date:20090819

Julian Lyndon-Smith wrote:
 Nope - but you are also running on an unsupported version of asterisk,
 so I am not surprised. From the readme:
 
 ===[ Installation Overview 
 ]===
 
 It is required that the proper version of Asterisk is installed prior to
 installing Skype For Asterisk. Skype For Asterisk is currently supported on:
 
Asterisk 1.4 versions = 1.4.25
Asterisk 1.6.0 versions = 1.6.0.6
Asterisk 1.6.1 versions = 1.6.1.5
 
Ah didn't spot that, if you are running 1.6.1, you need a version that 
isn't available yet.

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Re: [asterisk-users] [asterisk-announce] Asterisk project changes Music-On-Hold provider

Asterisk Development Team wrote:
 As posted on blogs.digium.com today:
 
 http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
 
 the Asterisk project has changed providers for Music-On-Hold (MOH)
 content distributed with/for Asterisk. In addition to the change for
 future Asterisk releases, we have also opted to rebuild historical
 releases with the new MOH content, in an effort to eliminate unnecessary
 distribution of the old MOH content.
 
Great to hear, although I am a bit suspicious, the asterisk-sounds 
package in 
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ still has 
a Mar 06 timestamp.

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Re: [asterisk-users] [asterisk-announce] Asterisk project changes Music-On-Hold provider

Please ignore my stupid reply to this, I was having issues with weasles 
at the time.

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Re: [asterisk-users] [asterisk-announce] Asterisk project changes Music-On-Hold provider

2009-08-19 Thread Tzafrir Cohen

On Wed, Aug 19, 2009 at 09:28:24AM +0100, Thomas Kenyon wrote:

 Great to hear, although I am a bit suspicious, the asterisk-sounds 
 package in 
 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ still has 
 a Mar 06 timestamp.

http://downloads.asterisk.org/pub/telephony/sounds/
has some newer files.

-- 
   Tzafrir Cohen
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+972-50-7952406   mailto:tzafrir.co...@xorcom.com
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[asterisk-users] CDR record for call originated from manager

2009-08-19 Thread DHAVAL INDRODIYA

hi,

i want CDR entry in database for a call which originated from manager via

action: originate

currently i didnt get this entry into my DB

any one have idea regarding this for getting this on DB

i enabled cdr_manager.conf entry to 'yes'

thanks in advance

regards
Dhaval
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[asterisk-users] ISDN Calling Sub Address and Called Sub Address for the branches

2009-08-19 Thread Alec Davis

Since 2004 asterisk/libpri have been able to receive the Calling Sub Address
in the ISDN setup message, and the dialplan was able to use it if required.
It's support is limited to only NSAP, not BCD or user formatted.
 
At the time 25/06/04 the questioned was asked, wouldn't it be a good idea to
be able to transmit it as well, but that never got implemented, as it wasn't
required at the time.
 
Further to this, there is also Called Sub Address that allows you to dial a
particular terminal device at an ISDN number, these days isn't a terminal
device a users extension.
 
Finishing off the limited support for SUBADDR, if your keen is at
https://issues.asterisk.org/view.php?id=15604
https://issues.asterisk.org/view.php?id=15604, this adds CallingSubAaddress
(Transmit) and CalledSubAddress (Transmit and Receive), still only in NSAP,
not User formatted.
 
This code may not ever make it into trunk, but if you find this code useful
please leave a comment on the mantis bug. This has been tested with the
exisiting 1.4 -1.6.2 branches, and is in use at 3 PRI/BRI sites with
asterisk 1.6.1.
 
The Digium team have other good ideas for 1.6.3 which will as I understand
it support SUBADDR over any transport, but it will be a while before most of
us are happy using the latest offering in production.
 
Alec Davis
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Re: [asterisk-users] MEETME how to lock the conference if no admin are connected

On Wed, 2009-08-19 at 09:16 +0200, BERGANZ François wrote:
 hello 
 
 
 is it possible to lock a conference IF no admin are connected ? 
 
 or how to do to have a conference offline?
snip
If I understand you correctly, we are doing something similar.  When
users call into a conference, they hear music on hold and cannot speak
to each other until the moderator joins the conference.  Our calls to
meetme are via macros but they should give you the idea:

[macro-confmod] ;conference moderator
exten = s,1,Macro(conference,${MACRO_EXTEN})
exten = s,n,MeetMe(${ARG1},cMaAsx)

[macro-confpart] ;conference participant
exten = s,1,Macro(conference,${MACRO_EXTEN})
exten = s,n,MeetMe(${ARG1},cIMswx,${ARG2})

I believe the critical options are the w for the participant (regular
users) which says wait until a marked user joins the conference and
the A for the moderator which designates the moderator as such a
marked user.

I don't understand what you mean by an offline conference.  Hope this
helps - John
 
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

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Making Christianity intelligible to secular society


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[asterisk-users] CAP_FOWNER=ep for asterisk

2009-08-19 Thread Raimund Sacherer


Hello,

I need CAP_FOWNER=ep for the asterisk process, i set it with setcap on  
the file /usr/sbin/asterisk, it's there when i look on it with getcap,  
but after starting and loocking with getpcaps there's only  
cap_net_admin+ep set.


So how exactly do I set CAP_FOWNER? Do I have to patch and recompile  
or is there another solution I did not see yet?


thanks,
best

--
Raimund Sacherer
-
RunSolutions
Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

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Re: [asterisk-users] Accessing to ekiga.net through Asterisk

2009-08-19 Thread SIP

Daniel,

I'm a little confused as to what I'm seeing here. You're bounding 
through two RFC1918 address networks -- 10.1.0.X and 192.168.2.X.   Is 
this some sort of dual NAT scenario?

Perhaps if you can explain a little more about your network setup.

N.



Daniel Bareiro wrote:
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 SIP wrote:

   
 Daniel,
 

 Hi SIP.

   
 Check your stunaddr setting. Is it misspelled, or do they really use
 stun.exiga.net instead of stun.ekiga.net ?
 

 Thanks to indicate that error to me. I doing the test again. I don't
 believe that this solves what I commented before about 192.168.1.2
 direction, but, just in case, I copy the output of debugging when trying
 to communicate to ekiga.net. The problem continues persisting after the
 correction.

 alderamin*CLI
 --- SIP read from 10.1.0.65:5060 ---
 INVITE sip:8...@10.1.0.10 SIP/2.0
 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke
 Max-Forwards: 70
 To: sip:8...@10.1.0.10
 From: Hector sip:2...@10.1.0.10;tag=typwm
 Call-ID: kafgeaflkmsd...@defiant.freesoftware.org
 CSeq: 709 INVITE
 Contact: sip:2...@10.1.0.65
 Content-Type: application/sdp
 Allow:
 INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
 Supported: replaces,norefersub,100rel
 User-Agent: Twinkle/1.2
 Content-Length: 247

 v=0
 o=twinkle 933572867 1938524932 IN IP4 10.1.0.65
 s=-
 c=IN IP4 10.1.0.65
 t=0 0
 m=audio 8000 RTP/AVP 8 0 3 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:3 GSM/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-15
 a=ptime:20

 -
 - --- (13 headers 12 lines) ---
 Sending to 10.1.0.65 : 5060 (NAT)
 Using INVITE request as basis request -
 kafgeaflkmsd...@defiant.freesoftware.org

 --- Reliably Transmitting (no NAT) to 10.1.0.65:5060 ---
 SIP/2.0 407 Proxy Authentication Required
 Via: SIP/2.0/UDP
 10.1.0.65;branch=z9hG4bKrqslryke;received=10.1.0.65;rport=5060
 From: Hector sip:2...@10.1.0.10;tag=typwm
 To: sip:8...@10.1.0.10;tag=as0a3a462b
 Call-ID: kafgeaflkmsd...@defiant.freesoftware.org
 CSeq: 709 INVITE
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
 Supported: replaces
 Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk,
 nonce=497d879d
 Content-Length: 0


 
 Scheduling destruction of SIP dialog
 'kafgeaflkmsd...@defiant.freesoftware.org' in 32000 ms (Method: INVITE)
 Found user '201'
 alderamin*CLI
 --- SIP read from 10.1.0.65:5060 ---
 ACK sip:8...@10.1.0.10 SIP/2.0
 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke
 Max-Forwards: 70
 To: sip:8...@10.1.0.10;tag=as0a3a462b
 From: Hector sip:2...@10.1.0.10;tag=typwm
 Call-ID: kafgeaflkmsd...@defiant.freesoftware.org
 CSeq: 709 ACK
 User-Agent: Twinkle/1.2
 Content-Length: 0


 -
 - --- (9 headers 0 lines) ---
 alderamin*CLI
 --- SIP read from 10.1.0.65:5060 ---
 INVITE sip:8...@10.1.0.10 SIP/2.0
 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKxpraybjr
 Max-Forwards: 70
 Proxy-Authorization: Digest
 username=201,realm=asterisk,nonce=497d879d,uri=sip:8...@10.1.0.10,response=9cb53107d4d15b7a2e7df8599e851b80,algorithm=MD5
 To: sip:8...@10.1.0.10
 From: Hector sip:2...@10.1.0.10;tag=typwm
 Call-ID: kafgeaflkmsd...@defiant.freesoftware.org
 CSeq: 710 INVITE
 Contact: sip:2...@10.1.0.65
 Content-Type: application/sdp
 Allow:
 INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
 Supported: replaces,norefersub,100rel
 User-Agent: Twinkle/1.2
 Content-Length: 247

 v=0
 o=twinkle 933572867 1938524932 IN IP4 10.1.0.65
 s=-
 c=IN IP4 10.1.0.65
 t=0 0
 m=audio 8000 RTP/AVP 8 0 3 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:3 GSM/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-15
 a=ptime:20

 -
 - --- (14 headers 12 lines) ---
 Sending to 10.1.0.65 : 5060 (NAT)
 Using INVITE request as basis request -
 kafgeaflkmsd...@defiant.freesoftware.org
 Found user '201'
 Found RTP audio format 8
 Found RTP audio format 0
 Found RTP audio format 3
 Found RTP audio format 101
 Peer audio RTP is at port 10.1.0.65:8000
 Found audio description format PCMA for ID 8
 Found audio description format PCMU for ID 0
 Found audio description format GSM for ID 3
 Found audio description format telephone-event for ID 101
 Capabilities: us - 0x4040a (gsm|alaw|ilbc|h261), peer - audio=0xe
 (gsm|ulaw|
 alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw)
 Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
 (telephone-event), combined - 0x1 (telephone-event)
 Peer audio RTP is at port 10.1.0.65:8000
 Looking for 8500 in from-internal (domain 10.1.0.10)
 list_route: hop: sip:2...@10.1.0.65

 --- Transmitting (no NAT) to 10.1.0.65:5060 ---
 SIP/2.0 100 Trying
 Via: SIP/2.0/UDP
 10.1.0.65;branch=z9hG4bKxpraybjr;received=10.1.0.65;rport=5060
 From: Hector sip:2...@10.1.0.10;tag=typwm
 To: sip:8...@10.1.0.10
 Call-ID: kafgeaflkmsd...@defiant.freesoftware.org
 CSeq: 710 INVITE
 User-Agent: Asterisk PBX
 Allow:

[asterisk-users] Individual PIN Code per Extension

2009-08-19 Thread James Mutuku

Hellos,

I have astersist 1.2 working with freepbx. I want to tie pin codes to
extensions(users). How do I do this?

-- 
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com

Has your organization implemented a customer relationship management
(CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM
can help you achieve better customer satisfaction and sales
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Re: [asterisk-users] Skype for Asterisk???

