-youhave,3)
exten => 3001,n,SayDigits(${digito})
exten => 3001,n,Set(ROAM=${digito})
exten => 3001,n,Set(DB(roam/ext)=${digito})
exten => 3001,n,playback(vm-goodbye)
exten => 3001,n,hangup
exten => _4XXX,1,Set(ROAM=${DB(roam/ext)})
exten => _4XXX,n,dial(SIP/${ROAM})

The idea was that the roaming phone first dials 3001, sets a 3 digits
extension (eg 123) and then I supposed that 4123 would work. But it
does not.
I am unsure about the 2 Set lines.
Can anyone help?
Regards

I'm thinking that _4XXX is an "over-complication".  _4XXX means you could
dial any number from 4000 through 4999 inclusive and get the extension at
SIP/${ROAM}. I'd change the line 2 

- exten => _4XXX,n,Dial(SIP/${ROAM},30,,mKkTt)
Or 
- exten => 4123,1,Dial(SIP/${ROAM},30,,mKkTt)


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Re: [asterisk-users] SIP Hardphone that works well with asterisk

[randulo]

On Fri, Feb 18, 2011 at 2:44 PM, Sherwood McGowan
 wrote:
> I'm VERY partial to Aastra's devices. Seriously, they don't take as long to
> boot as Polycoms, they're relatively inexpensive but are not CHEAP (like a


Great info.

> I do have a complaint about Aastra though...Because of my client's happiness
> with the brand, and because I personally think they're worth suggesting, I
> spoke with Aastra about becoming an authorized reseller...filled out the
> paperwork, scanned it and emailed it to the rep I was working with..and
> never heard another word...For a phone device company to never get back to a

Isn't this irritating? This is the era of recommendations on the net
and sales via email and e-commerce. The traditional phones mfrs are so
1990 - wanting to control everything. I guess they can "afford" to
piss consultants like you off, which is a shame IMO. And if they're
not reading this list, they're even more lame than I thought.
Meanwhile, they're probably spamming the heck out of people who aren't
interested.

/r

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Re: [asterisk-users] SIP Hardphone that works well with asterisk

[Sherwood McGowan]

I'm VERY partial to Aastra's devices. Seriously, they don't take as long to
boot as Polycoms, they're relatively inexpensive but are not CHEAP (like a
certain brand beginning with a "G", in my opinion), they have decent web
interfaces (also unlike the unnamed brand I non-mentioned a moment ago),
solid features, and have good expansion modules as well.

In almost every case where a client ends up choosing Aastra as the brand to
buy after reading my "shortlist" of brands/models I suggest, I've had less
overall issues/complaints about the phones themselves. Even Polycom based
clients/locations have more issues/complaints/annoyances they wish to bring
to my attention than the Aastra based ones.

I do have a complaint about Aastra though...Because of my client's happiness
with the brand, and because I personally think they're worth suggesting, I
spoke with Aastra about becoming an authorized reseller...filled out the
paperwork, scanned it and emailed it to the rep I was working with..and
never heard another word...For a phone device company to never get back to a
telecom consultant who wanted to not only put their brand at the top of the
shortlist, but wanted to push the brand as their preferred brand to use..I
mean come on, seriously, free advertising and purchase referrals? Why
wouldn't you just file the paperwork and reply to the completed application
with a "thank you, you are now an authorized reseller" or even just a
"thanks, but you don't meet the projected sales numbers we wish from our
resellers"?

Anyway, Aastra IS great in my book...

Cheers all!
Sherwood "I'm the Mick and that's my $0.05..keep the change" McGowan
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Re: [asterisk-users] Newbie´s question about Asterisk...

[Gopalakrishnan A.N]

Asterisk is open source and you can install in a normal PC itself and you
can avail all the features that proprietary system has.

If you want to integrate with any VoIP service then a PC with Asterisk is
enough or else if you want to integrate with PSTN lines then you need FXO
card to be installed, its a PCI card. Vendors like Sangoma, Digium (from
Asterisk) were selling these cards. And for internal for your agents if you
need analog hard phones then you need to have FXS card you can avail these
FXO and FXS cards in combination. These cards will fit in your PCI slot of
machine. Configuring these cards are also very easy.

If it is VoIP then you dont need these cards simply install Asterisk in a PC
and you are done.


