date:20051028

Re: [Asterisk-Users] Words for the Asterisk community to live by.

2005-10-28 Thread Dinesh Nair




On 10/28/05 03:41 Leif Madsen said the following:

I was sitting at my buddies house, and noticed a little sign that he
We provide service which is CHEAP, FAST & PERFECT.


a variation on this has been applied for a long time. CHEAP, FAST and 
QUALITY. pick any two.


--
Regards,   /\_/\   "All dogs go to heaven."
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo "The opinions here in no way reflect the opinions of my $a $b."  |
| done; done  |
+=+
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RE: [Asterisk-Users] SPA3000 as trunk - no caller ID - solved

2005-10-28 Thread Kerry Garrison


Well, we figured it out. It wasn't a factory reset that fixed it either.
Here is the info:

Corrected article:
http://voipspeak.net/index.php?option=com_content&task=view&id=24

The change that got it working was in the Peer Details. We said to put the
IP address of the asterisk server in the host field, but changing it to the
IP address of the SPA-3000 fixed the problem.
-Kerry

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of InetUID
Sent: Thursday, October 27, 2005 9:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID

I've had a very similar thing on my SPA-3000 and they only way to fix it was
a full default reset on the SPA and reconfigure it from scratch 8-(


Matt.

On 27/10/05, Kerry Garrison <[EMAIL PROTECTED]> wrote:
> Upgraded to 3.1.7
>
> Excerpts from Asterisk Log:
>
> Oct 27 07:43:50 DEBUG[1531]: cdr_mysql: SQL command as follows: INSERT 
> INTO cdr 
> (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,du
> ration
> ,billsec,disposition,amaflags,accountcode) VALUES ('2005-10-27 
> 07:43:50','\"Garrison Kerry\"
> <9496799285>','9496799285','s','from-sip-external',
> 'SIP/192.168.5.200-083279d0','','Congestion','',0,0,'NO ANSWER',3,'') 
> Oct 27 07:43:56 VERBOSE[1531]: -- Executing GotoIf("SIP/spa3000-8d99",
> "0?from-pstn-reghours|s|1:") in new stack Oct 27 07:43:56 DEBUG[1531]: 
> Check for res for spa3000 Oct 27 07:43:56 DEBUG[1531]: Call from user 
> 'spa3000' is 1 out of 0 Oct 27 07:43:56 DEBUG[1531]: build_route:
> Contact hop:
> Oct 27 07:43:56 DEBUG[1531]: Expression is '0'
> Oct 27 07:43:56 VERBOSE[1531]: -- Executing GotoIf("SIP/spa3000-8d99",
> "0?from-pstn-reghours|s|1:") in new stack
>
> The log is interesting in that it actually is pushing the CID across 
> but then something strange is happening, if I look at my CDR it shows 
> the
> following:
>
> The call comes in to SIP/192.168.5.200 Source is the correct source 
> phone number, Clid is correct CID, Dst is s, Disposition is NO ANSWER
> 6-7 seconds later it there is another entry The call comes in to 
> SIP/spa3000 Source is now empty, Clid is spa3000, Dst is 201, 
> Disposition is ANSWERED
>
> Here is a link to a screenshot of the SPA3000 settings:
> http://techdatapros.com/temp/spa3000.gif
>
> -Kerry
>
>
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Re: [Asterisk-Users] Outbound fax solution

2005-10-28 Thread Simon Woodhead

Fax over VoIP is just not reliable in my opinion. I'd run with doing it
directly to PSTN as the other poster suggested or via Hylafax. We've
used Hylafax behind Asterisk very succesfully in the past. 

SimonOn 10/29/05, KARIM MOUSLI <[EMAIL PROTECTED]> wrote:
my problem is to triger the transfer to sip provideri always get worng number error*** REPLY SEPARATOR  ***On 28/10/2005 at 20:27 Chris Mason (Lists) wrote:>Teliax works for me, generally. I don't know why but no other provider
>does. I suspect the other translate to G729 and send SIP.>>-->Chris Mason>NetConcepts>(264) 497-5670 Fax: (264) 497-8463>Int:  (305) 704-7249 Fax: (815)301-9759>Cell: 264-235-5670
>Yahoo IM: [EMAIL PROTECTED]>>___>--Bandwidth and Colocation sponsored by 
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Re: [Asterisk-Users] Asterisk CDR

2005-10-28 Thread Jerry Jones


cat asterisk.conf
[directories]
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run
astlogdir => /var/log/asterisk <<<



On Oct 29, 2005, at 12:45 AM, Kanishka Somaratne wrote:

Where does asterisk store the CDR information by default, just  
after a fresh instalation.

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[Asterisk-Users] Asterisk CDR

2005-10-28 Thread Kanishka Somaratne

Where does asterisk store the CDR information by default, just after a fresh 
instalation. 


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RE: [Asterisk-Users] Opinions on IAX JitterBuffer in old-school 1 .0.0?

2005-10-28 Thread steve



On Fri, 28 Oct 2005, Colin Anderson wrote:

> >Does sound like you have the fix - upgrade to a newer Asterisk.
> 
> *groan* Yes, it did solve the problem, 100%. I upgraded a single site to
> 1.0.9 and call quality is perfect. Now, on to the other 29thank GOD for
> SSH. 

As its such a big job, are you SURE you wouldn't rather move them all the 
CVS-HEAD, which is oh-so-nearly 1.2 beta2 and then 1.2 release...?

Steve

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Re: [Asterisk-Users] DTMF recognition unreliable or absent, ringing lost on 2nd half of "3-way" call.

2005-10-28 Thread Ryan


On Thu, Oct 27, 2005 at 02:10:51PM -0400, Dave Grey exclaimed:

>
>These appear to be a common problems, but after spending half a day  
>reading the wiki and list archives I have not gained much useful  
>knowledge beyond the fact that these are a common problems.  I am  
>hoping for some suggestions or pointers to further info.
>
>I have an ivr in my incoming context that does a background() and...  
>well, it is an ivr, no need to explain that, I guess.
>
>So, testing locally, it works wonderfully.  Testing through my DID,  
>provided by IPKall, it is decidedly hit-or-miss.  The digits seem to  
>be either not recognized at all or recognized incorrectly better than  
>half the time. Most often, I get the "invalid extension" playback  
>that I have assigned to the i,1 exten.  For a while, I had two test  
>extensions, one 2000 and one 2001.  Dialing 2001 usually sent me to  
>2000 instead. What is making it hard for me to debug is that it  
>*sometimes* works, recognizing the extension I dialed correctly.
>
>My peer entry in sip.conf for IPKall contains dtmfmode=rfc2833 as per  
>their recommendation.  I have tried setting relaxedtmf=yes in the  
>general section, with no noticeable change.  I turned it off again,  
>since the problem seems to be too much relaxation in any case.   
>Looking at the console, I dial 7056 and it sees 7055, I dial 7056  
>again and it sees 75, I dial 7056 a third time and it sees 706,  
>etc.  Seems random and all over the place.  Packet loss and/or  
>ordering?  Aside from the dtmf issue, incoming calls on the DID work  
>fine and sound excellent.
>

This is a known issue that is fixed in CVS HEAD. Search for my prior
emails to track down the bug numbers. Unfortunately this is going to
require the upstream providers to upgrade to truly fix the issue.
Basically the RTP packets are coming out of order and have the wrong
sequence numbers. I started on a band-aid solution using ip_queue, but I
have not had time to finish it up. I will post here if it ever
materializes.

-Ryan
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RE: [Asterisk-Users] Sipura SPA 2000 - error using second line

2005-10-28 Thread Kerry Garrison




Each line should be its own extension. Then you 
shouldn't get the conflict.


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Kanishka 
SomaratneSent: Friday, October 28, 2005 10:05 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Sipura SPA 
2000 - error using second line

HiI have a 
Sipura SPA 2000 unit and I have configured both the lines in the unit. both 
the lines are configured to use 729.when I make calls from the lines 
independently it works great. no problem at all.when line 1 is 
connected and when I try to make a call using line 2 while line 1 is 
connected I get codec error.what could be the problem , please 
help.I tried this with call the other codecs as well, i still get the 
same error, only when i am tring to make the second active 
callregardskanishka 
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[Asterisk-Users] Sipura SPA 2000 - error using second line

2005-10-28 Thread Kanishka Somaratne




HiI have a 
Sipura SPA 2000 unit and I have configured both the lines in the unit. both 
the lines are configured to use 729.when I make calls from the lines 
independently it works great. no problem at all.when line 1 is 
connected and when I try to make a call using line 2 while line 1 is 
connected I get codec error.what could be the problem , please 
help.I tried this with call the other codecs as well, i still get the 
same error, only when i am tring to make the second active 
callregardskanishka 
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Re: [Asterisk-Users] Mediatrix Gateways

2005-10-28 Thread Jerry Jones

Checkout the Adit 3104. Just installed one this week and now have 6  
more on order. Seems to be working fine and very straightforward,  
just make sure you have the latest code.
It will register with * and each port has a seperate account just  
like a seperate hardphone


it also is a router and firewall with optional vpn. Sip is an option  
also. Comes in 8,16, and 24 port flavors.



On Oct 28, 2005, at 10:25 PM, Rich Adamson wrote:





Was wondering if anyone has used any other gateways with Asterisk.

Such as AudioCodec or Mediatrix.

I would like to set Mediatrix Gateways at my remote sites with an  
Asterisk

Server at my head end and have the calls forwarded to the appropriate
gateways for terminations.

Questions I have is:

Do the Sip Phones at my remote Ends actually register with the  
gateways or
do they still register to my Asterisk Sip Server. Then the SIP  
Server will

route the calls to the appropriate gateway.

Do these gateways work with the Asterisk Servers.

Please let me know if anyone has ever dealt with this before.



Based on tests that I did over a year ago with the Mediatrix, I'd  
be very
careful using it without a firewall. At that time, it could only be  
configured
with SNMP and you could not change the community string. Therefore  
anyone

on the internet could access/change config parameters.

The Mediatrix 1204 did not make use of the register function at  
all. It
must be statically configured to point to asterisk. Same on the  
asterisk
side; asterisk had to be statically configured (no register) to  
point to

the Mediatrix.

To route a call to a specific port on the Mediatrix (eg, port 3),  
you have to
do strange things (like pass callerid of  to the Mediatrix,  
which is then

programmed to route any call with callerid of  to port 3).

Audio quality was very solid; no echo at all. Ongoing support is an  
issue as
they want to charge $$$ for each firmware release. Support is based  
only on
what the reseller of the box can offer; Mediatrix did not have any  
form of

end user support.

They have likely upgraded their firmware, so best check to see if  
the above

is still accurate.


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Re: [Asterisk-Users] Queue Login Out Question

2005-10-28 Thread Kyle Hagan


Im not using AgentCallback function.

Kyle

Johann wrote:


If your using agents, just add something like...

exten => 110,1,Exec(/usr/bin/local/writelog LOGIN ${EXTEN} ${TIMESTAMP})
exten => 110,2,AgentCallbackLogin(${CALLERIDNUM}|[EMAIL PROTECTED])

exten => 120,1,Exec(/usr/bin/local/writelog LOGOUT ${EXTEN} ${TIMESTAMP})
exten => 120,2,AgentCallbackLogin(${CALLERIDNUM})

110 is the login exte, 120 is the logout extension

Write a simple script writelog to do the writing, pass some agruments 
with the data when they login.  You could use an AGI script as well. 
Adjust as needed for your situation.



--johann

Kyle Hagan wrote:

We have 60+ members loged into the queue and talking to 5-10k people 
a day.


I need a better way to track them loggin in and out. The queue_log  
gets really big fast. And has data we dont need. Is there anyother 
way to track when someone loges in and out. I can write to a 
different file when they dial the number to login in the dial plan, 
but I dont see a way to write to a file when they hangup, it doesnt 
continue in the dial plan.


Kyle
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Re: [Asterisk-Users] Mediatrix Gateways

2005-10-28 Thread Rich Adamson


> Was wondering if anyone has used any other gateways with Asterisk.
> 
> Such as AudioCodec or Mediatrix.
> 
> I would like to set Mediatrix Gateways at my remote sites with an Asterisk
> Server at my head end and have the calls forwarded to the appropriate
> gateways for terminations.
> 
> Questions I have is:
> 
> Do the Sip Phones at my remote Ends actually register with the gateways or
> do they still register to my Asterisk Sip Server. Then the SIP Server will
> route the calls to the appropriate gateway.
> 
> Do these gateways work with the Asterisk Servers.
> 
> Please let me know if anyone has ever dealt with this before.

Based on tests that I did over a year ago with the Mediatrix, I'd be very 
careful using it without a firewall. At that time, it could only be configured
with SNMP and you could not change the community string. Therefore anyone
on the internet could access/change config parameters.

The Mediatrix 1204 did not make use of the register function at all. It
must be statically configured to point to asterisk. Same on the asterisk
side; asterisk had to be statically configured (no register) to point to
the Mediatrix.

To route a call to a specific port on the Mediatrix (eg, port 3), you have to
do strange things (like pass callerid of  to the Mediatrix, which is then
programmed to route any call with callerid of  to port 3).

Audio quality was very solid; no echo at all. Ongoing support is an issue as
they want to charge $$$ for each firmware release. Support is based only on
what the reseller of the box can offer; Mediatrix did not have any form of
end user support.

They have likely upgraded their firmware, so best check to see if the above
is still accurate.


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[Asterisk-Users] Geneys

2005-10-28 Thread Jonathan k. Creasy

Anyone using the Genesys framework with an Asterisk PBX? 
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Re: [Asterisk-Users] Outbound fax solution

2005-10-28 Thread KARIM MOUSLI

my problem is to triger the transfer to sip provider
i always get worng number error

*** REPLY SEPARATOR  ***

On 28/10/2005 at 20:27 Chris Mason (Lists) wrote:

>Teliax works for me, generally. I don't know why but no other provider 
>does. I suspect the other translate to G729 and send SIP.
>
>-- 
>Chris Mason
>NetConcepts
>(264) 497-5670 Fax: (264) 497-8463
>Int:  (305) 704-7249 Fax: (815)301-9759 
>Cell: 264-235-5670
>Yahoo IM: [EMAIL PROTECTED] 
>
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BEGIN:VCARD
VERSION:2.1
FN:Karim Mousli
EMAIL;INTERNET:[EMAIL PROTECTED]
TEL:+33 1 34 74 16 56
TEL;FAX:+33 1 34 74 30 21
TEL;CELL:+33 6 08 82 88 62
ORG:V-TEC virtual technologies, Clavister Professional Services
NOTE:Karim Mousli
ADR;PREF:4 place Des penitents;;;Meulan;;78250;FRANCE
ADR;WORK:;;4 place Des penitents;Meulan;;78250;FRANCE
END:VCARD
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Re: [Asterisk-Users] Outbound fax solution

2005-10-28 Thread Chris Mason (Lists)

Teliax works for me, generally. I don't know why but no other provider 
does. I suspect the other translate to G729 and send SIP.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


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RE: [Asterisk-Users] Having Meetme call another conference

2005-10-28 Thread Paul

If you make an outbound then transfer them to the meetme room your done.
Using a cisco phone like the 7960 if you have the flag in the mac file set
cnf_join_enable =1 you can call the party and transfer then then hangup and
the call is bridged.  If you have the cnf_join_enable = 0 and you hangup the
call is disconnected.

