[asterisk-users] Queue: Passing params to macros and gosubs

[Ed Greenberg]

I wrote:

Queue(ENGLISH,rh,,,20,,,log-answer)

but no variables are available in the subroutine context. I need to get 
${UNIQUEID} in there.


In a Dial command, I can write U(log-answer^${UNIQUEID}) but not in the 
gosub field of the Queue command.


Is it possible to pass variables from the dialplan to the subroutine 
context?


Thanks,

Ed Greenberg

Asterisk 1.8.10.1~dfsg-1ubuntu1 built by buildd @ yellow on a x86_64 
running Linux on 2012-04-24 12:47:04 UTC



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[asterisk-users] Rolm T1 not passing caller ID to asterisk

[Kohler, Ed]

We have a Rolm 9751 connecter to our asterisk box via a straight T1.  The Rolm 
cannot do PRI.  Has anyone figured out how to configure this link (probably on 
the Rolm side) to pass caller ID? Any Help or suggestions, aside from 
forklifting the Rolm, would be appreciated.

Thanks,

Edward Kohler
Network Technician
101Hines Hall
Siena College
515 Loudon Rd.
Loudonville, NY 12211
518-783-2391
Fax 518-783-2590
ekoh...@siena.edumailto:ekoh...@siena.edu

Siena College is a learning community advancing the ideals of a liberal arts 
education, rooted in its identity as a Franciscan and Catholic institution.

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please inform the sender by e-mail reply and destroy all copies of the original 
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[asterisk-users] Need provider recommendations for the UK

[Ed W]

Hi, I realise this is probably the wrong list for such a question, but I 
need a pointer to somewhere I can get some feedback on experience of 
(business class) voip providers for the UK?

Situation is that we are currently with Gradwell and use them for an 
inbound/outbound single line for a business and their quality has gone 
from excellent to abysmal in the last few weeks.  I'm sure they will 
work it out, but right now I just need a reliable provider that I can 
port a number to.  I'm not especially price sensitive, reliability is 
the main requirement.  IAX preferred, but not fussy.  Possibly multiple 
incoming numbers in future, single incoming at the moment - in general 
we rarely have more than 1 line in use, but occasionally hit 2-3 
simultaneous calls

Note, it's going to be important that we can port our number across from 
Gradwell...

Grateful if anyone can offer some really solid recommendations, or point 
me towards a more appropriate forum to request the same?

Thanks

Ed W

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[asterisk-users] Sangoma and BT single lines

[Ed W]

Hi, got a Sangoma A200 with a bunch of extension cards and having real 
problems getting it to deal with a normal single BT line

Symptoms are that incoming calls are fine.  Outgoing calls ring the far 
end, BUT asterisk never sees that the call is answered (ie no message in 
the logs files saying so), as a result the remove end can hear the PBX 
side talking, but there is no audio back from the remote side to us.  
When we hangup the log files show messages thave suggest it thinks the 
line is still ringing

Comparing with another line which works fine (this is a BT multi-line 
system with what they call PBX signalling on it) I see that as soon as 
the remote end answers then asterisk gets a log message stating the same 
and audio is fine on this line


Have now spent nearly 4 months trying to get the signalling sorted on 
this line.  Most recently we requested dual signalling on the line - 
the end result is now that outbound calls work and asterisk reports that 
the phone answers, however, when you hangup the call then asterisk 
obviously gets a bunch of extra line reversals and things there is an 
immediate incoming call on the back of that outgoing call...

Please - any suggestions on how to configure a Sangoma card for use with 
a normal BT single line?

Thanks

Ed W

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Re: [asterisk-users] Recommend Wireless IP Phone

[Ed W]

Wilton Helm wrote:
 Wi-Fi SIP phones aren't limited to hot spots.  I am in the process of
 setting up asterisk for SOHO.  At present I'm not even using VoIP
 trunking, only LAN to stns and I intend to use Wi-Fi instead of analog
 cordless phone.  I got the Engenius one, and it works, but I haven't
 played with it much.  I was disappointed that it only has a single
 line appearance, as part of my reason for going SIP was to allow the
 same features like say my 941.  I also got their 600 mw access point,
 but haven't had time to try it.  My goal is to cover out 3 acre
 property and the 1/2 mile road to the mailbox, including mountainous
 terrain.  Maybe I'll share more when I actually get it all put together.

I think you will get better range and longer battery life from a DECT
phone though... Probably more features and better quality also!

Any of the panasonic DECT phones seem to work very nicely (speaker
phone, features, R key works for call transfer, handset intercom etc -
mine lasts up to a week on a single charge and light use) and there are
several Siemens DECT phones with a builtin SIP gateway which avoids the
need for an external adaptor box

It's definitely possible to make wifi work for half a mile and you don't
even need a 600mw transmitter to do that - however, wifi is all about
receive strength, and so you are unlikely to get a significantly better
coverage with a high power hotspot which is suboptimally placed.  If you
do go that route then getting the antennas into a location where 90% of
the signal isn't already killed going through walls before it has to
travel some distance is the trick.  Probably also consider a repeater of
some sort rather than just one high power device

Good luck though!

Ed W
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Re: [asterisk-users] Polycom's lose BLF after Asterisk restart

[Ed W]

Mr Shunz wrote:
 Hi,

   
 We have an issue where Polycom's lose BLF functionality after a reboot.  The
 only way to fix it is to reboot the Polycoms.

 Anyone else have this issue?  We are using 1.4.18.

 If I run 'sip show subscriptions' all the subscriptions come back after the
 restart but the lights on the phones do not work.

 Any help would be appreciated.
 

 have the same issue with grandstreams and thomson (at least on st20XX)
 if we restart asterisk, phones don't renew subscriptions ...

 didn't search too hard, but i haven't found neither an option
 in asterisk nor on the phone to force resubscriptions ...
   

Can you reboot the phones remotely?  With snom it's quite easy to write
a script to reboot all phones - you can put that in your boot scripts

Ed
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Re: [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line

[Ed W]

robb wrote:
 I have a TDM400 working quite well, Digium dialled in and recompiled  
 chan_zap with some changes , to get BT Callerid working and  I have
 set hangup on polarity in the zaptel.conf which seems to work well

 this is a BT home line, not business, if you have a business line you
 should get the DCT set to 800ms and the disconnect clear should work


Would you be kind enough to share the changes you made to get callerID
working please?  Any chance of posting the relevant bits of your zap
config also?

My situation is that I have callerid working most of the time on a home
BT line.  Hangup is fairly reliably detected.  TDM400P

However, at a customers site on a bunch of business BT lines and the
same model of TDM400P we see unreliable hangups (not frequent, but
occasional times that lines are getting stuck off hook). Also callerId
is working about 50-60% of the time and when it doesn't work (or
genuinely that the callerId is witheld) there is a long pause for about
2-3 rings before Asterisk answers the zap line.  It would be desirable
to limit this pause because it makes it look like they are being slow to
answer all the calls!

Just wondering what changes you made?

Also, anyone understand why DCT is different between home and business
lines?  Can the Zap code be changed to avoid needing something tweaking
on the exchange?

Thanks

Ed W
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[asterisk-users] Polycom 330 not dialing 4 digit extensions beginning with 11xx

[Ed DeHart]

I have four Polycom 330 phones connected to an asterisk system.  There  
are other VoIP phones connected too.  All of the extensions are four  
digits beginning with 11.  From any of the phones, except the Polycom,  
picking up the handset to call extension 1103 for example works fine.   
With the Polycom 330, as I press the second 1 of 1103 it stops taking  
input and gives me an error.


I tried creating four digit extensions on another asterisk system  
where I have Polycom 501 phones connected and they too will not let me  
dial 1103.  I can dial 1203 or any other combinations of number, just  
not an extension that begins with 11.


Any suggestions?

-Ed


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[asterisk-users] modules/cdr_odbc.so

[Ed Nuñez]

Can anyone tell me if I can load the modules/cdr_odbc.so module without
having to re compile my 1.4.20 production Asterisk?

 

 

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Re: [asterisk-users] Asterisk Openfire Asterisk-IM Plugin Performance Observation

[Ed W]

 Is it better for production to run Openfire on a separate server than the PBX?
   


Since discovering linux vservers I put every service into its own 
install.  Each install can be very lightweight and vservers only add 
about 1MB to ram usage (I don't run a separate init process), so very 
lightweight. 

The advantage is that it's super simple to backup each server and you 
can test upgrades by simply copying the image, fire up a new instance, 
test your upgrade, then burn it down again...  Piece of cake to shuffle 
services between real machines also (preserving IP addresses also if 
that's required).  Backups can be done very easily (make the /vserver 
dir an LVM disk)

Good luck

Ed W

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Re: [asterisk-users] Asterisk with Nextone using H323

[Ed Nuñez]

Any reason in particular why you don't use SIP between your Asterisk and
NexTone?  This is how I have ours connected and it works well.  The only
issue I've experienced is that some of the carriers that only support g729
AB have trouble with the dtmf tones from g729A, but this is not SIP
specific.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guillermo
Salas M.
Sent: Tuesday, June 24, 2008 11:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk with Nextone using H323

El mar, 24-06-2008 a las 12:20 -0300, Everton Goularth escribió:
 I`m using chan_ooh323 in my asterisk server. This is my ooh323.conf:


Have you tried with chan_h323.so?

I've one gateways that uses h.323 and works only with chan_h323.so .

Regards,

-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net
SIP  : [EMAIL PROTECTED]
FWD  : 558563
USA  : 1 360 968 1701

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting


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Re: [asterisk-users] Problems configuring a PRI...

[Ed Nunez]

Here is my configuration with Global Crossing.  Hope this helps.

 

Zaptel.co

 

 

# Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 B8ZS/ESF ClockSource

span=1,1,0,esf,b8zs

# termtype: te

bchan=1-23

dchan=24

 

 

Zapata.conf

 

mode=mixed

 

signalling=pri_cpe

context=incoming-att

group=1

channel = 1-23

 

 

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher
Hoff
Sent: Tuesday, June 10, 2008 5:23 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problems configuring a PRI...

 

I'm trying to get a Qwest PRI configured and working with my lab Asterisk
server. They said that the switchtype is 5ess and the signaling is pri_cpe.
My entries into zaptel.conf are: 

span=1,0,0,esf,b8zs 
bchan=1-23 
dchan=24 
loadzone = us 
defaultzone=us 
channels=1-23 


And my entries in zapata.conf are: 

language=en 
context=telco-incoming 
switchtype=5ess 
signalling=pri_cpe 
rxwink=300 
usecallerid=yes 
hidecallerid=no 
callwaiting=no 
usecallingpres=yes 
callwaitingcallerid=yes 
threewaycalling=yes 
transfer=yes 
canpark=yes 
cancallforward=yes 
callreturn=yes 
echocancel=yes 
echocancelwhenbridged=yes 
rxgain=0.0 
txgain=0.0 
callgroup=1 
pickupgroup=1 
immediate=no 
group = 1 
switchtype = 5ess 
signalling = pri_cpe 
group = 1 
channel = 1-23 

I'm not able to make/receive calls, and the error I'm receiving is: 

[Jun 10 11:32:37] WARNING[31768]: chan_zap.c:2393 pri_find_dchan: No
D-channels available! Using Primary channel 24 as D-channel anyway! 
== Primary D-Channel on span 1 down 

Qwest says that the PRI is fine. I have a green light on the PRI card. 

Help!

 

___

 

Chris Hoff

Telecommunications Administrator

SEI LLC

Voice  +1 701 298 8865 Ext 2189

Mobile +1 701 361 5976

Fax +1 701 298 8860

Email [EMAIL PROTECTED]

 

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Re: [asterisk-users] MiixMonitor filename for queue calls.

[Ed Nunez]

I am using the following entry to define my filename

exten =
8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH},G
MT+8,%C%y%m%d%H%M)})

This will display  QUEUE-NOC (Caller ID number) (and time stamp)

I would also like to add the answering Agent ID to the file name.  Any idea
what this variable name is?

Thank you



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Smith
Sent: Saturday, June 07, 2008 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MiixMonitor filename for queue calls.

Hi Ed,

Glad to see you figured out your problem. I'm not sure what the 
differences are between your config and mine, but maybe this will help 
others too.

I add and remove my agents from the queue. So my agents.conf file is 
just the presistentagens=yes. Also I just run the command in the dial 
plan like below which saved mine items just fine. No configurations in 
the queue.conf file for the monitor type.

exten = 852,n,MixMonitor(/mercury/recordings/holding/${UNIQUEID}.gsm|b|)

 From there, in the hangup extension, I run a php script to take the CDR 
record and the file (rename it of course to 
queue-extension-callerid-callid-timestamp.gsm), and place it into the 
agents folder and the database for our agents/supervisors to review or 
download them.

