Re: [Asterisk-Users] gxp-2000 tftp cfg

[Kristof Hardy]

The VoIP Connection wrote:

It's here:
http://www.thevoipconnection.com/Downloads/GXP2000_1.0.1.9/gxp2000_config.txt


Very interesting, it wasn't available at Grandstream's site :-)
Thanks! I will adjust some things on the page now I have the new 
template.. http://voip-info.org/tiki-index.php?page=GXP-2000


Cheers
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Re: [Asterisk-Users] voip-info.org unreliable lately?

[Kristof Hardy]

Damon Estep wrote:
I assume the bandwidth is being donated or something, but surely someone 
would be willing to donate reliable bandwidth as the knowledge hosted on 
the site (which is also donated!) is worth way more than the bandwidth.


Sure it's the bandwidth? If the wiki is loaded, I see Server load on 
the bottom of the page, the numbers sometimes go as high as 80-100..?


Not sure if it's a Linux (guess so? :p) but if that represents the 
system load.. 80 is a 'bit' high indeed.


Cheers

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Re: [Asterisk-Users] tdm400p not working after cvs-head update

[Paradise Dove]

I have the same problem.
seems that tdm400b is not working on CVS HEAD

On 6/18/05, Steve Totaro [EMAIL PROTECTED] wrote:
  
 did you udate first? 
  
 - Original Message - 
 From: David Romero 
  
 To: Asterisk-Users@lists.digium.com 
 Sent: Friday, June 17, 2005 9:36 AM 
 Subject: [Asterisk-Users] tdm400p not working after cvs-head update 
 
 I hava tdm400p card on [EMAIL PROTECTED] box, this card works fine for weeks,
 
 today i did a CVS update to  the latest head files and the card is not
 working.
 
 
 Zaptel Configuration
 == 
 Channel map:
 Channel 01: FXS Kewlstart (Default) (Slaves: 01)
 Channel 02: FXS Kewlstart (Default) (Slaves: 02)
 Channel 03: FXS Kewlstart (Default) (Slaves: 03)
 Channel 04: FXS Kewlstart (Default) (Slaves: 04)
 4 channels configured.
 ZT_CHANCONFIG failed on channel 1: No such device or address (6)
  
 HELP!.
 
 thanks 
 
 David Romero
 ## 
 
  
  
 
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Re: [Asterisk-Users] Grandstream 100 pricing question

[Francesco Peeters]

On Tue, June 21, 2005 23:07, Kristof Hardy said:
 Francesco Peeters wrote:
 I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the
 pricing for these in Europe, so I'd like to hear from people here
 whether
 that is a reasonable price for them?

 Prices I know are around 99 EUR, incl VAT. But if you ask me, depending
 on how many you need, you should take a look into the GXP-2000. (+- 125
 EUR incl VAT)

 The difference in quality (and features) between these is big enough to
 justify the difference in price.

 Cheers..

Thanks for the info... I am considering getting these to experiment with,
so I can do some testing *before* I actually get in to the real thing. The
cheaper the test period, the better, so 2 of these (which later can be
reused in less used area's) look pretty interesting...

The shop I saw these also sells - pretty cheap -  little devices (forgot
the name, they look like a translucent blue ice-hockey puck) that do SIP
conversion for analog telephones or PBX extensions. (I am thinking
migration period here: first connect one of those to each of the two PBXs
as an extension, so you can use it to 'dial' in to the * server.

Then my migration plan - after initial testing - would then look like this:
1) Install * on both sites
2) IAX2 link
3) 'SIP-puck' on both PBX's and connect these to the * servers
(at this point all users can talk to eachother over landlines *and* SIP)
4) Start migrating inidividual users to SIP  *
5) retire old PBXs

-- 
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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[Asterisk-Users] DID not working? + sendmail problems

[Rick]

I have a box running [EMAIL PROTECTED] 1.0 and I noticed today when I
went to change some settings to do with the DIDs
that it is no longer detecting the different lines.



I have a Digium 4 port line card
and Im pretty sure that the DIDs
used to work when I used fxs_ks signaling on them.
However I changed to fxs_ls signaling because the ks wasnt detecting when people were hanging up
properly. Could changing the signaling be whats
causing the DIDs not to function properly?