Regards,
Paul Norris




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kurt x
Sent: Friday, October 28, 2005 3:15 PM
To: Asterisk
Subject: [Asterisk-Users] Having Meetme call another conference

Is it possible to have a bunch of people call a meetme room then have
that room call
into another conference off net.  T

Kurt
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[Asterisk-Users] asterisk outgoing does not work

2005-10-28 Thread KARIM MOUSLI




i have setup my sip.conf wich is working, between 2 xlite softphones
 
problem i can't get asterisk to dial to sip provider no matter what 
provider i choose
 
the prefix and telephone format is the main problem and i cant figure it 
even thoug i looked at example and di not work for me
 
i took exmple on nufone and net2phone configs !
 
bellow is my extenssions.conf
 
main question how to get to dial for example 1-805-111- or 
33134246424
 
please help
 

[globals]
nufone => SIP/nufone
NUMBER-OF-LINES = 9
[applications]
; Entries for IPSwitchBoard
exten => 77,1,MusicOnHold()
exten => _88XXX,1,Meetme(${EXTEN:2},dMq)
; Activate CFWD Dialing
exten => _*21*.,1,Answer()
exten => _*21*.,n,DBput(CFWD/${CALLERIDNUM}=${EXTEN:4})
exten => _*21*.,n,PlayBack(call-fwd-unconditional)
exten => _*21*.,n,PlayBack(activated)
exten => _*21*.,n,Hangup
; Deactivate CFWD Dialing
exten => **21,1,Answer()
exten => **21,n,DBdel(CFWD/${CALLERIDNUM})
exten => **21,n,PlayBack(call-fwd-unconditional)
exten => **21,n,PlayBack(de-activated)
exten => **21,n,Hangup
; Activate DUAL Dialing
exten => _*22*.,1,Answer()
exten => _*22*.,n,DBput(DUAL/${CALLERIDNUM}=${EXTEN:4})
exten => _*22*.,n,PlayBack(call-fwd-parallel)
exten => _*22*.,n,PlayBack(activated)
exten => _*22*.,n,Hangup
; Deactivate DUAL Dialing
exten => **22,1,Answer()
exten => **22,n,DBdel(DUAL/${CALLERIDNUM})
exten => **22,n,PlayBack(call-fwd-parallel)
exten => **22,n,PlayBack(de-activated)
exten => **22,n,Hangup
; Force Queue Closed
exten => _*11X.,1,Answer()
exten => _*11X.,n,DBPut(ExtensionOpen/${EXTEN:3}=no)
exten => _*11X.,n,UserEvent(ExtOpen,ActionID: extopen)
exten => _*11X.,n,Playback(ipm-ext-closed)
exten => _*11X.,n,Hangup()
; Force Queue Open
exten => _**11X.,1,Answer()
exten => _**11X.,n,DBPut(ExtensionOpen/${EXTEN:4}=yes)
exten => _**11X.,n,UserEvent(ExtOpen,ActionID: extopen)
exten => _**11X.,n,Playback(ipm-ext-open)
exten => _**11X.,n,Hangup()
; Set Queue back to Normal Open State
exten => _***11X.,1,Answer()
exten => _***11X.,n,DBdel(ExtensionOpen/${EXTEN:5})
exten => _***11X.,n,UserEvent(ExtOpen,ActionID: extopen)
exten => _***11X.,n,Playback(ipm-ext-normal)
exten => _***11X.,n,Hangup()
; Force Extension Closed
exten => *11,1,Answer()
exten => *11,n,DBPut(ExtensionOpen/${CALLERIDNUM}=no)
exten => *11,n,UserEvent(ExtOpen,ActionID: extopen)
exten => *11,n,Playback(ipm-ext-closed)
exten => *11,n,Hangup()
; Force Extension Open
exten => **11,1,Answer()
exten => **11,n,DBPut(ExtensionOpen/${CALLERIDNUM}=yes)
exten => **11,n,UserEvent(ExtOpen,ActionID: extopen)
exten => **11,n,Playback(ipm-ext-open)
exten => **11,n,Hangup()
; Set Extension back to Normal Open State
exten => ***11,1,Answer()
exten => ***11,n,DBdel(ExtensionOpen/${CALLERIDNUM})
exten => ***11,n,UserEvent(ExtOpen,ActionID: extopen)
exten => ***11,n,Playback(ipm-ext-normal)
exten => ***11,n,Hangup()
; Set Do not Disturb for Extension
exten => *77,1,Answer()
exten => *77,n,UserEvent(DND,ActionID: dndon/${CALLERIDNUM}/Phone)
exten => *77,n,DBput(dnd/${CALLERIDNUM}=yes)
exten => *77,n,Playback(vm-isunavail)
exten => *77,n,Playback(activated)
exten => *77,n,Hangup()
; Cancel Do not Disturb for Extension
exten => **77,1,Answer()
exten => **77,n,UserEvent(DND,ActionID: dndoff/${CALLERIDNUM})
exten => **77,n,DBdel(dnd/${CALLERIDNUM})
exten => **77,n,Playback(vm-isunavail)
exten => **77,n,Playback(de-activated)
exten => **77,n,Hangup()
; Speaking Clock
exten => *55,1,Answer()
exten => *55,n,Wait(1)
exten => *55,n,SayUnixTime(${EPOCH},,adbR)
exten => *55,n,Hangup()
; Echo Test
exten => *60,1,Playback(demo-echotest)
exten => *60,n,Echo()
exten => *60,n,Playback(demo-echodone)
; Log in Virtual User on Extension
exten => _*99*.,1,Answer()
exten => _*99*.,n(chkLoggedIn),DBget(EXT=user/${EXTEN:4})
exten => _*99*.,n,DBdel(userext/${EXT})
exten => _*99*.,n,DBdel(user/${EXTEN:4})
exten => _*99*.,n,Goto(notLoggedIn)
exten => 
_*99*.,chkLoggedIn+101(notLoggedIn),DBput(user/${EXTEN:4}=${CALLERIDNUM})
exten => _*99*.,n,DBput(userext/${CALLERIDNUM}=${EXTEN:4})
exten => _*99*.,n,PlayBack(agent-loginok)
exten => _*99*.,n,Hangup
; Log out Virtual User from Extension
exten => **99,1,Answer()
exten => **99,n,DBget(EXT=userext/${CALLERIDNUM})
exten => **99,n,DBdel(userext/${CALLERIDNUM})
exten => **99,n,DBdel(user/${EXT})
exten => **99,n,PlayBack(agent-loggedoff)
exten => **99,n,Hangup
; VoiceMailMain for Extension
exten => *,1,Answer()
exten => *,n(chkVoiceMail),DBGet(EXT=userext/${CALLERIDNUM})
exten => *,n,VoiceMailMain(s${EXT})
exten => *,chkVoiceMail+101,VoiceMailMain(s${CALLERIDNUM})
; VoiceMailMain for All SIPExtensions
exten => **,1,Answer()
exten => **,n,VoiceMailMain()
; VoiceMailMain
exten => asterisk,1,VoiceMailMain(s${CALLERIDNUM})
[phones]
; Now follows a lot of hint's - they make the buttons on SNOM phones work
exten => 1001,hint,SIP/1001 ; Karim Mousli
exten => 1002,hint,SIP/1002 ; Bruno maimbourg
exten => _.,1,SetLanguage(us)
exten => _.,n,Setvar(CALLEEIDNUM=${EXTEN})
exten => _.,n,Setvar(CALLERNUM=${CALLERI

[Asterisk-Users] anyone using these?

2005-10-28 Thread Jonathan k. Creasy

Voicetronix OpenSwitch6
http://www.telephonyware.com/telephonyware/tw3.html
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RE: [Asterisk-Users] Outbound fax solution

2005-10-28 Thread Kerry Garrison

Is there a VOIP provider that can handle fax calls? This would greatly
reduce the cost per fax.
-Kerry
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Howard
Sent: Friday, October 28, 2005 4:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Outbound fax solution

Kerry Garrison wrote:

> Got a client building a system that needs to send out hundreds of 
> faxes per day (not, not junk faxes). We have just implemented an 
> asterisk server for the client for their office and they asked if 
> there was an outbound fax solution that would utilize VOIP providers
> (<$0.02/minute) instead of internet based fax providers ($0.08/page). 
> Does anyone have any thoughts on this?


Your client should get their own PSTN trunks (you can easily push hundreds
of faxes per day out of a few analog lines) and move your faxing services
in-house.

My recommendation would be to get some new Class 1 fax modems like
MultiTechs or MainPines and use HylaFAX.

Lee.
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[Asterisk-Users] Warning - AgentCallBackLogin has changed, Possibly will cause 99% cpu

2005-10-28 Thread Julian Lyndon-Smith

As a result of a machine failure today we had to move to another server. 
The original server was running cvs-head from about June, the new server 
cvs-head as of today.


If you are using AgentCallBackLogin, please be aware that there was a 
change to the parameters used by this application made a couple of days 
ago. This is from the UPGRADE.txt file:


* The AgentCallBackLogin application now requires a second '|' before
  specifying an [EMAIL PROTECTED]  This is to distinguish the options
  string from the extension, so that they do not conflict.  See
  'show application AgentCallbackLogin' for more details.

This is *extremely* important - your dialplan will not work unless you 
change the application parameters as described. In fact, for us, using 
the old parameters caused a) Agent logins not to work and b) 99% cpu usage.


I know that this is documented in the readme, but it probably needs to 
be shouted from the rooftop that this is one change that you must 
absolutely must make when moving to 1.2 (or cvs-head as of today)


Julian
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Re: [Asterisk-Users] Outbound fax solution

2005-10-28 Thread Lee Howard


Kerry Garrison wrote:

Got a client building a system that needs to send out hundreds of 
faxes per day (not, not junk faxes). We have just implemented an 
asterisk server for the client for their office and they asked if 
there was an outbound fax solution that would utilize VOIP providers 
(<$0.02/minute) instead of internet based fax providers ($0.08/page). 
Does anyone have any thoughts on this?



Your client should get their own PSTN trunks (you can easily push 
hundreds of faxes per day out of a few analog lines) and move your 
faxing services in-house.


My recommendation would be to get some new Class 1 fax modems like 
MultiTechs or MainPines and use HylaFAX.


Lee.
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RE: [Asterisk-Users] Sipura 841 echo cancel question

2005-10-28 Thread Kris Boutilier









You’re probably looking for the ‘echotraining’
facility specified in /etc/asterisk/Zapata.conf. From that file:

 

; In some cases, the echo canceller
doesn't train quickly enough and there

; is echo at the beginning of the call. 
Enabling echo training will cause

; asterisk to briefly mute the channel,
send an impulse, and use the impulse

; response to pre-train the echo canceller
so it can start out with a much

; closer idea of the actual echo.  Value
may be "yes", "no", or a number of

; milliseconds to delay before training
(default = 400)

;

;echotraining=yes

 

Hope that helps.

 

Kris Boutilier

Information Services Coordinator

Sunshine Coast Regional District

 











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nora Lavelle
Sent: Friday, October 28, 2005
11:23 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Sipura
841 echo cancel question



 

Hi there, 

 

I’m new to asterisk and hoping you can help out. I
have a small deployment of asterisk running. 4 sipura 841 phones and a linux
box with a digium TDM400P.  When a user makes a call there is usually echo
for about 15 seconds and then it goes away. I have read through all the echo
stuff and to be honest totally confused.  Not sure what to set or how to
test. 

 

Any guidance totally appreciated !  Thanks in advance !


Nora

 








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Re: [Asterisk-Users] Top and asterisk performance

2005-10-28 Thread Julian Lyndon-Smith

Oh man, stupid of me - I forgot to include that info at the start. 
However, there are a couple of things that come out of the analysis:


1) I've restarted the * server. I now have

top - 00:29:08 up 13 days, 10:50,  1 user,  load average: 0.00, 0.01, 1.04
Tasks:  83 total,   1 running,  82 sleeping,   0 stopped,   0 zombie
Cpu(s):  0.3% us,  0.0% sy,  0.0% ni, 99.7% id,  0.0% wa,  0.0% hi,  0.0% si
Mem:   1034640k total,   270408k used,   764232k free,42524k buffers
Swap:  2031608k total,0k used,  2031608k free,   153256k cached

*much* better :)

2) There were about 10 channels that were constantly trying to call: It 
turns out that the agentcallbacklogin has had a change of parameters in 
the past couple of days, one that basically meant that the original 
dialplan caused the system to start eating cpu cycles. So my old dial 
plan seemed to cause the issues. I did realise that there was a problem 
early on, in that agents couldn't login. After a (quick!) bit of 
research, I discovered that the dialplan needed changing, changed it, 
did a "extensions reload" and voila! Agents could log in. I forgot about 
the 10 or so "crashed" (see below) channels.


foxtrot*CLI> show channels
Channel  Location State   Application(Data)
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:3BusyBusy()
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1   Down(None)
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:3BusyBusy()
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1   Down(None)
SIP/711-6416 [EMAIL PROTECTED]:1  Ring 
AgentCallbackLogin(711|@AgentQ
SIP/711-8145 [EMAIL PROTECTED]:1 Ring 
AgentCallbackLogin(|[EMAIL PROTECTED]
SIP/711-5ff9 [EMAIL PROTECTED]:1  Ring 
AgentCallbackLogin(711|@AgentQ
SIP/6035-3c31[EMAIL PROTECTED]:1 Up 
AgentCallbackLogin(6035|[EMAIL PROTECTED]
SIP/6033-a08c[EMAIL PROTECTED]:1 Up 
AgentCallbackLogin(6033|[EMAIL PROTECTED]
SIP/6025-c55c[EMAIL PROTECTED]:1 Up 
AgentCallbackLogin(6025|[EMAIL PROTECTED]
SIP/6035-4ef0[EMAIL PROTECTED]:1 Up 
AgentCallbackLogin(6035|[EMAIL PROTECTED]


At midnight, * automatically restarts, and now top is showing the 
results above. I can only assume that it was the result of those channels.