Kevin


Ed Nunez wrote:

 Can anyone give me input on the following issue?

  

 I have a queue with MixMonitor enabled. 

 This is also enabled in agents.conf.  

 On my extensions.conf, I am setting the monitor filename as fillows, 
 although I see the filename as desired in the console as I make my 
 test call, the system is only using the default file name to save the 
 mixmonitor file   (agented + uniqueID)

  

 Agents.conf

  

 [general]

 persistentagents=yes

  

 [agents]

 maxlogintries=3

 musiconhold = default

 updatecdr=yes

 recordagentcalls=yes

 recordformat=wav49

 urlprefix=http://pbx.netoneint.com/calls/

 savecallsin=/var/calls

  

 agent = 1000,1000,Ed Test1

 agent = 1001,1001,Ed Test2

  

  

 queues.conf

  

 [noi-noc]  

 monitor-format = wav49  

 monitor-type = MixMonitor  

  

 member = Agent/1001

 member = Agent/1000

  

  

 extensions.conf

  

 exten = 
 8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH)

 exten = 8484,1,answer

 exten = 8484,2,Queue(noi-noc)

  

  

 Console output

  

 -- Executing [EMAIL PROTECTED]:1] Set(Zap/1-1, 
 MONITOR_FILENAME=QUEUE-NOC-4073844200-Fri Jun  6 15:06:38 2008) in 
 new stack

 -- Executing [EMAIL PROTECTED]:2] Queue(Zap/1-1, noi-noc) in 
 new stack

 -- Started music on hold, class 'default', on Zap/1-1

 -- outgoing agentcall, to agent '1001', on 
 'Local/[EMAIL PROTECTED],1'

 -- Called Agent/1001

 -- Executing [EMAIL PROTECTED]:1] 
 Dial(Local/[EMAIL PROTECTED],2, SIP/1658) in new stack

 -- Called 1658

 -- SIP/1658-087e7610 is ringing

 -- Agent/1001 is ringing

 -- SIP/1658-087e7610 answered Local/[EMAIL PROTECTED],2

 -- Agent/1001 answered Zap/1-1

 -- Stopped music on hold on Zap/1-1

 [Jun  6 15:06:40] WARNING[3976]: app_queue.c:3014 try_calling: The 
 device state of this queue member, Agent/1001, is still 'Not in Use' 
 when it probably should not be! Please check UPGRADE.txt for correct 
 configuration settings.

   == Begin MixMonitor Recording Zap/1-1

   == Spawn extension (numberplan-custom-3, 1658, 1) exited non-zero on 
 'Local/[EMAIL PROTECTED],2'

   == Spawn extension (incoming-att, 8484, 2) exited non-zero on 'Zap/1-1'

 -- Hungup 'Zap/1-1'

   == End MixMonitor Recording Zap/1-1

  

  

  

  

  

 

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-- 
Kevin Smith

--- 
Mercury Network
Technical Support
Phone: 989.837.3790
Toll Free: 888.866.4638
www.mercury.net


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[asterisk-users] MiixMonitor filename for queue calls.

[Ed Nunez]

Can anyone give me input on the following issue?

 

I have a queue with MixMonitor enabled.  

This is also enabled in agents.conf.   

On my extensions.conf, I am setting the monitor filename as fillows,
although I see the filename as desired in the console as I make my test
call, the system is only using the default file name to save the mixmonitor
file   (agented + uniqueID)

 

Agents.conf

 

[general]

persistentagents=yes

 

[agents]

maxlogintries=3

musiconhold = default

updatecdr=yes

recordagentcalls=yes

recordformat=wav49

urlprefix=http://pbx.netoneint.com/calls/

savecallsin=/var/calls

 

agent = 1000,1000,Ed Test1

agent = 1001,1001,Ed Test2

 

 

queues.conf

 

[noi-noc]   

monitor-format = wav49   

monitor-type = MixMonitor   

 

member = Agent/1001

member = Agent/1000

 

 

extensions.conf

 

exten =
8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH)

exten = 8484,1,answer

exten = 8484,2,Queue(noi-noc)

 

 

Console output

 

-- Executing [EMAIL PROTECTED]:1] Set(Zap/1-1,
MONITOR_FILENAME=QUEUE-NOC-4073844200-Fri Jun  6 15:06:38 2008) in new
stack

-- Executing [EMAIL PROTECTED]:2] Queue(Zap/1-1, noi-noc) in new
stack

-- Started music on hold, class 'default', on Zap/1-1

-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'

-- Called Agent/1001

-- Executing [EMAIL PROTECTED]:1]
Dial(Local/[EMAIL PROTECTED],2, SIP/1658) in new stack

-- Called 1658

-- SIP/1658-087e7610 is ringing

-- Agent/1001 is ringing

-- SIP/1658-087e7610 answered Local/[EMAIL PROTECTED],2

-- Agent/1001 answered Zap/1-1

-- Stopped music on hold on Zap/1-1

[Jun  6 15:06:40] WARNING[3976]: app_queue.c:3014 try_calling: The device
state of this queue member, Agent/1001, is still 'Not in Use' when it
probably should not be! Please check UPGRADE.txt for correct configuration
settings.

  == Begin MixMonitor Recording Zap/1-1

  == Spawn extension (numberplan-custom-3, 1658, 1) exited non-zero on
'Local/[EMAIL PROTECTED],2'

  == Spawn extension (incoming-att, 8484, 2) exited non-zero on 'Zap/1-1'

-- Hungup 'Zap/1-1'

  == End MixMonitor Recording Zap/1-1

 

 

 

 

 

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Re: [asterisk-users] MiixMonitor filename for queue calls.

[Ed Nunez]

I have found the answer to my question.

 

For anyone intrested, the system was saving the file with my desired
filename in the default /monitor sub-directory and was also saving a second
copy of the file in the /calls sub-directory.  I commented out the 

 

;recordagentcalls=yes

 

Line in agents.con and this stoped the system from recording the seconfd
file in the /calls sub-directory.

 

Hope this information may be usefull to someone.

 

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nunez
Sent: Friday, June 06, 2008 3:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion';
[EMAIL PROTECTED]
Subject: [asterisk-users] MiixMonitor filename for queue calls.

 

Can anyone give me input on the following issue?

 

I have a queue with MixMonitor enabled.  

This is also enabled in agents.conf.   

On my extensions.conf, I am setting the monitor filename as fillows,
although I see the filename as desired in the console as I make my test
call, the system is only using the default file name to save the mixmonitor
file   (agented + uniqueID)

 

Agents.conf

 

[general]

persistentagents=yes

 

[agents]

maxlogintries=3

musiconhold = default

updatecdr=yes

recordagentcalls=yes

recordformat=wav49

urlprefix=http://pbx.netoneint.com/calls/

savecallsin=/var/calls

 

agent = 1000,1000,Ed Test1

agent = 1001,1001,Ed Test2

 

 

queues.conf

 

[noi-noc]   

monitor-format = wav49   

monitor-type = MixMonitor   

 

member = Agent/1001

member = Agent/1000

 

 

extensions.conf

 

exten =
8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH)

exten = 8484,1,answer

exten = 8484,2,Queue(noi-noc)

 

 

Console output

 

-- Executing [EMAIL PROTECTED]:1] Set(Zap/1-1,
MONITOR_FILENAME=QUEUE-NOC-4073844200-Fri Jun  6 15:06:38 2008) in new
stack

-- Executing [EMAIL PROTECTED]:2] Queue(Zap/1-1, noi-noc) in new
stack

-- Started music on hold, class 'default', on Zap/1-1

-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'

-- Called Agent/1001

-- Executing [EMAIL PROTECTED]:1]
Dial(Local/[EMAIL PROTECTED],2, SIP/1658) in new stack

-- Called 1658

-- SIP/1658-087e7610 is ringing

-- Agent/1001 is ringing

-- SIP/1658-087e7610 answered Local/[EMAIL PROTECTED],2

-- Agent/1001 answered Zap/1-1

-- Stopped music on hold on Zap/1-1

[Jun  6 15:06:40] WARNING[3976]: app_queue.c:3014 try_calling: The device
state of this queue member, Agent/1001, is still 'Not in Use' when it
probably should not be! Please check UPGRADE.txt for correct configuration
settings.

  == Begin MixMonitor Recording Zap/1-1

  == Spawn extension (numberplan-custom-3, 1658, 1) exited non-zero on
'Local/[EMAIL PROTECTED],2'

  == Spawn extension (incoming-att, 8484, 2) exited non-zero on 'Zap/1-1'

-- Hungup 'Zap/1-1'

  == End MixMonitor Recording Zap/1-1

 

 

 

 

 

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Re: [asterisk-users] sip show peers

[Ed Nunez]

Anyone has any good ideas on how to parse the CDR events and QUEUEs log
events from AMI connection?

Thank you



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: Friday, May 02, 2008 3:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip show peers

On Friday 02 May 2008 12:35:48 Philipp Kempgen wrote:
 Yeah, sometimes it would be helpful if asterisk -rx 'command' didn't
 pretty print but instead fall back to an easily parseable output
 format (like TSV with cslashes) if stdout isn't connected to a tty
 (isatty()).

The CLI is intended to be used by a human.  If you want machine parseable
output, I would suggest using AMI, as that's what it's meant for.

-- 
Tilghman

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Re: [asterisk-users] Phantom Rings

[Ed W]

 I'm fairly certain the problem is with the phone line.  I have all 
 callerID settings disabled as the Telco is unable to provide it along 
 with our rollover line setup due to limitations in their antiquated 
 switch.  The CLI and Logs all plainly show the calls as if they were 
 normal calls with the exception of a message about Failed to write 
 frame and no DTMF attempts, then the call is routed into the operator 
 queue.  The calls always came in on Zap1-1 so I tried swapping the 2 
 lines to see if it stayed on port 1 or if the phantom followed the 
 line.  As expected, the phantom rings followed the line and began 
 showing up on Zap2-1.  So it pretty has to be something in the telco, 
 but I'm not sure what.  Putting WaitForRing(3) before the Answer 
 command in my IVR menu eliminates most of them, but sometimes more of 
 them slip through.
   


I get a similar problem with a domestic analogue line in the UK.  I 
*speculate* that there is a short half ring being sent for some reason 
(line test or similar), but my card (Digium) seems to need about 5 
seconds to detect hangup on the remote end, so I get a phantom 2 rings 
at my end and then it stops...

No solution, but thought it might give you something to consider...

Ed W

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[asterisk-users] Avaya 4610sw

[Ed Nuñez]

I have loaded the SIP firmware for an Avaya 4610sw IP phone and have
successfully registered it to Asterisk BE and Asterisk 1.4.18.  I am however
experiencing two issues that I am hoping someone has already overcome.

 

The first one is that the phone looses it’s registration from Asterisk every
now and then.  I found a tip that may work and am now testing which is to
comment the line mailbox=(extension) from it’s sip.conf configuration.

 

The second issue is that if I make a call and place the call on hold, when I
pick up the line to resume the call, I hear no audio on neither the
originating or destination phone.  If I place the call on hold again, I can
hear music on hold on the destination phone.

 



 

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[asterisk-users] Avaya 4610sw

[Ed Nuñez]

I have an Avaya 4610SW IP phone which I have upgraded to SIP firmware.

 

I have successfully registered this phone to Asterisk BE as well as Asterisk 
1.4.18 

 

Almost everything is working well. Except for two issues.

 

One of the problems is that the phone looses registration every now and then ad 
I have to re register.   I have found a tip for this which I am testing if it 
will work, which is to comment out line mailbox=(extension).

 

The second and more serious issue is that when I place someone on hold, I am 
not able to resume the call.  I can hear the music on hold on the destination 
phone and the music on hold stops when I try to pick up the line from the Avaya 
again, but there is no audio between the two phones.  I can hang up and call 
again and I can hear both ways just fine.

 

I would appreciate any input on this.

 

Thank you

 

Ed Nu

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Re: [asterisk-users] How to hookup to cell phone for outbound calls?

[Ed W]

Sam Tam wrote:
 Well I think you need a GSM Gateway
 You can find some info on cyber-telecom.net
 For a cheap option you can try a CT-G1000 or CT-G2000 and then plug it in a
 X100P or something similar then it would be very economical.
   

Yep, this is the kind of thing I am after, except my hardware PBX has 
limited connectivity and ideally I want a USB or ethernet hookup to the 
box..?