My other problem is getting sendmail
to work. I get the following error.



 - Transcript of session follows -... while talking to my smtp server, details removed: DATA 550-Verification failed for [EMAIL PROTECTED] 550-unrouteable mail domain asterisk1.local 550 Sender verify failed550 5.1.1 users email address, details removed... User unknown 503 valid RCPT command must precede DATA



I cant find anywhere to change the
login/pass info to change it to auth with a valid account from the mail server.



Any help would be greatly appreciated.






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[Asterisk-Users] ASTERISK+SER+MWI

[harry gaillac]

Hello,

I wish to use asterisk as voicemail system with ser 
My phones are registered on SER and asterisk provide
conference ivr and voicemail system.

I want to use Asterisk Realtime Architecture to store
voicemessages and configuration .

Michael Shuler has written a patch for mwi.
http://bugs.digium.com/bug_view_page.php?bug_id=0002980

I need help to set up voicemail storage and how to
send mwi to clients registered on ser

Regards
Harry






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Re: [Asterisk-Users] Help on installing h323

[Nardis Dome]

Hi,

install oh323... see link for installation

http://lists.digium.com/pipermail/asterisk-users/2005-April/100061.html

--- craz sead [EMAIL PROTECTED] wrote:

 Hi all
 
 could somebody help me how to install and setup H323
 i
 would like to connect asterisk box with
 huawei/cisco,
 but i still dont understand about installing h323 on
 asterisk
 
 thaks
 
 
   
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[Asterisk-Users] 3-way conference using zap channels -- how is it done?

[Dimitris Kouimintzis]

My configuration is one TDM31B for internal subscribers and one AVM B1 v.4 for incoming/outgoing calls.

How do I:

a) transfer a call to an internal subscriberb) commence a 3-way conference (two internal - one external subscribers)

I've searched voip-info.org and google god but regarding to zap
channels documentation is not clear, if any, especially about 3-way
conference.

Thank you
-- Dimitris Kouimintzis

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[Asterisk-Users] zeroconf help

[stevanus]

hi,

recently I installed zeroconf for asterisk...
I've already followed the asterisk+zeroconf how to (which is too short), 
but it came with an error message...


asterisk: relocation error: /usr/lib/asterisk/modules/res_zeroconf.so: 
undefined symbol: DNSServiceRegister

Ouch ... error while writing audio data: : Broken pipe

it's weird since I've double checked the library and header from 
zeroconf and it seems that everything has been in the right place..


Is there anyone can help me? Well, it seems I hit another dead end this 
time...


Best regards,

Stevanus

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[Asterisk-Users] Core Dump

[Chee Foong]

Hello,
My asterisk crash from time to time, at least twice a day.
Reomendation from a site is to do a gdb on core dump

Below is what i get, but i have no idea what is going on
Does anybody have any idea?


(gdb)bt
#0  0x00181aed in _int_malloc () from /lib/tls/libc.so.6
#1  0x00180dfd in malloc () from /lib/tls/libc.so.6
#2  0x00177b03 in vasprintf () from /lib/tls/libc.so.6
#3  0x0808b663 in ast_cli (fd=1, fmt=0x1 Address 0x1 out of bounds) at
cli.c:54
#4  0x080a0725 in manager_event (category=2, event=0x80e812e Newexten,
fmt=0x80e6da0 Channel: %s\r\nContext: %s\r\nExtension: %s\r\nPriority:
%d\r\nApplication: %s\r\nAppData: %s\r\nUniqueid: %s\r\n) at manager.c:1420
#5  0x08089345 in pbx_extension_helper (c=0x9815758, con=0x1,
context=0x98158a8 macro-hangupcall, exten=0x981599c s,
priority=3, label=0x0, callerid=0x96bc008 Wait, action=159472384) at
pbx.c:1609
#6  0x08087767 in ast_spawn_extension (c=0x1, context=0x1 Address 0x1 out
of bounds,
exten=0x1 Address 0x1 out of bounds, priority=1, callerid=0x1 Address
0x1 out of bounds) at pbx.c:2206
#7  0x008c96a4 in macro_exec (chan=0x9815758, data=0x200) at app_macro.c:173
#8  0x0808938b in pbx_extension_helper (c=0x9815758, con=0x1,
context=0x98158a8 macro-hangupcall, exten=0x981599c s,
priority=1, label=0x0, callerid=0x1cef1d0 hangupcall, action=0) at
pbx.c:528
#9  0x08080c27 in ast_pbx_run (c=0x9815758) at pbx.c:2206
#10 0x00683d41 in ss_thread (data=0x9815758) at chan_zap.c:4975
#11 0x00669dac in start_thread () from /lib/tls/libpthread.so.0
#12 0x001eb9ea in clone () from /lib/tls/libc.so.6