[EMAIL PROTECTED] html]# vmstat 5 5
procs ---memory-- ---swap-- -io --system-- 
cpu
 r  b   swpd   free   buff  cache   si   sobibo   incs us 
sy id wa
 0  0  0 764280  42584 15319600 117   2914  5 
4 91  0
 0  0  0 764280  42584 15319600 0 9 2051   143  0 
0 99  1
 0  0  0 764280  42584 15319600 030 2026   124  0 
0 100  0
 0  0  0 764280  42584 15319600 0 0 2022   122  0 
0 100  0
 0  0  0 764280  42588 15319200 018 2023   134  0 
0 99  0



I also assume that the following is normal (something about threads ...)

[EMAIL PROTECTED] html]# ps -ef|fgrep asterisk
root 31462 1  0 00:00 ?00:00:00 /bin/sh 
/usr/sbin/safe_asterisk

root 31473 31462  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31514 31473  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31516 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31517 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31518 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31519 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31520 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31521 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31522 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31523 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31524 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31525 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31526 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31527 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31528 31514  0 00:00 ?00:00:01 /usr/sbin/asterisk -vvvg -c
root 31529 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31530 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31531 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31532 31514  0 00:00 ?00:00:00 /usr/sbin/asterisk -vvvg -c
root 31646 30821  0 00:34 pts/000:00:00 fgrep asterisk

Many thanks for all the help.

Julian.

Steve Kann wrote:

Julian Lyndon-Smith wrote:

We had to move from a old * server to a new one in a hurry (hardware 
failure). The old server was a dual pentium 700 with 512MB ram running 
fedora core 2, the new one is a single 3GHz Pentium with 1gb ram.


The same number of people are connected to the new server as the old, 
the s

Re: [Asterisk-Users] Mediatrix Gateways

2005-10-28 Thread contact

Hi,

I use Multitech gateways with asterisk. The pbx is taking care of all, register 
the phones, incoming and outgoing from voip gateways.

Best regards,

Chris HARIGA

--- Original Message ---
From: [EMAIL PROTECTED]
Sent: Fri 10/28/2005 1:42 pm
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Mediatrix Gateways

Was wondering if anyone has used any other gateways with Asterisk.

Such as AudioCodec or Mediatrix.

I would like to set Mediatrix Gateways at my remote sites with an Asterisk
Server at my head end and have the calls forwarded to the appropriate
gateways for terminations.

Questions I have is:

Do the Sip Phones at my remote Ends actually register with the gateways or
do they still register to my Asterisk Sip Server. Then the SIP Server will
route the calls to the appropriate gateway.

Do these gateways work with the Asterisk Servers.

Please let me know if anyone has ever dealt with this before.

Thanks.


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RE: [Asterisk-Users] I need GoIAX or VoipBuster [EMAIL PROTECTED] examples?

2005-10-28 Thread Kerry Garrison

http://voipspeak.net has GOIax example for AMP.
-Kerry

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Francesco
Peeters
Sent: Friday, October 28, 2005 3:28 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] I need GoIAX or VoipBuster [EMAIL PROTECTED] examples?

Hi all,

I am playing with [EMAIL PROTECTED] / AMP for the first time, and would like to
test GoIAX or VoipBuster (or both) through the use of the AMP interface.

I have tried several permutations of the possible configs, and sometimes it
logs in to GoIAX and VoipBuster, sometimes it doesn't, however it never
seems to be able to actually set up an IAX channel to start the call.

Using the VoipBuster client or IAXcomm for GoIAX I can dial out just fine.
The FOP shows the Extension calling a number, and shows one of the trunks
(currently always VoipBuster, even though the GoIAX trunk is primary in the
Out Route), but it never actually connects (I don't get a ringtone) and
after a while I get the all channels busy signal.

I can provide the current configs and full log data if anybody wants to have
a look, but I really would prefer to understand myself, and hope a quick
example will help me understand...

TIA!

--
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704 If your program
doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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Re: [Asterisk-Users] CallerID strings comprised of "%23..."

2005-10-28 Thread Darren Wiebe

I only have the answer to your last question.  From my experience, I 
would go for "arbitrary barf".  I don't think you are supposed to get 
anything if there is not a caller id passed.


Darren

Dave Grey wrote:

Well, I am batting close to zero where responses to my questions are  
concerned, but I suppose I will just keep swinging.


I just set up an account with callpacket.com, and noticed that on  
incoming calls through this provider the values of CALLERID(name) and  
CALLERID(num) are "%23%23%23%23%23%23%23%23%23%23" when the caller  
has either blocked callerid (tested with *67), or, apparently, sent  
values that are unexpected (tested via friend who is, for whatever  
reason, doing SetCallerID("caller 6398A" <>) on his outbound calls).


I have "speak caller ID" macro that does a system() call to a script  
on the local machine, and I have been tinkering with ways of handling  
the different possible strings in some reasonably intelligent way.   
My question is -- is %23 the escape for the # character here, as I  
suspect, and if so, is there a way I can tell asterisk to interpret  
it as such, or do I need to convert it back on my own?


Is the "%23%23%23%23%23%23%23%23%23%23" (or "###") any kind  
of an industry standard string, that evaluates to something sensible  
on a consumer CID display, or is it just some arbitrary barf that  
callpacket has chosen to send in those cases?


Thanks for any info.

lyd
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[Asterisk-Users] Outbound fax solution

2005-10-28 Thread Kerry Garrison




Got a client building a system that needs to send out 
hundreds of faxes per day (not, not junk faxes). We have just implemented an 
asterisk server for the client for their office and they asked if there was an 
outbound fax solution that would utilize VOIP providers 
(<$0.02/minute) instead of internet based fax providers ($0.08/page). 
Does anyone have any thoughts on this?
 
-Kerry
 
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[Asterisk-Users] I need GoIAX or VoipBuster [EMAIL PROTECTED] examples?

2005-10-28 Thread Francesco Peeters

Hi all,

I am playing with [EMAIL PROTECTED] / AMP for the first time, and would like
to test GoIAX or VoipBuster (or both) through the use of the AMP
interface.

I have tried several permutations of the possible configs, and sometimes
it logs in to GoIAX and VoipBuster, sometimes it doesn't, however it never
seems to be able to actually set up an IAX channel to start the call.

Using the VoipBuster client or IAXcomm for GoIAX I can dial out just fine.
The FOP shows the Extension calling a number, and shows one of the trunks
(currently always VoipBuster, even though the GoIAX trunk is primary in
the Out Route), but it never actually connects (I don't get a ringtone)
and after a while I get the all channels busy signal.

I can provide the current configs and full log data if anybody wants to
have a look, but I really would prefer to understand myself, and hope a
quick example will help me understand...

TIA!

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
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Re: [Asterisk-Users] Help with Zultys

2005-10-28 Thread John Novack


Google finds:

http://www.zultys.com/index.jsp

Linc Fessenden wrote:


Hi everyone!
I just got a zultys zip 2 today with no manuals.  Can anyone tell me 
how to get in and config the=is thing please?  I know there has to be 
some *super secret code* to enable dhcp on it somehow and then a login 
as password for the web interface or something?  HELP!!???



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[Asterisk-Users] CallerID strings comprised of "%23..."

2005-10-28 Thread Dave Grey

Well, I am batting close to zero where responses to my questions are  
concerned, but I suppose I will just keep swinging.


I just set up an account with callpacket.com, and noticed that on  
incoming calls through this provider the values of CALLERID(name) and  
CALLERID(num) are "%23%23%23%23%23%23%23%23%23%23" when the caller  
has either blocked callerid (tested with *67), or, apparently, sent  
values that are unexpected (tested via friend who is, for whatever  
reason, doing SetCallerID("caller 6398A" <>) on his outbound calls).


I have "speak caller ID" macro that does a system() call to a script  
on the local machine, and I have been tinkering with ways of handling  
the different possible strings in some reasonably intelligent way.   
My question is -- is %23 the escape for the # character here, as I  
suspect, and if so, is there a way I can tell asterisk to interpret  
it as such, or do I need to convert it back on my own?


Is the "%23%23%23%23%23%23%23%23%23%23" (or "###") any kind  
of an industry standard string, that evaluates to something sensible  
on a consumer CID display, or is it just some arbitrary barf that  
callpacket has chosen to send in those cases?


Thanks for any info.

lyd
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[Asterisk-Users] SIP Host "Unspecified"

2005-10-28 Thread Mark Hulber


In recent CVS Head build when I run: "sip show peers" my dynamic peers show:

Name/username  HostDyn Nat ACL Port Status   
sipura2_2/sipura2_2(Unspecified)D   N  0UNKNOWN  
sipura2_1/sipura2_1(Unspecified)D   N  0UNKNOWN  
sipura1_2/sipura1_2(Unspecified)D  0UNKNOWN  
sipura1_1/sipura1_1(Unspecified)D  0UNKNOWN  

So I can no longer see which IP they are at and I can't see the qualify 
times.  If this the new designed behavior?


Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running 
Linux on 2005-10-28 19:30:31 UTC


MARK.


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Re: [Asterisk-Users] Top and asterisk performance

2005-10-28 Thread Mark Hulber


It might be helpful to show what's using the CPU (the rest of Top).

MARK.

Julian Lyndon-Smith wrote:
We had to move from a old * server to a new one in a hurry (hardware 
failure). The old server was a dual pentium 700 with 512MB ram running 
fedora core 2, the new one is a single 3GHz Pentium with 1gb ram.


The same number of people are connected to the new server as the old, 
the same number of inbound calls to the isdn30 etc (on average 20 
calls active at any time (SIP and ZAP)). Basically, just a server 
swapout.


I must be reading top wrong, because the old server had a idle of 
approx 30%, whereas the new server is


top - 13:35:21 up 12 days, 23:57,  1 user,  load average: 7.11, 7.20, 
7.21

Tasks:  98 total,   9 running,  89 sleeping,   0 stopped,   0 zombie
Cpu(s): 99.0% us,  1.0% sy,  0.0% ni,  0.0% id,  0.0% wa,  0.0% hi,  
0.0% si

Mem:   1034640k total,   144792k used,   889848k free,21952k buffers
Swap:  2031608k total,0k used,  2031608k free,61248k cached

Notice the 99.0% us. This fluctuates between 80 and 99%.

The other difference is that the new server is on cvs-head as of today 
- I did say that it was an emergency :) whereas the old server was 
cvs-head from june sometime.


Is it just me, or is there a problem ?

Julian.
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[Asterisk-Users] Why can't I dial - just using SIP internally

2005-10-28 Thread Angus Comber


Hello

I have setup a couple of sip accounts - here is my sip.conf:
context=default
disallow=all
allow=ulaw
allow=alaw
allow=gsm

[200]
username=200
type=friend
secret=1234
port=5060
nat=never
[EMAIL PROTECTED]
dtmfmode=rfc2833
context=default
callerid="Angus" <200>
host=dynamic
insecure=very
group=1
callgroup=1
pickupgroup=1

[201]
username=201
type=friend
secret=1234
port=5060
nat=never
dtmfmode=rfc2833
context=default
callerid="Lisa" <201>
host=dynamic
insecure=very
group=1
callgroup=1
pickupgroup=1





my extensions.conf:

[frompstnanalog]
exten => 787367,1,Dial(SIP/200,1)
exten => 787367,2,Voicemail(su200)
exten => 787367,3,Hangup


[default]
;exten => _X.,1,Dial(ZAP/g1/${EXTEN},20,Ttm)
;exten => _X.,2,Hangup

exten => _2XX,1,Dial(SIP/${EXTEN},20,Ttm)
exten => _2XX,2,Voicemail(su${EXTEN})
exten => _2XX,3,Hangup

exten => *97,1,Answer
exten => *97,2,VoicemailMain([EMAIL PROTECTED])
exten => *97,3,Hangup


I have setup two IP phones, they register OK but cannot dial each other.  I
had to switch on sip debug to get anything on the asterisk console:

pbx*CLI>
<-- SIP read from 192.168.0.21:5060:
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK-17vz3vxf4uhz;rport
From: "Angus" ;tag=oa5ljlnorj
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Max-Forwards: 70
Contact: 
P-Key-Flags: keys="3"
User-Agent: snom190-3.56m
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 342

v=0
o=root 2065976712 2065976712 IN IP4 192.168.0.21
s=call
c=IN IP4 192.168.0.21
t=0 0
m=audio 1 RTP/AVP 0 8 3 18 4 9 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:9 g722/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

--- (18 headers 16 lines)---
Using INVITE request as basis request -
[EMAIL PROTECTED]
Sending to 192.168.0.21 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.0.21:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.0.21:5060;branch=z9hG4bK-17vz3vxf4uhz;rport;received=192.168.0.21
From: "Angus" ;tag=oa5ljlnorj
To: ;tag=as7203b20e
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: 
Proxy-Authenticate: Digest realm="asterisk", nonce="24b5d1a5"
Content-Length: 0


---
Scheduling destruction of call '[EMAIL PROTECTED]' in
15000 ms
Found user '200'
pbx*CLI>
<-- SIP read from 192.168.0.21:5060:
ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK-17vz3vxf4uhz;rport
From: "Angus" ;tag=oa5ljlnorj
To: ;tag=as7203b20e
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK
Max-Forwards: 70
Contact: 
Content-Length: 0


--- (9 headers 0 lines)---
pbx*CLI>
<-- SIP read from 192.168.0.21:5060:
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK-ass0mb36ivsw;rport
From: "Angus" ;tag=oa5ljlnorj
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
Max-Forwards: 70
Contact: 
P-Key-Flags: keys="3"
User-Agent: snom190-3.56m
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Proxy-Authorization: Digest
username="200",realm="asterisk",nonce="24b5d1a5",uri="sip:[EMAIL 
PROTECTED];user=phone",response="a5598b627eb4c3bad2084bd553daad3f",algorithm=md5
Content-Type: application/sdp
Content-Length: 342

v=0
o=root 2065976712 2065976712 IN IP4 192.168.0.21
s=call
c=IN IP4 192.168.0.21
t=0 0
m=audio 1 RTP/AVP 0 8 3 18 4 9 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:9 g722/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

--- (19 headers 16 lines)---
Using INVITE request as basis request -
[EMAIL PROTECTED]
Sending to 192.168.0.21 : 5060 (non-NAT)
Found user '200'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 9
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.21:1
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format g729
Found description format g723
Found description format g722
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10f
(g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe
(gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 201 in default (domain 192.168.0.20)
Reliably Tra

Re: [Asterisk-Users] chan_bluetooth and audio problem

2005-10-28 Thread Vlasis Hatzistavrou


Hello,

We had similar problems with chan_bluetooth and various mobile devices.