The scenario is basically a small commercial PBX (small form factor) 
which can be supplied with IP phones and will talk out via a cell phone 
channel (or via a satellite phone if that's the only option available, 
but this is out of scope of this question).  So basically I want to 
figure out some options to hookup a GSM cellphone channel to a small 
form factor asterisk PBX which has limited expansion options (ethernet, 
USB and mini-PCI - although prefer to use the later for a wifi card...)

I only need a single channel of GSM right now (and a single SIM)

Any thoughts?  Remember this needs to be production quality and priced 
sensible for a commodity market

Thanks for pointers to hardware

Ed W

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[asterisk-users] Matching + characters in dial plan

[Ed W]

Can someone please explain how to match a + character in a dial plan (so 
that I can swap it for the 00 country escape code).

In Europe at least the + is a common shortcut for the international 
prefix (which is 00 in my country).  However, my trunk chokes on the + 
character and all my speed-dials are setup with a + at the start of 
them... Trying to fix the phone rather than the addressbook...

Thanks

Ed W

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[asterisk-users] How to hookup to cell phone for outbound calls?

[Ed W]

Hi

I need a small PBX for use on the move.  This means that outbound calls 
will need to be made over the cell phone network.

Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot 
then what hardware options do I have to get an outbound cellular 
channel?  Options need to be rock solid, so no bluetooth to a cell phone 
kind of solutions need apply. 

Can any of the 3G usb devices out there offer outbound analogue calls 
(ie other than via voip)?

Cheers

Ed W

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Re: [asterisk-users] problems with zaptel and Udev

[Ed Nunez]

I had the same issue and updated my Zaptel drivers to version 1.4.17 and
it's rebooting fine now.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of robert
boardman
Sent: Sunday, January 13, 2008 12:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] problems with zaptel and Udev

Hi

I have had a Centos 5 installed with asterisk and zaptel for a couple of
weeks, I had to reboot eh machine today, and when it rebooted it got
stuck at Starting udev if I remove thew tdm400 it boots OK, but no zaptel

has anyone seen this , and can offer any advice?

Thanks Robb


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[asterisk-users] Asterisk ports and CentOS firewall

[Ed Nunez]

If I enable the firewall on my Server, which ports should I open for
Asterisk to work properly.  Is it enough to just open the SIP ports?

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[asterisk-users] Disable VAD on Polycom 330 or 301

[Ed Nuñez]

Does anyone know an easy way to disable VAD on Polycom Phones?

Thank you




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Re: [asterisk-users] Poll: Asterisk IMAP feedback (was: Is anyonesuccessfully using IMAP storage)

[Ed W]

Anthony Rodgers wrote:
We tried with MS Exchange but couldn't get it to work (MS Exchange 
doesn't support a master account).

It used to?  Not out the box though...

Ed W
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[asterisk-users] WARNING[26913]: channel.c:786 channel_find_locked: Avoided deadlock for '0x82d9668', 10 retries!

[Ed Nuñez]

Is anyone familiar with this error message?

WARNING[26913]: channel.c:786 channel_find_locked: Avoided deadlock for
'0x82d9668', 10 retries!

Why does it happen, and how can I prevent from happening.




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Re: [asterisk-users] ChanSpy issue

[Ed Nuñez]

Good point, but the deal is that I have a remote call center with their own
Nortel PBX.  I get these calls from my DID provided via Zap and I send them
VoIP to the gateway connected to the Nortel PBX.  This is what I refer to my
SIP trunk.  When I specify Sip/SIPTRUNK(SIPTRUNK) is the name of the
trunk.  Asterisk only monitors one call at a time in the whole trunk, and
you can Cycle through the calls by pressing *. 




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent: Wednesday, September 26, 2007 8:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy issue

I am not an expert on chanspy, but it seems to me spying on the trunk
would not work very well, would not you hear multiple conversations
mixed if more than one extension were calling?  Seems best to me to
spy on an extension.  YOu also can do a show channels to see who is
talking to whom.

on Wednesday 09/26/2007 Wai Wu([EMAIL PROTECTED]) wrote
  The parameter to Chanspy should be the whole or part of the channel name.
I do not understand what you mean by sip trunk. It make perfect sense that
you can hear both streams of voice when you use the phone's extension as
Asterisk usually uses SIP/extension+xxx as the channel name of the call.
  
  
  -Original Message-
  From: [EMAIL PROTECTED] on behalf of Ed Nuñez
  Sent: Wed 9/26/2007 4:48 PM
  To: [EMAIL PROTECTED]
  Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: Re: [asterisk-users] ChanSpy issue
   
   
  
  Hello list
  
   
  
  I am having an issue with Chanspy/SIP that I'm hoping someone has come
  across and resolved in the past.
  
   
  
  I am sending calls that come in TDM through T1 ZAP channels and go out to
a
  SIP trunk.
  
   
  
  If I spy on the SIP channel, I can hear the person on the SIP side of the
  call just fine, but the person on the ZAP channel fades in and out.
  
  If I spy on the ZAP channel, and can hear both sides just fine, but I
don't
  know who I am spying on since I have other calls coming in on the same
T1.
  
   
  
  If I spy on a SIP extension instead of a SIP trunk, I hear both sides
just
  fine.
  
   
  
  I am using a recent version of Asterisk 1.2 and I am using g729 licenses.
  
   
  
  This is the command I am using to spy.
  
   
  
  exten = 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4))
  
   
  
   
  
  
  
   
  
  
  !DOCTYPE HTML PUBLIC -//W3C//DTD HTML 3.2//EN
  HTML
  HEAD
  META HTTP-EQUIV=Content-Type CONTENT=text/html; charset=iso-8859-1
  META NAME=Generator CONTENT=MS Exchange Server version 6.5.7638.1
  TITLERE: [asterisk-users] ChanSpy issue/TITLE
  /HEAD
  BODY
  !-- Converted from text/plain format --
  
  PFONT SIZE=2The parameter to Chanspy should be the whole or part of
the channel name. I do not understand what you mean by quot;sip
trunkquot;. It make perfect sense that you can hear both streams of voice
when you use the phone's extension as Asterisk usually uses
quot;SIP/extension+xxxquot; as the channel name of the call.BR
  BR
  BR
  -Original Message-BR
  From: [EMAIL PROTECTED] on behalf of Ed NuñezBR
  Sent: Wed 9/26/2007 4:48 PMBR
  To: [EMAIL PROTECTED]BR
  Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'BR
  Subject: Re: [asterisk-users] ChanSpy issueBR
  BR
  BR
  BR
  Hello listBR
  BR
  BR
  BR
  I am having an issue with Chanspy/SIP that I'm hoping someone has
comeBR
  across and resolved in the past.BR
  BR
  BR
  BR
  I am sending calls that come in TDM through T1 ZAP channels and go out to
aBR
  SIP trunk.BR
  BR
  BR
  BR
  If I spy on the SIP channel, I can hear the person on the SIP side of
theBR
  call just fine, but the person on the ZAP channel fades in and out.BR
  BR
  If I spy on the ZAP channel, and can hear both sides just fine, but I
don'tBR
  know who I am spying on since I have other calls coming in on the same
T1.BR
  BR
  BR
  BR
  If I spy on a SIP extension instead of a SIP trunk, I hear both sides
justBR
  fine.BR
  BR
  BR
  BR
  I am using a recent version of Asterisk 1.2 and I am using g729
licenses.BR
  BR
  BR
  BR
  This is the command I am using to spy.BR
  BR
  BR
  BR
  exten =gt; 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4))BR
  BR
  BR
  BR
  BR
  BR
  BR
  BR
  BR
  BR
  BR
  /FONT
  /P
  
  /BODY
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 [EMAIL PROTECTED]

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Re: [asterisk-users] Ast_log

[Ed Nuñez]

The Asterisk log file is normally located in 
 /var/log/asterisk
But you may want to read your asterisk.conf file to make sure the path in
which your system store it.

You will see something like this

astlogdir = /var/log/asterisk



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Wednesday, September 26, 2007 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Ast_log


Hi all,
Anyone know where the asterisk log file is stored? I have some failed
calls into my Asterisk box, and I just want to find out why those calls
failed. Thnx.

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Re: [asterisk-users] ChanSpy issue

[Ed Nuñez]

 

Hello list

 

I am having an issue with Chanspy/SIP that I’m hoping someone has come
across and resolved in the past.

 

I am sending calls that come in TDM through T1 ZAP channels and go out to a
SIP trunk.

 

If I spy on the SIP channel, I can hear the person on the SIP side of the
call just fine, but the person on the ZAP channel fades in and out.

If I spy on the ZAP channel, and can hear both sides just fine, but I don’t
know who I am spying on since I have other calls coming in on the same T1.

 

If I spy on a SIP extension instead of a SIP trunk, I hear both sides just
fine.

 

I am using a recent version of Asterisk 1.2 and I am using g729 licenses.

 

This is the command I am using to spy.

 

exten = 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4))

 

 



 

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Re: [asterisk-users] Call Center SoftPhone with Auto Answer

[Ed Pastore]

On Sep 17, 2007, at 11:11 AM, Joao Pereira wrote:

 But still, the user can choose not to answer the phone.
 I want to force the users to accept the calls.

Wouldn't that be the same as paging/intercom, then?
http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom

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Re: [asterisk-users] Flash IDE

[Ed W]

Juan Sandro wrote:
 Hi

 We have a number offices accommodating 4-6 people each hence it is very
 important for PBX to be fanless and silent. We have been looking at using
 IDE flash disks also called DOM. The performance tests we have done so far
 satisfy our requirements, however we are concerned with DOM durability.

 We have installed debian and vanilla asterisk on 1GB DOM. All seems to work
 fine at the moment however will DOM last? How long it will last? Is anyone
 able to share similar experience? Any other information/tips?
   

I worried a lot about the same, in the end I went for a small laptop 
drive for safety (it's inaudible)

However, this came up on slashdot recently and if you search around the 
logic seems to be that:

- Flash rewrites quite a few times
- The good stuff has wear levelling so that most roughly speaking the 
whole thing should work until it suddenly all fails
- Given a big enough drive with a fair bit of free space then you should 
find it hard to wear it out in less than quite a few years even if you 
are hitting it quite hard (probably multiples of this).  Simply do the 
maths to get the rough life

So basically it seems that given a large enough flash drive with decent 
wear levelling the lifetime should be completely ample...

...Thats the theory anyway.

I feel quite bullish about the whole thing, but I think I would avoid 
the *really* discounted cheapo flash drives since they may not have the 
correct wear levelling.  Decent brand names should be fine though (and 
you can google for details on their specs)

Ed W

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[asterisk-users] Queue stats

[Ed Nuñez]

Can anyone recommend a good commercial solution for queue statistics?  




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Re: [asterisk-users] Tuning a ZyWALL for Asterisk

[Ed Pastore]

Anyone?


On Aug 24, 2007, at 3:35 PM, Ed Pastore wrote:

 I understand this question is over-broad, but hopefully you can have
 patience with a newbie and toss me a bone...

 I am in the testing stage of deploying Asterisk. I have successfully
 configured it to work behind the NAT of my ZyXEL ZyWALL 35 firewall.
 However, I think there is a lot of tuning I can do to get better
 reliability, bandwidth management, and maybe QoS from the firewall. I
 have some clues as to how to do some of this, but both telephony and
 routing are not strong points for me (I mostly work on systems,
 servers, and LANs).

 Is there any sort of reference material that will guide me in setting
 up my ZyWALL for VoIP? I don't see much help from ZyXEL, and I only
 see scattered posts around the net, but I know a lot of people are
 using ZyWALLs with Asterisk.

 If there isn't a reference, then can anyone chime in with some
 particulars on what you've done?

 Any hints would be greatly appreciated. Thanks!

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[asterisk-users] Tuning a ZyWALL for Asterisk

[Ed Pastore]

I understand this question is over-broad, but hopefully you can have  
patience with a newbie and toss me a bone...

I am in the testing stage of deploying Asterisk. I have successfully  
configured it to work behind the NAT of my ZyXEL ZyWALL 35 firewall.  
However, I think there is a lot of tuning I can do to get better  
reliability, bandwidth management, and maybe QoS from the firewall. I  
have some clues as to how to do some of this, but both telephony and  
routing are not strong points for me (I mostly work on systems,  
servers, and LANs).

Is there any sort of reference material that will guide me in setting  
up my ZyWALL for VoIP? I don't see much help from ZyXEL, and I only  
see scattered posts around the net, but I know a lot of people are  
using ZyWALLs with Asterisk.

If there isn't a reference, then can anyone chime in with some  
particulars on what you've done?