thanks

Regard
CCF


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Re: [Asterisk-Users] Grandstream 100 pricing question

[Ming-Wei Shih]

On Wed, Jun 22, 2005 at 08:35:58AM +0200, Francesco Peeters wrote:
 The shop I saw these also sells - pretty cheap -  little devices (forgot
 the name, they look like a translucent blue ice-hockey puck) that do SIP
 conversion for analog telephones or PBX extensions. (I am thinking
 migration period here: first connect one of those to each of the two PBXs
 as an extension, so you can use it to 'dial' in to the * server.

Is this a webshop in Europe? Care to share the URL?

Regards

Ming-Wei
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[Asterisk-Users] 3month Internship between February end July 2006

[thomas DEILLON]

Hello,

I make a sawdwish course in network and software engeneering at CPE lyon and 
in my company I'm working on Asterisk from 1 year.
So, I'm looking for a internship (3 month) in a english country on a Asterisk 
project.

Thanks,

Thomas DEILLON
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Re: [Asterisk-Users] Grandstream 100 pricing question

[Francesco Peeters]

On Wed, June 22, 2005 10:02, Ming-Wei Shih said:
 On Wed, Jun 22, 2005 at 08:35:58AM +0200, Francesco Peeters wrote:
 The shop I saw these also sells - pretty cheap -  little devices (forgot
 the name, they look like a translucent blue ice-hockey puck) that do SIP
 conversion for analog telephones or PBX extensions. (I am thinking
 migration period here: first connect one of those to each of the two
 PBXs
 as an extension, so you can use it to 'dial' in to the * server.

 Is this a webshop in Europe? Care to share the URL?

 Regards

 Ming-Wei

Sorry, it's a real shop in Zoetermeer, the Netherlands, I visit sometimes...

They do have a website, but I doubt whether they have all their stuff on
there (they often have these small lots of special kit in the store) and
whether they have a webstore...

http://www.telec.com



-- 
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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[Asterisk-Users] using DBGet inside extensions.ael

[Chris Stenton]

Anyone know how you can use DBGet inside extensions.ael?

Thanks

Chris

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[Asterisk-Users] Sip client

[gale81]

Hello!

If I want to build a Sip client application in Java .
What kind of Java Api would I use to connect to the server and to implement
the sip signaling?
Thanks Ale



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[Asterisk-Users] Asterisk to NEC NEAX

[Ramkumar]

Hi,



How can I make calls from Asterisk client to NEC NEAX 2400
traditional phone ?



Is it possible to have a connection between Asterisk and NEC
NEAX 2400, since NEC-NEAX2400 is an IP-PBX and supports SIP.



Please help me to find a solution ;;;



Thanks  Regards

Ram Kumar

Customer Support Engineer

Barcode Gulf LLC

Dubai , UAE

Mobile : + 971 50 5594178

Email :
[EMAIL PROTECTED]








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[Asterisk-Users] PPPD problem please help

[Daniel Nyström]

After a big help from Peter Svensson, I got ISDN Data-calls up and 
running.
But now when everything seems connected, pppd has been authorized by 
other peer and even got an IP address, the whole connection seems to 
stop working.
Very unregulary, the PPPD's EchoReq's stop being answered, and of course 
all TCP/IP-traffic as well.

It takes between 0-2 minutes till the connection breaks.
I've setup 'lcp-echo-interval 30' and 'lcp-echo-failure 4' (something 
like that). And it could reply to 1-2 EchoReq's sometimes, sometimes none.