I suppose that chan_bluetooth is in a very early stage. We tried to 
contact the author of the channel with debugging information etc but 
without luck...


There is also the chance that the project may be stalled...

Best regards,
Vlasis Hatzistavrou.

José Luis Gómez wrote:


Hello.
I'm having problem with motorola v635 and asterisk. I can make a call
but I can't hear any audio and the other side of the call can hear me
(one way audio).
I'm using usb to bluetooth adaptor (noganet).
I'm using gentoo with kernel 2.6.13-r2, asterisk 1.0.9 and
chan_bluetooth 0.0.1_pre20050212.
What's may be wrong?
   
I show you my files:

- bluetooth.conf:
[general]
interface = 0
[00:15:A8:A8:19:82]
name= V635
type= HS
channel = 3
autoconnect = yes
# If I put channel 7, the other side of the call can't hear me (no
audio). The audio stay on the phone (I can hear the call on phone).

- hcid.conf
options {
   autoinit yes;
   security auto;
   pairing multi;
   pin_helper /usr/bin/bluepin;
}
device {
   name "Asterisk";
   class 0x200404;
   iscan enable;
   pscan enable;
   lm accept;
   lp rswitch,hold,sniff,park;
}

- rfcomm.conf
rfcomm0 {
   bind yes;
   device xx:xx:xx:xx:xx:xx;
   channel 7;
   comment "motoV635";
}

Thanks in advance.
 José Luis


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[Asterisk-Users] David Choo/eServices/eSpore is overseas

2005-10-28 Thread David Choo


I will be out of the office starting  29/10/2005 and will not return until
13/11/2005.

Dear Sir / Mdm,

I'm currently out of office.

During this period of time, I have minimal access to internet and email
cccess. As such, I might not be able to reply to your queries promptly. I
apologise for the inconvenience caused.

In the meantime, for any technical assitance, please contact the Espore
Technical Support Hotline at +65-68422725 and select option 2.

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[Asterisk-Users] problem with asterisk-realtime-odbc

2005-10-28 Thread J Mauricio Mejia Vargas

hello:  I need guide with the following question.

I am trying to use asterisk - realtime using ODBC, and when I make a call to
number  the CLI show the next error:

""Oct 28 15:17:34 WARNING[7034]: config.c:893 find_engine: Realtime mapping for
'sippeers' found to engine 'odbc', but the engine is not available
Oct 28 15:17:34 NOTICE[7034]: chan_sip.c:9835 handle_request_register:
Registration from '' failed for '192.168.10.109' - Wrong
password
Oct 28 15:17:36 WARNING[7034]: config.c:893 find_engine: Realtime mapping for
'sipusers' found to engine 'odbc', but the engine is not available
Oct 28 15:17:36 WARNING[7034]: config.c:893 find_engine: Realtime mapping for
'sippeers' found to engine 'odbc', but the engine is not available""

I make the next configuration:
(software) 
asterisk 1.2.0-beta1; asterisk ADDONS 1.2.0-beta1; UNIXodbc 2,2,11; UNIXodbc -
devel 2,2,11; Fedora 4; MySQL 4,1,11 FC4 

steps that have become.  
***

1.I created database with name "asterisk", inside this I make a table
"sip_buddies", a user of the database with name "asterisk" without password has
all the permissions.  
2. I have entered data to the database manually to the table "sip_buddies" 
3. I modified the files" ODBC.INI, ODBCINST.INI" located in /etc  with the next
information
**
file ODBC.INI:  
[MySQL-asterisk]
Description = MySQL Asterisk database
Trace   = Yes
TraceFile   = /var/log/odbc.log
Driver  = MySQL
SERVER  = localhost
USER= asterisk
PASSWORD=
#PORT= 3306
DATABASE= asterisk
Socket  = /var/lib/mysql/mysqld.sock
**
file ODBCINST.INI
[PostgreSQL]
Description = ODBC for PostgreSQL
Driver  = /usr/lib/libodbcpsql.so
Setup   = /usr/lib/libodbcpsqlS.so
FileUsage   = 1
[MySQL]
Description = ODBC for MySQL
Driver  = /usr/lib/libmyodbc.so
Setup   = /usr/lib/libodbcmyS.so
FileUsage   = 1
***
4.  After that I modified the files "res_odbc.conf, res_mysql.conf" located in
/etc/asterisk.  
*
file: res_odbc.conf
[general]
dbhost = 192.168.10.29
dbname = asterisk
dbuser = asterisk
dbpass = ""
dbport = 3306
dbsock = /tmp/mysql.sock

file: res_mysql.conf
;[asterisk]
;dsn => asterisk
;username => myuser
;password => mypass
;pre-connect => yes
[mysql1]
dsn => MySQL-asterisk
username => asterisk
;password => mypass
pre-connect => yes
*
5. after that I setup one telephone GRANDSTREAM 102 using info from the
database. 
6. execute asterisk -rc
7. I look the error:
Oct 28 15:17:34 WARNING[7034]: config.c:893 find_engine: Realtime mapping for
'sippeers' found to engine 'odbc', but the engine is not available
Oct 28 15:17:34 NOTICE[7034]: chan_sip.c:9835 handle_request_register:
Registration from '' failed for '192.168.10.109' - Wrong
password
Oct 28 15:17:36 WARNING[7034]: config.c:893 find_engine: Realtime mapping for
'sipusers' found to engine 'odbc', but the engine is not available
Oct 28 15:17:36 WARNING[7034]: config.c:893 find_engine: Realtime mapping for
'sippeers' found to engine 'odbc', but the engine is not available

If you know where the problem is...please tell to me.


-- 
Atentamente

J Mauricio Mejia Vargas
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[Asterisk-Users] Metreos

2005-10-28 Thread Dean Collins









http://www.metreos.com/products/application-suite.shtml

 

Interesting company, commercializing apps
for cisco, funny how people are paying for this stuiff when most of it is
available on asterisk for free.

 

Cheers,

Dean

 






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Re: [Asterisk-Users] Help with Zultys

2005-10-28 Thread Dustin Wenz

The ZIP2 does DHCP out of the box. So if it's new, all you have to do is plug it in and it will get an address from your DHCP server. Run an nmap scan in your DHCP range to find it if you don't know what address was assigned; the MAC will probably start with "00:01:e1". From there you can use the http interface to configure it.I'm using one right now with Asterisk, and it's pretty straightforward, though I can't seem to get the "Subscribe Message Waiting Service" feature to work.    - .DustinOn Oct 28, 2005, at 11:56 AM, Linc Fessenden wrote:Hi everyone!I just got a zultys zip 2 today with no manuals.  Can anyone tell me how to get in and config the=is thing please?  I know there has to be some *super secret code* to enable dhcp on it somehow and then a login as password for the web interface or something?  HELP!!???-- -Linc FessendenIn the Beginning there was nothing, which exploded - Yeah right...___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___
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Re: [Asterisk-Users] ANNOUNCEMENT : A2Billing - AreskiCC V3 new release

2005-10-28 Thread Areski K

[Asterisk-Users] Having Meetme call another conference

2005-10-28 Thread kurt x

Is it possible to have a bunch of people call a meetme room then have
that room call
into another conference off net.  T

Kurt
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Re: [Asterisk-Users] Top and asterisk performance

2005-10-28 Thread Steve Kann


Julian Lyndon-Smith wrote:

We had to move from a old * server to a new one in a hurry (hardware 
failure). The old server was a dual pentium 700 with 512MB ram running 
fedora core 2, the new one is a single 3GHz Pentium with 1gb ram.


The same number of people are connected to the new server as the old, 
the same number of inbound calls to the isdn30 etc (on average 20 
calls active at any time (SIP and ZAP)). Basically, just a server 
swapout.


I must be reading top wrong, because the old server had a idle of 
approx 30%, whereas the new server is


top - 13:35:21 up 12 days, 23:57,  1 user,  load average: 7.11, 7.20, 
7.21

Tasks:  98 total,   9 running,  89 sleeping,   0 stopped,   0 zombie
Cpu(s): 99.0% us,  1.0% sy,  0.0% ni,  0.0% id,  0.0% wa,  0.0% hi,  
0.0% si

Mem:   1034640k total,   144792k used,   889848k free,21952k buffers
Swap:  2031608k total,0k used,  2031608k free,61248k cached

Notice the 99.0% us. This fluctuates between 80 and 99%.

The other difference is that the new server is on cvs-head as of today 
- I did say that it was an emergency :) whereas the old server was 
cvs-head from june sometime.


Is it just me, or is there a problem ?



Well, I suppose it depends on what's using all that CPU -- you leave 
that part out.


Assuming it's asterisk threads causing the 100% cpu usage, and the load 
average of 7, then, yes, that's a lot of CPU.


But, you could have some other program/processes doing that, and if 
they're batch processes (they're clearly not niced), they may end up 
with a lower dynamic priority and not affect asterisk too much.


-SteveK



Julian.
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Re: [Asterisk-Users] Queue Login Out Question

2005-10-28 Thread Johann


If your using agents, just add something like...

exten => 110,1,Exec(/usr/bin/local/writelog LOGIN ${EXTEN} ${TIMESTAMP})
exten => 110,2,AgentCallbackLogin(${CALLERIDNUM}|[EMAIL PROTECTED])

exten => 120,1,Exec(/usr/bin/local/writelog LOGOUT ${EXTEN} ${TIMESTAMP})
exten => 120,2,AgentCallbackLogin(${CALLERIDNUM})

110 is the login exte, 120 is the logout extension

Write a simple script writelog to do the writing, pass some agruments 
with the data when they login.  You could use an AGI script as well. 
Adjust as needed for your situation.



--johann

Kyle Hagan wrote:

We have 60+ members loged into the queue and talking to 5-10k people a day.

I need a better way to track them loggin in and out. The queue_log  gets 
really big fast. And has data we dont need. Is there anyother way to 
track when someone loges in and out. I can write to a different file 
when they dial the number to login in the dial plan, but I dont see a 
way to write to a file when they hangup, it doesnt continue in the dial 
plan.


Kyle
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[Asterisk-Users] Mediatrix Gateways

2005-10-28 Thread gorand

Was wondering if anyone has used any other gateways with Asterisk.

Such as AudioCodec or Mediatrix.

I would like to set Mediatrix Gateways at my remote sites with an Asterisk
Server at my head end and have the calls forwarded to the appropriate
gateways for terminations.

Questions I have is:

Do the Sip Phones at my remote Ends actually register with the gateways or
do they still register to my Asterisk Sip Server. Then the SIP Server will
route the calls to the appropriate gateway.

Do these gateways work with the Asterisk Servers.

Please let me know if anyone has ever dealt with this before.

Thanks.


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[Asterisk-Users] Sipura 841 echo cancel question

2005-10-28 Thread Nora Lavelle









Hi there, 

 

I’m new to asterisk and hoping you can help out. I
have a small deployment of asterisk running. 4 sipura 841 phones and a linux
box with a digium TDM400P.  When a user makes a call there is usually echo
for about 15 seconds and then it goes away. I have read through all the echo
stuff and to be honest totally confused.  Not sure what to set or how to
test. 

 

Any guidance totally appreciated !  Thanks in advance !


Nora

 






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Re: [Asterisk-Users] Asterisk GUI/web interfaces that don't change config files

2005-10-28 Thread Dan Littlejohn

On 10/28/05, Dustin Wildes <[EMAIL PROTECTED]> wrote:
> Stay tuned for PhoneCALL's 2.7-RC1 release scheduled soon. We're adding
> a new Security Manager that allows you to set the levels of editing for
> your users/admins.
>
>
> Chris Bagnall wrote:
>
> >Hello all,
> >
> >I'm trying to find an Asterisk web interface (or windows gui interface) to
> >asterisk that won't allow users to go making changes to config files. I've
> >trawled through the very extensive list in the wiki, but there doesn't seem
> >to be a clear defining line between applications that are purely status
> >viewers and ones that will allow config changes.
> >
> >I'm looking for the user to be able to do fairly simple things like see the
> >last few people who called them, find out if other extensions are busy, add
> >entries to the CLID directory and so on.
> >
> >Thanks in advance folks.
> >
> >Regards,
> >
> >Chris
> >
> >
>
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ARI allows for a user level experience.
  http://www.littlejohnconsulting.com/?q=node/11

Dan Littlejohn
[EMAIL PROTECTED]
www.littlejohnconsulting.com
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Re: [Asterisk-Users] Echo Canceller question- is there a viablesolution?

2005-10-28 Thread Matthew Fredrickson

Don't thank me, it's Mgernoth and kb1_kanobe that get the props for all 
of this.  They've been doing a lot of work to improve the software echo 
cancelers lately.


Matthew Fredrickson

On Oct 28, 2005, at 1:29 AM, [EMAIL PROTECTED] wrote:


Hello Matthew,
It is always nice to see improvements.  I look forward to testing your
patches.

It just seems that so many other hardware manufacturers have tackled 
the

problem, I am surprised digium has not put more research into getting
the issue solved in software, which is possible, as opposed to coming 
up

with alternate solutions.

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Fredrickson
Sent: Thursday, October 27, 2005 1:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Echo Canceller question- is there a
viablesolution?


On Oct 27, 2005, at 12:38 AM, [EMAIL PROTECTED] wrote:

My question is, what is the direction in relation to analog boards and



such?


Right now, it looks like the current fad of the asterisk group is
hardware echo cancelation.  However, there is work that is occurring on
the software echo cans to improve them.  In fact, I just committed
basically an update to
KB1
(which was until now the latest and greatest version of MEC2) that is
supposed to provide somewhat significant improvements.



Quite a few people tend to have difficulties with echo, and although
the WIKI has some very helpful advice, from a business standpoint I
would think that it would be an important step to come up with a final



solution to the problem.

Many companies who make the higher end equipment seem to have tackled
the issue on their hardware.

Do we know if digium is spending time on solving the issue?  For
example, having a tool to run on a digium analog or t1 board to
analyze the line statistics and come up with the proper gain settings
could be extremely helpful.

Such a tool would require a firm knowledge of the causes and solutions



to echo however, but I would assume that digium should have a grasp on



this.