On Fri, Feb 18, 2011 at 5:38 PM, Terry Brummell  wrote:

> Yes, I use Elastix myself too.  Funny that I didn’t mention that one!
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA
> *Sent:* Friday, February 18, 2011 6:11 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...
>
>
>
> i prefer to go with Elastix very easy to setup and maintain and reach UI
> rather than freePBX
>
> cheers
> Dhaval
>
> On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell 
> wrote:
>
> Dean’s link has references to Trixbox.  TB has a bad, bad, very bad
> reputation for being very insecure.  Alternatives to TB are FreePBX & PBX in
> a Flash.  All are Asterisk based and very easy to set up.
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dean Collins
> *Sent:* Thursday, February 17, 2011 7:29 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...
>
>
>
> If you already have experience with linux asterisk will be easy for you.
>
>
>
> Other people will reply with official links but here is how I use Asterisk
> in my small home office www.cognation.net/asterisk
>
>
>
>
>
> Cheers,
>
> Dean
>
>
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Francisco Javier
> Cintrón Olguín
> *Sent:* Thursday, February 17, 2011 7:26 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Newbie´s question about Asterisk...
>
>
>
> Hi, My name is Francisco from México.
>
> Here, in my work we have a very very old panasonic PBX(12 years old). We
> are growing and we need to increase our external lines(from 3 to 4) and our
> internal lines(from 6 to 10). Besides we need voice mail and voice menu too.
>
>
> We asked for a quote to our panasonic dealer. The whole thing cost about
> 4,500 dollars.
>
> My boss just saw a thing called Asterisk this morning looking for options
> in Google. He asked my to investigate what this thing called Asterisk is and
> if we could save some money using it instead of the panasonic solution. So,
> here I am.
>
> I have some experience as linux sysadmin(we have 1 oracle linux server and
> 1 linux print server) nevertheless I don´t have any idea where and how to
> start this evaluation?
>
>
> Please
> Would you give us a clue where to see If Asterisk could work for us?
>
> Thanks for your kind help.
>
>
> --
> _
> -- 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
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> _
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>   http://www.asterisk.org/hello
>
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>



-- 
Thank you  with regards,
Gopalakrishnan A.N.
VoIP call - sip:sai...@gtalk2voip.com
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Re: [asterisk-users] Newbie´s question about Asterisk...

[Terry Brummell]

Yes, I use Elastix myself too.  Funny that I didn't mention that one! 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA
Sent: Friday, February 18, 2011 6:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]Newbie´s question about Asterisk...

 

i prefer to go with Elastix very easy to setup and maintain and reach UI rather 
than freePBX 

cheers
Dhaval

On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell  wrote:

Dean's link has references to Trixbox.  TB has a bad, bad, very bad reputation 
for being very insecure.  Alternatives to TB are FreePBX & PBX in a Flash.  All 
are Asterisk based and very easy to set up.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Collins
Sent: Thursday, February 17, 2011 7:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]Newbie´s question about Asterisk...

 

If you already have experience with linux asterisk will be easy for you.

 

Other people will reply with official links but here is how I use Asterisk in 
my small home office www.cognation.net/asterisk 

 

 

Cheers,

Dean

 

 



From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Francisco Javier 
Cintrón Olguín
Sent: Thursday, February 17, 2011 7:26 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Newbie´s question about Asterisk...

 

Hi, My name is Francisco from México. 

Here, in my work we have a very very old panasonic PBX(12 years old). We are 
growing and we need to increase our external lines(from 3 to 4) and our 
internal lines(from 6 to 10). Besides we need voice mail and voice menu too. 

We asked for a quote to our panasonic dealer. The whole thing cost about 4,500 
dollars. 

My boss just saw a thing called Asterisk this morning looking for options in 
Google. He asked my to investigate what this thing called Asterisk is and if we 
could save some money using it instead of the panasonic solution. So, here I 
am. 

I have some experience as linux sysadmin(we have 1 oracle linux server and 1 
linux print server) nevertheless I don´t have any idea where and how to start 
this evaluation?


Please
Would you give us a clue where to see If Asterisk could work for us?

Thanks for your kind help. 