BUT!
The really strange thing is; when my pppd has determined that the serial 
link is closed (the ISDN connection), it will send an Terminate Request!
And that Terminate Request are confirmed by other peer! How strange is 
that?!
I've spoken to my ISP and even got their log from my call. And it all 
seems correct from their side. The EchoReq's doesn't seems to reach them 
though. But the Terminal Request does afterwards.


I think this is a Nobel Prize issue. :)
--
Daniel

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[Asterisk-Users] call divert to TRUNK , if one number is unregistered?

[Erdem HAKİ]

I have a question.



I have two numbers on Asterisk like 902121234567 and 902123645789
and i want to divert first numbers call to Trunk if second number is
unregistered. Is it possible? İf yes, how?



Flow Diagram:



*Two numbers are registered on Asterisk



902121234567 registered to
Asterisk

   

902123645789 registered to
Asterisk



*One number is registered, other one is not registered



902121234567 registered to
Asterisk

   

902123645789-x  not registered to
Asterisk



*So first number want to make a call second one (desired
situation)



902121234567  Asterisk à Trunk





Thanks for your interest.



Erdem HAKI  [EMAIL PROTECTED]






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Re: [Asterisk-Users] GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?

[Leandro Morgado]

Steve Underwood wrote:

 Robert Rozman wrote:

 Hi,

 I'm getting unreliable dtmf recognition (it works fine for 4-5
 digits, errors (duplicates) on more), when transferred inband from
 gsm gateway to NT port of quadbri under bristuffed Asterisk.

 Since Asterisk is claimed to have good dtmf recognizer, I suspect
 there are some settings to workarouned... I've tried dtmf relax, but
 didn't help, so I suspect gain settings

 Is there any other possible cause of unreliable dtmf inband
 recognition ? Where can I set gain on voice channel (I guess majority
 of settings under bristuff in zaptel.conf are dummy) ?

 Any other advice on this problem or similar experience ?

 Thanks in advance,


 I kind of amazed if works at all when getting DTMF out of a GSM phone.
 You really shouldn't expect it to.

We have sucessfully read incoming DTMF from:

a) Nokia32 Analog GSM connected to TDM400 (had to use relaxdtmf with
chan_zap)
b) Ateus BRI ISDN GSM connected to AVM Fritz (had to patch chan_capi
0.3.5 to support relaxdtmf)


Question (I'm from a software eng. background, not telco):
What would be the reason for not receiving DTMF from a GSM
phone/gateway? Do you have the time to explain why? (I'm really
interested in learning :)

Thanks,

Leandro

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Re: [Asterisk-Users] Sip client

[Dave Walker]

[EMAIL PROTECTED] wrote:


Hello!

If I want to build a Sip client application in Java .
What kind of Java Api would I use to connect to the server and to implement
the sip signaling?
Thanks Ale



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Have you tried JAIN?
http://www.google.co.uk/search?q=sip+java+api
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Re: [Asterisk-Users] Group/Broadcast Voicemail

[Zen Kato]

Hi,

I also changed as following sequences;

app_voicemail.c

1. Line 3724 tmp[256] to tmp[4096]  vm_exec
2. Line 3760 tmp[256] to tmp[4096]  append_mailbox
3. Line 3796 tmp[256] to tmp[4096]  vm_box_exists
4. Line 3290 tmp[256] to tmp[4096]  vm_execmain
5. Line 80   tmp[256] to tmp[4096]  #define BASEMAXLINE
6. Line 82   tmp[256] to tmp[4096]  #define BASEMAXLINE

I tried to copy to 99 mailboxes, but no luck, only could copy to 51 mailboxes. 

-- Executing VoiceMail(SIP/1021-6bd9, u010302030303040305030603
070308030903100311031203130314031503160317031803190320032103
220323032403250326032703280329033003310332033303340335033603
370338033903400341034203430344034503460347034803490350035103
520353035403550356035703580359036003610362036303640365036603
670368036903700371037203730374037503760377037803790380038103
820383038403850386038703880389039003910392039303940395039603
970398039903) in new stack

(snip)..
-- User ended message by pressing #
-- Playing 'auth-thankyou' (language 'en')
Jun 22 17:15:20 NOTICE[11044]: app_voicemail.c:1244 copy_message: Copying 
message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 22 17:15:25 WARNING[11044]: app.c:994 ast_lock_path: Failed to lock path 
'': File exists
.(snip)..
Jun 22 17:15:25 NOTICE[11044]: app_voicemail.c:1244 copy_message: Copying 
message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Unable to create lock file: No such file or directory

I would like to copy to 100-150 mailboxes for one CPU.