It just seems difficult to suggest to companies to use an asterisk
based solution (if they do not use pri) when there is the possibility
that an installation will have issues with echo.

At this point, it feels more like a trial experience to eliminate echo



in various environments.


Unfortunately, that's the way it is right now.  Getting to the point
where you have enough knowledge to be able to work on these things is
not an insignificant task.
It seems like we're slowly getting there, and now that we have some 
more

interest on improving the software echo cans we might be a little be
closer to getting to the point where it "just works".



I have used local tone from the CO to help narrow things down, but a
tool that would loop dial a line and do an analysis could reduce the
implementation time from days to hours.


Well, there isn't anything that does the "whole job" right now.
There's a bunch
of pieces that go together, and if you have the necessary knowledge of
how to put the pieces together, you can get pretty close to it "just
working".
  It's not that
bad though, one can also see it as job security as well :-)

Matthew Fredrickson

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RE: [Asterisk-Users] Opinions on IAX JitterBuffer in old-school 1 .0.0?

2005-10-28 Thread Colin Anderson

>Does sound like you have the fix - upgrade to a newer Asterisk.

*groan* Yes, it did solve the problem, 100%. I upgraded a single site to
1.0.9 and call quality is perfect. Now, on to the other 29thank GOD for
SSH. 
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Re: [Asterisk-Users] Grandstream GXP-2000

2005-10-28 Thread Erick Baum

We have 50 of these phones in one location and a couple remote phones.  The problem seems to be caused by the volume settings on the phone.  We have noticed that the echo seems to be worse when the volume is very high on the phone (not using speakerphone).  We're still testing, but that's what we've been able to come up with so far.

On 10/28/05, Chris Bagnall <[EMAIL PROTECTED]> wrote:
> Erick, we're also using 1.0.1.12, having some echo problems,> mostly with in/out going ZAP calls (on quadBRI, w/asterisk
> 1.0.9), the internal SIP calls seem to work fine. (but you> have to make sure your volume isn't too high) Also the> GXP-2000 has the annoying feature that calls get disturbed> when you touch the wire (going from handset to phone).
We currently have 3 sites with about 15 GXP-2000s at 2, and 3 at the third.The 2 larger sites are running 1.0.1.9, the third site is running 1.0.1.12.
On straight SIP-SIP calls or outbound calls via an IAX->PSTN gateway, thereis no discernable echo at all. Ever.One of the larger sites is connected with 2 BRIs for incoming calls viazaphfc. Generally, there is echo for the first 2 or 3 seconds of a call,
then it dies out completely. There are exceptions - I've had reports of echolasting for the duration of a phone call, but this has usually been down toa callcentre at the other end of the line (probably using an autodialler)
where the first few seconds of an inbound call is silence. I guess this mustbe upsetting the echo training in Zaptel.The other larger site is connected using a TDM400 with 3 POTS lines. Thereis nearly always echo on incoming calls on these lines for about 30 seconds,
after which they appear to be fine.In conclusion - I'm not sure if the GXP2000s are the problem in yourscenario.Regards,Chris--C.M. Bagnall, Director, Minotaur I.T. LimitedThis email is made from 100% recycled electrons
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http://lists.digium.com/mailman/listinfo/asterisk-users-- | Erick Baum| Teal Networks, Inc.| http://www.teal.net 
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Re: [Asterisk-Users] Webui to show registered phones

2005-10-28 Thread Ben Higley

I use the same type of thing in my PHP without going through the Manager
port:

I had to do "chmod u+s /usr/sbin/asterisk" in order for my apache server
to be able to connect and get the response...


> Thanks for that Adam, fantastic!
>
> I did need to add one line to get it to work
>
> #!/usr/bin/perl
> #
>
> ##get lists of registered peers from asterisk
> $iaxpeers = `/usr/sbin/asterisk -rx \"iax2 show peers\"`;
> $sippeers = `/usr/sbin/asterisk -rx \"sip show peers\"`;
>
> ##replace newline characters with html break
> $iaxpeers =~ s/\n//g;
> $sippeers =~ s/\n//g;
>
>
> ##output the webpage
> print <<;
> Content-type: text/html\n\n ##THIS
> 
>  
>Registered devices
>  
>  
>CURRENT SIP USERS
>$sipppeers
>CURRENT IAX USERS
>$iaxpeers
>  
>
> ;
>
>
> Adam Moffett wrote:
>
>>
>>> Hi all, does anyone know if there is any app/webui that can show
>>> phones that are currently registered to *.  I guess this sort of
>>> funcionality counld be grabbed from the CLI with iax2 show peers and
>>> sip show peers, but having little programming knowledge wouldn't know
>>> where to start.
>>>
>>> I'm asking because we currently have several sip phones onsite and
>>> lots of remote iax2 users who would like to see availability without
>>> dialing.
>>>
>>> Bails
>>
>>
>>
>> Below is a simple perl script that might do the trick.  save it to
>> [something].cgi and most any distribution's apache web server should
>> be able to run it.
>>
>> If the web server isn't running on the same machine as asterisk then
>> it's a little more difficult.  An option might be to configure ssh to
>> allow authentication based on known RSA keys (so there's no password
>> prompt).  That is actually pretty easy to do, and you can google for
>> instructions on that.  Then the script can use ssh to talk to a shell
>> on the asterisk server which will in turn execute asterisk -rx and
>> give you the output.
>>
>> By the way, I haven't actually tested this except on the command line
>> and my html is lousy.  So while I'm sure the script will run, I pretty
>> much guarantee the resulting web page to look like crap.
>>
>>
>>
>>
>> #!/usr/bin/perl
>> #
>>
>> ##get lists of registered peers from asterisk
>> $iaxpeers = `/usr/sbin/asterisk -rx \"iax2 show peers\"`;
>> $sippeers = `/usr/sbin/asterisk -rx \"sip show peers\"`;
>>
>> ##replace newline characters with html break
>> $iaxpeers =~ s/\n//g;
>> $sippeers =~ s/\n//g;
>>
>>
>> ##output the webpage
>> print <> 
>>  
>>Registered devices
>>  
>>  
>>CURRENT SIP USERS
>>$sipppeers
>>CURRENT IAX USERS
>>$iaxpeers
>>  
>>
>> 
>> EOF
>>
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>
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[Asterisk-Users] Montreal Meet Asterisk Get-Together

2005-10-28 Thread Joshua Colp - Asterlink









Hello Folks,

 

I thought I’d make a sorta announcement as I’ll be
in Montreal on
a partial vacation/partial hangout/partial meet and greet thing. I thought it
might be nice for all the people in the area, and perhaps those attending the
Meet Asterisk thing to get together for supper and talk. If anyone is
interested, respond to this post and we’ll decide on a place/time/date
etc. Meet the minds behind the sillyness on IRC! Muahahaha… yeah

 

Joshua Colp






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Re: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?

2005-10-28 Thread Daniel Varella de Oliveira


 It costs here more or less R$600,00 (about US$264,55)

 Our friend, Dave Cotton post a message with a good price for outside of 
Brazil. US$295,00 is a good price, I think.

 I know that guy in Sao Paolo (the correct is São Paulo), that the site 
http://www.thehightechstore.com/plugcell.htmlannounced. His name is 
Douglas Prado and he is the owner of Contacto Telecom company. Contacto is 
the unique distributor of Plugcell in region of São Paulo. If you contact 
him, tell about me (He knows me as Daniel ex-Nooracom company in Rio de 
Janeiro). Maybe you can get a discount on your negotiation. hehehehe.
-- 

[ ]'s

Daniel Varella de Oliveira
Tecnologia IP Ltda
Tel.: +55 (21)3139-4091 / r. 108
Rio de Janeiro - Brasil
www.tecnologiaip.com.br




On Friday 28 October 2005 12:22, Tomasz Chmielewski wrote:
> Daniel Varella de Oliveira schrieb:
>  > Tomasz,
>  >
>  >  I'm from Brazil, and we are using here a solution that is based on a
>
> box where we can connect a GSM cellphone and use this directly to a
> phone or PBX extension.
>
>  >  I think that you can use some Digium's card (FXS or FXO) on your
>
> server, connect this GSM box there, and route your cellphone calls
> through this box.
>
>  >  There are boxes with just one channel and others up to six channels.
>  >  They have a lot compatibilities with the most common cellphones.
>
> looks interesting.
>
> do you know by chance how much such a single-cell box cost (more or less)?



pgpYL3LSzOWLh.pgp
Description: PGP signature
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[Asterisk-Users] Help with Zultys

2005-10-28 Thread Linc Fessenden


Hi everyone!
I just got a zultys zip 2 today with no manuals.  Can anyone tell me how 
to get in and config the=is thing please?  I know there has to be some 
*super secret code* to enable dhcp on it somehow and then a login as 
password for the web interface or something?  HELP!!???


--
-Linc Fessenden

In the Beginning there was nothing, which exploded - Yeah right...

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Re: [Asterisk-Users] Webui to show registered phones

2005-10-28 Thread Tzafrir Cohen

On Fri, Oct 28, 2005 at 09:39:59AM -0400, Adam Moffett wrote:
> 
> >Hi all, does anyone know if there is any app/webui that can show 
> >phones that are currently registered to *.  I guess this sort of 
> >funcionality counld be grabbed from the CLI with iax2 show peers and 
> >sip show peers, but having little programming knowledge wouldn't know 
> >where to start.
> >
> >I'm asking because we currently have several sip phones onsite and 
> >lots of remote iax2 users who would like to see availability without 
> >dialing.
> >
> >Bails 
> 
> 
> Below is a simple perl script that might do the trick.  save it to 
> [something].cgi and most any distribution's apache web server should be 
> able to run it.

That script requires running it as root. Why not use Asterisk::Manager
and connect to the manager interface? The script would also be much more
efficient.

If you find Asterisk::Manager from asterisk-perl
(http://asterisk/gnuinter.net ) to be broken, let me know and I'll send
you my up-to-date version. I'll publish it once I'll get some signs of
life from the author of asterisk-perl to know that that code is actually
legal to distribute.

> 
> If the web server isn't running on the same machine as asterisk then 
> it's a little more difficult.  

Manager interface?

-- 
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Re: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?

2005-10-28 Thread Steve Kennedy

On Fri, Oct 28, 2005 at 05:43:03PM +0100, David Cook wrote:

> You might want to investigate a Nokia 22 
> (http://europe.nokia.com/nokia/0,8764,56024,00.html). This provides a single 
> GSM line which is interfaced to the PBX by an anlogue trunk/extension. From 
> memory they cost around £100-150. I am going to revisit this as a solution to 
> our ever increasing PSTN-GSM call spend as soon as we have our Asterisk PBX 
> in place.

The big problem with SIM based GSM gateways is that the CLI will always
be that of the SIM not the real caller (there's no way to pass CLI, well
at least not in the UK).

Also in the UK it is illegal to run GSM gateways for 3rd parties (so you
can run them for your own company, but as a VoIP provider etc you cant
use a GSM gateway for anyone else, including customers).

Steve

-- 
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Re: [Asterisk-Users] Console detach.

2005-10-28 Thread Tzafrir Cohen

On Fri, Oct 28, 2005 at 09:59:00AM +0200, Pepe Aracil wrote:
> Hello.
> 
> I have installed asterisk 1.0.9.dfsg-5 in debian sarge.

Did you install the binary package from unstable or rebuilt it?

> 
> if I run /etc/init.d/asterisk stop and then /etc/init.d/asterisk start . 
> Asterisk don't detach from console where i started it. It beguin to write 
> all warning,debug,AGI dialog,... to the console.
> 
> If I start ast. manually without init script as root. It works fine and it 
> detach from console, but if i start asterisk with "-U asterisk" i have the 
> same problem.

My first guess would be that it has some files owned by root that the
standard invocation fails to open.

  strace -f -eopen /etc/init.d/asterisk start

comes to mind.

-- 
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Re: [Asterisk-Users] OT: Suggestions for E1 Service in the UK

2005-10-28 Thread Steve Kennedy

On Fri, Oct 28, 2005 at 05:40:05PM +0100, Charles Trevor wrote:

> > Well, the major incumbent is BT.
> > Are you sitting down ?
> > Installation :
> > Per channel  1 year contract  3/5y contract  3/5y+commitment
> > First 15 channels (min 8)GBP 125 GBP 80GBP 0
> > 16-30 (per channel)  GBP  30 GBP 15GBP 0
> > Annual Rental (per channel)  GBP 182.32   DDI Non Quota
> >   GBP 208.32   DDI Quota
> Affiniti (Kingston Communication) is another choice. Their min channel
> number is 6, and their pricing is more reasonable than BT. Dont have
> exact prices to hand, but they are better. They are also a far easier
> company to deal with than BT, who I have had no end of problems with.

They'll have better pricing on channels etc, however they'll buy BT
tails which are distance priced and will have to get on Kingston's
network.


Steve

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[Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?

2005-10-28 Thread David Cook

> If anyone knows of smaller-scale units that work on GSM900 and 1800, I'd
> also love to hear about them.

You might want to investigate a Nokia 22 
(http://europe.nokia.com/nokia/0,8764,56024,00.html). This provides a single 
GSM line which is interfaced to the PBX by an anlogue trunk/extension. From 
memory they cost around £100-150. I am going to revisit this as a solution to 
our ever increasing PSTN-GSM call spend as soon as we have our Asterisk PBX in 
place.

David Cook

JP Computer Services
Delivering Business Benefit
http://www.jpcompserv.co.uk

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Re: [Asterisk-Users] OT: Suggestions for E1 Service in the UK

2005-10-28 Thread Charles Trevor


> 
> Well, the major incumbent is BT.
> 
> Are you sitting down ?
> 
> Installation :
> 
> Per channel  1 year contract  3/5y contract  3/5y+commitment
> 
> First 15 channels (min 8)GBP 125 GBP 80GBP 0
> 16-30 (per channel)  GBP  30 GBP 15GBP 0
> 
> 
> Annual Rental (per channel)  GBP 182.32   DDI Non Quota
>   GBP 208.32   DDI Quota
> 
> jd
> 

Affiniti (Kingston Communication) is another choice. Their min channel
number is 6, and their pricing is more reasonable than BT. Dont have
exact prices to hand, but they are better. They are also a far easier
company to deal with than BT, who I have had no end of problems with.