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[asterisk-users] Assigning an extension to a roaming phone

[Axelle]

Hi,
I'm trying to automatically have the dialplan assign an extension to a
roaming phone on my network.
I tried the following without success:

exten => 3001,1(readop),BackGround(beep)
exten => 3001,n,Read(digito,vm-youhave,3)
exten => 3001,n,SayDigits(${digito})
exten => 3001,n,Set(ROAM=${digito})
exten => 3001,n,Set(DB(roam/ext)=${digito})
exten => 3001,n,playback(vm-goodbye)
exten => 3001,n,hangup
exten => _4XXX,1,Set(ROAM=${DB(roam/ext)})
exten => _4XXX,n,dial(SIP/${ROAM})

The idea was that the roaming phone first dials 3001, sets a 3 digits
extension (eg 123) and then I supposed that 4123 would work. But it
does not.
I am unsure about the 2 Set lines.
Can anyone help?
Regards

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[asterisk-users] lua -asterisk manual

[Borin]

Please could someone advise good manual for using lua for asterisk dialplan.
There is not much docu about it.
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[asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2

[Borin]

Hello,
trying to load ael module in asterisk ver 1.6.2 got the following:

asterisk*CLI> module load pbx_ael.so
Unable to load module pbx_ael.so
Command 'module load pbx_ael.so' failed.
[Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error
loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined
symbol: ast_compile_ael2
[Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module
'pbx_ael.so' could not be loaded.

I did not find in google what it could be and what should be done to solve
this. I also tried the same on ast ver 1.8.2.3, got the same. I am usind
debian as OS and install asterisk from sources that I took on digium site.
Did anyone have the same issue?

Regards, Kate
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Re: [asterisk-users] Newbie´s question about Asterisk...

[DHAVAL INDRODIYA]

i prefer to go with Elastix very easy to setup and maintain and reach UI
rather than freePBX

cheers
Dhaval

On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell  wrote:

> Dean’s link has references to Trixbox.  TB has a bad, bad, very bad
> reputation for being very insecure.  Alternatives to TB are FreePBX & PBX in
> a Flash.  All are Asterisk based and very easy to set up.
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dean Collins
> *Sent:* Thursday, February 17, 2011 7:29 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...
>
>
>
> If you already have experience with linux asterisk will be easy for you.
>
>
>
> Other people will reply with official links but here is how I use Asterisk
> in my small home office www.cognation.net/asterisk
>
>
>
>
>
> Cheers,
>
> Dean
>
>
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Francisco Javier
> Cintrón Olguín
> *Sent:* Thursday, February 17, 2011 7:26 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Newbie´s question about Asterisk...
>
>
>
> Hi, My name is Francisco from México.
>
> Here, in my work we have a very very old panasonic PBX(12 years old). We
> are growing and we need to increase our external lines(from 3 to 4) and our
> internal lines(from 6 to 10). Besides we need voice mail and voice menu too.
>
>
> We asked for a quote to our panasonic dealer. The whole thing cost about
> 4,500 dollars.
>
> My boss just saw a thing called Asterisk this morning looking for options
> in Google. He asked my to investigate what this thing called Asterisk is and
> if we could save some money using it instead of the panasonic solution. So,
> here I am.
>
> I have some experience as linux sysadmin(we have 1 oracle linux server and
> 1 linux print server) nevertheless I don´t have any idea where and how to
> start this evaluation?
>
>
> Please
> Would you give us a clue where to see If Asterisk could work for us?
>
> Thanks for your kind help.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Newbie´s question about Asterisk...

[Terry Brummell]

Dean's link has references to Trixbox.  TB has a bad, bad, very bad reputation 
for being very insecure.  Alternatives to TB are FreePBX & PBX in a Flash.  All 
are Asterisk based and very easy to set up.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Collins
Sent: Thursday, February 17, 2011 7:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]Newbie´s question about Asterisk...

 

If you already have experience with linux asterisk will be easy for you.

 

Other people will reply with official links but here is how I use Asterisk in 
my small home office www.cognation.net/asterisk 

 

 

Cheers,

Dean

 

 



From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Francisco Javier 
Cintrón Olguín
Sent: Thursday, February 17, 2011 7:26 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Newbie´s question about Asterisk...

 

Hi, My name is Francisco from México. 

Here, in my work we have a very very old panasonic PBX(12 years old). We are 
growing and we need to increase our external lines(from 3 to 4) and our 
internal lines(from 6 to 10). Besides we need voice mail and voice menu too. 

We asked for a quote to our panasonic dealer. The whole thing cost about 4,500 
dollars. 

My boss just saw a thing called Asterisk this morning looking for options in 
Google. He asked my to investigate what this thing called Asterisk is and if we 
could save some money using it instead of the panasonic solution. So, here I 
am. 