Any hints would be greatly appreciated. Thanks!

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[asterisk-users] Stable-Stable Asterisk

[Ed Pastore]

Hi, folks.

I've been on the Asterisk Announce list for a while now, and it seems  
to me that the release versions of Asterisk are a bit bleeding-edge.  
They qualify as stable, but I wouldn't call them production stable  
since half the time a new one comes out, a fix for it comes out the  
next day.

So... that said, what's a good version to linger on? I don't *need*  
anything particularly fancy, feature-wise, but would like to keep it  
as secure and stable as possible. And I certainly don't mind fancy  
features. :)

Also (please forgive a newbie), how can I tell what version of  
Asterisk I'm running? My current install was set up by a vendor and  
I'm still learning the ropes. Where's the best place to look to find  
the build number?

I do know that I'm running some version of 1.2, and am also not sure  
if I should stay there, or move up to 1.4.

Thanks!

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[asterisk-users] Music on hold 1.2

[Ed Nuñez]

What is a good solution for playing music on hold on the 1.2 branch.  I do not 
want to use mpg123 because last time I used it in a production server it caused 
many problems.   The MPG123 process was taking about 60% of my Xeon CPU.

 

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Re: [asterisk-users] kore dump

[Ed Nuñez]

For anyone interested on the crashes I was experiencing when using ChanSpy
from SIP extension to SIP extensions with the group option.  For the last
couple of days, I’ve been monitoring from Zap extensions to SIP extensions,
and the system has not crashed once.  The problem only happens when I spy
from SIP.

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vadim
Berezniker
Sent: Tuesday, June 26, 2007 2:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] kore dump

 

use the safe_asterisk script

 

it will restart asterisk if it crashes and it enables core dumps (your core
size limit is probably set to 0 when you start asterisk).

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Tuesday, June 26, 2007 2:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: [asterisk-users] kore dump

 

I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server.

 

My PBX has experienced several core dumps the last couple of days and I am
not sure if this is what’s causing it, but it always seems to happen when a
particular extension on a grandstream phone uses ChanSpy SIP group.

 

I have not been able to locate where the core dump file is being saved.   I
can’t find it in my TMP directory.

 

I would also like to know if Asterisk can be setup to automatically re start
if there is a core dump.  I was thinking of setting up a cron job to launch
Asterisk every minute.  If it’s running, no harm done, and if it crashes,
the cron job will make sure that it’s started every 60 seconds.

 

Any suggestions?

 

 

Thank you

 

Ed Nuñez

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[asterisk-users] network routing

[Ed Nuñez]

I have installed the Asterisk BE B.2.2 image file in a new server.  I need to 
make network routing changes.  However in their version of rPath (pound key) 
Digium has removed the netconfig command.  I am able to manually add the route 
with 

 

Route add default gw xxx.xxx.xxx.xxx however when I reboot I lose my routing.  
Does anyone know which conf file I need to edit in order to make this routing 
change permanent?

 

Thank you

 

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Re: [asterisk-users] network routing

[Ed Nuñez]

This allows me to edit the IP Address of the NIC card, but not edit my IP
routing.

 

Thanks 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ~Russell
Sent: Thursday, June 28, 2007 2:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] network routing

 

try to edit /etc/sysconfig/network-scripts/ifcfg-eth0  if u have eth0   

if not try ifcfg-eth1 for eth1




On 6/29/07, Ed Nuñez  [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 
wrote:

I have installed the Asterisk BE B.2.2 image file in a new server.  I need
to make network routing changes.  However in their version of rPath (pound
key) Digium has removed the netconfig command.  I am able to manually add
the route with 

 

Route add default gw xxx.xxx.xxx.xxx however when I reboot I lose my
routing.  Does anyone know which conf file I need to edit in order to make
this routing change permanent?

 

Thank you

 


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Re: [asterisk-users] network routing

[Ed Nuñez]

Thanks, that worked

 

· I was using GATEWAYDEV=eth1

And that was not working.

 

Thanks again

 

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ~Russell
Sent: Thursday, June 28, 2007 3:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] network routing

 

How many GW you need to add ?  if it is one .. then add 

GATEWAY=xxx.xxx.xxx.xxx into /etc/sysconfig/network

thanks
Russell



On 6/29/07, Ed Nuñez [EMAIL PROTECTED] wrote:

I have installed the Asterisk BE B.2.2 image file in a new server.  I need
to make network routing changes.  However in their version of rPath (pound
key) Digium has removed the netconfig command.  I am able to manually add
the route with 

 

Route add default gw xxx.xxx.xxx.xxx however when I reboot I lose my
routing.  Does anyone know which conf file I need to edit in order to make
this routing change permanent?

 

Thank you

 


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Re: [asterisk-users] kore dump

[Ed Nuñez]

What is a god Windows application to read core dump files?



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of J. Oquendo
Sent: Tuesday, June 26, 2007 4:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] kore dump

Vadim Berezniker wrote:

 use the safe_asterisk script

 it will restart asterisk if it crashes and it enables core dumps (your 
 core size limit is probably set to 0 when you start asterisk).

 *From:* [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] *On Behalf Of *Ed 
 Nuñez
 *Sent:* Tuesday, June 26, 2007 2:22 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion; 
 [EMAIL PROTECTED]
 *Subject:* [asterisk-users] kore dump

 I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server.

 My PBX has experienced several core dumps the last couple of days and 
 I am not sure if this is what’s causing it, but it always seems to 
 happen when a particular extension on a grandstream phone uses ChanSpy 
 SIP group.

 I have not been able to locate where the core dump file is being 
 saved. I can’t find it in my TMP directory.

 I would also like to know if Asterisk can be setup to automatically re 
 start if there is a core dump. I was thinking of setting up a cron job 
 to launch Asterisk every minute. If it’s running, no harm done, and if 
 it crashes, the cron job will make sure that it’s started every 60 
 seconds.

 Any suggestions?

 Thank you

 Ed Nuñez

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If that fails you could always try something like:
*/2 * * * * /bin/ps -C /usr/bin/asterisk || { /usr/bin/asterisk  }

or so...

--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
echo infiltrated.net|sed 's/^/sil@/g' 

Wise men talk because they have something to say; fools, because they have
to say something. -- Plato





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Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy

[Ed Nuñez]

To configure the Cisco for RFC 2833 add the following line to the desired
dial-peer

dtmf-relay rtp-nte

Hope this helps.

Ed Nuñez



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Tuesday, June 26, 2007 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco
SIP Proxy

This is usually a Cisco issue.

You need to set the Cisco to use RFC2833 DTMF.  Check the Cisco docs.

tracinet wrote:
 Jason,
 I am at least having similar issues with rfc2833 DTMF:
 
 http://bugs.digium.com/view.php?id=10058
 
 
 On 6/20/07, Jason Ma [EMAIL PROTECTED] wrote:

 Hi buddies,
 I encountered DTMF issue when I tried to place call from x-lite to a
 sip conference serice,here is the diagram.
 X-liteAsterisk---Cisco SIP proxySIP Conference service

 The Call can be established,and I can hear from x-lite the prompt of
 the conference,but when I input any digits,nothing happened,the
 conference service did not recognize my input.At the same time,in the
 log of asterisk,I can find that asterisk recognized all the
 digitsI tried rfc2833,inband,info in the dtmfmode
 parameter,but did not work ,I'm not sure whether asterisk send the
 right dtmf to cisco proxy,how can I track that?

 I made another test,dialing from x-lite registered with Cisco proxy to
 voicemail service of Asterisk.
 x-liteCisco SIP proxyAsterisk---Voicemail service

 Both the call and dtmf worked fine,I can input my mailbox number and
 password and listen my  voicemail.both rfc2933 and inband worked
 in this situation,but not info.

 My Asterisk is 1.4.4 with asterisk now,I did not configure dtmfmode in
 the section of  xlite and the trunk to cisco proxy,just configure the
 dtmfmode in sip.conf.

 When I used rfc2833,I can see the log in asterisk as :

 [2007-06-19 16:01:40] DTMF[8925] channel.c: DTMF begin '2' received on
 SIP/-08269470
 [2007-06-19 16:01:41] DTMF[8925] channel.c: DTMF end '2' received on
 SIP/-08269470, duration 160 ms
 [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF begin '1' received on
 SIP/-08269470
 [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF end '1' received on
 SIP/-08269470, duration 140 ms

 and when I used inband,I can see :

 [2007-06-19 15:55:21] DTMF[8852] channel.c: DTMF end '2' received on
 SIP/-09d916c0, duration 0 ms
 [2007-06-19 15:55:22] DTMF[8852] channel.c: DTMF end '1' received on
 SIP/-09d916c0, duration 0 ms

 Is that right?Can I check what digits that asterisk sent out ?

 How can I track where is wrong with the dtmf?Did asterisk send dtmf to
 Cisco proxy correctly?
 I really have no idea about that.Please advise.Thank you very much

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[asterisk-users] kore dump

[Ed Nuñez]

I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server.

 

My PBX has experienced several core dumps the last couple of days and I am not 
sure if this is what's causing it, but it always seems to happen when a 
particular extension on a grandstream phone uses ChanSpy SIP group.

 

I have not been able to locate where the core dump file is being saved.   I 
can't find it in my TMP directory.

 

I would also like to know if Asterisk can be setup to automatically re start if 
there is a core dump.  I was thinking of setting up a cron job to launch 
Asterisk every minute.  If it's running, no harm done, and if it crashes, the 
cron job will make sure that it's started every 60 seconds.

 

Any suggestions?

 

 

Thank you

 

Ed Nuñez

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[asterisk-users] 1.4.5

[Ed Nuñez]

I am seeing a peculiar message on my console screen on my new installation of 
Asterisk 1.4.5I would appreciate any comments.

 

Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS

Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS

Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS

Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS

Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS

Really destroying SIP dialog '[EMAIL PROTECTED] Method: OPTIONS

Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS

Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS

 

 

 

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Re: [asterisk-users] inband DTMF for g729

[Ed Nuñez]

I have a similar issue with Qwest SIP.  They only support rfc2833 in g729AB,
and Asterisk is only G729A.  Sprint works fine for me.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Fredrickson
Sent: Friday, June 22, 2007 3:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] inband DTMF for g729

Sounds like you need a new SIP carrier.  G.729 has a way of  
destroying inband DTMF tones.

---
Matthew Fredrickson
Software Engineer
Digium, Inc.

On Jun 22, 2007, at 1:20 PM, Gary Chen wrote:

 Does anybody know why Asterisk does not support inband DTMF for G.729?
 Our SIP carrier use inband dtmf for G.729. This causes problem for  
 us to use it for our Asterisk IVR system.

 Any suggestion to solve this problem?

 Gary
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Re: [asterisk-users] install Asterisk-addons 1.4.2

[Ed Nunez]

I have Asterisk 1.4.5 and addons 1.4.1.  Can anyone tell me if I can just
install addons 1.4.2 on this system without re installing Asterisk?

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
clive.chan(Alpha Trilogies Networks)
Sent: Wednesday, June 20, 2007 9:06 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] install Asterisk-addons 1.4.2

 

Hi, 

I am trying to install the Asterisk-addons-1.4.2, and when I make install it
prompt me such error messages

make[1]: Entering directory `/usr/src/asterisk-addons/asterisk-ooh323c'

cp .libs/libchan_h323.so.1.0.1 /usr/lib/asterisk/modules/chan_ooh323.so

cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory

make[1]: *** [install] Error 1

make[1]: Leaving directory `/usr/src/asterisk-addons/asterisk-ooh323c'

make: *** [install] Error 2

 

 

 

How to solve it out?

 

clive chan

Alpha Trilogies Networks Sdn Bhd 

Tel : 04 - 647 288 Ext: 338

Tel : 04 - 647 2999

Mobile : 012 - 408 6376

email : [EMAIL PROTECTED]

 

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Re: [asterisk-users] ChanSpy SIP

[Ed Nuñez]

For anyone experiencing the same problem, I was able to make SpyChan work on
SIP extensions using the b and v options.

 

exten = _**.,1,ChanSpy(IAX2/1654|bv(4))

 

 

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nunez
Sent: Tuesday, June 19, 2007 8:05 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ChanSpy SIP

 

Has anyone succesfully tried using ChanSpy on SIP channels with the latest
Asterisk 1.4?  I tried ChanSpy(SIP/5060) to monitor SIP extension 5060 and
the console displays, Monitoring Sip/5060, but I don't hear anything.  I am
able to monitor Zap channels.