Charlie

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RE: [Re] Re: [Asterisk-Users] Echo canceller on TE406 & Asterisk

2005-10-28 Thread Robert Augustyn

Darren,
Can you elaborate on what echocan did you use and how?
Thanks.
robert 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Darren Wright
> Sent: Friday, October 28, 2005 7:35 AM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: RE: [Re] Re: [Asterisk-Users] Echo canceller on 
> TE406 & Asterisk
> 
> I have given up totally on Digium based echo cancel, hardware 
> or software.  The KB1 is the best so far, but still 
> unacceptable.  I installed a hardware echocan FACING the T1 
> card in the asterisk box, and
> all is perfect.   No complaints from any of my clients since 
> taking that
> leap.
> 
> -Darren
> 
> 
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Re: [Asterisk-Users] Re: Asterisk Redundency

2005-10-28 Thread Ray Van Dolson

On Wed, Oct 26, 2005 at 10:12:09AM -0400, Matt wrote:
> Does anyone know if SIPURA SPA-2002's support DNS SRV records?

Yep, it does (as does its brother PAP2-NA).

Ray
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Re: [Asterisk-Users] Not saving voicemail message

2005-10-28 Thread Richard Smith

Thank you Hadley, that was the problem.

Cheers,

Richard

- Original Message -
From: "Hadley Rich" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Friday, October 28, 2005 12:20 AM
Subject: Re: [Asterisk-Users] Not saving voicemail message


> On Friday 28 October 2005 12:06, Richard Smith wrote:
> > [EMAIL PROTECTED] 1.2.0 beta4 writes to the respective voicemail directory
and
> > when the call is hung-up the .wav file disappears.
>
> Sounds like voicemail.conf is setup to delete the message after it is
emailed
> to the user.
>
> You may also want to refer here
> http://www.catb.org/~esr/faqs/smart-questions.html
>
> HTH
>
> hads
>
> --
> You may already be a loser.
> -- Form letter received by Rodney Dangerfield.
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[Asterisk-Users] Top and asterisk performance

2005-10-28 Thread Julian Lyndon-Smith

We had to move from a old * server to a new one in a hurry (hardware 
failure). The old server was a dual pentium 700 with 512MB ram running 
fedora core 2, the new one is a single 3GHz Pentium with 1gb ram.


The same number of people are connected to the new server as the old, 
the same number of inbound calls to the isdn30 etc (on average 20 calls 
active at any time (SIP and ZAP)). Basically, just a server swapout.


I must be reading top wrong, because the old server had a idle of approx 
30%, whereas the new server is


top - 13:35:21 up 12 days, 23:57,  1 user,  load average: 7.11, 7.20, 7.21
Tasks:  98 total,   9 running,  89 sleeping,   0 stopped,   0 zombie
Cpu(s): 99.0% us,  1.0% sy,  0.0% ni,  0.0% id,  0.0% wa,  0.0% hi,  0.0% si
Mem:   1034640k total,   144792k used,   889848k free,21952k buffers
Swap:  2031608k total,0k used,  2031608k free,61248k cached

Notice the 99.0% us. This fluctuates between 80 and 99%.

The other difference is that the new server is on cvs-head as of today - 
I did say that it was an emergency :) whereas the old server was 
cvs-head from june sometime.


Is it just me, or is there a problem ?

Julian.
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[Asterisk-Users] Cell phone extension woes

2005-10-28 Thread Chris Miller



I've got a cell phone setup as an extension in a queue. On occasion the 
cell phone will drop the call due to loss of, or bad, signal. Is there a 
clean way in the dial plan to reintroduce a call back into the queue 
when the call is dropped on the extension side? I realize this would 
occur even during a normal (extension side) call hangup, but as long as 
asterisk terminates the call when the caller hangs up, this would be fine.


Thoughts?

Chris
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Re: [Asterisk-Users] fxotune fails with valid TDM/FXO card

2005-10-28 Thread Chris Miller


Mojo with Horan & Company, LLC wrote:

The recent suggestion on the list was to not use 1.0.9 zaptel


You mean the driver, or the version of fxotune? fxotune has been removed 
from the prior versions of the zaptel driver, it's only included in 1.2 
now. As for the driver, is anyone using the 1.2 zaptel driver with 
Asterisk 1.0.9? The way the downloads are grouped together on the 
Asterisk web page, I was led to believe they shouldn't be mixed.


Chris
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Re: [Asterisk-Users] X100P doesn't show Caller-ID

2005-10-28 Thread Rich Adamson


> Can anyone out there help me ?
> 
> Beside countless  benefits, without Caller-ID on my
> SIP Phone's screen  *  is a real pain in my neck.
> 
> I have * 1.2.0-Beta (Latest CVS) and a X100P card.
> Sometimes I can get the Caller-ID and sometimes I can
> not. Even from the same number. Any suggestions ?

This has been a rather common issue with the old x100p card.
Not sure why, but when I was using two of them, both had the
same kind of issue. I replaced these with the TDM04b, and it
too will sometimes miss the callerid. The TDM is much better
then the x100p, but will still miss a few.


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Re: [Asterisk-Users] x100p (FXO) not being seen by asterisk (is my bestguess) .

2005-10-28 Thread Mr. James W. Laferriere

Hello All ,

On Thu, 27 Oct 2005, Mr. James W. Laferriere wrote:
> On Thu, 27 Oct 2005, Phil Pritchard wrote:
> > only new to asterisk, but have had some hardware exp.
> >stay away from irq9 its tied to irq2 and will always be shared, Paul has
> > the go.. in bios disable serial and or usb (if not using) and make sure irda
> > is not enabled. another one is the lpt port if your not using that, there is
> > another irq you can steel..
>   ALL & I mean all serial/parrallel/...'everything I can find'... has 
> been 
>   turned off in the bios .  And I have recompiled a kernel with those 
> same 
>   items turned off in it .  That d??ned module wants to load at irq 9 no 
>   matter what I do .  Of course there is no way to set irq's to a 
>   particular pci slot in the bios .
>   Does anyone now howto set irq say at the boot: or in modprobe.conf ?
> > dont share interrupts, as a rule(if you can help it)... it usually leads to
> > system instability and usually under load.
>   Quite well understand this point .  Have heard it on this list many 
>   times .  And am doing my best NOT too .
> > UBCD ...(www.ultimatebootcd.com).  has some nice tools that can probe a 
> > system
> > to give a second appinion on interrupt conflicts, ram and hard drive
> > errors.
> > its my best tool for hardware problems..
>   IMO ,  The mirrors have the su??iest download schemes I have seen in 
>   some time .\IMO
>   I have yet to burn that image but as soon as I do I'll boot it on that 
>   piece of junk I bought for near next to nothing .  Which is almost what 
>   it is worth ,  Nothing .
>   Thank you for your input ,  Every bit helps .  JimL

Finally got that da??ed wcfxo to load on a irq by itself (*).  Had to 
turn off the last item of the onbord devices the ether & buy an ether 
card to get connectivity .  But even with the suggestion by 'Paul' to 
use a two line cord & finally using a singular irq ,  The config's I 
sent last time have not changed .  The x100p/wcfxo combination see the 
line ringing (**) .  But asterisk does NOT see it on the console nor 
does it pick up the line .  Quite frustrating when everything should be 
ok per every example I've seen & still nothing positive to show for it .

ANY suggestions/questions/... Please pipe up .  Tia ,  JimL

(*)
Oct 28 09:43:47 asterisk-test kernel: Zapata Telephony Interface Registered on 
major 196
Oct 28 09:43:47 asterisk-test kernel: PCI: Found IRQ 5 for device :01:01.0
Oct 28 09:43:47 asterisk-test kernel: Registered Span 1 ('WCFXO/0') with 1 
channels
Oct 28 09:43:47 asterisk-test kernel: Span ('WCFXO/0') is new master
Oct 28 09:43:47 asterisk-test kernel: New regoffset: 7
Oct 28 09:43:47 asterisk-test kernel: wcfxo: DAA mode is 'FCC'
Oct 28 09:43:47 asterisk-test kernel: Found a Wildcard FXO: Wildcard X101P
Oct 28 09:43:47 asterisk-test kernel: BATTERY!
Oct 28 09:43:47 asterisk-test kernel: Registered tone zone 0 (United States / 
North America)

(**)
Oct 28 09:46:35 asterisk-test kernel: wcfxo: RING!
Oct 28 09:46:37 asterisk-test kernel: wcfxo: NO RING!
Oct 28 09:46:41 asterisk-test kernel: wcfxo: RING!
Oct 28 09:46:43 asterisk-test kernel: wcfxo: NO RING!
Oct 28 09:46:47 asterisk-test kernel: wcfxo: RING!
Oct 28 09:46:49 asterisk-test kernel: wcfxo: NO RING!

-- 
+--+
| James   W.   Laferriere | SystemTechniques | Give me VMS |
| NetworkEngineer | 3542 Broken Yoke Dr. |  Give me Linux  |
| [EMAIL PROTECTED] | Billings , MT. 59105 |   only  on  AXP |
+--+
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[Asterisk-Users] Asterisk with Zultys SIP gateway

2005-10-28 Thread Dustin Wenz

I'm trying to configure an Asterisk server to make inbound and  
outbound calls through a Zultys MX250 operating as a SIP media gateway.


This is my first experience with Asterisk, but from what I  
understand, I need to register the Asterisk system as a SIP device  
with the MX250. This is what I have in sip.conf under [general]:

register => asterisk:[EMAIL PROTECTED]

Doing this results in repeated messages at the Asterisk console:
NOTICE[2795]: chan_sip.c:4790 sip_reg_timeout:-- Registration for  
'[EMAIL PROTECTED]' timed out, trying again


I'm confident that the request is reaching the Zultys system, because  
if I deliberately omit the password, I get:
WARNING[2795]: chan_sip.c:8656 handle_response_register: Forbidden -  
wrong password on authentication for REGISTER for 'asterisk' to  
'mx250.mycompany.com'


Does anyone have experience integrating Asterisk with a Zultys phone  
system that might be able to offer some tips?


Thanks!

- .Dustin

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[Asterisk-Users] 2 problems

2005-10-28 Thread Manuel Silva




Hello!
 
I have installed 2 servers, one with SER integrated 
with PostgreSQL (Fedora Core 3) and the other with Asterisk 
(Fedora Core 4). I can talk Softphone -> SER -> SER -> Softphone (in 
case I try to contact a person that as a different SIP server). Now 
the goal is to, use Asterisk as a gateway, with SIP trunking between the 
Asterisk and a PBX (with IP module).
 
So if I try to make a call from a softphone to 
a PSTN connected phone, it will be like this: 
softphone->SER->Asterisk->PBX->destination.
 
I have already configured SER to forward calls to 
Asterisk, and it seems to work fine. Now, my problem is how to configure 
Asterisk to forward a call from SER to the PBX, or how to establish an 
IP trunking between Asterisk and the PBX. The other problem is how do I 
configure Asterisk, to forward a call from a PSTN phone.
 
If anyone can help me, please!! :-)
  
 
Manuel Silva
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[Asterisk-Users] Prevent transcoding

2005-10-28 Thread Simon Woodhead


Hi folks,

Is anyone aware of a way to prevent transcoding or better still apply 
some kind of weighting to codec selection based on other channels in the 
call? Let's say we support g729 and gsm, a peer supports both and a 
client supports one of them. We're seeing calls frequently coming in on 
one and being transcoded to the other whilst it would be much more 
efficient to pass it straight through for the negligible bandwidth 
saved. Is there any way of achieving this?


Thanks,
Simon

PS - before anyone suggests re-inviting, not doing so is intentional!

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Re: [Asterisk-Users] Webui to show registered phones

2005-10-28 Thread bails


Thanks for that Adam, fantastic!

I did need to add one line to get it to work

#!/usr/bin/perl
#

##get lists of registered peers from asterisk
$iaxpeers = `/usr/sbin/asterisk -rx \"iax2 show peers\"`;
$sippeers = `/usr/sbin/asterisk -rx \"sip show peers\"`;

##replace newline characters with html break
$iaxpeers =~ s/\n//g;
$sippeers =~ s/\n//g;


##output the webpage
print <<;
Content-type: text/html\n\n ##THIS


  Registered devices


  CURRENT SIP USERS
  $sipppeers
  CURRENT IAX USERS
  $iaxpeers


;


Adam Moffett wrote:



Hi all, does anyone know if there is any app/webui that can show 
phones that are currently registered to *.  I guess this sort of 
funcionality counld be grabbed from the CLI with iax2 show peers and 
sip show peers, but having little programming knowledge wouldn't know 
where to start.


I'm asking because we currently have several sip phones onsite and 
lots of remote iax2 users who would like to see availability without 
dialing.


Bails 




Below is a simple perl script that might do the trick.  save it to 
[something].cgi and most any distribution's apache web server should 
be able to run it.


If the web server isn't running on the same machine as asterisk then 
it's a little more difficult.  An option might be to configure ssh to 
allow authentication based on known RSA keys (so there's no password 
prompt).  That is actually pretty easy to do, and you can google for 
instructions on that.  Then the script can use ssh to talk to a shell 
on the asterisk server which will in turn execute asterisk -rx and 
give you the output.


By the way, I haven't actually tested this except on the command line 
and my html is lousy.  So while I'm sure the script will run, I pretty 
much guarantee the resulting web page to look like crap.





#!/usr/bin/perl
#

##get lists of registered peers from asterisk
$iaxpeers = `/usr/sbin/asterisk -rx \"iax2 show peers\"`;
$sippeers = `/usr/sbin/asterisk -rx \"sip show peers\"`;

##replace newline characters with html break
$iaxpeers =~ s/\n//g;
$sippeers =~ s/\n//g;


##output the webpage
print <
 
   Registered devices
 
 
   CURRENT SIP USERS
   $sipppeers
   CURRENT IAX USERS
   $iaxpeers
 


EOF

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Re: [Asterisk-Users] Ouch - Error while writing audio data - broken pipe

2005-10-28 Thread Rich Adamson


> I'm getting the following error when starting Asterisk: Error while writing 
> audio data: broken pipe. In my processesses I have tons of mpg123 instances 
> running, probaby because of asterisk trying to start ad nauseum.
> 
> What could be creating this?  I am running Beta 1.2, trying to see if 
> Realtime Config could meet my needs in the near future.

One of the more frequent causes is an error in zapata.conf (regardless of
where its read from).



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[Asterisk-Users] Ouch - Error while writing audio data - broken pipe

2005-10-28 Thread Michaël Gaudette

I'm getting the following error when starting Asterisk: Error while writing 
audio data: broken pipe. In my processesses I have tons of mpg123 instances 
running, probaby because of asterisk trying to start ad nauseum.


What could be creating this?  I am running Beta 1.2, trying to see if 
Realtime Config could meet my needs in the near future.