I have some experience as linux sysadmin(we have 1 oracle linux server and 1 
linux print server) nevertheless I don´t have any idea where and how to start 
this evaluation?


Please
Would you give us a clue where to see If Asterisk could work for us?

Thanks for your kind help. 

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Re: [asterisk-users] Trunk grouping

[Gopalakrishnan A.N]

If it is a E1/T1 trunk then all the channels are grouped in one group, once
done place your Asterisk server inside your squid network. I hope this
helps!

On Fri, Feb 18, 2011 at 12:46 PM, Malvin Rito  wrote:

>  Hi List,
>
>
>
> Were upgrading our network switches and need to create multiple VLAN
> groups, but since our Squid Proxy (Transparent Proxy) Server should be
> accessible to all VLAN groups we need to setup a trunk grouping inside our
> Squid Proxy Box. Is anyone has a documentation or code on how to implement
> trunk grouping?
>
>
>
> Your thoughts will be highly appreciated.
>
>
>
> Regards,
>
> Malvin
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Thank you  with regards,
Gopalakrishnan A.N.
VoIP call - sip:sai...@gtalk2voip.com
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Re: [asterisk-users] Trunk grouping

[Damian Ryszka]

Dnia Fri, 18 Feb 2011 15:16:40 +0800
"Malvin Rito"  napisał(a):

> Were upgrading our network switches and need to create multiple
> VLAN groups, but since our Squid Proxy (Transparent Proxy) Server
> should be accessible to all VLAN groups we need to setup a trunk
> grouping inside our Squid Proxy Box. Is anyone has a documentation
> or code on how to implement trunk grouping?

If I correctly understand your problem, you should configure VLANs at
your linux box. You'll have to add a IP address to all configured
VLAN interfaces, then bind your squid proxy to those addresses.
Google for 8021q module and vconfig utility.

Regards,
-- 
Damian Ryszka aka Rychu
rychu(at)sileman.net.pl

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Re: [asterisk-users] Asterisk with TE 121 DADHI incoming calls fail

[Faisal Hanif]

This is not Digium's customer support address but free public emailing list
for asterisk user's contributed by community volunteers.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jan Zieher
Sent: Friday, February 18, 2011 2:19 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk with TE 121 DADHI incoming calls fail

 

Dear Customer Support,
i connected the asterisk to a e1 interface of our hipath4000. outgoing calls
from a sip peer of my asterisk to an up0 telephone which iss connected to
the hipath4000 are working. If you want to dial from an up0 device to the e1
interface where asterisk is connected to, you have to use the prefix 83. But
when you enter the 3rd cipher this error appears at the cli

CODE:
 SELECT ALL

[Feb 16 11:29:07] ERROR[2555]: chan_dahdi.c:13691 dahdi_pri_error: 1 !! Not
yet handling pre-handle message type SEGMENT (0x60)
[Feb 16 11:29:07] ERROR[2555]: chan_dahdi.c:13691 dahdi_pri_error: 1 !!
Don't know how to pre-handle message type SEGMENT (0x60)
[Feb 16 11:29:07] ERROR[2555]: chan_dahdi.c:13691 dahdi_pri_error: 1 !! Not
yet handling pre-handle message type SEGMENT (0x60)
[Feb 16 11:29:07] ERROR[2555]: chan_dahdi.c:13691 dahdi_pri_error: 1 !!
Don't know how to pre-handle message type SEGMENT (0x60)



My Config files:
extensions.conf

CODE:
 SELECT ALL

[general]
static=yes
writeprotect=no

[isdn]

; Ankommende anrufe
exten => 833762,1,Dial(SIP/3762,45,r)

; Rausgehende Anrufe
exten => _0[1-9].,1,Dial(DAHDI/g1/${EXTEN:1})

[default]
include => isdn



/etc/dahdi/system.conf

CODE:
 SELECT ALL

span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31

# Global data

loadzone= de
defaultzone = de



chan_dahdi.conf

CODE:
 SELECT ALL

[trunkgroups]

[channels]
language=de
switchtype=euroisdn

pridialplan=unknown
prilocaldialplan=unknown
internationalprefix = 00
nationalprefix = 0
;localprefix = VORWAHL
;privateprefix = VORWAHL+MSN
;unknownprefix =
priindication = outofband

facilityenable = yes
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
immediate=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
callgroup=1
pickupgroup=1
mohinterpret=default
mohsuggest=default
overlapdial=yes

group=1
signalling = pri_cpe
channel => 1-15,17-31
context = default



I would be gratefully, if you have an idea or some advices to me.
Thanks ! 