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[asterisk-users] ChanSpy SIP

[Ed Nunez]

Has anyone succesfully tried using ChanSpy on SIP channels with the latest 
Asterisk 1.4?  I tried ChanSpy(SIP/5060) to monitor SIP extension 5060 and the 
console displays, Monitoring Sip/5060, but I don't hear anything.  I am able to 
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RE: [asterisk-users] g729

[Ed Nuñez]

Oddly enough the call was being recorded.  In any case in case anyone is
having the same problem, here is what did to get rid of the errors.  I am
now using Monitor instead of MixMonitor as Jaswinder suggested.

Thanks

exten =
_1NXXNXX,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
RID:6}-${EXTEN}-${TIMESTAMP}-OUT)
exten = _1NXXNXX,2,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
exten = _1NXXNXX,3,Set(CDR(UserField)=${MONITOR_FILENAME})
exten = _1NXXNXX,4,Set(CALLERID(number)=14073844200)
exten = _1NXXNXX,5,Monitor(${CALLFILENAME}.wav49||mb)
exten = _1NXXNXX,6,Dial(SIP/[EMAIL PROTECTED]) 




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729

I think asterisk first converts audio stream to slin for recording to
a wav file . Since you are using hardware g729 transcoder i think this
is what is causing the problem . Is the calla actually being recorded
?  I suggest that you use monitor application since it can directly
record g729 audio stream and run some cron script with sox mixing the
IN and OUT files in 1 file .

On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
 Yes

 This is my extensions.conf entry.

 exten = _1NXNXXX,1,Set(DYNAMIC_FEATURES=automon)
 exten =

_1NXXNXX,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
 RID}-${EXTEN}-${TIMESTAMP}-OUT)
 exten =

_1NXXNXX,3,Set(TOUCH_MONITOR=CONTINEX-${CALLERID}-${EXTEN}-${TIMESTAMP}-
 OUT)
 exten = _1NXXNXX,4,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
 exten = _1NXXNXX,5,Set(CDR(UserField)=${MONITOR_FILENAME})
 exten = _1NXXNXX,6,Set(CALLERID(number)=14073844200)
 exten = _1NXXNXX,7,MixMonitor(${CALLFILENAME}.wav49)
 exten = _1NXXNXX,8,Dial(SIP/[EMAIL PROTECTED],,wW)




 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
 Singh
 Sent: Wednesday, June 06, 2007 4:28 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] g729

 Are you trying to record the conversation as well ?

 On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
 
 
 
 
  I installed a hardware g729 codec card in my asterisk, and I'm getting
the
  following error when calling from a g729 sip extension to a SIP trunk
also
  set to g729.  The call goes through just fine, but these error messages
 keep
  flying by until I disconnect the call.
 
 
 
  Any ideas?
 
 
 
  ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin
  failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
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RE: [asterisk-users] g729

[Ed Nuñez]

Just wanted to update anyone interested in this issue.

If I monitor a g729 SIP channel using ChanSpy, I am getting the same error
as when I use MixMon.




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Thursday, June 07, 2007 12:14 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] g729

Oddly enough the call was being recorded.  In any case in case anyone is
having the same problem, here is what did to get rid of the errors.  I am
now using Monitor instead of MixMonitor as Jaswinder suggested.

Thanks

exten =
_1NXXNXX,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
RID:6}-${EXTEN}-${TIMESTAMP}-OUT)
exten = _1NXXNXX,2,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
exten = _1NXXNXX,3,Set(CDR(UserField)=${MONITOR_FILENAME})
exten = _1NXXNXX,4,Set(CALLERID(number)=14073844200)
exten = _1NXXNXX,5,Monitor(${CALLFILENAME}.wav49||mb)
exten = _1NXXNXX,6,Dial(SIP/[EMAIL PROTECTED]) 




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729

I think asterisk first converts audio stream to slin for recording to
a wav file . Since you are using hardware g729 transcoder i think this
is what is causing the problem . Is the calla actually being recorded
?  I suggest that you use monitor application since it can directly
record g729 audio stream and run some cron script with sox mixing the
IN and OUT files in 1 file .

On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
 Yes

 This is my extensions.conf entry.

 exten = _1NXNXXX,1,Set(DYNAMIC_FEATURES=automon)
 exten =

_1NXXNXX,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
 RID}-${EXTEN}-${TIMESTAMP}-OUT)
 exten =

_1NXXNXX,3,Set(TOUCH_MONITOR=CONTINEX-${CALLERID}-${EXTEN}-${TIMESTAMP}-
 OUT)
 exten = _1NXXNXX,4,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
 exten = _1NXXNXX,5,Set(CDR(UserField)=${MONITOR_FILENAME})
 exten = _1NXXNXX,6,Set(CALLERID(number)=14073844200)
 exten = _1NXXNXX,7,MixMonitor(${CALLFILENAME}.wav49)
 exten = _1NXXNXX,8,Dial(SIP/[EMAIL PROTECTED],,wW)




 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
 Singh
 Sent: Wednesday, June 06, 2007 4:28 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] g729

 Are you trying to record the conversation as well ?

 On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
 
 
 
 
  I installed a hardware g729 codec card in my asterisk, and I'm getting
the
  following error when calling from a g729 sip extension to a SIP trunk
also
  set to g729.  The call goes through just fine, but these error messages
 keep
  flying by until I disconnect the call.
 
 
 
  Any ideas?
 
 
 
  ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin
  failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
 
  Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
  Translation to slin failed, dropping frame for spies
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RE: [asterisk-users] g729

[Ed Nunez]

Yes, that is correct.  I am using mixmon and using wav49.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 4:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729

Are you trying to record the conversation as well ?

On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:




 I installed a hardware g729 codec card in my asterisk, and I'm getting the
 following error when calling from a g729 sip extension to a SIP trunk also
 set to g729.  The call goes through just fine, but these error messages
keep
 flying by until I disconnect the call.



 Any ideas?



 ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin
 failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies
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RE: [asterisk-users] g729

[Ed Nuñez]

Yes

This is my extensions.conf entry.

exten = _1NXNXXX,1,Set(DYNAMIC_FEATURES=automon)   
exten =
_1NXXNXX,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
RID}-${EXTEN}-${TIMESTAMP}-OUT)
exten =
_1NXXNXX,3,Set(TOUCH_MONITOR=CONTINEX-${CALLERID}-${EXTEN}-${TIMESTAMP}-
OUT)
exten = _1NXXNXX,4,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
exten = _1NXXNXX,5,Set(CDR(UserField)=${MONITOR_FILENAME})
exten = _1NXXNXX,6,Set(CALLERID(number)=14073844200)
exten = _1NXXNXX,7,MixMonitor(${CALLFILENAME}.wav49)
exten = _1NXXNXX,8,Dial(SIP/[EMAIL PROTECTED],,wW)




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 4:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729

Are you trying to record the conversation as well ?

On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:




 I installed a hardware g729 codec card in my asterisk, and I'm getting the
 following error when calling from a g729 sip extension to a SIP trunk also
 set to g729.  The call goes through just fine, but these error messages
keep
 flying by until I disconnect the call.



 Any ideas?



 ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin
 failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies

 Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
 Translation to slin failed, dropping frame for spies
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[asterisk-users] Voip-info.org

[Ed Nuñez]

Is anyone else having trouble going into voip-info.org today? 

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[asterisk-users] g729

[Ed Nuñez]

I installed a hardware g729 codec card in my asterisk, and I'm getting the 
following error when calling from a g729 sip extension to a SIP trunk also set 
to g729.  The call goes through just fine, but these error messages keep flying 
by until I disconnect the call.

 

Any ideas?

 

ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, 
dropping frame for spies

Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: Translation 
to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: Translation 
to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: Translation 
to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation 
to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: Translation 
to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation 
to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: Translation 
to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: Translation 
to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: Translation 
to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation 
to slin failed, dropping frame for spies

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[asterisk-users] FW: autologoff

[Ed Nuñez]

Is the autologoff function supported in Asterisk BE B.1-3?  I have
configured my agents.conf with a 5 second timeout, but the agents extension
continues ringing until the call eventually goes to voicemail.



Agents.conf
[general]
persistentagents=yes


[agents]
autologoff = 5
multiplelogin = no
recordagencalls = yes
monitor-join = yes
createlink = yes
updatecdr = yes
musiconhold = default
recordformat = wav49
savecallsin = /var/spool/asterisk/monitor/

agent = 1650,1650,Tareq 
agent = 1656,1656,Ed 
agent = 2000,2000,test agent
agent = 1704,1704,Reload Test



queues.conf
[general]
persistentmembers=yes


[noi-cust-serv-spanish]
strategy = leastrecent
announce-frequency = 30
announce-holdtime = yes
announce-round-seconds = 10
timeout=180
monitor-format=wav49
monitor-join=yes
joinempty = strict
leavewhenempty = strict
musiconhold = default
eventwhencalled = yes
servicelevel=180
reportholdtime =yes
maxlen=0; maximum ammount of calls waiting
queue-youarenext = queue-youarenext;   (You are now first
in line.)
queue-thereare = queue-thereare;   (There are)
queue-callswaiting = queue-callswaiting;   (calls waiting.)
queue-holdtime = queue-holdtime;   (The current est.
holdtime is)
queue-minutes = queue-minutes  ;   (minutes.)
queue-seconds = queue-seconds  ;   (seconds.)
queue-thankyou = queue-thankyou;   (Thank you for your
patience.)
queue-lessthan = queue-less-than   ;   (less than)
queue-reporthold = queue-reporthold

member = Agent/1656


autologoff - with this option you set for how long the phone has to ring
with no answer, before the agent to be logged off. You have to set the
maximum period of time in seconds. By default this option is set to 15
seconds.


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[asterisk-users] FW: Re install

[Ed Nuñez]

 

I had to re install the my Asterisk BE with the latest version, and when I try 
to load my g.729 codec license I do not see the folders in the path that they 
are described in the instructions given to us with the license or in your 
online documentation.  I installed the disk 1 immage (rPath), and I am not able 
to perform the g.729 installation or registration.

 

 

 

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RE: [asterisk-users] FW: Re install

[Ed Nuñez]

I was able to fid the modules directoty, but when I run 





-r-x--  1 root root 1288344 May 21 11:35 register
 
/root/register
 

 

 

I get the following error

 

 

-bash: /root/register: cannot execute binary file

 

 

I have changed the file attributes as you can see on the ls -l

 

 



 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Monday, May 21, 2007 11:25 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] FW: Re install

 

 

I had to re install the my Asterisk BE with the latest version, and when I
try to load my g.729 codec license I do not see the folders in the path that
they are described in the instructions given to us with the license or in
your online documentation.  I installed the disk 1 immage (rPath), and I am
not able to perform the g.729 installation or registration.

 

 

 

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[asterisk-users] AsteriskNow!

[Ed Nuñez]

 

Does anyone know how to gain access directly to the configuration files in 
AsteriskNow?  I have dual NICs and need to change the binding in the config 
file.  I believe they blocked ssh2 access by default.

 

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[asterisk-users] RE: Autologoff

[Ed Nuñez]

 

I am having an issue with the autologoff fuction in agents.conf

 

I am running Asterisk BE and I am testing with agent 1656.  I log in, and then 
make a call into the queue.  The agent's phone rings, and if I answer it, all's 
fine/  I am trying to have this agent automatically be logged off if he does 
not answer the queue callback within 5 seconds, however the agents extension 
keeps ringing until the call eventually goes to the extension's voice mail, 
which I am also trying to avoid.

 

Here is my agents.conf

 

[general]

 

persistentagents=yes

 

 

[agents]

 

autologoff=5

multiplelogin=no

recordagencalls=yes

monitor-join=yes

createlink=yes

updatecdr=yes

musiconhold=default

recordformat=wav49

urlprefix=http://xxx.xxx.xxx.xxx/calls/

savecallsin=/var/www/html/calls

 

agent = 1650,1650,

agent = 1656,1656,Ed

 

 

Here is my queues.conf

 

[general]

persistentmembers=yes

 

 

[noi-cust-serv-spanish]

strategy = leastrecent

announce-frequency = 90

announce-holdtime = yes

announce-round-seconds = 10

timeout=180

monitor-format=wav49

monitor-join=yes

joinwhenempty = strict

leavewhenempty = yes

musiconhold = default

eventwhencalled = yes

queue-youarenext = queue-youarenext;   (You are now first in 
line.)

queue-thereare = queue-thereare;   (There are)

queue-callswaiting = queue-callswaiting;   (calls waiting.)

queue-holdtime = queue-holdtime;   (The current est. 
holdtime is)

queue-minutes = queue-minutes  ;   (minutes.)

queue-seconds = queue-seconds  ;   (seconds.)

queue-thankyou = queue-thankyou;   (Thank you for your 
patience.)

queue-lessthan = queue-less-than   ;   (less than)

 

member = Agent/1656

 

 

 

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[asterisk-users] Autologoff

[Ed Nuñez]

I am having an issue with the autologoff fuction in agents.conf

 

I am running Asterisk BE and I am testing with agent 1656.  I log in, and then 
make a call into the queue.  The agent's phone rings, and if I answer it, all's 
fine/  I am trying to have this agent automatically be logged off if he does 
not answer the queue callback within 5 seconds, however the agents extension 
keeps ringing until the call eventually goes to the extension's voice mail, 
which I am also trying to avoid.