2009-08-19 Thread David fire

a few days ago slashdot (sorry i havent the link now) wrote about skype has
a very huge problem whit a licence in a core codec, and if they dont get an
aregment whit the codec owner they will close the doors...
David

2009/8/19 Thomas Kenyon dig...@sanguinarius.co.uk

 Julian Lyndon-Smith wrote:
  Nope - but you are also running on an unsupported version of asterisk,
  so I am not surprised. From the readme:
 
  ===[ Installation Overview
 ]===
 
  It is required that the proper version of Asterisk is installed prior to
  installing Skype For Asterisk. Skype For Asterisk is currently supported
 on:
 
 Asterisk 1.4 versions = 1.4.25
 Asterisk 1.6.0 versions = 1.6.0.6
 Asterisk 1.6.1 versions = 1.6.1.5
 
 Ah didn't spot that, if you are running 1.6.1, you need a version that
 isn't available yet.

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(\__/)
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Re: [asterisk-users] Skype for Asterisk???

2009-08-19 Thread Michael Graves

I wonder if that was not a codec specific issue, but rather the matter
of their license to the p2p technology provided by JoltID? Since Skype
has recently dveloped their own codec (SILK) they could easily drop
support for any codec that they previously licensed from outside. I
think that the failure to ge a new license on a codec would not be a
major issue for them.

Failure to renew the license on the p2p transport technology is a much
more significant problem.

Michael

--Original Message Text---
From: David fire
Date: Wed, 19 Aug 2009 09:04:59 -0300

a few days ago slashdot (sorry i havent the link now) wrote about skype
has a very huge problem whit a licence in a core codec, and if they
dont get an aregment whit the codec owner they will close the doors...
David


2009/8/19 Thomas Kenyon dig...@sanguinarius.co.uk
Julian Lyndon-Smith wrote:
 Nope - but you are also running on an unsupported version of asterisk,
 so I am not surprised. From the readme:

 ===[ Installation Overview 
 ]===

 It is required that the proper version of Asterisk is installed prior to
 installing Skype For Asterisk. Skype For Asterisk is currently supported on:

Asterisk 1.4 versions = 1.4.25
Asterisk 1.6.0 versions = 1.6.0.6
Asterisk 1.6.1 versions = 1.6.1.5


Ah didn't spot that, if you are running 1.6.1, you need a version that
isn't available yet.


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sip:mgra...@mstvp.onsip.com
skype mjgraves
Twitter mjgraves


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Re: [asterisk-users] Skype for Asterisk???

Michael Graves wrote:
 I wonder if that was not a codec specific issue, but rather the matter 
 of their license to the p2p technology provided by JoltID? Since Skype 
 has recently dveloped their own codec (SILK) they could easily drop 
 support for any codec that they previously licensed from outside. I 
 think that the failure to ge a new license on a codec would not be a 
 major issue for them.
 
 Failure to renew the license on the p2p transport technology is a much 
 more significant problem.
 
 Michael
 
That's probably what it was, It does appear to be trying to remove Jolt 
support.

http://www.theregister.co.uk/2009/07/31/skype_joltid/


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[asterisk-users] Newbie: Mac OS X - Asterisk Gui 2.0 (svn) loops at Verifying Dialplan Contexts needed for GUI

2009-08-19 Thread Administrator

Hi All,

This is my first post. I searched the archives and found something similar
and I tried some of those suggestions. I changed the file permissions on the
scripts directory to 777 (which doesn't seem secure), I also manually ran
the detectdahdi.sh script. The response is None.

I am running Mac OS X 10.5.7 with Asterisk 1.4.26.1 which I compiled from
source. The Asterisk Gui (2.0) was built from the 8/16/09 source.

The console spits out a long list of events but I have snipped what I think
is a relevant portion. After this there seems to be a lot of repeating.

Here it is:

Parsing '/usr/local/etc/asterisk/manager.conf': Found
  == HTTP Manager 'manager' logged on from 192.168.1.21
  == Parsing '/usr/local/etc/asterisk/http.conf': Found
  == Saving '/usr/local/etc/asterisk/http.conf': Saved
  == Parsing '/usr/local/etc/asterisk/http.conf': Found
  == Parsing '/usr/local/etc/asterisk/http.conf': Found
  == Saving '/usr/local/etc/asterisk/http.conf': Saved
  == Parsing '/usr/local/etc/asterisk/extensions.conf': Found
  == Parsing '/usr/local/etc/asterisk/guipreferences.conf': Found
-- Executing [executecomm...@asterisk_guitools:1]
System(Local/executecomm...@asterisk_guitools-95f0,2, sh
/var/lib/asterisk/scripts/detectdahdi.sh) in new stack
-- Executing [executecomm...@asterisk_guitools:1]
System(Local/executecomm...@asterisk_guitools-355c,2, dahdi_genconf) in
new stack
/bin/sh: dahdi_genconf: command not found
[Aug 19 08:43:53] WARNING[86135]: app_system.c:107 system_exec_helper:
Unable to execute 'dahdi_genconf'
  == Spawn extension (asterisk_guitools, executecommand, 1) exited non-zero
on 'Local/executecomm...@asterisk_guitools-355c,2'
None
-- Executing [executecomm...@asterisk_guitools:2]
Hangup(Local/executecomm...@asterisk_guitools-95f0,2, ) in new stack
  == Spawn extension (asterisk_guitools, executecommand, 2) exited non-zero
on 'Local/executecomm...@asterisk_guitools-95f0,2'
  == Parsing '/usr/local/etc/asterisk/http.conf': Found
[Aug 19 08:43:53] ERROR[86135]: utils.c:966 ast_carefulwrite: write()
returned error: Broken pipe
[Aug 19 08:43:53] ERROR[86135]: utils.c:966 ast_carefulwrite: write()
returned error: Broken pipe
[Aug 19 08:43:53] ERROR[86135]: utils.c:966 ast_carefulwrite: write()
returned error: Broken pipe
[Aug 19 08:43:53] ERROR[86135]: utils.c:966 ast_carefulwrite: write()
returned error: Broken pipe
[Aug 19 08:43:53] ERROR[86135]: utils.c:966 ast_carefulwrite: write()
returned error: Broken pipe
  == Parsing '/usr/local/etc/asterisk/meetme.conf': Found
-- Executing [executecomm...@asterisk_guitools:1]
System(Local/executecomm...@asterisk_guitools-eee5,2, sh
/var/lib/asterisk/scripts/detectdahdi.sh) in new stack
None
-- Executing [executecomm...@asterisk_guitools:2]
Hangup(Local/executecomm...@asterisk_guitools-eee5,2, ) in new stack
  == Spawn extension (asterisk_guitools, executecommand, 2) exited non-zero
on 'Local/executecomm...@asterisk_guitools-eee5,2'
[Aug 19 08:43:53] ERROR[86135]: utils.c:966 ast_carefulwrite: write()
returned error: Broken pipe
[Aug 19 08:43:53] ERROR[86135]: utils.c:966 ast_carefulwrite: write()
returned error: Broken pipe
[Aug 19 08:43:53] ERROR[86135]: utils.c:966 ast_carefulwrite: write()
returned error: Broken pipe
[Aug 19 08:43:53] ERROR[86135]: utils.c:966 ast_carefulwrite: write()
returned error: Broken pipe
[Aug 19 08:43:53] ERROR[86135]: utils.c:966 ast_carefulwrite: write()
returned error: Broken pipe
  == Parsing '/usr/local/etc/asterisk/asterisk.conf': Found
  == Parsing '/usr/local/etc/asterisk/http.conf': Found
  == Saving '/usr/local/etc/asterisk/http.conf': Saved
  == Parsing '/usr/local/etc/asterisk/http.conf': Found
  == Parsing '/usr/local/etc/asterisk/http.conf': Found
  == Saving '/usr/local/etc/asterisk/http.conf': Saved
  == Parsing '/usr/local/etc/asterisk/extensions.conf': Found
-- Executing [executecomm...@asterisk_guitools:1]
System(Local/executecomm...@asterisk_guitools-549b,2, sh
/var/lib/asterisk/scripts/detectdahdi.sh) in new stack
None
-- Executing [executecomm...@asterisk_guitools:2]
Hangup(Local/executecomm...@asterisk_guitools-549b,2, ) in new stack
  == Spawn extension (asterisk_guitools, executecommand, 2) exited non-zero
on 'Local/executecomm...@asterisk_guitools-549b,2'
  == Parsing '/usr/local/etc/asterisk/guipreferences.conf': Found
-- Executing [executecomm...@asterisk_guitools:1]
System(Local/executecomm...@asterisk_guitools-734c,2, dahdi_genconf) in
new stack
-- Executing [executecomm...@asterisk_guitools:1]
System(Local/executecomm...@asterisk_guitools-aebb,2, sh
/var/lib/asterisk/scripts/detectdahdi.sh) in new stack
  == Parsing '/usr/local/etc/asterisk/http.conf': Found
/bin/sh: dahdi_genconf: command not found
[Aug 19 08:43:56] WARNING[86135]: app_system.c:107 system_exec_helper:
Unable to execute 'dahdi_genconf'
  == Spawn extension (asterisk_guitools, executecommand, 1) exited non-zero
on 'Local/executecomm...@asterisk_guitools-734c,2'
  == Parsing

Re: [asterisk-users] Multi operator platform Asterisk {manage}

On Wed, 19 Aug 2009, ABBAS SHAKEEL wrote:

 Rsync looks really great !

Yep. Something I wish I learned about earlier in my career :)

 How you are get CDR records etc from the remote servers for reporting 
 purpose . I was thinking to have one centralized database ? but your 
 comments let me think about distributed database ?

I like having a database at each datacenter for autonomy and performance.

Periodically, the master database executes a script with the following 
snippet (error checking and irrelevant details removed for brevity):

# for each host
 for HOST in ${HOST_LIST}
 do

${DATE} +%T Mark the records on ${HOST} to be collected.
 mysql\
 ${USER_AUTH}\
 --databa...@database_database@\
 --execute=update cdrs set disposition = 'COLLECTING'\
 --execute= where disposition is NULL;\
 --host=${HOST}\
 ${END_OF_LIST}

${DATE} +%T Dump the marked records from ${HOST}.
 mysqldump\
 ${USER_AUTH}\
 --databases @database_datab...@\
 --host=${HOST}\
 --no-create-info\
 --skip-opt\
 --tables cdrs\
 --where=disposition = 'COLLECTING'\
 /tmp/${HOST}.sql\
 ${END_OF_LIST}

${DATE} +%T Load the records from ${HOST} into our database.
 mysql\
 ${USER_AUTH}\
 --databa...@database_database@\
 --host=localhost\
 /tmp/${HOST}.sql\
 ${END_OF_LIST}

${DATE} +%T Compressing the dump file from ${HOST}.
 gzip /tmp/${HOST}.sql
 mv /tmp/${HOST}.sql.gz /tmp/${HOST}.sql.gz-${TIMESTAMP}

${DATE} +%T Delete the collected records from ${HOST}.
 mysql ${USER_AUTH}\
 --databa...@database_database@\
 --execute=delete from cdrs where disposition = 
'COLLECTING';\
 --host=${HOST}\
 ${END_OF_LIST}

${DATE} +%T Set the disposition on this host.
 mysql ${USER_AUTH}\
 --database=${DATABASE_DATABASE}\
 --execute=update cdrs set disposition = 'COLLECTED'\
 --execute= where disposition = 'COLLECTING';\
 --host=localhost\
 ${END_OF_LIST}

# end of hosts loop
 done

-- 
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: How to copy track from CD for MOH without getting Junk at beginning of frame ...