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Re: [asterisk-users] Dial(Local/...) vs. Goto()?

[Gilles]

On Fri, 18 Feb 2011 14:10:54 +0500, "Faisal Hanif" 
wrote:
>The difference you will feel when using callback files or AMI.

Thanks Faisal.

"Two of the most common areas where Local channels are used include
members configured for queues, and in use with callfiles. There are
also other uses where you want to ring two destinations, but with
different information, such as different callerID for each outgoing
request."
https://wiki.asterisk.org/wiki/display/AST/Introduction+to+Local+Channels


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[asterisk-users] Asterisk with TE 121 DADHI incoming calls fail

[Jan Zieher]

Dear Customer Support,
i connected the asterisk to a e1 interface of our hipath4000. outgoing 
calls from a sip peer of my asterisk to an up0 telephone which iss 
connected to the hipath4000 are working. If you want to dial from an up0 
device to the e1 interface where asterisk is connected to, you have to 
use the prefix 83. But when you enter the 3rd cipher this error appears 
at the cli


CODE:SELECT ALL 


   |[Feb 16 11:29:07] ERROR[2555]: chan_dahdi.c:13691 dahdi_pri_error:
   1 !! Not yet handling pre-handle message type SEGMENT (0x60)
   [Feb 16 11:29:07] ERROR[2555]: chan_dahdi.c:13691 dahdi_pri_error: 1
   !! Don't know how to pre-handle message type SEGMENT (0x60)
   [Feb 16 11:29:07] ERROR[2555]: chan_dahdi.c:13691 dahdi_pri_error: 1
   !! Not yet handling pre-handle message type SEGMENT (0x60)
   [Feb 16 11:29:07] ERROR[2555]: chan_dahdi.c:13691 dahdi_pri_error: 1
   !! Don't know how to pre-handle message type SEGMENT (0x60)
   |



My Config files:
extensions.conf

CODE:SELECT ALL 


   |[general]
   static=yes
   writeprotect=no

   [isdn]

   ; Ankommende anrufe
   exten => 833762,1,Dial(SIP/3762,45,r)

   ; Rausgehende Anrufe
   exten => _0[1-9].,1,Dial(DAHDI/g1/${EXTEN:1})

   [default]
   include => isdn
   |



/etc/dahdi/system.conf

CODE:SELECT ALL 


   |span=1,1,0,ccs,hdb3,crc4
   # termtype: te
   bchan=1-15,17-31
   dchan=16
   echocanceller=mg2,1-15,17-31

   # Global data

   loadzone= de
   defaultzone = de
   |



chan_dahdi.conf

CODE:SELECT ALL 


   |[trunkgroups]

   [channels]
   language=de
   switchtype=euroisdn

   pridialplan=unknown
   prilocaldialplan=unknown
   internationalprefix = 00
   nationalprefix = 0
   ;localprefix = VORWAHL
   ;privateprefix = VORWAHL+MSN
   ;unknownprefix =
   priindication = outofband

   facilityenable = yes
   usecallerid=yes
   callwaiting=yes
   usecallingpres=yes
   callwaitingcallerid=yes
   threewaycalling=yes
   transfer=yes
   canpark=yes
   cancallforward=yes
   callreturn=yes
   immediate=no
   echocancel=yes
   echocancelwhenbridged=yes
   echotraining=yes
   callgroup=1
   pickupgroup=1
   mohinterpret=default
   mohsuggest=default
   overlapdial=yes

   group=1
   signalling = pri_cpe
   channel => 1-15,17-31
   context = default
   |



I would be gratefully, if you have an idea or some advices to me.
Thanks !
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Re: [asterisk-users] Dial(Local/...) vs. Goto()?

[Faisal Hanif]

The difference you will feel when using callback files or AMI.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Friday, February 18, 2011 1:31 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dial(Local/...) vs. Goto()?

Hello,

I was wondering: What does Dial(Local/...) offer that a Goto() doesn't?

For instance:

;exten => h,n,Goto(callback,start)
exten => h,n,Dial(Local/start@callback)

[callback]
exten => start,1,Verbose(In callback)


Thank you.


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Re: [asterisk-users] DTMF and Snom

[Faisal Hanif]

Well you simple use dtmfmode=info in peer configuration of Snome phone.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, February 18, 2011 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] DTMF and Snom

 

Hello list,

I'm having some troubles with DTMF tones. When pressing numbers on a Snom
phone, the DTMF-signal takes too long.