 

Here is my agents.conf

 

[general]

 

persistentagents=yes

 

 

[agents]

 

autologoff=5

multiplelogin=no

recordagencalls=yes

monitor-join=yes

createlink=yes

updatecdr=yes

musiconhold=default

recordformat=wav49

urlprefix=http://64.211.222.226/calls/

savecallsin=/var/www/html/calls

 

agent = 1650,1650,Tareq Tujjar

agent = 1656,1656,Ed Nuñez

 

 

Here is my queues.conf

 

[general]

persistentmembers=yes

 

 

[noi-cust-serv-spanish]

strategy = leastrecent

announce-frequency = 90

announce-holdtime = yes

announce-round-seconds = 10

timeout=180

monitor-format=wav49

monitor-join=yes

joinwhenempty = strict

leavewhenempty = yes

musiconhold = default

eventwhencalled = yes

queue-youarenext = queue-youarenext;   (You are now first in 
line.)

queue-thereare = queue-thereare;   (There are)

queue-callswaiting = queue-callswaiting;   (calls waiting.)

queue-holdtime = queue-holdtime;   (The current est. 
holdtime is)

queue-minutes = queue-minutes  ;   (minutes.)

queue-seconds = queue-seconds  ;   (seconds.)

queue-thankyou = queue-thankyou;   (Thank you for your 
patience.)

queue-lessthan = queue-less-than   ;   (less than)

 

member = Agent/1656

 

 

 

 

 

 

 

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RE: Re Re: [asterisk-users] TC400B

[Ed Nuñez]

The g729 licenses are US$10 a pop and you can buy them directly from
www.Digium.com



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad
Sent: Wednesday, May 02, 2007 5:10 AM
To: asterisk-users@lists.digium.com
Subject: Re Re: [asterisk-users] TC400B

Dear Andres;

How much it cost the 4 licenses of G729 and from where
I have to buy them?

Also, what if I need to do IP Trunk between Asterisk
and another IP PBX in another side (in case I need 30
ports for this IP Trunk, and I need to use G729 or
G723 codec), then also I need to buy a license for
this? How much?

I was think that no licenses in Asterisk, now I see
something new :) -

Regards
Bilal

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RE: [asterisk-users] Calls in ulaw, not gsm as desired

[Ed Nuñez]

Reload will reload your sip.conf file!  As well as iax.conf,
extensions.conf, queues.conf, voicemail.conf, users.conf

 

 

 

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall
Sent: Tuesday, May 01, 2007 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls in ulaw, not gsm as desired

 

I was in the asterisk console and I typed reload. Is this not enough to
reload the sip.conf file?

Rob

Andreas Sikkema wrote: 

However, even once I reloaded the extensions, its still only 
using ulaw.


 
You didn't reload the sip config? Maybe that's your problem?
 
  

 

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[asterisk-users] Confference function

[Ed Nuñez]

I would like to know if anyone here knows the answer to the following question

 

I need to implement the following conferencing feature for my agents.

 

1.   Agent receives call from caller

2.   Agent conferences a verification service

3.   After finishing the verification, agent needs to drop third party 
(Verification service) and continue on the line with caller.

 

My problem right now is being able to disconnect the third party and keeping 
the caller on the line.  Would this be a function of Asterisk or the SIP / IAX 
phone?  Any comments would be appreciated.

 

Thank you

 

Ed Nuñez 

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RE: [asterisk-users] Asterisk Pix firewalls

[Ed Nuñez]

Don

 

This may not be a solution to your question, but I would like to share that
I’ve been having one way audio issues when connecting point to sight to a
PIX 515E using SIP.  I changed to IAX and this is working perfectly now.  It
was paynless to configure IAX2, so you might want to consider it.

 

Ed

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don E. Wisdom
Sent: Tuesday, April 24, 2007 8:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk  Pix firewalls

 

Hi,
I asked this last week but i didn't get any answer   So i will elaborate on
my question.   I need to setup a pix 515 firewall (running 7.2.2 OS) to
allow sip traffic thru it from a sip phone wherever i may be.  The pix is
where all my servers are colocated and i will need to connect thru it from
softphones / hardphones wherever i happen to be traveling.   I need help
setting up the pix for inbound and outbound sip/iax traffic.   Any help
would be greatly appreciated.
Thanks
--Don 

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[asterisk-users] SIP over VON

[Ed Nuñez]

Hello all

 

I would like to know if anyone here has had any experience trying to set up
SIP or IAX over VPN.  I am testing with Cisco VPN client and when I call the
Asterisk server in my office I get one way audio.

 

Thanks

 

Ed Nunez 

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Re: [asterisk-users] How can I improve call quality?

[Ed W]

Check first using something like testmyvoip.com to get an idea of your 
situation (stress the internet by opening up lots of simultaneous 
downloads during the test)


Repeat: Try the above before you do anything else...

Ed W
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Re: [asterisk-users] UK zaptel and zapata.conf for TDM400P

[Ed W]

Hi


usecallerid=yes
cidsignalling=v23
cidstart=polarity


Although this is what the wiki recommends, I just couldn't get the 
cidstart=polarity to play well with immediate=yes, I kept loosing the 
callerid?


This is what I ended up with and now it avoids the annoying 2 rings 
before the internal extensions start to ring.  However, I still have a 
problem in that if someone hangs up while still in ringing state then 
asterisk continues to ring for 2 more rings (roughly).  This is annoying 
because BT appear to do a line test every 30 hours or so and so my lines 
ring for 2 rings at random times of day or night



[EMAIL PROTECTED] asterisk]# more zapata.conf
;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]

[channels]

language=en
context=from-zaptel
signalling=fxs_ks
rxwink=300  ; Atlas seems to use long (250ms) winks

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

ukcallerid=yes
cidsignalling=v23
cidstart=ring
;cidstart=polarity ; Added for UK CLI detection
sendcalleridafter=0
immediate=yes ; as we recieve cli info before not after first ring.

answeronpolarityswitch=no

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Re: [asterisk-users] How can I improve call quality?

[Ed W]

Hi


Also - are there any useful stats/logs that I can examine to see the
quality of calls?
  


You didn't mention that you have any QOS on your router, so we can 
basically guarantee that your problem is the internet connection.


Remember that all the research on networking has been how to saturate a 
single connection and download as fast as possible, so when some spod 
hits a website and reads a web page then he grabs basically the whole 
connection for a short space of time.  During that time your voip 
packets tend to loose out and get delayed - the jitter buffer does some 
stuff to try and compensate, but ultimately it will loose


Add some kind of priorisation to the T1 line and your quality should go 
up dramatically


Check first using something like testmyvoip.com to get an idea of your 
situation (stress the internet by opening up lots of simultaneous 
downloads during the test)


Cheap fix is to get a separate DSL line and run the voice over that...

Ed W


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Re: [asterisk-users] How can I improve call quality?

[Ed W]

Our 'net link is a dedicated 2Mb fibre connection (of which we have ever
used 50% max bandwidth).  


Remember in computer terms this means that you used 100% of the 
connection, 50% of the time  Your voice will loose out against the 
big data packets and spoil the voice quality big time


Ed W
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[asterisk-users] Hardware suggestions for 8-10 lines in the UK

[Ed W]

Hi

I have previously had good success on smaller installations with TDM400P 
cards.  I now have a UK customer looking for 8-10+ lines and it seems 
like a PRI would be most economical + reliable?


Has anyone here used PRI interfaces in the UK and can confirm that it 
works well (using Trixbox for preference?).  I have had some niggling 
issues with the TDM400P cards which puts me off adding lots of these in 
a single box.  Am I right in thinking that ISDN or PRI will be a better 
and more reliable option?


Any suggestions on 2U servers that should work well?  For example the 
DELL 2850 seem to lack any spare HD type power outlets which is 
irritating...


I also need two internal fax machines.  Can anyone confirm that the best 
solution is just some linksys ATA's at the fax end, then switching them 
directly down the PRI card?  Is there a better option to guarantee best 
quality?  It would be convenient not to have extra analogue lines in the 
building if we go down the PRI route...


Grateful for any thoughts

Ed W


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Re: [asterisk-users] Hardware suggestions for 8-10 lines in the UK

[Ed W]

Best bet would be to talk to insert telco of preference and ask them what 
they recommend. For anything more than about 8 channels a PRI is likely to be the 
most cost-effective route if you need physical lines on-site, especially if that's 
likely to grow beyond 8 lines in the future.
  


Just for reference:

- Called BT and their prices are basically the same for PRI as they are 
for analogue (~£10/month), but you need to buy a minimum of 8 lines for 
a PRI.  You can then increase decrease in single lines.  DIDs are quite 
cheap also (practically free)


- Installation cost is cheaper on PRI than analogue, but they will waive 
the installation cost if you buy something else from them such as 
broadband or a mobile phone


Sounds like PRI is the way we will go therefore as long as the equipment 
is reliable.  Can anyone recommend PRI cards which are known to work 
flawlessly with Euro ISDN 30?  (FWIW, BT tell me they now supply all new 
lines as Euro standard instead of the v85 that it claims on the voip wiki)



You've obviously had better success with the TDM400Ps than I have in the UK. 
Certainly the call quality on asterisk installs we've done around 
Northampton/Milton Keynes has improved markedly since we switched to delivering 
calls via SIP or IAX rather than using physical lines on-site. The clients are 
also happy - each less line they have on-site is a saving of at least £10/month.
  


Agreed.  My experience is that quality is higher on Voip than it is via 
a TDM400p.  However, my experience hasn't been that VoiP is as reliable 
as copper lines and so unless you can tolerate the odd outage once per 
month or two then you might want to stick to copper for the main 
carrier?  Does this match with the experience from others?



Still after recommendations on a server box (2U with space for a couple 
of PCI cards would be sensible), the PRI card and also any ATA adaptors 
which are known to work well with fax units


Cheers

Ed W

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[asterisk-users] Chanspy

[Ed Nuñez]

In my asterisk, I have calls coming in on a Zap channel and going out SIP.
My problem is that when I spy on the SIP channel, I hear the calling parting
breaking in and out, and the called party sounds just fine (SIP).  If I spy
on the Zap channel , I hear both sides just fine.  I am spying from my SIP
extension.

Any ideas?


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Re: [asterisk-users] SIP RTP Tunnel

[Ed Greenberg]

Also set canreinvite=no between Asterisk and the provider.

[EMAIL PROTECTED] wrote:
Hola Sanjay, 


this works pretty well in one direction. The Sip User who is registered at the 
Asterisk. But the Sip user who calls from sipXYZ.com still sends it data 
diretly to sip user 1.

Any idea?

Thanx!!

-Original Message-
From: Sanjay Rajdev [mailto:[EMAIL PROTECTED] 
Sent: Donnerstag, 29. März 2007 18:27

To: kalle odenthal; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP RTP Tunnel

Try setting canreinvite = no in sip.conf or the database (where you have 
sipuser setting).

Regards,
Sanjay Rajdev

- Original Message -
From: kalle odenthal [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, March 30, 2007 5:52:47 AM (GMT+0530) Asia/Calcutta
Subject: [asterisk-users] SIP RTP Tunnel

Hello,

is it possible to rout ALL RTP Data over Asterisk, like

SIP1 ---RTP--- Asterisk ---RTP--- SIP2

I know it seems quite useless. But I want to simulate a IAX - SIP connection and have no Phonecard installed on my computer ;) 

Thanx, 


Kalle




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Re: [asterisk-users] Sending Email From the dialplan

[Ed Greenberg]

Al Bochter wrote:

I have looked around with no luck.

Does anyone know of a way to send an email from the dialplan.
The system that I am working on has none thing to do with VoiceMail.

This is something like the SMS command but using sending email

I am working on a prepaid alarm dispatch program for Asterisk if 
anyone has any input please let me know.
I will be more than happy to write the code as Open Source for others 
to use code. With help from the list.


I think you need an AGI program, or just a system call to an external 
script.