I also need someone's help.

Regards,

Zen Kato


 I did change char tmp[4096], *ext; to 4096 but there's also the same 
 line under vm_execmain but I really don't know anything about 
 programming. I only saw the same line.
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Re: [Asterisk-Users] Grandstream 100 pricing question

[Pavel Jezek]

I had gxp-2000 for testing some days, but features are (in current 
firmware) _very_ limited!
phone does not have missed, dialed numbers, phone book, speakerphone is 
useless...

phone have nice backlight display and in-line power :-)
but if you like features, grandstream is not for you...
PJ

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[Asterisk-Users] Variables to emailbody of voicemail

[Gabriel Perez S.]

Hi,

I am trying to send a variable to emailbody of voicemail.conf.  That is
to say, I try to send I number that it is entered by the numeric keypad
of the telephone and arrives in the body of the mail with the recording
of the message.

In the single documentation they mark some variables that can be formed
in voicemail.conf but encounter does not form to send a new variable.
Try to replace the value of one of these variables with script AGI that
gathers the numbers but it did not give result.  Some idea?

Thanks.

-- 
Gabriel Perez S.
System  Network Development
-
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[Asterisk-Users] ZapRAS

[Daniel Nyström]

I'm trying to use ZapRAS to enable ppp connection through my E1.
After the ZapRAS command is executed, all sound is crappy on all lines!
The only solution is to reboot the machine (or halt it, and then power 
it on since Digium's hardware doesn't like reboots).

Anyone know how this can happen?!
I'm using * 1.0.6 on Dell PowerEdge 1850 which are told (too late 
though) not to work very well, but should that really be the problem 
with ZapRAS?!

--
Daniel
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[Asterisk-Users] Dialplan Q: Dialing with Capi

[Patrik Schindler]

Hello,

I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX and CAPI 
as channels. A call comes in via IAX2 and should be redirected to CAPI.

So I wrote the following dialplan:

[fromiax]
exten = _8XXX,1,Answer
exten = _8XXX,2,Dial(CAPI/265:B${EXTEN:1},,r)

[fromcapi]
exten = 265,1,Answer
exten = 265,2,Dial(IAX2/PoC/[EMAIL PROTECTED])
exten = 265-BUSY,1,Busy
exten = 265-NOANSWER,1,Busy

[default]
exten = s,1,Answer
exten = s,2,Congestion


The asterisk on the described side is connected to a classic company pbx 
(from Ericsson) and has local Extensions _XXX. The CAPI Interface gets 
signalled with MSN 260-265.

Yesterday I rewrote the dialplan to allow calls from any company phone 
(_XXX) to 265 will be rerouted to PoC/11 (the IAX peer). This works fine, 
the IAX peer will be called with MSN 265 as callerid, so the called party 
can see the number for callback.

I'm encountering the following problem with this setup, when the iax peer 
dials 8253, which should be redirected to CAPI/253 (8 is a extension 
prefix specific to the iax peer configuration).

-- Accepting AUTHENTICATED call from xx.xx.xx.xx, requested format = 4, 
actual format = 4
-- Executing Answer(IAX2/[EMAIL PROTECTED]/3, ) in new stack
-- Executing Dial(IAX2/[EMAIL PROTECTED]/3, CAPI/265:B253||r) in new stack
Jun 22 11:47:22 NOTICE[13337]: chan_capi.c:1173 capi_request: didn't find 
capi device with outgoing msn = 265. you should check your config!
Jun 22 11:47:22 NOTICE[13337]: app_dial.c:746 dial_exec: Unable to create 
channel of type 'CAPI'
== Everyone is busy/congested at this time
-- Hungup 'IAX2/[EMAIL PROTECTED]/3'

When I change the dialplan to the outgoing MSN will be 260 than it works 
as expected: 253 will be called, but the called party gets 260 as callerid 
instead of the 265.