On Wed, 19 Aug 2009, Lee, John (Sydney) wrote:

 I was copying tracks from CD into mp3 files so that I could use it in
 Asterisk 1.4.21.2 MOH.

 Are there any Asterisk+Audio expert that can offer me some advice?

Don't use MP3. Why would you want to burn CPU cycles decompressing the 
same stuff over and over?

-- 
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] DAHDI - better to have card?

2009-08-19 Thread Boehm, Matthew

 Boehm, Matthew wrote:
 
  MeetMe requires an external timing source. Right now, using the
dummy
  driver. Is it possible to use the card solely for timing purposes?
 Any
  benefit to doing so? Or should I just sell the cards?
 
 DAHDI 2.2.0 provides timing without using the dummy driver and without
 needing any cards. You can use a card for timing, but you'd need to
 hook
 up a crossover cable between two of the ports to get them out of red
 alarm status... but it's really not necessary any longer.

So you say I don't need a card for timing, but will having the card,
strictly for timing purposes, help at all with the quality of the
conferences or help if a conference reaches 50+ people?

Thanks,
Matthew

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[asterisk-users] outbound calls not ringing

2009-08-19 Thread Ott Rose


I put a post on here about my issues with outbound calls not ringing but i 
haven't resolved it. so i am trying again.

When i dial any outside number i dont get a ring tone at all. when the person 
picks up and starts to talk i can hear them fine. it sounds great. How do I 
start to troubleshot this?

_
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[asterisk-users] mysql error (err 2002)

2009-08-19 Thread harry R

Hi

I Have a problem with mysql/asterisk realtime interaction.
each time I try to connect a sip phone or to use this CLI command - realtime
mysql status - I obtain this error message :
Mysql Realtime: Failed to connect database server asteriskdb on localhost
(err 2002)

here a sample of my res_mysql.conf file :
[general]
dbhost = localhost
dbname = asteriskdb
dbuser = asterisk
dbpass = asterisk
dbport = 3306

and a sample of my extconfig.conf file :
[settings]
sipusers = mysql,general,sip_terminal
sippeers = mysql,general,sip_terminal

My mysql database named asteriskdb and I have a table for sip named
sip_terminal.

So any tips to resolve my error message will be welcome.

regards.

Harry.
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Re: [asterisk-users] outbound calls not ringing

2009-08-19 Thread Danny Nicholas

Have you tried putting a (,r) on your Dial command (dial
dahdi/1/18005551212,60,r) ?

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Wednesday, August 19, 2009 8:55 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] outbound calls not ringing

 

I put a post on here about my issues with outbound calls not ringing but i
haven't resolved it. so i am trying again.

When i dial any outside number i dont get a ring tone at all. when the
person picks up and starts to talk i can hear them fine. it sounds great.
How do I start to troubleshot this?

  _  

With Windows Live, you can organize, edit, and share your photos. Click
http://www.windowslive.com/Desktop/PhotoGallery  here.

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Re: [asterisk-users] Newbie: How to copy track from CD for MOH without getting Junk at beginning of frame ...

2009-08-19 Thread Alex Balashov

Steve Edwards wrote:
 On Wed, 19 Aug 2009, Lee, John (Sydney) wrote:
 
 I was copying tracks from CD into mp3 files so that I could use it in
 Asterisk 1.4.21.2 MOH.

 Are there any Asterisk+Audio expert that can offer me some advice?
 
 Don't use MP3. Why would you want to burn CPU cycles decompressing the 
 same stuff over and over?

Yep, agreed.  Convert the file to the native codec(s) in which it will 
be played.


-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [asterisk-users] mysql error (err 2002)

2009-08-19 Thread Steve Howes

On 19 Aug 2009, at 14:59, harry R wrote:
 Mysql Realtime: Failed to connect database server asteriskdb on  
 localhost (err 2002)

 here a sample of my res_mysql.conf file :
 [general]
 dbhost = localhost
 dbname = asteriskdb
 dbuser = asterisk
 dbpass = asterisk
 dbport = 3306

mysql -uasterisk -pasterisk asteriskdb

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Re: [asterisk-users] DAHDI - better to have card?

2009-08-19 Thread Benny Amorsen

Kevin P. Fleming kpflem...@digium.com writes:

 DAHDI 2.2.0 provides timing without using the dummy driver and without
 needing any cards. You can use a card for timing, but you'd need to hook
 up a crossover cable between two of the ports to get them out of red
 alarm status... but it's really not necessary any longer.

Does that mean you don't have to load any kernel modules to get MeetMe
running these days?

This would be very handy, as it is rather difficult to get custom kernel
modules working with OpenVZ-based virtualization.


/Benny


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Re: [asterisk-users] Skype for Asterisk???

2009-08-19 Thread Terry Wilson

 Have you posted a bug describing the issues you are having at 
 http://betareports.digium.com/mantis/
 yet? I would love to have the opportunity to actually fix any bugs
 that people find.  :-)

 I installed the 1.0 release of Skype for Asterisk and last night on my
 production box running Asterisk 1.26.1 i got segfaults and 32 core  
 dumps,
 all happened in a time frame between 01:04 - 01:08 at night (so 4
 minutes).

 Anyone else seeing this?

I haven't seen (or heard of) it happening.  Please post a bug report  
on http://betareports.digium.com/mantis/ with a backtrace from one of  
the core dumps along with the relevant information about your setup,  
dialplan, chan_skype.conf, etc.  If there is a crash, I need to fix  
it. :-)

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Re: [asterisk-users] Zaptel - DAHDI: now echo

Tzafrir Cohen wrote:
 On Tue, Aug 18, 2009 at 10:00:55AM -0400, Dave Fullerton wrote:
 
 Here's my $0.02. If you don't want an echo canceller, specify 
 echocanceller=none,x-y and have dahdi_cfg print a warning (at any 
 verbosity level) when an echo canceller is not specified for a channel.
 Personally, I would also like to see an option that says Use the 
 hardware canceller, like echocanceller=hw,x-y. This would have the 
 added benefit of being able to display an error/warning when the 
 hardware canceller is specified but no hw canceller is present. It goes 
 against my grain to not specify a canceller to mean use a harware one if 
 it happens to exist.
 
 Though this means you have to explicitly configure hardware echo
 cancellers to work, which is not as before. This leaves even more room
 for error.
 

It is true that this method would require more configuration work and 
that it would probably throw people off who were used to the old method. 
However, I don't agree that it leaves more room for error. The current 
system, IMHO, has a certain amount of ambiguity to it. If I inherit a 
production system from someone, I can't tell for sure what the echo 
canceller setup is just by looking at system.conf. I have to look at 
system.conf and then know if hardware echo can is present. Aside from 
opening the case or looking at dmesg output, I'm not even sure how to 
see if a hardware echocan is present or not.
The post that started this thread is another example of that ambiguity. 
Not defining an echo canceller to mean don't use one, or use a hardware 
one if there is one I think leaves room for confusion and error.

-Dave

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Re: [asterisk-users] Zaptel - DAHDI: now echo

2009-08-19 Thread Jeff LaCoursiere


On Wed, 19 Aug 2009, Dave Fullerton wrote:

 Tzafrir Cohen wrote:
 On Tue, Aug 18, 2009 at 10:00:55AM -0400, Dave Fullerton wrote:

 Here's my $0.02. If you don't want an echo canceller, specify
 echocanceller=none,x-y and have dahdi_cfg print a warning (at any
 verbosity level) when an echo canceller is not specified for a channel.
 Personally, I would also like to see an option that says Use the
 hardware canceller, like echocanceller=hw,x-y. This would have the
 added benefit of being able to display an error/warning when the
 hardware canceller is specified but no hw canceller is present. It goes
 against my grain to not specify a canceller to mean use a harware one if
 it happens to exist.

 Though this means you have to explicitly configure hardware echo
 cancellers to work, which is not as before. This leaves even more room
 for error.


 It is true that this method would require more configuration work and
 that it would probably throw people off who were used to the old method.
 However, I don't agree that it leaves more room for error. The current
 system, IMHO, has a certain amount of ambiguity to it. If I inherit a
 production system from someone, I can't tell for sure what the echo
 canceller setup is just by looking at system.conf. I have to look at
 system.conf and then know if hardware echo can is present. Aside from
 opening the case or looking at dmesg output, I'm not even sure how to
 see if a hardware echocan is present or not.
 The post that started this thread is another example of that ambiguity.
 Not defining an echo canceller to mean don't use one, or use a hardware
 one if there is one I think leaves room for confusion and error.

 -Dave


I feel like I must be missing something here.  In 1.4, to my knowledge, if 
hardware echo cancellation was present, it would be used automatically. 
Further, software echo was enabled by default.  If hardware was available 
the software would turn itself off automatically.

What was wrong with this setup?  There was no ambiguity, and there was no 
confusion.

Have I assumed the above in error all this time?

So in 1.6 the hardware echo is on if available, and its only that you must 
enable software cancellation if you want it by adding the appropriate 
module.  Is that right?

It seems then that we would be back to the 1.4 situation if asterisk 
shipped with one of the SEC modules enabled by default, and you could 
change it or turn it off if you wanted.  Kevin seemed to confirm that this 
was the plan.  Sounds good to me.

Cheers,

j

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Re: [asterisk-users] DAHDI - better to have card?

2009-08-19 Thread David Backeberg

On Wed, Aug 19, 2009 at 10:11 AM, Benny Amorsenbenny+use...@amorsen.dk wrote:
 Kevin P. Fleming kpflem...@digium.com writes:

 DAHDI 2.2.0 provides timing without using the dummy driver and without
 needing any cards. You can use a card for timing, but you'd need to hook
 up a crossover cable between two of the ports to get them out of red
 alarm status... but it's really not necessary any longer.

 Does that mean you don't have to load any kernel modules to get MeetMe
 running these days?

Yes, and no. There's a new-er application called ConfBridge() which
will let you attempt to do conferencing without depending on a
kernel-based timer. I think you need to jump to the 1.6.2 tree to get
the application, and there isn't much documentation in the world yet.
I attempted to write some up at:

http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge

If somebody knows more about ConfBridge() than I do, please feel free
to update the wiki on voip-info.

If you're using tradational MeetMe(), you still need some kind of
kernel module, whether it's dahdi_dummy or the appropriate kernel
module for a real Digium card.

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Re: [asterisk-users] Zaptel - DAHDI: now echo

Jeff LaCoursiere wrote:
 On Wed, 19 Aug 2009, Dave Fullerton wrote:
 
 Tzafrir Cohen wrote:
 On Tue, Aug 18, 2009 at 10:00:55AM -0400, Dave Fullerton wrote:

 Here's my $0.02. If you don't want an echo canceller, specify
 echocanceller=none,x-y and have dahdi_cfg print a warning (at any
 verbosity level) when an echo canceller is not specified for a channel.
 Personally, I would also like to see an option that says Use the
 hardware canceller, like echocanceller=hw,x-y. This would have the
 added benefit of being able to display an error/warning when the
 hardware canceller is specified but no hw canceller is present. It goes
 against my grain to not specify a canceller to mean use a harware one if
 it happens to exist.
 Though this means you have to explicitly configure hardware echo
 cancellers to work, which is not as before. This leaves even more room
 for error.

 It is true that this method would require more configuration work and
 that it would probably throw people off who were used to the old method.
 However, I don't agree that it leaves more room for error. The current
 system, IMHO, has a certain amount of ambiguity to it. If I inherit a
 production system from someone, I can't tell for sure what the echo
 canceller setup is just by looking at system.conf. I have to look at
 system.conf and then know if hardware echo can is present. Aside from
 opening the case or looking at dmesg output, I'm not even sure how to
 see if a hardware echocan is present or not.
 The post that started this thread is another example of that ambiguity.
 Not defining an echo canceller to mean don't use one, or use a hardware
 one if there is one I think leaves room for confusion and error.