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Re: [asterisk-users] dial a pager and enter DTMF

[Ed Greenberg]

Supa wrote:
Probably just a simple syntax issue, but does anyone know how to dial 
a number and the once phone has been answered, play DTMF tones and 
then disconnect. I am trying to use this for page notification.


Ive been trying the following string with out luck:

exten = s,2,Dial(SIP/TelaSip-gw4/5198881212|D(12345678)


I think what you are missing is the timeout parameter.

The dial command is... 


Dial(type/identifier, timeout, options)

So you would want something like:
   exten = s,2,Dial(SIP/TelaSip-gw4/5198881212|120|D(12345678))
which would give you a two minute timeout

Also, you are indeed missing a right paren on the end, which I added in 
the line above.


/edg
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Re: [asterisk-users] Best phone for easy provisioning

[Ed W]

Paul Hales wrote:

I know Brett and Jurgen have been pretty happy with the Snom's - Brett
even wrote an auto-provision utility for the Snom's at one time.
  


Yes, look at the latest Trixbox for the basic SNOM templates and then 
off you go.


You setup a tftp server (easy), the phone looks for two files, one 
generic snom file and another with it's MAC address in the name.  So you 
have generic stuff in one file and specific phone stuff in the other.  
You can get the initial phone config done using DHCP, or just log into 
each phone and change the URL to point to your tftp server.


Get the phone roughly setup manually and then simply look at the web 
config utility, it has an option to dump it's entire config out and so 
you just cut and paste the entries you want to override into the tftp 
files.  Piece of cake.


You can easily reboot all the phones by sending them a certain SIP 
message, and so it's very easy to redeploy a new config, or reboot all 
the phones when you reboot the server.  The phones themselves re-read 
the config every X minutes so they pickup new config quickly even 
without a reboot.


I have a bunch of 360s which I negotiated for about the same price as 
the 320s.  Drop me a line if you want the name of a UK firm to buy them 
from.  They work nicely out of the box including the flashy lights 
showing busy extensions.  The only thing which doesn't work without a 
patch (it appears) is line pickup by pushing the BLF keys.  I can live 
without that, but it would be nice to have.


Happy to post my configs if anyone wants to write up the notes on the wiki?

Ed
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Re: [asterisk-users] Diagnosing poor call quality

[Ed W]

Check from the sites in question using testmyvoip.com or whatever the 
site is called.


In the UK I found that some strange things sometimes happen.  At one 
point I was sure that BT were perhaps misclassifying IAX packets as 
P2P... However, not had a problem with SIP.


Beware that ADSL uses vastly more bandwidth than you expect on small 
packets, eg if you are classifying using a cheap router then you 
probably need to at least half your claimed bandwidth in order to make 
the prioritisation work correctly.  I added some (hack) patches to fix 
the linux calculation for HTB on the linux QOS list a year or two back.  
If you have a linux router you could use those to improve the 
calculation quality for QOS - or else I found a Draytek router does 
impressively well at getting it right for small sites...


Very likely you will find that the issue is variable jitter on the 
line.  The link above should help you figure this out


Good luck

Ed W
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Re: [asterisk-users] Re: Help - Poor Voice Quality

[Ed W]

Hi

Yes, I know that I am using IAX2 and not SIP for my connection to 
teliax.  IAX2 is the preferred protocol for connection to teliax.  I 
have the firewall configured to prioritorize port 4569 for IAX2.



1) 4569 is only the IAX setup port.  Edit rtp.conf to limit the rtp 
ports to some subset and then prioritise those instead
2) Uplink bandwidth is always the constraint on these lines.  This is 
highlighted in this case
3) Shorewall can't correctly prioritise bandwidth whenever using some 
kind of DSL service or whenever the packets are encapsulated such as the 
cable service.  Read the linux QOS faq for more info and as a workaround 
slash the theoretical bandwidth in half in your shaping script.  This 
should get you working and you can tweak later
4) Monitor the QOS buckets as you make/break calls to check that all the 
packets are classified correctly.  Otherwise your voip packets might be 
accidently in the bulk box


Basically VOIP goes from perfect to horrible when the jitter rises and 
packet loss goes up.  Probably this is happening in your case


Good luck

Ed W
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[asterisk-users] Glitches in voicemail prompts

[Ed W]

I changed from using a recent asterisk system standalone to a Trixbox
install and now I get clicks and minor dropouts on the voicemail
prompts.  System load is non-existant on this machine, interrupts
*appear* to be fine, and as near as I can tell the glitch is at the same
point in the prompt each time...

Any suggestions on how to debug this further?

To my ear it sounds like what happens when you get an overflow in some
decoder code and the levels have wrapped around?

Any thoughts?

Ed W

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[asterisk-users] Dial plan constructions suggestions?

[Ed W]

Can I ask for some advice on dial-plan construction please

I have setup my dialplan to use 9 to get a zap trunk, leaving everything 
else for internal extensions.


However, this creates a problem in that my callerid is correct, but 
doesn't work to re-dial the incoming caller.  So if I simply click 
missed calls on my Snom phone and hit redial then it tries to dial an 
internal extension.


So I then setup Asterisk to add a 9 to the incoming callerid for all 
calls which come via the Zap trunk, but now this creates some issues 
with applications like Snapanumber and perhaps HudLite, which are trying 
to map the caller ID to numbers in the addressbook (and I don't really 
want my internal Outlook address books to have everyone listed with a 
9 in front of their number)


How are others handling this?

I have considered simply dropping the prefix digit and working around 
any clashes in internal and external numbers (not very hard).


Grateful for any thoughts

Ed W
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Re: [asterisk-users] Dial plan constructions suggestions?

[Ed W]

Hi

There was a thread about this not too long ago, so the archives may 
have a bit more on it...


The way I handle it is by forcing the caller to dial the full number 
starting with zero (normally 10 or 11 digits in the UK - which I'm 
guessing you're from too)



Yes, I use something similar on another box, but there I support shorter 
dial codes as well.  It's not to hard to make 8 dial 0208 or 
7 dial 0207, etc.  I happen to also map some of the 1xx codes 
across as well.


It's still not a complete solution though because on this other box I 
have a business line and a personal line and I send calls to different 
lines based on the type of call (or more usually the time of day...).  I 
want to have seperate billing basically.  When the call comes in it 
makes sense to have the caller tagged with (in my dialplan) 9 for a 
personal call, and I use 3 (for no good reason) for my business line.  
I actually have one phone which defaults to business line if I don't add 
a prefix, another DECT phone which is my personal phone, but I can see 
on either where the call is coming from and also force the call to use a 
different route just by dialing the prefix.



Basically it's tricky.  I do already use custom ring tones for each 
line, so I guess I could drop the prefix, but it's nice to have it so 
that I can see at a glance whether it's a business call or not...


Any other suggestions?

Any suggestions on other software than Snap which does callerId lookup 
from Thunderbird (not Outlook).  For example is HUDLite ever going to 
support Thunderbird...?


Cheers

Ed W
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Re: [asterisk-users] Re: Dial plan constructions suggestions?

[Ed W]

Hi


I had the same situation, in that I wanted to be able to use the
Voicemail 'dial back' feature, and had a few phones with internal
CID-based dial features, that I wanted to be allowed to be used. Your
normal context is set up to operate with a '9' (or whatever) in front;
so it is clear that you will need a different context from which to
dial, a context that doesn't have the '9' at the beginning.
  



I appreciate your point, but it's not that hard to avoid having the 9 
prefix at all (in a simple dialplan at least).  So to be honest one 
might as well dump the whole dial 9 thing completely in the scenario 
you describe?


I think the solution here is really that the CID type applications 
become aware of prefix digits and strip them.  Anyone know of good 
solutions to this?


Any backend solutions to get Asterisk to hook into Exchange server etc?

Cheers

Ed W
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Re: [asterisk-users] Disconnect Supervision UK / BT solution?

[Ed W]

Chris Earle (CBL) wrote:

Sorry -- you're right, I didn't express the scenario properly ..

The disconnect supervision problem is when I 'forward'/divert an incoming
POTS call out another FXO channel to a mobile phone or POTS line.
(POTS - Sangoma|Asterisk - POTS/mobile)

When the incoming POTS hangs up and/or the mobile the person was connected
to .. Asterisk/Sangoma doesn't hang the Zap channels up.
  


Just to clarifydoes it all work ok if you are using SIP or IAX for the 
forwarded channels?  Eg local SIP phones?


I only have incoming zap lines in my config and with the exception of 
hangup on ringing I have found hangup detection to work fine.  I have a 
fax machine forwarding in my config as well and again no problems yet 
with hangup on that.


Does it fail to work *every* time, or just intermittently?  Does 
CallerId work ok in your setup?  (can be a clue to help diagnose your setup)


Ed W
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Re: [asterisk-users] Disconnect Supervision UK / BT solution?

[Ed W]

Matt Brown wrote:

Well,

I have just phoned BT today who said they can increase the CPC value 
on the line - however it needs to be done at the exchange - and has 
been booked for Tues.


I suppose I will know wether this worked on Tues :-) - I shall post my 
findings.


I would be keen to hear your findings - however, I'm still not clear 
exactly what the problem is in your case.  There are numerous kinds of 
disconnect problems - which one are you having (so we know which one the 
CPC fixes...)


Cheers

Ed W
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Re: [asterisk-users] Disconnect Supervision UK / BT solution?

[Ed W]

Does anyone have any thoughts/confirmation about this finally being a viable
solution?  This disconnect supervision problem has plagued TDM and Sangoma
cards for a long time!
  


Just to be clear, what is the exact disconnect problem that you see?

I have three TDM cards in two different systems, one using PBX lines and 
one on a private BT line.  Both of them have trouble detecting a caller 
who is ringing, but then hangs up before being answered by the asterisk 
system


However, *all* of them are absolutely fine at spotting a normal hangup 
once the call is connected and I see no random disconnects during calls 
either.


Can you confirm that this is what you mean, or whether it's something else?

Ed W

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[asterisk-users] Asterisk not hanging up

[Ed W]

I have a problem with calls not hanging up if for some reason the 
physical phone dies or gets unplugged


I can demonstrate this in practice by making a call from a handset, then 
unplugging the handset from the power.  The call remains active and 
asterisk never seems to disconnect it. 

More annoyingly when power is re-applied the handset comes back to life, 
won't receive incoming calls (because asterisk thinks it's busy), but 
likewise the handset itself doesn't think it's in a call so it can't 
retrieve the call or do a proper hangup.


I have no NAT in place and the handsets are all set to register/login 
and qualify=yes set (which I had hoped would sort this...)


The handsets are SNOM 360s but I don't think this is directly relevant.  
Asterisk is setup to use FreePBX dialplan (but again don't think this 
is relevant?)


Can someone please suggest a way to ensure that the calls get hungup - 
we had a 9 hour call earlier before someone noticed It's rare, but 
the consequences are potentially quite dire.



Cheers

Ed W

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Re: [asterisk-users] TDM404B VS TDM2401B

[Ed W]

Hi


i'm not very happy with TDM404B voice quality, low volume


Check the gain set in the zap config file.  You can increase the in/out 
gain quite a bit over standard.


Echo is frequently a symptom of wrong country settings, hence wrong 
impedence settings.  Also endpoints matter


Ed W
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Re: [asterisk-users] TDM2400 Hardware Echo Cancel

[Ed W]

Hi


Echo cancel almost works, but the users
hear 
what they describe as a 'crackle' coming back when they talk. 
  


Just a thought, but check that your gain levels are not too high?

Ed
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Re: [asterisk-users] Asterisk not hanging up calls

[Ed W]

Simon Tennant wrote:

I have noticed that Asterisk (version 1.2.13) is not hanging up a call
when the wifi handset moves out of range.

My setup is Nokia E61 connected to wifi access point (private IP range)
and then to server on internet (public IP).

I have been testing using the talking clock application, and walking out
of range does not hang up the call.
  


I can reproduce the same problem by simply unplugging a normal SIP 
handset from the power during a call (or it crashes and locks up).  When 
the handset gets re-booted the call is left in progress, new incoming 
calls aren't taken (because asterisk thinks that the handset is still in 
a call) and other problems


I added an L() entry on the dialplan to limit calls to something 
sensible in the meantime, but would like to get a proper workaround?


Any thoughts

Ed W
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[asterisk-users] TDM 400P in the UK - doesn't see ringing calls hanging up before answer

[Ed W]

Using a TDM400P in the UK nearly works fine, but I have a last remaining 
problem in that if the incoming is ringing and then the caller hangs up, 
asterisk takes another couple of rings before it spots the hangup.