My capi.conf looks like this:

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=de

[interfaces]
msn=260
incomingmsn=*
outgoingmsn=260,261,263,264,265
controller=1
softdtmf=1
context=fromcapi
language=de
echocancel=yes
devices=2

Okay, I could also set msn=265; but in the future, more msns should be 
redirected to VoIP channels, so this is not a long term solution.

Any help is very welcome! Thanks for asterisk!

:wq! PoC
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[Asterisk-Users] Detecting the active queue agent...

[Guilherme Lopes]

Hello

I need to know who's the agent that picks up a call in a queue, preferably in
real-time so I can invoke a script that sends info to the agent's PC. It
doesn't seem easy to do unless I use the info on asterisk's log file (using
swatch or something? is this possible?) ... is there any other way? Anyone?

NOTE: I've succesfully tried to use firefly as an IAX2 extension to 
make an URL

popup appear on the agent BUT only using Dial ... Queue doesn't seem to do the
same thing, even having a field for that purpose. Anyway, I would much 
prefer a

more 'neutral' solution in order to use whichever softphone I choose.

Guilherme



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[Asterisk-Users] OT: Asterisk and Mambo - help wanted

[Kristian Kielhofner]

Hello everyone,

	So, this isn't exactly what it seems.  I am not looking to integrate 
Asterisk and Mambo.  I am the maintainer/creator of AstLinux, and I have 
recently decided that I should really have a better web site for it.  I 
would like to use Mambo so that I can do updates easily, from anywhere, 
without having to waste time learning PHP/HTML/etc.  Mambo CMS seems the 
best and most powerful way to do this.  It's not that easy, however, to 
go from the default Mambo site to a site suitable for an open source 
project such as AstLinux.  I just need some help to get a layout, theme, 
etc. going.  Updates and maintenance I can handle (probably).


	So, what I am looking for is someone who is familiar with Mambo (and 
preferably Asterisk, too) and would be willing to help me jump start 
astlinux.org/.com.  Because AstLinux is an open source project, I will 
be unable to directly compensate anyone (monetarily) for their work at 
this time.  However, any people that help out are more than welcome to 
plug their own projects, companies, names, etc. on the site (within reason).


Interested?  Comments?  Questions?  Suggestions?  Drop me a line.

Thanks!

--
Kristian Kielhofner
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Re: [Asterisk-Users] Dialplan Q: Dialing with Capi

[jurczak]

You could try the new chan_capi-cm-0.5.1 in which you dont need to have any 
msn defined in capi.conf

On Wed, 22 Jun 2005 12:28:58 +0200 (MEST), Patrik Schindler wrote
 Hello,
 
 I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX 
 and CAPI as channels. A call comes in via IAX2 and should be 
 redirected to CAPI.
 
 So I wrote the following dialplan:
 
 [fromiax]
 exten = _8XXX,1,Answer
 exten = _8XXX,2,Dial(CAPI/265:B${EXTEN:1},,r)
 
 [fromcapi]
 exten = 265,1,Answer
 exten = 265,2,Dial(IAX2/PoC/[EMAIL PROTECTED])
 exten = 265-BUSY,1,Busy
 exten = 265-NOANSWER,1,Busy
 
 [default]
 exten = s,1,Answer
 exten = s,2,Congestion
 
 The asterisk on the described side is connected to a classic company 
 pbx 
 (from Ericsson) and has local Extensions _XXX. The CAPI Interface 
 gets signalled with MSN 260-265.
 
 Yesterday I rewrote the dialplan to allow calls from any company 
 phone 
 (_XXX) to 265 will be rerouted to PoC/11 (the IAX peer). This works 
 fine, the IAX peer will be called with MSN 265 as callerid, so the 
 called party can see the number for callback.
 
 I'm encountering the following problem with this setup, when the iax 
 peer dials 8253, which should be redirected to CAPI/253 (8 is a 
 extension prefix specific to the iax peer configuration).
 
 -- Accepting AUTHENTICATED call from xx.xx.xx.xx, requested format = 
 4, actual format = 4 -- Executing Answer(IAX2/[EMAIL PROTECTED]/3, ) in 
 new stack -- Executing Dial(IAX2/[EMAIL PROTECTED]/3, CAPI/265:B253||r) 
 in 
 new stack
 Jun 22 11:47:22 NOTICE[13337]: chan_capi.c:1173 capi_request: didn't 
 find capi device with outgoing msn = 265. you should check your config!
 