 -Dave

 
 I feel like I must be missing something here.  In 1.4, to my knowledge, if 
 hardware echo cancellation was present, it would be used automatically. 
 Further, software echo was enabled by default.  If hardware was available 
 the software would turn itself off automatically.
 
 What was wrong with this setup?  There was no ambiguity, and there was no 
 confusion.
 
 Have I assumed the above in error all this time?
 
 So in 1.6 the hardware echo is on if available, and its only that you must 
 enable software cancellation if you want it by adding the appropriate 
 module.  Is that right?
 
 It seems then that we would be back to the 1.4 situation if asterisk 
 shipped with one of the SEC modules enabled by default, and you could 
 change it or turn it off if you wanted.  Kevin seemed to confirm that this 
 was the plan.  Sounds good to me.

Sort of, except it's not a difference between 1.4 and 1.6, it's a 
difference between Zaptel and DAHDI (which also works in asterisk 1.4). 
In Zaptel you compiled in a software echo canceller and that was used if 
a hardware canceller was not present (you didn't have to specify). In 
DAHDI, you must explicitly specify what software echo canceller you want 
to use for each channel in system.conf. If you do not specify, then you 
either do not get an echo canceller, or you automatically use the 
hardware canceller-if it is present. My suggestion is that you always 
have to explicitly state what echo canceler you wish to use for each 
channel, whether it be software, hardware or none at all.

-Dave

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Re: [asterisk-users] Channels don't go away with soft hangup

2009-08-19 Thread Luis Morales

There is another way,

Try to recompile your asterisk with this options:

1) edit asterisk Makefile and add:
BUSYDETECT+= -DBUSYDETECT_COMPARE_TONE_AND_SILENCE -DBUSYDETECT_MARTIN

2) Run
make clean
./configure --prefix=/usr
make
make install

Regards,



On Wed, Aug 19, 2009 at 6:26 PM, Luis Moralesfaston...@gmail.com wrote:
 Take a look:


 1) Verify the cable pin out for rj-11 conector for analog por 6 and 9.
 The pin out must be equal to any port that work fine.

 2) If not (1) try to reproduce the scenary, call from iax client to
 any celular. Activate the debug level and verbose level. (core set
 debug 255, core set verbose 255 from asterisk cli) and look what
 happens. It's posible that your celular device not send the correct
 signal when the call finish.

 Good luck!


 On Wed, Aug 19, 2009 at 5:33 PM, Raimund Sachererr...@runsolutions.com 
 wrote:
 Hi Luis,
 the problem is it is basically the first time i saw it (or recognized it)
 so, it does definitly not happen regularly, I have more problems with our
 xircom analog usb switch which handles our outgoing mobile connections, this
 stuff has problems detecting busydetect but this I will debug another time.
 This time I had the problem with 2 landlines, out of like 100ths of calls
 today.
 What I want to know is if there is a harder way to force a hangub than soft
 hangup which does not interrupt the other calls.
 best
 --
 Raimund Sacherer
 -
 RunSolutions
 Open Source It Consulting
 -
 Email:�...@runsolutions.com
 tel: 625 40 32 08

 Parc Bit - Centro Empresarial Son Espanyol
 Edificio Estel - Local 3D
 07121 -  Palma de Mallorca
 Baleares
 On Aug 19, 2009, at 12:46 PM, Luis Morales wrote:

 Did you try use busytect option enabeled into zaptel.conf file ?

 Another way must be recompile your asterisk an enable BUSYDETECT
 options for hangup.

 Regards,


 On Wed, Aug 19, 2009 at 8:55 AM, Raimund Sachererr...@runsolutions.com
 wrote:

 Hello List,

 our setup:

 Callcenter

 IBM Hardware, 1x TE420, 1x xircom analog switch, 4x different cellular

 providers on the xircom analog port, ~60 agents

 Debian 5.0.1 (Lenny)

 Asterisk 1.4.21.2 Debian Package recompiled with additional app_queue

 segfault fix

 Zaptel 1.4.11 Debian Package


 My Problem is I have two channels (Zap/9-1 and Zap/6-1) which have a

 duration of over 4 hours.

 I am pretty sure that there is not much talking going on:

 Zap/6-1

       Frames in: 104

      Frames out: 101

  Time to Hangup: 0

    Elapsed Time: 4h23m26s

 Zap/9-1

       Frames in: 7196

      Frames out: 7186

  Time to Hangup: 0

    Elapsed Time: 4h23m28s


 I tried to terminate the channels with soft hangup, but they are not

 going away, so, what are my possibilities without interrupting the

 service?


 thanks and best regards,

 Ray


 --

 Raimund Sacherer

 -

 RunSolutions

     Open Source It Consulting

 -


 Parc Bit - Centro Empresarial Son Espanyol

 Edificio Estel - Local 3D

 07121 -  Palma de Mallorca

 Baleares


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 --
 -
 Luis Morales
 Consultor de Tecnologia
 Cel: +(58)416-4242091
 -
 Empieza por hacer lo necesario, luego lo que es posible... y de
 pronto estarás haciendo lo imposible

 Leonardo Da'Vinci
 -

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 --
 -
 Luis Morales
 Consultor de Tecnologia
 Cel: +(58)416-4242091
 -
 Empieza por hacer lo necesario, luego lo que es posible... y de
 pronto estarás haciendo lo imposible

 Leonardo Da'Vinci
 -




-- 
-
Luis

Re: [asterisk-users] Zaptel - DAHDI: now echo

Dave Fullerton wrote:

 It is true that this method would require more configuration work and 
 that it would probably throw people off who were used to the old method. 
 However, I don't agree that it leaves more room for error. The current 
 system, IMHO, has a certain amount of ambiguity to it. If I inherit a 
 production system from someone, I can't tell for sure what the echo 
 canceller setup is just by looking at system.conf. I have to look at 
 system.conf and then know if hardware echo can is present. Aside from 
 opening the case or looking at dmesg output, I'm not even sure how to 
 see if a hardware echocan is present or not.

The dahdi_scan tool will tell you whether hardware echocans are present
or not, among other methods.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] DAHDI - better to have card?

Benny Amorsen wrote:
 Kevin P. Fleming kpflem...@digium.com writes:
 
 DAHDI 2.2.0 provides timing without using the dummy driver and without
 needing any cards. You can use a card for timing, but you'd need to hook
 up a crossover cable between two of the ports to get them out of red
 alarm status... but it's really not necessary any longer.
 
 Does that mean you don't have to load any kernel modules to get MeetMe
 running these days?

No, it's still DAHDI, it's still a kernel module. What's not required
are any additional modules beyond dahdi_dummy, and DAHDI automatically
uses timing provided by hardware (if present) or internal timing.

However, MeetMe requires more than just timing, it also requires
conference mixing, which is provided by DAHDI in any case, regardless of
timing source. ConfBridge does not use DAHDI for mixing, and can use
non-DAHDI sources of timing, but it's still very new.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] outbound calls not ringing

On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
 I put a post on here about my issues with outbound calls not ringing
 but i haven't resolved it. so i am trying again.
 
 When i dial any outside number i dont get a ring tone at all. when the
 person picks up and starts to talk i can hear them fine. it sounds
 great. How do I start to troubleshot this?
snip
What type of phones are giving you the problem? If I recall correctly,
our SIP phones had this problem depending on how the destination handled
signaling.  We resolved it by adding progressinband=no (as opposed to
the default never - at least I think it is the default) but this
produces the problem of duplicate ring tones at times.  Hope this helps
- John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Zaptel - DAHDI: now echo

Kevin P. Fleming wrote:
 Dave Fullerton wrote:
 
 It is true that this method would require more configuration work and 
 that it would probably throw people off who were used to the old method. 
 However, I don't agree that it leaves more room for error. The current 
 system, IMHO, has a certain amount of ambiguity to it. If I inherit a 
 production system from someone, I can't tell for sure what the echo 
 canceller setup is just by looking at system.conf. I have to look at 
 system.conf and then know if hardware echo can is present. Aside from 
 opening the case or looking at dmesg output, I'm not even sure how to 
 see if a hardware echocan is present or not.
 
 The dahdi_scan tool will tell you whether hardware echocans are present
 or not, among other methods.
 

I tried that, but I didn't see anything that specified whether the echo 
canceller was present. Here's the output, can you tell me what I should 
be looking for?

r...@srv210394:~# dahdi_scan
[1]
active=yes
alarms=OK
description=T2XXP (PCI) Card 0 Span 1
name=TE2/0/1
manufacturer=Digium
devicetype=Wildcard TE220 (4th Gen)
location=Board ID Switch 0
basechan=1
totchans=24
irq=16
type=digital-T1
syncsrc=2
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=B8ZS,AMI
framing_opts=ESF,D4
coding=B8ZS
framing=ESF
[2]
active=yes
alarms=OK
description=T2XXP (PCI) Card 0 Span 2
name=TE2/0/2
manufacturer=Digium
devicetype=Wildcard TE220 (4th Gen)
location=Board ID Switch 0
basechan=25
totchans=24
irq=16
type=digital-T1
syncsrc=2
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=B8ZS,AMI
framing_opts=ESF,D4
coding=B8ZS
framing=ESF

Thanks,

Dave

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Re: [asterisk-users] mysql error (err 2002)

2009-08-19 Thread harry R

mysql -uasterisk -pasterisk asteriskdb

When I do that in a linux terminal it works.
But I always have this err 2002.
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Re: [asterisk-users] Zaptel - DAHDI: now echo

Dave Fullerton wrote:

 The dahdi_scan tool will tell you whether hardware echocans are present
 or not, among other methods.

 
 I tried that, but I didn't see anything that specified whether the echo 
 canceller was present. Here's the output, can you tell me what I should 
 be looking for?
 
 r...@srv210394:~# dahdi_scan
 [1]
 active=yes
 alarms=OK
 description=T2XXP (PCI) Card 0 Span 1
 name=TE2/0/1
 manufacturer=Digium
 devicetype=Wildcard TE220 (4th Gen)

It would show up in the 'devicetype' line, which should end with 'with
VPM400M', 'with VPMOCT064' or 'with VPMOCT128' in the case of a
dual/quad span card, depending on which module is attached.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] CAP_FOWNER=ep for asterisk

2009-08-19 Thread Tilghman Lesher

On Wednesday 19 August 2009 05:54:32 Raimund Sacherer wrote:
 I need CAP_FOWNER=ep for the asterisk process, i set it with setcap on
 the file /usr/sbin/asterisk, it's there when i look on it with getcap,
 but after starting and loocking with getpcaps there's only
 cap_net_admin+ep set.

 So how exactly do I set CAP_FOWNER? Do I have to patch and recompile
 or is there another solution I did not see yet?

You'd need to patch and recompile.  I really don't think this is really all
that safe of a modification.  Is there another way (such as through groups)
that you can do what you want here?

-- 
Tilghman  Teryl
with Peter, Cottontail, Midnight, Thumper,  Johnny (bunnies)
and Harry, BB,  George (dogs)

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Re: [asterisk-users] mysql error (err 2002)

On Wed, 19 Aug 2009, harry R wrote:

 mysql -uasterisk -pasterisk asteriskdb

 When I do that in a linux terminal it works. But I always have this err 
 2002.

Going out on a shaky limb I know little about...