This is annoying in that I sometimes get phantom calls late at night 
(possibly due to call waiting or the exchange doing a half ring to see 
if we are live).  Also I get phantom calls on either the voicemail or 
when I answer there is just dial-tone because the caller hungup before 
the call was answered


I have fiddled with a number of settings relating to polarity reversal 
because I believe that might be relevant to BT's implementation, but 
it's not made any difference from the default config. 

Any suggestions on how to fix this from UK users?  I have tried most of 
the suggestions in the voip wiki to no effect (haven't tried calling BT 
and asking them to change any settings yet)


Thanks for any thoughts

Ed W
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Re: [asterisk-users] Asterisk not hanging up

[Ed W]

Philipp Kempgen wrote:

Ed W wrote:

  

I have a problem with calls not hanging up if for some reason the
physical phone dies or gets unplugged



Have you tried the RTP timeout settings in sip.conf?
  


Sounds exactly like what I need!  Thanks

Is there no default set then??

Cheers

Ed W
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Re: [asterisk-users] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)

[Ed Rubright - mail lists]

Mark Coccimiglio wrote:

Marty,
   Where are you paying $1000 for a 1600 series Cisco?  I can get you 
20% off that price on any quantity (note: Sarcasam).  Its not the 
1990's anymore.  You can get them on eBay ($50-150) for only slightly 
more then the Linksys.  The performance is rock solid.  Three-quarters 
of the world have used them for decades.  I know of units running 2 
and 3 YEARS between reboots.  The power company reboots my equipment 
more then I do.  Ok it is true that Cisco does not support the models 
anymore, but you can't buy a services contract for a linksys router 
either.  It can sometimes be a little difficult to configure without 
any technical knowledge but that is what most of us get paid for.  It 
does impress the customer when you bring in the grey box labled 
Cisco.  As for performance just try to put 50 people behind a 
linksys/netgear/dlink.  I've used 1605R supporting +100 users.  Not 
even a blink.  Finally, untill everyone is using 10Mps FTTH the 
broad band link is still the slowest part of the connection.  Not to 
shabby for antiquated technology.


Mark C

Martin Joseph wrote:


On 2007-01-06 00:48:11 -0800, Mark Coccimiglio [EMAIL PROTECTED] said:


Mike
I'm using a Cisco 1605R [running IOS 12.3(5a)] small office router 
with Fair-Weight queueing enabled.  Works great.  The nice thing 
about Fair-Weight queueing is that it dynamically adapts to lower 
the priority of higher demand traffic (e.g. large downloads).  If 
you want quality stick with quality stuff.


Mark C



Reread the subject line please.  $1000 (US) isn't inexpensive by any 
stretch.


Marty


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Mark,

Do these 1600 series Cisco routers you mention that you find on eBay for 
$50-$150 support Layer3 routing?  I have a managed switch setup on my 
home network with several VLANs defined. (work subnet, home subnet, VOIP 
subnet)   I currently have to use a Linux box to route between the 
VLANs.  I'd like to move to Gigabit routing, but I'd need to replace the 
Linux box(more processor power and new NICs) and that gets expensive.


I'd much rather have a router or smart switch for that matter that does 
Gigabit Layer3 routing all in one unit. 


Do you have any recommendationsthat wouldn't break the bank?

Thanks,
Ed
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RE: [asterisk-users] MixMonitor and Queues

[Ed Nuñez]

In queues.conf you must have the following under the queues you want to record.

monitor-format=wav49 ; you may also use wav or gsm formats
monitor-join=yes; if you have the latest sox installed, 
thiswill join the in and out files into one.

In agents.conf

recordagencalls=yes
monitor-join = yes
recordformat=wav49
savecallsin=/var/www/html/calls ;this is the path where call will be 
recorded.

That's all

If you want to change the file name place this in your extensions.conf on a 
line prior to sending the call to the queue.

exten= 1097,4,Set(MONITOR_FILENAME=QUEUE-${CALLERID}-${TIMESTAMP})


Ed Nuñez
IT/Telecom Engineer
 
4037 Metric Drive
Winter Park, FL
 
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore
Sent: Wednesday, December 13, 2006 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MixMonitor and Queues

Greetings, all.

I would like to record calls that are entered into queues and I'm not 
quite sure how to do it.  Here's how I'm currently set up:

- Call comes in and is placed into Queue #1 (which rings all phones for 
15 sec).
- If call drops out of this queue, it is placed into Queue #2 (which 
plays MoH until the call is picked up).

I've tinkered with MixMonitor and I have my queues set up, but I'm not 
sure how to combine the two.  Ideally, I'd like to only record once the 
call comes out of queue (no point in recording hold music, unless I want 
to hear people mumble about how lousy a company we are for placing them 
on hold ;)  )

On a semi-related note, is it possible to determine the extension that 
pull the call out of queue before the call is bridged?  The reason I ask 
is that I'd like to put the receiving extension in the name of the file 
that MixMonitor creates.  If not, no biggie.

Recap:

Two queues.  First rings for 15 seconds then drops into the second. 
Second plays music on hold till the call is answered.  I want to record 
the call when it's pulled out of either queue using MixMonitor.  Bonus 
points if I can determine the answering extension before MixMonitor 
starts (if possible).

Any help would be greatly appreciated.

Thanks,
Jay
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RE: [asterisk-users] MixMonitor and Queues

[Ed Nuñez]

I've been trying to find where to download the Web Vmail application and 
instructions on how to install it for Asterisk BE.  Any ideas?

Thanks

Ed Nuñez
IT/Telecom Engineer
 
4037 Metric Drive
Winter Park, FL
 
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore
Sent: Wednesday, December 13, 2006 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MixMonitor and Queues

Greetings, all.

I would like to record calls that are entered into queues and I'm not 
quite sure how to do it.  Here's how I'm currently set up:

- Call comes in and is placed into Queue #1 (which rings all phones for 
15 sec).
- If call drops out of this queue, it is placed into Queue #2 (which 
plays MoH until the call is picked up).

I've tinkered with MixMonitor and I have my queues set up, but I'm not 
sure how to combine the two.  Ideally, I'd like to only record once the 
call comes out of queue (no point in recording hold music, unless I want 
to hear people mumble about how lousy a company we are for placing them 
on hold ;)  )

On a semi-related note, is it possible to determine the extension that 
pull the call out of queue before the call is bridged?  The reason I ask 
is that I'd like to put the receiving extension in the name of the file 
that MixMonitor creates.  If not, no biggie.

Recap:

Two queues.  First rings for 15 seconds then drops into the second. 
Second plays music on hold till the call is answered.  I want to record 
the call when it's pulled out of either queue using MixMonitor.  Bonus 
points if I can determine the answering extension before MixMonitor 
starts (if possible).

Any help would be greatly appreciated.

Thanks,
Jay
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RE: [asterisk-users] Asterisk manager

[Ed Nuñez]

Your line number nine should also specify a file name to monitor to and the 
format, like this

exten = 9,2,Monitor(from-${CALLEDID}-at-${TIMESTAMP},wav)

or better yet, use MixMon instead, because this will merge the two files into 
just one.  (both sides of the call)

Ed Nuñez
IT/Telecom Engineer
 
4037 Metric Drive
Winter Park, FL
 
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nik600
Sent: Tuesday, December 12, 2006 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk manager

Hi

i am trying to record a call with

exten = 9,1,Answer
exten = 9,2,Monitor
exten = 9,3,Dial(SIP/200)

This will record the call, but asterisk generates 2 files in
/var/spool/asterisk/monitor/

-in.wav
-out.wav

Can i have only one file?
Can i customize the path where to save the files?

Thanks
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[asterisk-users] downloading asterisk GUI

[Ed Nuñez]

This may be a Linux newby question, but here it goes.

 

I was reading the instructions on downloading and installing Asterisk GUI, but 
I can't get this to work.

 

svn checkout http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui

 

What would be the equivalent command in CentOS 4?

 

http://astrecipes.net/?n=217

 

 

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RE: [asterisk-users] downloading asterisk GUI

[Ed Nuñez]

Thanks

 

 

Ed Nuñez

IT/Telecom Engineer

  

4037 Metric Drive

Winter Park, FL

 

(o) 407-384-4200 x 1656

(f) 407-384-4222

(c) 732-925-0730

  _  

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mail list
Sent: Friday, December 08, 2006 4:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] downloading asterisk GUI

 

yum install subversion

On 09/12/06, Kovar Petr [EMAIL PROTECTED] wrote:

svn is application called subversion, you should download and install it 
first.



- Original Message - 

From: Ed Nuñez mailto:[EMAIL PROTECTED]  

To: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com  

Sent: Friday, December 08, 2006 7:18 PM

Subject: [asterisk-users] downloading asterisk GUI

 

This may be a Linux newby question, but here it goes.

 

I was reading the instructions on downloading and installing Asterisk 
GUI, but I can't get this to work.

 

svn checkout  http://svn.digium.com/svn/asterisk-gui/trunk 
http://svn.digium.com/svn/asterisk-gui/trunk  asterisk-gui

 

What would be the equivalent command in CentOS 4?

 

http://astrecipes.net/?n=217 

 

 


  _  


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image001.gif
Description: image001.gif
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[asterisk-users] queue agent Monitor

[Ed Nuñez]

Hello list.

 

Does anyone know if and how I can use in my context the following variable 
found in the CDR field?

 

DSTCHANNEL

 

I am trying to make the answering agent part of the monitor file name, but it 
is not working. 

 

exten= 0072,1,Answer

exten= 0072,2,Ringing

exten= 0072,3,Wait(2)

exten= 0072,4,set(MONITORFILENAME=${DST_CHANNEL}${CALLERID}-${TIMESTAMP})

exten= 0072,5,Queue(NOC)

exten= 0072,6,Hangup

include = parkedcalls

#include users.conf

 

This is what I am getting for a file name.

 

4072493400-20061207-160632.wav

 

Caller - timestamp.wav 

But I want to see

Agent(1656)-caller-timestamp.wav

 

 

Thank you

 

 

 

Ed Nuñez

IT/Telecom Engineer

  

4037 Metric Drive

Winter Park, FL

 

(o) 407-384-4200 x 1656

(f) 407-384-4222

(c) 732-925-0730



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RE: [asterisk-users] queue agent Monitor

[Ed Nuñez]

I just tried that and it doesn't work.  This may be perhaps because the file 
name needs to be defined before the call is sent to the queue.

 

When I saw you answer I thought it would work because it sounded very logical.  
:-)

 

This is the macro I use to send the call to the extension

 

Just in case I put the line before and after the extension.

 

[macro-extensions] 

exten = s,1,set(MONITOR_FILENAME=${EXTEN}-${CALLERID}-${TIMESTAMP})

exten = s,2,Dial(${ARG1}|30|t,,wW)

exten = s,3,set(MONITOR_FILENAME=${EXTEN}-${CALLERID}-${TIMESTAMP})

exten = s,4,Voicemail(u${ARG2})

exten = s,104,Voicemail(b${ARG2})

 

 

 

Ed Nuñez

 

 

 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
Sent: Thursday, December 07, 2006 3:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] queue agent Monitor

 

The queue application sends the call to an agent.  Use the agent

extension's dialplan to set up the monitor, that way you will have the

actual agent extension.

 

On Thu, 2006-12-07 at 14:18 -0600, Ed Nuñez wrote:

 Hello list.

 

  

 

 Does anyone know if and how I can use in my context the following

 variable found in the CDR field?

 

  

 

 DSTCHANNEL

 

  

 

 I am trying to make the answering agent part of the monitor file name,

 but it is not working. 

 

  

 

 exten= 0072,1,Answer

 

 exten= 0072,2,Ringing

 

 exten= 0072,3,Wait(2)

 

 exten= 0072,4,set(MONITORFILENAME=

 ${DST_CHANNEL}${CALLERID}-${TIMESTAMP})

 

 exten= 0072,5,Queue(NOC)

 

 exten= 0072,6,Hangup

 

 include = parkedcalls

 

 #include users.conf

 

  

 

 This is what I am getting for a file name.

 

  

 

 4072493400-20061207-160632.wav

 

  

 

 Caller - timestamp.wav 

 

 But I want to see

 

 Agent(1656)-caller-timestamp.wav

 

  

 

  

 

 Thank you

 

  

 

  

 

  

 

 Ed Nuñez

 

 IT/Telecom Engineer

 

  

 

 4037 Metric Drive

 

 Winter Park, FL

 

  

 

 (o) 407-384-4200 x 1656

 

 (f) 407-384-4222

 

 (c) 732-925-0730

 

 

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