Regards,

Mike 


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Re: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?

2005-10-28 Thread Steve Kennedy

On Fri, Oct 28, 2005 at 04:40:07PM +0200, Dave Cotton wrote:

> On Fri, 2005-10-28 at 16:22 +0200, Tomasz Chmielewski wrote:
> > looks interesting.
> > do you know by chance how much such a single-cell box cost (more or less)?
> I found it here http://www.thehightechstore.com/plugcell.html
> at 295$USD

You can buy a Siemens MC35 (or is it TC35) for around GBP 100. It's a
GSM module with serial port, external aerial and handset interface.


Steve

-- 
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Re: [Asterisk-Users] OT: Suggestions for E1 Service in the UK

2005-10-28 Thread John Daragon


Geoff Manning wrote:

We are looking to acquire E1 service in Fleet right outside of London. I am
in the States so I am not aware of the key players. We currently get ADSL
from Eclipse but were interested in a quote for E1.

What is a typical E1 line go for nowadays and who can I get it from?


Well, the major incumbent is BT.

Are you sitting down ?

Installation :

Per channel  1 year contract  3/5y contract  3/5y+commitment

First 15 channels (min 8)GBP 125 GBP 80GBP 0
16-30 (per channel)  GBP  30 GBP 15GBP 0


Annual Rental (per channel)  GBP 182.32   DDI Non Quota
 GBP 208.32   DDI Quota

jd

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Re: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?

2005-10-28 Thread Dave Cotton

On Fri, 2005-10-28 at 16:22 +0200, Tomasz Chmielewski wrote:

> 
> 
> looks interesting.
> 
> do you know by chance how much such a single-cell box cost (more or less)?
> 
> 

I found it here http://www.thehightechstore.com/plugcell.html

at 295$USD


-- 
Dave Cotton <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] call queue

2005-10-28 Thread gwynpen

set 
persistentmembers = yes
in your queues.conf and the logins of your callback agents 
will survive a restart. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Baris Simsek
> Sent: Friday, October 28, 2005 2:44 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] call queue
> 
> hello,
> 
> I want to learn that, is it 'MUST' to login call queue?
> 
> I have 3 call queues, and i want to distribute incoming call 
> to the one of them. But i don't want to callbacklogin. 
> Because of, after a restart, all agents have to do callbacklogin.
> 
> thanks...
> 
> --
> Baris Simsek
> Project Manager
> Empatiq Communication Technologies
> 
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Re: [Asterisk-Users] IAX channel options

2005-10-28 Thread Andrew Kohlsmith

On Friday 28 October 2005 10:11, Michael Welter wrote:
> This works fine for the Qwest line, but Asterisk doesn't absorb the 'W'
> for the IAX call--the 'W' is sent as part of the dial string.

Dial(IAX2/${NUMBER:1})

IAX2 isn't limited to numeric numbers.  IAX2 can send text, URLs, binary data, 
hell even Video.  You got bit thinking in POTS-only terms.  :-)

-A.
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Re: [Asterisk-Users] IAX channel options

2005-10-28 Thread Rich Adamson

That should be a lowercase "w". The placement is correct, just not
a lowercase "w".



> I have an installation with four Qwest POTS lines.  For some unknown 
> reason, Qwest drops the first digit in the dial string, and the call 
> fails.  To fix that problem, I put a 'W' in the dial string:
> 
> QWEST=Zap/g2
> 
> exten => _9303NXX,1,Dial(${QWEST}/W${EXTEN:1})
> 
> The client has since complained that, when all four lines are busy, he 
> cannot make a local call.  So I provided the ability to roll over to 
> another system to complete the call:
> 
> TELECOMMATTERS=IAX2/[EMAIL PROTECTED]
> 
> exten => _9303NXX,1,ChanIsAvail(${QWEST}&${TELECOMMATTERS})
> exten => _9303NXX,2,Cut(MYCHANNEL=AVAILCHAN,,1)
> exten => _9303NXX,3,Dial(${MYCHANNEL}/W${EXTEN:1})
> exten => _9303NXX,4,Hangup
> exten => _9303NXX,102,Congestion
> 
> This works fine for the Qwest line, but Asterisk doesn't absorb the 'W' 
> for the IAX call--the 'W' is sent as part of the dial string.
> 
> Is there a solution for this?
> 
> TIA
> 
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---End of Original Message-


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Re: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?

2005-10-28 Thread Tomasz Chmielewski


Daniel Varella de Oliveira schrieb:

> Tomasz,
>
>  I'm from Brazil, and we are using here a solution that is based on a 
box where we can connect a GSM cellphone and use this directly to a 
phone or PBX extension.
>  I think that you can use some Digium's card (FXS or FXO) on your 
server, connect this GSM box there, and route your cellphone calls 
through this box.

>
>  There are boxes with just one channel and others up to six channels.
>  They have a lot compatibilities with the most common cellphones.


looks interesting.

do you know by chance how much such a single-cell box cost (more or less)?


--
Tomek
http://wpkg.org
WPKG - software deployment and upgrades with Samba

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[Asterisk-Users] IAX channel options

2005-10-28 Thread Michael Welter

I have an installation with four Qwest POTS lines.  For some unknown 
reason, Qwest drops the first digit in the dial string, and the call 
fails.  To fix that problem, I put a 'W' in the dial string:


QWEST=Zap/g2

exten => _9303NXX,1,Dial(${QWEST}/W${EXTEN:1})

The client has since complained that, when all four lines are busy, he 
cannot make a local call.  So I provided the ability to roll over to 
another system to complete the call:


TELECOMMATTERS=IAX2/[EMAIL PROTECTED]

exten => _9303NXX,1,ChanIsAvail(${QWEST}&${TELECOMMATTERS})
exten => _9303NXX,2,Cut(MYCHANNEL=AVAILCHAN,,1)
exten => _9303NXX,3,Dial(${MYCHANNEL}/W${EXTEN:1})
exten => _9303NXX,4,Hangup
exten => _9303NXX,102,Congestion

This works fine for the Qwest line, but Asterisk doesn't absorb the 'W' 
for the IAX call--the 'W' is sent as part of the dial string.


Is there a solution for this?

TIA

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[Asterisk-Users] h323 no audio from the sip phone to the outside world.

2005-10-28 Thread mik sib

Hi all,

through oh323 i can register to my gatekeeper and make
and receive calls.

My gatekeeper routes the incoming call as well as the
outgoing.

The problem is simply that i can't ear nothing from my
SIP ipPhones. I can ear my voice during a call from a
normal telephone in my SIP phone but no viceversa.

How can i debug this situation ? I've no errors in the
log or at the asterisk startup.
How to understand what's happening ?
I've tryed different phones also.
any idea ?
thank you very much
Mik


Here's my oh323.conf
 Configuration of OpenH323 channel driver
--
Version: 0.7.3
Listening on address: 10.0.0.253:1720
Gatekeeper used: [EMAIL PROTECTED]
(Registered)
FastStart/H245Tunnelling/H245inSetup: ON/ON/ON
Supported formats in pref. order: ulaw<0>
Jitter buffer limits (min/max): 20-100 ms
TCP port range: 1 - 2
UDP (RAS) port range: 1 - 2
UDP (RTP) port range: 1 - 2
IP Type-of-Service value: 0
User input mode: rfc2833
Max number of inbound H.323 calls: 100
Max number of outbound H.323 calls: 100
Max number of simultaneous H.323 calls: 100
Max call rate (ingress direction): 1.00/30
Default language: en
Default music class: default
Default context: voip-h323

doing a call with the ip phone to the outside world
through the gatekeeper

[2]WrapperAPI::h323_make_call: Making call.
[2]WrapH323EndPoint::MakeCall: Making call to
0258115040
[4]WrapH323EndPoint::CreateConnection: Creating a
H323Connection [32066]
[2]WrapH323Connection::WrapH323Connection: Creation of
WrapH323Connection based on user data.
[2]WrapH323Connection::WrapH323Connection: Call is
outgoing.
[4]WrapH323Connection::WrapH323Connection:
WrapH323Connection created.
[3]WrapH323EndPoint::MakeCall: Call token is
ip$localhost/32066
[3]WrapH323EndPoint::MakeCall: Call reference is 32066
[2]WrapH323Connection::OnSendSignalSetup: Sending
SETUP message...
[3]WrapH323Connection::OnSendSignalSetup: Setting
display name 0432281316 Fabio Violino
[3]WrapH323Connection::OnSendSignalSetup: Setting
calling party number test419
[2]WrapH323Connection::OnAlerting: Ringing phone for
"0258115040" ...
[3]WrapH323EndPoint::OpenAudioChannel: Direction =>
RECODER, Buffer => 320
[2]WrapH323EndPoint::OpenAudioChannel: Media format:
FrameSize 8, FrameTime 8, TimeUnits 8
[2]WrapH323EndPoint::OpenAudioChannel: Codec info:
FrameRate 160
[2]WrapH323EndPoint::OpenAudioChannel: Packet size:
160
[2]WrapH323EndPoint::OpenAudioChannel: Frames per
packet: 20
[2]WrapH323EndPoint::OpenAudioChannel: LID Codec
G.711-uLaw-64k
[3]WrapH323EndPoint::OpenAudioChannel: The sound
channel with the application is asterisk-oh323(fd=45)
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskSoundChannel::PAsteriskSoundChannel:
Object initialized.
[3]PAsteriskSoundChannel::Open: os_handle 45,
mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize
160
[3]WrapH323EndPoint::OpenAudioChannel: Opened sound
channel "Asterisk" for recording using 1x320 byte
buffers.
[3]WrapH323Connection::OnEstablished:
WrapH323Connection [ip$localhost/32066] established
(FastStartDisabled/noH245Tunneling)
[3]WrapH323EndPoint::OnConnectionEstablished:
Connection [ip$localhost/32066] established.
[3]WrapH323EndPoint::GetConnectionInfo:
[ip$localhost/32066] RTP Media:
10.0.0.253:10004-0.0.0.0:0
[3]WrapH323EndPoint::OpenAudioChannel: Direction =>
PLAYER, Buffer => 320
[2]WrapH323EndPoint::OpenAudioChannel: Media format:
FrameSize 8, FrameTime 8, TimeUnits 8
[2]WrapH323EndPoint::OpenAudioChannel: Codec info:
FrameRate 160
[2]WrapH323EndPoint::OpenAudioChannel: Packet size:
160
[2]WrapH323EndPoint::OpenAudioChannel: Frames per
packet: 20
[2]WrapH323EndPoint::OpenAudioChannel: LID Codec
G.711-uLaw-64k
[3]WrapH323EndPoint::OpenAudioChannel: The sound
channel with the application is asterisk-oh323(fd=43)
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskSoundChannel::PAsteriskSoundChannel:
Object initialized.
[3]PAsteriskSoundChannel::Open: os_handle 43,
mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize
160
[3]WrapH323EndPoint::OpenAudioChannel: Opened sound
channel "Asterisk" for playing using 1x320 byte
buffers.
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[5]PAsteriskSoundChannel::Read: Data read [320 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskS

Re: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?

2005-10-28 Thread Jean-Michel Hiver


Anders Svensson a écrit :


Only the pricing is not that fantastic
 


It's actually not that bad compared with other GSM gateways.

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RE: [Asterisk-Users] Webui to show registered phones

2005-10-28 Thread Sherwood McGowan

You could always (I'll actually do it, I have similar scripts written) just
whip up a php script that connects to a Asterisk Manager Proxy (to limit the
possibility of crashing the server by making too many Manager API
connections), and have it issue the following commands:

Action: Command
Command: Sipshowpeers

And read the socket stream back using a loop and the flush() command. 

Here's an example of the code, only this one is using "sip show peer
" to get the status of a single customer (If you use qualify=yes for
a customer, the server will send requests for an ack to the UA every so many
minutes, to ensure that it's reachable)


Here ya go, hope you guys enjoy it! There's going to be a lot of this
functionality being put in ARTCP and SACP (Asterisk RealTime Control Panel
and Static Asterisk Control Panel, respectively). That way, non-technical
people can get output that they can understand (just do a str_replace() with
some arrays for the $needle, $haystack, and $replacment) without possibly
killing your server.

\n";
} else {
echo "";
socket_write($socket, "Action: SIPshowpeer\r\n");
socket_write($socket, "Peer: " . trim($number) ."\r\n\r\n");
echo "";
while($buffer = socket_read($socket, 512, PHP_NORMAL_READ)){
if(trim($buffer) == "Response: Error") {
echo "Received error - ".trim($buffer).".
Number probably not found";
break;
}
if(trim($buffer) == "--END COMMAND--"||substr_count($buffer,
"Server:") >0) { 
echo $buffer;
break;
}
echo $buffer;
flush();
}
echo "";
socket_write($socket, "Action: Logoff\r\n\r\n");
socket_close($socket);
}
?>



->-Original Message-
->From: [EMAIL PROTECTED] 
->[mailto:[EMAIL PROTECTED] On Behalf Of 
->Adam Moffett
->Sent: Friday, October 28, 2005 9:46 AM
->To: Asterisk Users Mailing List - Non-Commercial Discussion
->Subject: Re: [Asterisk-Users] Webui to show registered phones
->
->
->> Hi all, does anyone know if there is any app/webui that can show 
->> phones that are currently registered to *.  I guess this sort of 
->> funcionality counld be grabbed from the CLI with iax2 show 
->peers and 
->> sip show peers, but having little programming knowledge 
->wouldn't know 
->> where to start.
->>
->> I'm asking because we currently have several sip phones onsite and 
->> lots of remote iax2 users who would like to see 
->availability without 
->> dialing.
->>
->> Bails
->
->P.S.:
->
->show peers will show you the devices that are registered.  If 
->you're interested in how many active calls there are I think 
->you want iax2 show channels and sip show channels.
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Re: [Asterisk-Users] Grandstream GXP-2000

2005-10-28 Thread Faris Raouf


Erick Baum wrote:
We're having a rather serious echo problem using the Grandstream 
GXP-2000's with Asterisk 1.0.9.  I'm wondering if there is something I'm 
overlooking that might be an easy fix.  The echo seems to be worst on 
internal SIP to SIP calls but you do get it every once in a while on 
outgoing calls through the PRI.  It's not the speakerphone echo problem, 
we're running the 1.0.1.12  firmware that pretty much 
fixes that.  It seems like most of the echo cancellation functions are 
for outgoing calls through the phone company.  Is this a more likely a 
phone problem?  We've got about 50 of these phones all doing the same 
thing.