I always specify all of the connection options in scripts so I don't get 
caught by something in some user's my.cnf file. Does:

mysql\
--database=asteriskdb\
--host=localhost\
--password=asterisk\
--user=asterisk\
${END_OF_LIST}

also work?

-- 
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] Zaptel - DAHDI: now echo

Kevin P. Fleming wrote:
 Dave Fullerton wrote:
 
 The dahdi_scan tool will tell you whether hardware echocans are present
 or not, among other methods.

 I tried that, but I didn't see anything that specified whether the echo 
 canceller was present. Here's the output, can you tell me what I should 
 be looking for?

 r...@srv210394:~# dahdi_scan
 [1]
 active=yes
 alarms=OK
 description=T2XXP (PCI) Card 0 Span 1
 name=TE2/0/1
 manufacturer=Digium
 devicetype=Wildcard TE220 (4th Gen)
 
 It would show up in the 'devicetype' line, which should end with 'with
 VPM400M', 'with VPMOCT064' or 'with VPMOCT128' in the case of a
 dual/quad span card, depending on which module is attached.
 

I guess I just found a bug then, because the card above is a TE220B. 
Here's a portion of the dmesg output:

wct4xxp :02:08.0: PCI INT A - GSI 16 (level, low) - IRQ 16
Found TE2XXP at base address dfcfff80, remapped to f8872f80
TE2XXP version c01a016c, burst ON
Octasic optimized!
FALC version: 0005, Board ID: 00
Reg 0: 0x35b55400
Reg 1: 0x35b55000
Reg 2: 0x
Reg 3: 0x
Reg 4: 0xff01
Reg 5: 0x
Reg 6: 0xc01a016c
Reg 7: 0x1000
Reg 8: 0x
Reg 9: 0x00ff00ff
Reg 10: 0x004a
TE2XXP: Launching card: 0
TE2XXP: Setting up global serial parameters
Found a Wildcard: Wildcard TE220 (4th Gen)

About to enter spanconfig!
Done with spanconfig!
dahdi: Registered tone zone 0 (United States / North America)
About to enter startup!
TE2XXP: Span 1 configured for ESF/B8ZS
wct2xxp: Setting yellow alarm on span 1
timing source auto card 0!
VPM400: Not Present
timing source auto card 0!
firmware: requesting dahdi-fw-oct6114-064.bin
VPM450: echo cancellation for 64 channels
VPM450: hardware DTMF disabled.
VPM450: Present and operational servicing 2 span(s)
Completed startup!


-Dave

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Re: [asterisk-users] mysql error (err 2002)

2009-08-19 Thread Steve Howes


On 19 Aug 2009, at 16:37, harry R wrote:

 mysql -uasterisk -pasterisk asteriskdb

 When I do that in a linux terminal it works.
 But I always have this err 2002.

I'd try and tcpdump it if you can find a way. Might be something odd  
happening.

S

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Re: [asterisk-users] outbound calls not ringing

2009-08-19 Thread Ott Rose

we are using Aastra 57i

i don't see that setting. where is it at?

From: jsulli...@opensourcedevel.com
To: asterisk-users@lists.digium.com
Date: Wed, 19 Aug 2009 11:07:21 -0400
Subject: Re: [asterisk-users] outbound calls not ringing

On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
I put a post on here about my issues with outbound calls not ringing
but i haven't resolved it. so i am trying again.

When i dial any outside number i dont get a ring tone at all. when the
person picks up and starts to talk i can hear them fine. it sounds
great. How do I start to troubleshot this?
snip
What type of phones are giving you the problem? If I recall correctly,
our SIP phones had this problem depending on how the destination handled
signaling. We resolved it by adding progressinband=no (as opposed to
the default never - at least I think it is the default) but this
produces the problem of duplicate ring tones at times. Hope this helps
- John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society

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[asterisk-users] * 1.4 - 1.6, zaptel - dahdi

This has got to be an FAQ, so if someone can point me to where it is
answered, I would be greatly appreciated. The documentation for all this
stuff is scattered and (to me at least) either very sketchy or hard to
find.

What I want is a guide for how to convert from 1.4 with zaptel to 1.6
with dahdi. All the recent kernel vulnerabilities are forcing me to
upgrade my home server from no-longer-supported Fedora 8 up to Fedora
11, and that means upgrading asterisk as well.

It appears that, although the dahdi-tools package is part of Fedora 11,
the kernel modules are not. I couldn't get the dahdi tools such as
dahdi_scan to work at all until I installed the dahdi-kmdl package from
ATrpms. Does that match others' experiences?

This also means a translation from extensions.conf to AEL. I tried using
the tool whose name I forget that produces an ael file from an old
extensions.conf file, but I get tons of errors when I try to load the
resulting file. In particular, there are comments about Macros but no
detailed documentation as to how to translate something like GoTo(s|
3). I try using goto s|3 and get an error that there is no label s
in the current context. Is there something else I should be doing to
accomplish this, or do I really have to create an explicit label for
every extension that is the current target of a Goto() call?

Thanks for any advice. A line-by-line translation of my dialplan looks
to be a very tedious and time-consuming task with a steep learning
curve. There has got to be a better way, I know I'm not the first to
have to do this.

--Greg



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Re: [asterisk-users] mysql error (err 2002)

2009-08-19 Thread Miguel Molina

harry R escribió:
 mysql -uasterisk -pasterisk asteriskdb

 When I do that in a linux terminal it works.
 But I always have this err 2002.
Greeting missing.
Elaborate missing.
Err 0x1b5a9f4c

You're not talking to machines here. :-)

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center


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Re: [asterisk-users] outbound calls not ringing

sip.conf

On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote:
 
 we are using Aastra 57i
 
 i don't see that setting. where is it at?
 
  From: jsulli...@opensourcedevel.com
  To: asterisk-users@lists.digium.com
  Date: Wed, 19 Aug 2009 11:07:21 -0400
  Subject: Re: [asterisk-users] outbound calls not ringing
  
  On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
   I put a post on here about my issues with outbound calls not
 ringing
   but i haven't resolved it. so i am trying again.
   
   When i dial any outside number i dont get a ring tone at all. when
 the
   person picks up and starts to talk i can hear them fine. it sounds
   great. How do I start to troubleshot this?
  snip
  What type of phones are giving you the problem? If I recall
 correctly,
  our SIP phones had this problem depending on how the destination
 handled
  signaling. We resolved it by adding progressinband=no (as opposed to
  the default never - at least I think it is the default) but this
  produces the problem of duplicate ring tones at times. Hope this
 helps
  - John
  -- 
  John A. Sullivan III
  Open Source Development Corporation
  +1 207-985-7880
  jsulli...@opensourcedevel.com
  
  http://www.spiritualoutreach.com
  Making Christianity intelligible to secular society
  
  
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John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] * 1.4 - 1.6, zaptel - dahdi

2009-08-19 Thread Singer X.J. Wang

Can I make a non related suggestions?

Ditch Fedora and use CentOS.

Greg Woods wrote:
 This has got to be an FAQ, so if someone can point me to where it is
 answered, I would be greatly appreciated. The documentation for all this
 stuff is scattered and (to me at least) either very sketchy or hard to
 find.

 What I want is a guide for how to convert from 1.4 with zaptel to 1.6
 with dahdi. All the recent kernel vulnerabilities are forcing me to
 upgrade my home server from no-longer-supported Fedora 8 up to Fedora
 11, and that means upgrading asterisk as well.

 It appears that, although the dahdi-tools package is part of Fedora 11,
 the kernel modules are not. I couldn't get the dahdi tools such as
 dahdi_scan to work at all until I installed the dahdi-kmdl package from
 ATrpms. Does that match others' experiences?

 This also means a translation from extensions.conf to AEL. I tried using
 the tool whose name I forget that produces an ael file from an old
 extensions.conf file, but I get tons of errors when I try to load the
 resulting file. In particular, there are comments about Macros but no
 detailed documentation as to how to translate something like GoTo(s|
 3). I try using goto s|3 and get an error that there is no label s
 in the current context. Is there something else I should be doing to
 accomplish this, or do I really have to create an explicit label for
 every extension that is the current target of a Goto() call?

 Thanks for any advice. A line-by-line translation of my dialplan looks
 to be a very tedious and time-consuming task with a steep learning
 curve. There has got to be a better way, I know I'm not the first to
 have to do this.

 --Greg



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Re: [asterisk-users] Accessing Asterisk gosub arguments in extensions.lua

2009-08-19 Thread Brian Camp



On Monday 17 August 2009 18:03:10 pm Tilghman Lesher wrote:

  How does one go about accessing gosub arguments from Asterisk in
  extensions.lua?

  You cannot.  The various methods of dialplan creation are not
  designed to be
  interoperable.  Some people have made various methods work (such as
  between
  extensions.conf and extensions.ael) but those methods are not
  guaranteed to
  work, nor are they supported.


Why does Asterisk even have functions in pbx_lua like app.return and 
app.gosub, which jump between extensions.conf and extensions.lua if its 
not supported or guaranteed to work.  Theres nothing in extensions.lua 
or pbx_lua.c, which so far as I can tell are the only available sources 
of documentation for pbx_lua, that suggests extensions.conf 
interoperability doesn't work.

  Use one method and stick with it.

Deciding to switch between a 10,000+ line dialplan in an assembly-like 
language (extensions.conf) over to one in Lua in one all at once is not 
a good position to be in.

-Brian

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Re: [asterisk-users] Zaptel - DAHDI: now echo

Dave Fullerton wrote:

 I guess I just found a bug then, because the card above is a TE220B. 
 Here's a portion of the dmesg output:

You are correct sir; I wrote the code that was supposed to report the
VPM presence via dahdi_scan, but clearly did not test it properly
because it didn't work :-(

I've just committed a fix to dahdi-linux to correct this problem, and it
will be in the next dahdi-linux point release. Thanks!

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] * 1.4 - 1.6, zaptel - dahdi

2009-08-19 Thread Axel Thimm

Hi,

On Wed, Aug 19, 2009 at 09:56:38AM -0600, Greg Woods wrote:
 This has got to be an FAQ, so if someone can point me to where it is
 answered, I would be greatly appreciated. The documentation for all this
 stuff is scattered and (to me at least) either very sketchy or hard to
 find.
 
 What I want is a guide for how to convert from 1.4 with zaptel to 1.6
 with dahdi. All the recent kernel vulnerabilities are forcing me to
 upgrade my home server from no-longer-supported Fedora 8 up to Fedora
 11, and that means upgrading asterisk as well.
 
 It appears that, although the dahdi-tools package is part of Fedora 11,
 the kernel modules are not. I couldn't get the dahdi tools such as
 dahdi_scan to work at all until I installed the dahdi-kmdl package from
 ATrpms. Does that match others' experiences?

I just finished a transition from the old ATrpms setup for asterisk to
a new one that allows you to use asterisk 1.4 on F11 w/o fear of
asterisk 1.6 overwriting it. It's not what I recommend, one should try
to move to 1.6, but upgrading both the underlying OS and the
asterisk/dahdi/zaptel framework can be seperated that way. For
RHEL/CentOS/Scientific Linux there will even be asterisk12/zaptel12
support, and there are shiny new 1.6 packages there as well.