I have the following in sip.conf :

dtmfmode = rfc2833


which works well for Grandstream, Yealink and Cisco phones. But not for
Snom.

Snom support tells me I should use SIP info.

Is it possible to have something like this :

dtmfmode = rfc2833, info

??

Because all the other phones types are set to rfc2833, I cannot change to
just dtmfmode = info

What is a proper solution in my case here ?!


Thank you !

Jonas.

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Re: [asterisk-users] SIP Hardphone that works well with asterisk

[Ishfaq Malik]

On Fri, 2011-02-18 at 08:10 +0100, Hans Witvliet wrote:
> On Fri, 2011-02-18 at 00:51 +0100, Albert wrote:
> > On 18.02.2011 00:30, Andrew Joakimsen wrote: 
> > > On Sat, Feb 12, 2011 at 07:31, ast guy  wrote:
> > > > Hi,
> > > >  I have been out of touch with asterisk for quit some time and needed 
> > > > some
> > > > recommendations. I am looking for SIP hardphone that works well with
> > > > asterisk server.
> > > > 
> > > Polycom phones are still working well and durable as a brick.
> > > 
> > > Gransdstream phones still feel cheap.
> > > 
> > > Cisco still have the same shenanigans going on with their firmware 
> > > downloads.
> > > 
> > > 
> > Linksys SPA921, SPA922, SPA941, SPA942 are also working pretty well.
> > 
> same for snom, we have 320, 360, but other probably as well.
> hw
> 
We use Snoms quite extensively and generally don't have too many
problems with them. They also have a very good method for remote
configuration.

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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[asterisk-users] Dial(Local/...) vs. Goto()?

[Gilles]

Hello,

I was wondering: What does Dial(Local/...) offer that a Goto()
doesn't?

For instance:

;exten => h,n,Goto(callback,start)
exten => h,n,Dial(Local/start@callback)

[callback]
exten => start,1,Verbose(In callback)


Thank you.


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[asterisk-users] FAX on PRI to MFCR2

[leonimar cape]

Hi,

I am having issues sending and receiving fax on my asterisk setup.

Currently I have a server that has 2 x E1 TDM cards one is sangoma and the 
other 

one is openvox. Both support echo cancellation.

One of the e1 is connected to our telco provider via mfcr2 where all our 
incoming calls originate. On the other end is a pri connection going to HICOM 
PABX where the local attached to a fax is connected. Fax passing thru this 
connection are not getting thru and always getting drop.

FYI: all of the 8xE1s are currently up with two using mfcr2 and the rest is 
ISDN 

pri. 


Server is spec is 
Dual Xeon 3.00GHz
2 GIG RAM

Load Average: less than 1


Dahdi configuration:


context=from-pstn
signalling=fxs_ks
rxwink=300  ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
faxdetect=both
faxbuffers=>6,full
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1

;Uncomment these lines if you have problems with the disconection of your 
analog 
lines
;busydetect=yes
;busycount=3


immediate=no




### PRI 
group=0,11
context=a2billing
switchtype=euroisdn
priindication=inband
overlapdial=yes
nsf=none
signalling=pri_net
channel => 1-15,17-31
context = default
group = 63

### MFCR2
group=2,12
context=outrt-005-IN_E1P2_PBX_JLB
signalling = mfcr2
mfcr2_variant=ph
mfcr2_max_ani=10
mfcr2_max_dnis=4
mfcr2_get_ani_first=yes
mfcr2_category=national_subscriber
mfcr2_logdir=span2
;mfcr2_call_files=yes
channel => 32-46
channel => 48-62
context = default

I am not sure if I miss anything in the configuration or there something I need 
to enable or include in the extensions.conf

Any information is much appreciated.

Regards,

Mac



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[asterisk-users] DTMF and Snom

[Jonas Kellens]

Hello list,

I'm having some troubles with DTMF tones. When pressing numbers on a 
Snom phone, the DTMF-signal takes too long.


I have the following in sip.conf :

dtmfmode = rfc2833


which works well for Grandstream, Yealink and Cisco phones. But not for 
Snom.


Snom support tells me I should use SIP info.

Is it possible to have something like this :

dtmfmode = rfc2833, info

??

Because all the other phones types are set to rfc2833, I cannot change 
to just dtmfmode = info


What is a proper solution in my case here ?!


Thank you !

Jonas.
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