--
| Erick Baum



Hi Eric,

I only have two of these but have not come across an echo problem with 
them on SIP at all. Nothing unusual needs to be done to the config at 
all. So although I don't know how to help you, you can be assured that 
the problem is solvable and not down to the actual phones themselves, if 
you see what I mean?


Faris.

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Re: [Asterisk-Users] E1/T1 failover hardware

2005-10-28 Thread astgroups

We use the following device for Asterisk fail-over and our T1s. I
believe they have an E1 version also:
http://www.red-fone.com/fonebridge.html



On Thu, 2005-10-20 at 11:23, John Daragon wrote:
> Warning ! I know zip about electronics.
> 
> I've been looking for a device to handle the switching of an E1 
> connection from one Asterisk box to another in the event of a 
> catastrophic server failure.  All of the solutions I've seen so far have 
> been designed to handle the situation where the telco line faults so 
> that the local PBX can switch to a secondary E1.
> 
> I've come across this application note :
> 
> http://www.maxim-ic.com/appnotes.cfm/appnote_number/2857
> 
> which describes "T1/E1/J1, N+1 Redundancy With Analog Switches"
> 
> These parts are obviously designed to be built into E1 boards - hence, I 
> think, the protection circuitry.
> 
> Here's the question, then :  what (apart from jumping through regulatory 
> hoops) is to stop a simple array of MOSFETS (and a bit of control 
> circuitry) implementing a failover switch controlled (say) by a pin on a 
> serial or parallel port ?
> 
> jd

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[Asterisk-Users] Re: Grandstream GXP-2000

2005-10-28 Thread Doug Meredith

Erick Baum <[EMAIL PROTECTED]> wrote:

>We're having a rather serious echo problem using the Grandstream GXP-2000's
>with Asterisk 1.0.9. I'm wondering if there is something I'm overlooking
>that might be an easy fix. The echo seems to be worst on internal SIP to SIP
>calls but you do get it every once in a while on outgoing calls through the
>PRI. It's not the speakerphone echo problem, we're running the

Crazy idea, but are the two end-points on the SIP-SIP calls nearby?
Is it possible that the mic on phone A is actually picking up party B?

Doug
-- 
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com

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RE: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?

2005-10-28 Thread Anders Svensson

Only the pricing is not that fantastic

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Boris Bakchiev
Sent: den 28 oktober 2005 15:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] "GSM cards" / "mobile phone cards" for
Asterisk?

Get VoiceBlue VoIP GSM gateway.

It works very well with asterisk.
I have been using it for the last 4 month and its fantastic!


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomasz
Chmielewski
Sent: Friday, October 28, 2005 10:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] "GSM cards" / "mobile phone cards" for
Asterisk?

I was wondering if there is something like that on this Earth:

Some of our users are "mobile users" - they are rarely in one place for 
longer than 15 minutes.
They use mobile phones a lot.

 From our mobile operator we have an offer which allows us to call for 
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Re: [Asterisk-Users] Webui to show registered phones

2005-10-28 Thread Adam Moffett



Hi all, does anyone know if there is any app/webui that can show 
phones that are currently registered to *.  I guess this sort of 
funcionality counld be grabbed from the CLI with iax2 show peers and 
sip show peers, but having little programming knowledge wouldn't know 
where to start.


I'm asking because we currently have several sip phones onsite and 
lots of remote iax2 users who would like to see availability without 
dialing.


Bails


P.S.:

show peers will show you the devices that are registered.  If you're 
interested in how many active calls there are I think you want iax2 show 
channels and sip show channels.

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RE: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?

2005-10-28 Thread Boris Bakchiev

Get VoiceBlue VoIP GSM gateway.

It works very well with asterisk.
I have been using it for the last 4 month and its fantastic!


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomasz
Chmielewski
Sent: Friday, October 28, 2005 10:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] "GSM cards" / "mobile phone cards" for
Asterisk?

I was wondering if there is something like that on this Earth:

Some of our users are "mobile users" - they are rarely in one place for 
longer than 15 minutes.
They use mobile phones a lot.

 From our mobile operator we have an offer which allows us to call for 
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Re: [Asterisk-Users] Webui to show registered phones

2005-10-28 Thread Adam Moffett



Hi all, does anyone know if there is any app/webui that can show 
phones that are currently registered to *.  I guess this sort of 
funcionality counld be grabbed from the CLI with iax2 show peers and 
sip show peers, but having little programming knowledge wouldn't know 
where to start.


I'm asking because we currently have several sip phones onsite and 
lots of remote iax2 users who would like to see availability without 
dialing.


Bails 



Below is a simple perl script that might do the trick.  save it to 
[something].cgi and most any distribution's apache web server should be 
able to run it.


If the web server isn't running on the same machine as asterisk then 
it's a little more difficult.  An option might be to configure ssh to 
allow authentication based on known RSA keys (so there's no password 
prompt).  That is actually pretty easy to do, and you can google for 
instructions on that.  Then the script can use ssh to talk to a shell on 
the asterisk server which will in turn execute asterisk -rx and give you 
the output.


By the way, I haven't actually tested this except on the command line 
and my html is lousy.  So while I'm sure the script will run, I pretty 
much guarantee the resulting web page to look like crap.





#!/usr/bin/perl
#

##get lists of registered peers from asterisk
$iaxpeers = `/usr/sbin/asterisk -rx \"iax2 show peers\"`;
$sippeers = `/usr/sbin/asterisk -rx \"sip show peers\"`;

##replace newline characters with html break
$iaxpeers =~ s/\n//g;
$sippeers =~ s/\n//g;


##output the webpage
print <
 
   Registered devices
 
 
   CURRENT SIP USERS
   $sipppeers
   CURRENT IAX USERS
   $iaxpeers
 


EOF

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Re: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?

2005-10-28 Thread maka

You can try DIAX, which I see has some added GSM/PSTN gateway support.
I have yet to try it myself, it looks nice though. -
http://www.laser.com/dante/diax/diaxhlp.htm#gsm

cheersOn 10/28/05, Daniel Varella de Oliveira <[EMAIL PROTECTED]> wrote:
Tomasz, I'm from Brazil, and we are using here a solution that is based on a boxwhere we can connect a GSM cellphone and use this directly to a phone or PBXextension. I think that you can use some Digium's card (FXS or FXO) on your server,
connect this GSM box there, and route your cellphone calls through this box. There are boxes with just one channel and others up to six channels. They have a lot compatibilities with the most common cellphones.
 Take a look to this site:http://www.zenitetecnologia.com.br/english/index.jsp Zenite is one of the best companies that build this boxes here in Brazil.
 I hope that I could help you.--[ ]'sDaniel Varella de OliveiraTecnologia IP LtdaTel.: +55 (21)3139-4091 / r. 108www.tecnologiaip.com.br
On Friday 28 October 2005 10:26, Tomasz Chmielewski wrote:> I was wondering if there is something like that on this Earth:>> Some of our users are "mobile users" - they are rarely in one place for
> longer than 15 minutes.> They use mobile phones a lot.>>  From our mobile operator we have an offer which allows us to call for> free between our mobile phones.>> So the idea is to put a SIM card inside the Asterisk box, equipped with
> a special card, a card which would be a mobile phone really.>> This would allow all office users to reach our mobile users without the> need of buying additional phones for the office users.
> Office users would call Asterisk over IAX, and asterisk would call> "mobile users" using a free GSM/mobile.>> Does anyone have an idea if such cards exist, and if so, if they work> with Asterisk?
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Re: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?

2005-10-28 Thread Daniel Varella de Oliveira

Tomasz,

 I'm from Brazil, and we are using here a solution that is based on a box 
where we can connect a GSM cellphone and use this directly to a phone or PBX 
extension.
 I think that you can use some Digium's card (FXS or FXO) on your server, 
connect this GSM box there, and route your cellphone calls through this box.

 There are boxes with just one channel and others up to six channels.
 They have a lot compatibilities with the most common cellphones.

 Take a look to this site: 
http://www.zenitetecnologia.com.br/english/index.jsp
 Zenite is one of the best companies that build this boxes here in Brazil.

 I hope that I could help you.
-- 

[ ]'s

Daniel Varella de Oliveira
Tecnologia IP Ltda
Tel.: +55 (21)3139-4091 / r. 108
www.tecnologiaip.com.br

On Friday 28 October 2005 10:26, Tomasz Chmielewski wrote:
> I was wondering if there is something like that on this Earth:
>
> Some of our users are "mobile users" - they are rarely in one place for
> longer than 15 minutes.
> They use mobile phones a lot.
>
>  From our mobile operator we have an offer which allows us to call for
> free between our mobile phones.
>
> So the idea is to put a SIM card inside the Asterisk box, equipped with
> a special card, a card which would be a mobile phone really.
>
> This would allow all office users to reach our mobile users without the
> need of buying additional phones for the office users.
> Office users would call Asterisk over IAX, and asterisk would call
> "mobile users" using a free GSM/mobile.
>
> Does anyone have an idea if such cards exist, and if so, if they work
> with Asterisk?

pgpW1VfgS5q3m.pgp
Description: PGP signature
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Re: SV: [Asterisk-Users] call queue

2005-10-28 Thread Baris Simsek


yep, thats it.. thank you.

Arne Morten Johansen wrote:


What about making queuemembers phones instead of agents?

Queues.conf: 


[qeuename]
.Blabla.
member => SIP/PhoneName


-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Baris Simsek
Sendt: 28. oktober 2005 14:44
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] call queue

hello,

I want to learn that, is it 'MUST' to login call queue?

I have 3 call queues, and i want to distribute incoming call to the one 
of them. But i don't want to callbacklogin. Because of, after a restart, 
all agents have to do callbacklogin.


thanks...

 




--
Baris Simsek


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[Asterisk-Users] when is 1.2 being released?

2005-10-28 Thread Adam Moffett


does anyone know when 1.2 will no longer be beta?

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[Asterisk-Users] OT: Suggestions for E1 Service in the UK

2005-10-28 Thread Geoff Manning

We are looking to acquire E1 service in Fleet right outside of London. I am
in the States so I am not aware of the key players. We currently get ADSL
from Eclipse but were interested in a quote for E1.

What is a typical E1 line go for nowadays and who can I get it from?

Thanks,
Geoff

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Re: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?

2005-10-28 Thread Mark Elkins

On Fri, 2005-10-28 at 14:26 +0200, Tomasz Chmielewski wrote:

> So the idea is to put a SIM card inside the Asterisk box, equipped with 
> a special card, a card which would be a mobile phone really.

> Does anyone have an idea if such cards exist, and if so, if they work 
> with Asterisk?

You can get "Fixed" Cell units... basically a Cell Phone which provides a Trunk
line instead of screen and keypad. This looks then like an analogue trunk line.
I believe that there is an Italian PCI card that has 4 cell units built
into it. I believe that such units can also plug into an Ethernet and
run SIP.

or - Wait for the Sony Ericsson P990i cell phone which comes with Wifi -
and stick on a SIP client.. and run wireless.
-- 
  .  . ___. .__  Posix Systems - Sth Africa.  e.164 VOIP ready
 /| /|   / /__   [EMAIL PROTECTED]  -  Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496

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Re: [Asterisk-Users] Console detach.

2005-10-28 Thread Rene Caspari

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

* Pepe Aracil [2005-10-28 10:05]:
> I have installed asterisk 1.0.9.dfsg-5 in debian sarge.
> 
> if I run /etc/init.d/asterisk stop and then /etc/init.d/asterisk start . 
> Asterisk don't detach from console where i started it. It beguin to write 
> all warning,debug,AGI dialog,... to the console.
Try to edit /etc/default/asterisk:
Uncomment "RUNASTSAFE=yes".


bye, rene
- -- 
  ___  _   _ ___   __  ___   _  _ ___ ___ 
 / _ \| |_| | _ ) / _|/ _ \ / __/   | \| | __|   |
Rene Caspari|  _  |  _  | . \( (_|  _  |\__ \ _ | .' | _| | |
http://rene.|_| |_|_| |_|_|\_|\__|_| |_|/|_||_|\_|___||_|
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)

iD8DBQFDYh/Oi7H4n8pAp5MRAkf6AKDMybia3q0LOgULUJuDMYIL43XxMwCdEfYE
200aZOmbmJWZfTpzGCGcGhw=
=b0tu
-END PGP SIGNATURE-
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SV: [Asterisk-Users] call queue

2005-10-28 Thread Arne Morten Johansen

What about making queuemembers phones instead of agents?

Queues.conf: 

[qeuename]
.Blabla.
member => SIP/PhoneName


-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Baris Simsek
Sendt: 28. oktober 2005 14:44
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] call queue

hello,

I want to learn that, is it 'MUST' to login call queue?

I have 3 call queues, and i want to distribute incoming call to the one 
of them. But i don't want to callbacklogin. Because of, after a restart, 
all agents have to do callbacklogin.

thanks...

-- 
Baris Simsek
Project Manager
Empatiq Communication Technologies

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[Asterisk-Users] PhoneCALL v2.7 goes MultiLingual

2005-10-28 Thread Dustin Wildes


Hello Everyone!
PhoneCALL version 2.7 http://www.vecsector.com/phonecall is finally 
approaching, which will be a major improvement over the past releases 
thanks to everyone's input & feature requests!
One of the newest features to PhoneCALL is the ability for the entire 
interface to be translated to any language you want/need.  We currently 
have guys working on a Spanish & Russian language file.  It works by 
auto-detecting your language settings of your browser, and by selecting 
your language.


The language file is rather simple to edit (if you are bilingual 
*grin*), and I'm asking for help translating PhoneCALL to your language 
of choice.

Here is a sample format:

///
// GENERAL
//
ALL==All
ADD==Add
EDIT==Edit
DEL==Delete
FIELD==Field
VALUE==Value
NAME==Name
DESCRIPTION==Description
TENANT==Tenant


If you are interested, please contact me & I'll get you started on a 
language file.

We'd love to get as many languages as possible!  :-)

Thank you for your time!



Dustin Wildes
VecSector, LLC
[EMAIL PROTECTED]
http://www.vecsector.com/phonecall
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RE: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?

2005-10-28 Thread Chris Bagnall

> So the idea is to put a SIM card inside the Asterisk box, 
> equipped with a special card, a card which would be a mobile 
> phone really.

There are a number of places that sell GSM gateways (which is what you're
referring to). What I've yet to see are GSM gateways for small business
users that take only one SIM card. Most of the ones I've seen are larger
units designed for 4-16 SIM cards.

If anyone knows of smaller-scale units that work on GSM900 and 1800, I'd
also love to hear about them.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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