ATM there are no userland bits to download at ATrpms at all, as I'm
trying to move the fax application out of the asterisk14 rpm to using
agx-ast-addons instead, but I'll release the packages and make an
announcement about it soon. If you feel like a guinea pig, you can
contact me off list or on ATrpms-devel to help testing.
-- 
Axel.Thimm at ATrpms.net


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Re: [asterisk-users] outbound calls not ringing

2009-08-19 Thread Ott Rose


here is my sip.conf. i don't see it.
;;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications 
to ;
; this file must be done via the web gui. There are alternative files to make   
 ;
; custom modifications, details at: http://freepbx.org/configuration_files  
 ;
;;
;

[general]

; These files will all be included in the [general] context
;
#include sip_general_additional.conf

;sip_general_custom.conf is the proper file location for placing any sip general
;options that you might need set. For example: enable and force the sip 
jitterbuffer.
;If these settings are desired they should be set the sip_general_custom.conf 
file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat'ing when going
;through a firewall.  For nat'ing you'd need to add the following lines:
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf

;sip_nat.conf is here for legacy support reasons and for those that upgrade
;from previous versions.  If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf

;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf

; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf

;sip_custom_post.conf If you have extra parameters that are needed for a
;extension to work to for example, those go here.  So you have extension
;1000 defined in your system you start by creating a line [1000](+) in this
;file.  Then on the next line add the extra parameter that is needed.
;When the sip.conf is loaded it will append your additions to the end of
;that extension.
;
#include sip_custom_post.conf


 From: jsulli...@opensourcedevel.com
 To: asterisk-users@lists.digium.com
 Date: Wed, 19 Aug 2009 12:17:15 -0400
 Subject: Re: [asterisk-users] outbound calls not ringing
 
 sip.conf
 
 On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote:
  
  we are using Aastra 57i
  
  i don't see that setting. where is it at?
  
   From: jsulli...@opensourcedevel.com
   To: asterisk-users@lists.digium.com
   Date: Wed, 19 Aug 2009 11:07:21 -0400
   Subject: Re: [asterisk-users] outbound calls not ringing
   
   On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
I put a post on here about my issues with outbound calls not
  ringing
but i haven't resolved it. so i am trying again.

When i dial any outside number i dont get a ring tone at all. when
  the
person picks up and starts to talk i can hear them fine. it sounds
great. How do I start to troubleshot this?
   snip
   What type of phones are giving you the problem? If I recall
  correctly,
   our SIP phones had this problem depending on how the destination
  handled
   signaling. We resolved it by adding progressinband=no (as opposed to
   the default never - at least I think it is the default) but this
   produces the problem of duplicate ring tones at times. Hope this
  helps
   - John
   -- 
   John A. Sullivan III
   Open Source Development Corporation
   +1 207-985-7880
   jsulli...@opensourcedevel.com
   
   http://www.spiritualoutreach.com
   Making Christianity intelligible to secular society
   
   
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 -- 
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 Open Source Development Corporation
 +1 207-985-7880
 jsulli...@opensourcedevel.com
 
 http://www.spiritualoutreach.com
 Making Christianity intelligible to secular society
 
 
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 asterisk-users

Re: [asterisk-users] * 1.4 - 1.6, zaptel - dahdi

I'm a 1.2 Luddite, but...

On Wed, 19 Aug 2009, Greg Woods wrote:

 What I want is a guide for how to convert from 1.4 with zaptel to 1.6 
 with dahdi.

If you didn't find it on voip-info.org, please start an article.

 All the recent kernel vulnerabilities are forcing me to upgrade my home 
 server from no-longer-supported Fedora 8 up to Fedora 11, and that means 
 upgrading asterisk as well.

Not to start a flame war, but I'm very happy with CentOS 5.3.

Any particular reason you can't compile 1.4 on F11?

 This also means a translation from extensions.conf to AEL.

Really? AEL is compiled by pbx_ael.so into plain old extensions.conf 
style dialplan. When was extensions.conf deprecated? Did I miss the memo?

-- 
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] outbound calls not ringing

Oops! - You're using FreePBX - someone who knows more about FreePBX will
have to help you as I don't.  May I also suggest that you bottom post in
future responses rather than top post; that makes it a little easier to
follow.  Good luck - John

On Wed, 2009-08-19 at 16:59 +, Ott Rose wrote:
 here is my sip.conf. i don't see it.
 ;;
 ; Do NOT edit this file as it is auto-generated by FreePBX. All
 modifications to ;
 ; this file must be done via the web gui. There are alternative files
 to make;
 ; custom modifications, details at:
 http://freepbx.org/configuration_files   ;
 ;;
 ;
 
 [general]
 
 ; These files will all be included in the [general] context
 ;
 #include sip_general_additional.conf
 
 ;sip_general_custom.conf is the proper file location for placing any
 sip general
 ;options that you might need set. For example: enable and force the
 sip jitterbuffer.
 ;If these settings are desired they should be set the
 sip_general_custom.conf file.
 ;
 ; jbenable=yes
 ; jbforce=yes
 ;
 ;It is also the proper place to add the lines needed for sip nat'ing
 when going
 ;through a firewall.  For nat'ing you'd need to add the following
 lines:
 ; nat=yes , externip= , localhost= , and optionally fromdomain= .
 ;
 #include sip_general_custom.conf
 
 ;sip_nat.conf is here for legacy support reasons and for those that
 upgrade
 ;from previous versions.  If you have this file with lines in it
 please make
 ;sure they are not duplicated in sip_general_custom.conf, if so remove
 them
 ;from sip_nat.conf as sip_general_custom.conf will have precedence.
 #include sip_nat.conf
 
 ;sip_registrations_custom.conf is for any customizations you might
 need to do to
 ;the automatically generated registrations that FreePBX makes.
 ;
 #include sip_registrations_custom.conf
 #include sip_registrations.conf
 
 ; These files should all be expected to come after the [general]
 context
 ;
 #include sip_custom.conf
 #include sip_additional.conf
 
 ;sip_custom_post.conf If you have extra parameters that are needed for
 a
 ;extension to work to for example, those go here.  So you have
 extension
 ;1000 defined in your system you start by creating a line [1000](+) in
 this
 ;file.  Then on the next line add the extra parameter that is needed.
 ;When the sip.conf is loaded it will append your additions to the end
 of
 ;that extension.
 ;
 #include sip_custom_post.conf
 
 
  From: jsulli...@opensourcedevel.com
  To: asterisk-users@lists.digium.com
  Date: Wed, 19 Aug 2009 12:17:15 -0400
  Subject: Re: [asterisk-users] outbound calls not ringing
  
  sip.conf
  
  On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote:
   
   we are using Aastra 57i
   
   i don't see that setting. where is it at?
   
From: jsulli...@opensourcedevel.com
To: asterisk-users@lists.digium.com
Date: Wed, 19 Aug 2009 11:07:21 -0400
Subject: Re: [asterisk-users] outbound calls not ringing

On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
 I put a post on here about my issues with outbound calls not
   ringing
 but i haven't resolved it. so i am trying again.
 
 When i dial any outside number i dont get a ring tone at all.
 when
   the
 person picks up and starts to talk i can hear them fine. it
 sounds
 great. How do I start to troubleshot this?
snip
What type of phones are giving you the problem? If I recall
   correctly,
our SIP phones had this problem depending on how the destination
   handled
signaling. We resolved it by adding progressinband=no (as
 opposed to
the default never - at least I think it is the default) but this
produces the problem of duplicate ring tones at times. Hope this
   helps
- John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Accessing Asterisk gosub arguments in extensions.lua

2009-08-19 Thread Tilghman Lesher

On Wednesday 19 August 2009 11:29:58 Brian Camp wrote:
 On Monday 17 August 2009 18:03:10 pm Tilghman Lesher wrote:
   How does one go about accessing gosub arguments from Asterisk in
   extensions.lua?
  
   You cannot.  The various methods of dialplan creation are not
   designed to be
   interoperable.  Some people have made various methods work (such as
   between
   extensions.conf and extensions.ael) but those methods are not
   guaranteed to
   work, nor are they supported.

 Why does Asterisk even have functions in pbx_lua like app.return and
 app.gosub, which jump between extensions.conf and extensions.lua if its
 not supported or guaranteed to work.

Because it's a generic interface to call applications.  It's that you have
app_stack.so loaded, which creates those two applications, and LUA doesn't
limit which applications you can run.  Applications are black boxes, as far as
LUA is concerned.  Would you want it that if you load MyCustomApp, that LUA
won't let you run it, without modifying the LUA source?

 Theres nothing in extensions.lua 
 or pbx_lua.c, which so far as I can tell are the only available sources
 of documentation for pbx_lua, that suggests extensions.conf
 interoperability doesn't work.

There's nothing that says it will, either.

 Deciding to switch between a 10,000+ line dialplan in an assembly-like
 language (extensions.conf) over to one in Lua in one all at once is not
 a good position to be in.

Here's where I would suggest that if you've got 10,000 lines in a dialplan,
you've probably already gone the wrong direction and you should probably
refactor a good amount of that into a database, with proper abstraction
techniques.

-- 
Tilghman  Teryl
with Peter, Cottontail, Midnight, Thumper,  Johnny (bunnies)
and Harry, BB,  George (dogs)

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Re: [asterisk-users] CAP_FOWNER=ep for asterisk

2009-08-19 Thread Tilghman Lesher

On Wednesday 19 August 2009 10:43:37 Tilghman Lesher wrote:
 On Wednesday 19 August 2009 05:54:32 Raimund Sacherer wrote:
  I need CAP_FOWNER=ep for the asterisk process, i set it with setcap on
  the file /usr/sbin/asterisk, it's there when i look on it with getcap,
  but after starting and loocking with getpcaps there's only
  cap_net_admin+ep set.
 
  So how exactly do I set CAP_FOWNER? Do I have to patch and recompile
  or is there another solution I did not see yet?

 You'd need to patch and recompile.  I really don't think this is really all
 that safe of a modification.  Is there another way (such as through groups)
 that you can do what you want here?

As an addendum, cap_net_admin actually has +eip, because if you ever use
core restart now, those capabilities would otherwise be dropped.  This also
means that whereever Asterisk forks off a separate process to do something
(System, AGI, MOH, etc.), it has to drop those privileges before the exec().
If you proceed with your modification, you should do similar, in order to
avoid possible security issues.  BTW, this gets much simpler starting in 1.6.1
with the ast_safe_fork() API call, which does all of those safety procedures
and more, in one place.

-- 
Tilghman  Teryl
with Peter, Cottontail, Midnight, Thumper,  Johnny (bunnies)
and Harry, BB,  George (dogs)

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Re: [asterisk-users] * 1.4 - 1.6, zaptel - dahdi

2009-08-19 Thread David Backeberg

On Wed, Aug 19, 2009 at 11:56 AM, Greg Woodsg...@gregandeva.net wrote:
 This has got to be an FAQ, so if someone can point me to where it is
 answered, I would be greatly appreciated. The documentation for all this
 stuff is scattered and (to me at least) either very sketchy or hard to
 find.

 What I want is a guide for how to convert from 1.4 with zaptel to 1.6
 with dahdi. All the recent kernel vulnerabilities are forcing me to
 upgrade my home server from no-longer-supported Fedora 8 up to Fedora
 11, and that means upgrading asterisk as well.

It sounds like you would really rather not convert from 1.4 to 1.6 right now.
So don't.

I'm going to suggest something controversial. Build from source, and
don't even spend more than 1 minute looking for an rpm.

http://downloads.asterisk.org/pub/telephony/asterisk/

tar -x
./configure
make
make install

Done.

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Re: [asterisk-users] * 1.4 - 1.6, zaptel - dahdi

On Wed, 2009-08-19 at 12:25 -0400, Singer X.J. Wang wrote:
 Can I make a non related suggestions?
 
 Ditch Fedora and use CentOS.

Might be a possibility except that this is a catch-all home server. It
is used for things other than asterisk, so there are other reasons why I
need a more up-to-date distro.

--Greg



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Re: [asterisk-users] * 1.4 - 1.6, zaptel - dahdi

2009-08-19 Thread Gordon Henderson

On Wed, 19 Aug 2009, Greg Woods wrote:

 On Wed, 2009-08-19 at 12:25 -0400, Singer X.J. Wang wrote:
 Can I make a non related suggestions?

 Ditch Fedora and use CentOS.

 Might be a possibility except that this is a catch-all home server. It
 is used for things other than asterisk, so there are other reasons why I
 need a more up-to-date distro.

Compile from source. That keeps you more or less distribution independant. 
It's only painful once :)

And I'd suggest to stick to 1.4 and do the migration from Zaptel to Dahdi 
inside the latest released 1.4 - unless you really need stuff that 1.6 is 
offering. One step at a time and all that...

Gordon

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Re: [asterisk-users] * 1.4 - 1.6, zaptel - dahdi

On Wed, 2009-08-19 at 09:56 -0600, Greg Woods wrote:
 This has got to be an FAQ, so if someone can point me to where it is
 answered, I would be greatly appreciated. The documentation for all this
 stuff is scattered and (to me at least) either very sketchy or hard to
 find.
 
Well at least I'm not crazy, since so far all the suggestions have been
ways to avoid doing the conversion at all (I may consider some of them).
Unfortunately converting to a whole different OS is likely to be just as
big a task (if not bigger) than converting * from 1.4 to 1.6,
considering that there is more than asterisk involved. I suspect the
zaptel/dahdi conversion won't be that hard. It's the extensions.conf -
extensions.ael conversion that's biting me right now. I confess though
that I am surprised to find that I had to install the dahdi-kmdl package
from ATrpms given that Fedora does package dahdi-tools and dahdi-linux. 

If at all possible I want to use the standard packaged version as it
makes security updates much easier. If I wanted to maintain all my apps
from source, I'd use Gentoo which is at least designed to be installed
that way :-)  Or more likely I'd avoid the OS upgrade at all and compile
the apps I use from source on the old OS in order to be able to apply
security fixes. But I'm much more willing to compile things from source
on my desktop than on my main home server; I need easily maintainable
apps there. That, after all, is the whole reason for upgrading the OS in
the first place.

I heard one person say that extensions.conf should still work in 1.6; is
that true? Mine seems to be completely ignored. Is there some other
config file that I need to edit to make this happen? If I could do that,
I could get the OS switch done and it would buy me some time to do the
AEL conversion.

--Greg



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Re: [asterisk-users] * 1.4 - 1.6, zaptel - dahdi

On Wed, 19 Aug 2009, Greg Woods wrote:

 If at all possible I want to use the standard packaged version as it 
 makes security updates much easier.

I used to be a use the source Luke kind of guy. Now I'm a yum-aholic.

But, when the pain of using packages exceeds the hassle of the source, 
I'll use the source without hesitation.

FWIW, I've never used an Asterisk package. I started with the source and 
probably will always stay this way. I don't like being dependent on others 
for my mission critical stuff.

-- 
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] * 1.4 - 1.6, zaptel - dahdi

Greg Woods wrote:

 I heard one person say that extensions.conf should still work in 1.6; is
 that true? Mine seems to be completely ignored. Is there some other
 config file that I need to edit to make this happen? If I could do that,
 I could get the OS switch done and it would buy me some time to do the
 AEL conversion.

extensions.conf is fully supported in Asterisk 1.6.x; there have been
some syntax changes (which are covered in the UPGRADE files) that might
be causing failures at loading time, but unless you are explicitly *not*
 loading pbx_config.so, it will be parsed and loaded.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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[asterisk-users] Dial plan sample for detecting Voice Mail

2009-08-19 Thread Bharath B. Reddy Bynagari

Hi, 

 

I am trying to implement a macro-screen mentioned at
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

 

I put the following code in my extensions_additional.conf

screen-from: You have a call;

screen-accept: Press 1 to accept this call or any other key to reject.;

 

[macro-screen]

exten = s,1,Wait(0.2)

exten = s,1,Playback(screen-from)

exten = s,1,Playback(${ARG1})

exten = s,1,Read(ACCEPT|screen-accept|1)

exten = s,1,GotoIf($[${ACCEPT} = 1 ] ?yes:no)

exten = s,1(yes),SetVar(MACRO_RESULT=CONTINUE)

exten = s,1(no),System(/bin/rm ${ARG1})

 

; end of [macro-screen]

 

[multi-dir-callback]

include = multi-dir-callback-custom

exten = _X.,1,Macro(screen,)

exten = _X.,1,Answer

exten = _X.,n,Playback(beep)

;exten =
_X.,n,DeadAGI(agi://127.0.0.1/callback_handler?num2=${EXTEN}callid=${CALL_I
D}num1=${num1}CallStatus=${DIALSTATUS}state=${STATE})a

exten =
_X.,n,DeadAGI(agi://127.0.0.1/callback_handler?num2=${EXTEN}callid=${CALL_I
D}num1=${num1}state=${STATE})

exten = _X.,n,Goto(${EXTEN},1)

exten =
hangup,1,DeadAGI(agi://127.0.0.1/callback_handler?num2=${EXTEN}callid=${CAL
L_ID}num1=${num1}state=${STATE})

 

; end of [multi-dir-callback]

 

It is not even recognizing the Screen macro? What I am I doing wrong?

 

Thanks

 

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[asterisk-users] AsteriskGUI Create VoiceMenu SNAFU

2009-08-19 Thread Gary Baribault

Hi All,

 I'm new to Asterisk, but am a relatively accomplished Linux guy 
(RH5.1) .. I'm using the Asterisk-GUI to try and create a menu for 
incoming calls on an Analog Trunk. I have recorded some .WAV files for 
the menu, but when I try to upload the files, I get an AG101 message. So 
I copied the files to /var/lib/asterisk/sounds/record .. when I go to 
the Voice Menu Prompts selection down the left side of the Asterisk-GUI, 
I see my four files with the options to record again, play and delete. 
If I then go to the Voice Menu option to configure a Voice Menu, and 
click on the Create New VoiceMenu, I get nothing .. nada .. kaput .. how 
the heck do I create a menu for an incoming call on a Trunk?

 When I started this project I knew it would be fun .. I would learn 
a lot! The problem is that one of our administrators is absolutely a 
newbi to Linux, so I have to make this work with the GUI .. any help or 
suggestions would be appreciated!

Thanks

Gary B

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Re: [asterisk-users] * 1.4 - 1.6, zaptel - dahdi

On Wed, 2009-08-19 at 11:49 -0700, Steve Edwards wrote:

 But, when the pain of using packages exceeds the hassle of the source, 
 I'll use the source without hesitation.

Agreed. I use trunk for MythTV, for instance, because there are features
not in the latest packaged version that I really do need (VDPAU and
HD-PVR support, for those who know about such things). And it's a pain
to upgrade, which of course needs to be done relatively often when you
are following the development version. So I do know how to do this and
what's involved. But I'm not convinced I've reached that point yet with
asterisk. It's true that I wouldn't need the development branch, but
it's still more hassle to apply security fixes when compiling from
source. And in this particular case, the conversion I am contemplating
is one that is going to have to be done eventually. When the developers
want to convert the config language, sooner or later they will stop
supporting the old stuff and it won't be possible to get the newest
supported features without converting. So I don't want to fall TOO far
behind. I'd rather convert now when I can do it in a leisurely manner,
before I absolutely HAVE to.

 I don't like being dependent on others 
 for my mission critical stuff.

Unless you are one of the developers, you're still dependent on others
to maintain the source. Granted it does eliminate the folks in the
middle doing the packaging. And in my case, nothing on my home server
could really be described as mission critical. It's closer to what the
MythTV folks call the WAF (Wife Approval Factor :-) She likes the
features that asterisk offers (separate voice mail boxes, phones in
every room, intercom ability to call between phones, having a backup
VOIP line out so we can make two calls at once, etc.) but none of that
is critical to her use of the phone. We can live without asterisk for
short periods of time, so it's not really mission critical. I have an
old-fashioned answering machine that will take messages if the computer
fails to answer (it's crashed, asterisk isn't running, we've powered it
off for a vacation, etc.) and I can patch the cordless phone base
station through to the wallplate (although then the VOIP phones don't
work).

As always, in the end, we make a tradeoff between security concerns,
reliability concerns, having the latest and greatest features, and ease
of long-term maintenance. What that means exactly is determined on a
case-by-case basis. In my case, I'll run the old system until I can find
a way to convert to the versions packaged with Fedora 11 relatively
quickly, to minimize my window of vulnerability and down time. Then for
the longer term I'll work on the AEL conversion. I was just hoping to
find something that would aid in that effort, but so far all I have seen
are suggestions on how to avoid having to do it at all.

--Greg

 


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Re: [asterisk-users] * 1.4 - 1.6, zaptel - dahdi

On Wed, 2009-08-19 at 14:27 -0700, Steve Edwards wrote:

  When the developers want to convert the config language, sooner or later 
  they will stop supporting the old stuff and it won't be possible to get 
  the newest supported features without converting. 

 I don't see that happening in my lifetime :)

Murphy's Law says it will happen at the time that is least convenient
for me :-)


 This appears to be somewhat inconsistent in your determination to update 
 to 1.6. Are packages for 1.4 available for F11?

Well, Axel did just say that he might be providing some through ATrpms,
so that is a possibility. The standard Fedora repositories have only
1.6. Kevin Fleming also implied that my extensions.conf file ought to
work with only minor changes with 1.6, so I'll probably see why that's
not working as a first step. That looks like the fastest way to get
going under F11.


 I'm not saying I could fix a security issue, but I can download, compile, 
 and install long before anybody could package it for me.

Yes, but that also requires you to follow some sort of devel list so
that you know when you need to do this. As opposed to yum/apt-get which
always automatically knows when there is an update.


 
 In my home, Myth is definitely mission critical :)

We've got a Comcrap DVR as a backup. That would at least catch the shows
where failure would drop the WAF the most. But for me, failures due to
MythTV or Asterisk software are rare. I have had some hardware failures
that put one or the other of these out of commission for weeks at a
time, and it does suck to go back to the 20th century :-)


 I was just hoping to 
  find something that would aid in that effort, but so far all I have seen 
  are suggestions on how to avoid having to do it at all.
 
 Not at all. I'm just saying if the available packages are doing it for 
 you, compiling the source is pretty trivial.

Right, but you're still telling me how I can avoid having to convert. I
still haven't seen anyone point to something that would HELP me convert.

--Greg



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Re: [asterisk-users] * 1.4 - 1.6, zaptel - dahdi

On Wed, 19 Aug 2009, Greg Woods wrote:

 When the developers want to convert the config language, sooner or later 
 they will stop supporting the old stuff and it won't be possible to get 
 the newest supported features without converting. So I don't want to 
 fall TOO far behind.

I don't see that happening in my lifetime :)

While extensions.conf sucks, I've used AEL enough to know it's basically a 
hack that usually works. In 1.2 inconsistencies abound and error checking 
is a joke. A misplaced semi-colon can cause large chunks of your dialplan 
to disappear without warning. Lua is supposed to be better so maybe your 
energies would be better focused there instead of AEL.

 I'd rather convert now when I can do it in a leisurely manner, before I 
 absolutely HAVE to.

This appears to be somewhat inconsistent in your determination to update 
to 1.6. Are packages for 1.4 available for F11?

 I don't like being dependent on others for my mission critical stuff.

 Unless you are one of the developers, you're still dependent on others 
 to maintain the source.

I'm not saying I could fix a security issue, but I can download, compile, 
and install long before anybody could package it for me.

 And in my case, nothing on my home server could really be described as 
 mission critical. It's closer to what the MythTV folks call the WAF 
 (Wife Approval Factor :-)

In my home, Myth is definitely mission critical :)

 As always, in the end, we make a tradeoff between security concerns, 
 reliability concerns, having the latest and greatest features, and ease 
 of long-term maintenance. What that means exactly is determined on a 
 case-by-case basis. In my case, I'll run the old system until I can find 
 a way to convert to the versions packaged with Fedora 11 relatively 
 quickly, to minimize my window of vulnerability and down time. Then for 
 the longer term I'll work on the AEL conversion. I was just hoping to 
 find something that would aid in that effort, but so far all I have seen 
 are suggestions on how to avoid having to do it at all.

Not at all. I'm just saying if the available packages are doing it for 
you, compiling the source is pretty trivial. If the packages catch up 
with F11 you can always install them then.

-- 
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] Dial plan sample for detecting Voice Mail

2009-08-19 Thread Danny Nicholas

Did you #include extensions_additional.conf in your extensions.conf file?
Verify this by doing dialplan show macro-screen from CLI.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharath B.
Reddy Bynagari
Sent: Wednesday, August 19, 2009 2:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dial plan sample for detecting Voice Mail

 

Hi, 

 

I am trying to implement a macro-screen mentioned at
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

 

I put the following code in my extensions_additional.conf

screen-from: You have a call;

screen-accept: Press 1 to accept this call or any other key to reject.;

 

[macro-screen]

exten = s,1,Wait(0.2)

exten = s,1,Playback(screen-from)

exten = s,1,Playback(${ARG1})

exten = s,1,Read(ACCEPT|screen-accept|1)

exten = s,1,GotoIf($[${ACCEPT} = 1 ] ?yes:no)

exten = s,1(yes),SetVar(MACRO_RESULT=CONTINUE)

exten = s,1(no),System(/bin/rm ${ARG1})

 

; end of [macro-screen]

 

[multi-dir-callback]

include = multi-dir-callback-custom

exten = _X.,1,Macro(screen,)

exten = _X.,1,Answer

exten = _X.,n,Playback(beep)

;exten =
_X.,n,DeadAGI(agi://127.0.0.1/callback_handler?num2=${EXTEN}callid=${CALL_I
D}num1=${num1}CallStatus=${DIALSTATUS}state=${STATE})a

exten =
_X.,n,DeadAGI(agi://127.0.0.1/callback_handler?num2=${EXTEN}callid=${CALL_I
D}num1=${num1}state=${STATE})

exten = _X.,n,Goto(${EXTEN},1)

exten =
hangup,1,DeadAGI(agi://127.0.0.1/callback_handler?num2=${EXTEN}callid=${CAL
L_ID}num1=${num1}state=${STATE})

 

; end of [multi-dir-callback]

 

It is not even recognizing the Screen macro? What I am I doing wrong?

 

Thanks

 

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Re: [asterisk-users] * 1.4 - 1.6, zaptel - dahdi

2009-08-19 Thread Axel Thimm

On Wed, Aug 19, 2009 at 03:48:22PM -0600, Greg Woods wrote:
  This appears to be somewhat inconsistent in your determination to
  update to 1.6. Are packages for 1.4 available for F11?
 
 Well, Axel did just say that he might be providing some through ATrpms,
 so that is a possibility.

Actually there already were asterisk-1.4.25.1 F11 packages at ATrpms
until last week

http://www.google.com/search?q=asterisk-1.4.25.1-78.fc11

(same for addons, sounds etc.) but since we wanted to make
asterisk-1.6 a first class citizen these packages needed to be renamed
to asterisk14 to not be automatically overridden. So the new packages
have been renamed to asterisk14-1.4.26.1-84,
asterisk-addons14-1.4.9-24, asterisk14-app_ldap-2.0rc1-5 and the
sounds packages have been made 1.4/1.6 compatible.
-- 
Axel.Thimm at ATrpms.net


pgpAGqlhb6Z6Q.pgp
Description: PGP signature
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Re: [asterisk-users] * 1.4 - 1.6, zaptel - dahdi

2009-08-19 Thread Ira

At 02:27 PM 8/19/2009, you wrote:
Not at all. I'm just saying if the available packages are doing it for
you, compiling the source is pretty trivial. If the packages catch up
with F11 you can always install them then.

My first experience with Linux and Asterisk was putting a whatever 
TrixBox was before it was TrixBox disk in a machine and saying 
install About 3 weeks later I removed all the TrixBox and I've been 
installing Asterisk from source ever since, including a clean install 
on a fresh Atom machine when I started with CentOS 5 and then 
compiled 1.6.2 and Dahdi, moved all my configuration and other 
necessary files across and 2 days later after figuring all the 
updates needed in extensions.conf  and sip.conf it was all working 
better than ever. Would have taken much less time if I was more 
careful reading the update docs and not mis-spelling something in a 
couple of places. The PBX was down for about 15 minutes while I moved 
the wires and TDM400 between machines and then it was up again.

All that just to say, as far as I can tell working with source is no 
harder than anything else. Wget into a folder, tar to unpack it, 
make, then make install, then reboot or just restart gracefully for 
Asterisk only changes. The only occasional gotcha is when Yum updates 
the kernel you have to remake Dahdi or it won't work after a reboot.

And everything I know about Linux fits on some notes on half a sheet of paper.

Ira


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Re: [asterisk-users] outbound calls not ringing

2009-08-19 Thread Duncan Turnbull

Generally with FreePBX the ring options are set in the General Options - 
you can set the Dial options which are normally tr, but I guess that 
isn't working for you.

The SIP files you could edit would have custom in their name, otherwise 
your changes will be overwritten when you reload freepbx

You could put this in sip_general_custom.conf which will be included

Cheers Duncan

John A. Sullivan III wrote:
 Oops! - You're using FreePBX - someone who knows more about FreePBX will
 have to help you as I don't.  May I also suggest that you bottom post in
 future responses rather than top post; that makes it a little easier to
 follow.  Good luck - John

 On Wed, 2009-08-19 at 16:59 +, Ott Rose wrote:
   
 here is my sip.conf. i don't see it.
 ;;
 ; Do NOT edit this file as it is auto-generated by FreePBX. All
 modifications to ;
 ; this file must be done via the web gui. There are alternative files
 to make;
 ; custom modifications, details at:
 http://freepbx.org/configuration_files   ;
 ;;
 ;

 [general]

 ; These files will all be included in the [general] context
 ;
 #include sip_general_additional.conf

 ;sip_general_custom.conf is the proper file location for placing any
 sip general
 ;options that you might need set. For example: enable and force the
 sip jitterbuffer.
 ;If these settings are desired they should be set the
 sip_general_custom.conf file.
 ;
 ; jbenable=yes
 ; jbforce=yes
 ;
 ;It is also the proper place to add the lines needed for sip nat'ing
 when going
 ;through a firewall.  For nat'ing you'd need to add the following
 lines:
 ; nat=yes , externip= , localhost= , and optionally fromdomain= .
 ;
 #include sip_general_custom.conf

 ;sip_nat.conf is here for legacy support reasons and for those that
 upgrade
 ;from previous versions.  If you have this file with lines in it
 please make
 ;sure they are not duplicated in sip_general_custom.conf, if so remove
 them
 ;from sip_nat.conf as sip_general_custom.conf will have precedence.
 #include sip_nat.conf

 ;sip_registrations_custom.conf is for any customizations you might
 need to do to
 ;the automatically generated registrations that FreePBX makes.
 ;
 #include sip_registrations_custom.conf
 #include sip_registrations.conf

 ; These files should all be expected to come after the [general]
 context
 ;
 #include sip_custom.conf
 #include sip_additional.conf

 ;sip_custom_post.conf If you have extra parameters that are needed for
 a
 ;extension to work to for example, those go here.  So you have
 extension
 ;1000 defined in your system you start by creating a line [1000](+) in
 this
 ;file.  Then on the next line add the extra parameter that is needed.
 ;When the sip.conf is loaded it will append your additions to the end
 of
 ;that extension.
 ;
 #include sip_custom_post.conf


 
 From: jsulli...@opensourcedevel.com
 To: asterisk-users@lists.digium.com
 Date: Wed, 19 Aug 2009 12:17:15 -0400
 Subject: Re: [asterisk-users] outbound calls not ringing

 sip.conf

 On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote:
   
 we are using Aastra 57i

 i don't see that setting. where is it at?

 
 From: jsulli...@opensourcedevel.com
 To: asterisk-users@lists.digium.com
 Date: Wed, 19 Aug 2009 11:07:21 -0400
 Subject: Re: [asterisk-users] outbound calls not ringing

 On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
   
 I put a post on here about my issues with outbound calls not
 
 ringing
 
 but i haven't resolved it. so i am trying again.

 When i dial any outside number i dont get a ring tone at all.
 
 when
 
 the
 
 person picks up and starts to talk i can hear them fine. it
 
 sounds
 
 great. How do I start to troubleshot this?
 
 snip
 What type of phones are giving you the problem? If I recall
   
 correctly,
 
 our SIP phones had this problem depending on how the destination
   
 handled
 
 signaling. We resolved it by adding progressinband=no (as
   
 opposed to
 
 the default never - at least I think it is the default) but this
 produces the problem of duplicate ring tones at times. Hope this
   
 helps
 
 - John
 -- 
 John A. Sullivan III
 Open Source Development Corporation
 +1 207-985-7880
 jsulli...@opensourcedevel.com

 http://www.spiritualoutreach.com
 Making Christianity intelligible to secular society


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[asterisk-users] PRI Connected to definity errors

2009-08-19 Thread C F

We have setup asterisk to handle our calls before between telco and an
Avaya definity. The PRI keeps locking up every so often.
In addition I keep getting this error when trying to call the avaya:
-- Channel 0/2, span 1 got hangup request, cause 102
-- Hungup 'Zap/2-1'
When that error happens I get a fast busy (congestion) tone.

Any one can point me in the right direction?

TIA

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Re: [asterisk-users] Newbie: How to copy track from CD for MOH without getting Junk at beginning of frame ...

2009-08-19 Thread Lee, John (Sydney)


 Yep, agreed.  
 Convert the file to the native codec(s) in which it will be played.

Alex, could you please elaborate on this?  I am no audio guy.
On Media player, I can rip it into mp3 or wav or windows media audio.
Which one should I use?

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Re: [asterisk-users] Newbie: How to copy track from CD for MOH without getting Junk at beginning of frame ...

2009-08-19 Thread Eric Fort

Probably none of the ones you list, though I believe wav files are
uncompressed.  Use SOX http://sox.sourceforge.net/ under Linux, Windows or
OSX and RIP/Convert the files to match the codec you are using for calls.
 If you are accepting calls that use the GSM codec then have a set of MOH
files encoded as .gsm, if you are accepting calls that use the g.723 codec
then encode your MOH files as g.723, if using speex, use speex, etc...  use
files already encoded in the formats in which you originate and terminate
calls.  That way the processor isn't repeating the process of transcoding on
every call!
Eric Fort
FortConsulting

On Wed, Aug 19, 2009 at 5:25 PM, Lee, John (Sydney)
john@compuware.comwrote:


  Yep, agreed.
  Convert the file to the native codec(s) in which it will be played.
 
 Alex, could you please elaborate on this?  I am no audio guy.
 On Media player, I can rip it into mp3 or wav or windows media audio.
 Which one should I use?

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Re: [asterisk-users] Newbie: How to copy track from CD for MOH without getting Junk at beginning of frame ...