 Jun 22 11:47:22 NOTICE[13337]: app_dial.c:746 dial_exec: Unable to 
 create channel of type 'CAPI' == Everyone is busy/congested at this time
 -- Hungup 'IAX2/[EMAIL PROTECTED]/3'
 
 When I change the dialplan to the outgoing MSN will be 260 than it 
 works as expected: 253 will be called, but the called party gets 260 
 as callerid instead of the 265.
 
 My capi.conf looks like this:
 
 [general]
 nationalprefix=0
 internationalprefix=00
 rxgain=0.8
 txgain=0.8
 language=de
 
 [interfaces]
 msn=260
 incomingmsn=*
 outgoingmsn=260,261,263,264,265
 controller=1
 softdtmf=1
 context=fromcapi
 language=de
 echocancel=yes
 devices=2
 
 Okay, I could also set msn=265; but in the future, more msns should 
 be redirected to VoIP channels, so this is not a long term solution.
 
 Any help is very welcome! Thanks for asterisk!
 
 :wq! PoC
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[Asterisk-Users] Re: app_changrab.c released on pbxfreeware.org

[Nenad Radosavljevic]

Hi !

Managed to fix app_changrab.c to compile and start working under 1.0.X.

it is working on my installation, but is tested well enough.

Regards,
   Nenad

Here is diff -u :

--

--- app_changrab.c.orig 2005-06-20 22:10:50.0 +0200
+++ app_changrab.c  2005-06-22 11:43:54.0 +0200
@@ -8,11 +8,12 @@
 */

/*uncomment below or build with -DAST_10_COMPAT for 1.0 */
-//#define AST_10_COMPAT
+#define AST_10_COMPAT

#include asterisk/file.h
#include asterisk/logger.h
#include asterisk/channel.h
+#include asterisk/channel_pvt.h
#include asterisk/pbx.h
#include asterisk/utils.h
#include asterisk/musiconhold.h
@@ -24,9 +25,11 @@
#include string.h
#include pthread.h

-#include asterisk.h
+#include ../asterisk.h

+#ifndef AST_10_COMPAT
ASTERISK_FILE_VERSION(__FILE__, $Revision: 1.39 $)
+#endif

static char *tdesc = Take over an existing channel and bridge to it.;
static char *app = ChanGrab;
@@ -85,10 +88,13 @@
   ast_log(LOG_WARNING, No Such Channel: %s\n,(char *) data);
   return -1;
   }
-
+#ifndef AST_10_COMPAT
   if(flags  oldchan-_bridge  strchr(flags,'b'))
   oldchan = oldchan-_bridge;
-
+#else
+   if(flags  oldchan-bridge  strchr(flags,'b'))
+   oldchan = oldchan-bridge;
+#endif
   if(flags  strchr(flags,'r')  oldchan-_state == AST_STATE_UP) {
   return -1;
   }
@@ -102,9 +108,10 @@
   if((f = ast_read(newchan))) {
   ast_frfree(f);
   memset(config,0,sizeof(struct ast_bridge_config));
+#ifndef AST_10_COMPAT
   ast_set_flag((config.features_callee), 
AST_FEATURE_REDIRECT);
   ast_set_flag((config.features_caller), 
AST_FEATURE_REDIRECT);

-
+#endif
   if(oldchan  !oldchan-pbx)
   ast_hangup(oldchan);




Unfortunatly it won't compile under 1.0.7 :(

I have uncommented #define AST_10_COMPAT but I don't see any usage of it 
in

app_changrab.c.

Complains about missing asterisk.h ( I think it should be #include
../asterisk.h )

It also complains about ASTERISK_FILE_VERSION() function, and about 
_bridge

member of ast_channel structure and some othet things:

Here is compile log:

 . 




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Re: [Asterisk-Users] Grandstream 100 pricing question

[Peter Svensson]

On Wed, 22 Jun 2005, Pavel Jezek wrote:

 I had gxp-2000 for testing some days, but features are (in current 
 firmware) _very_ limited!
 phone does not have missed, dialed numbers, phone book, speakerphone is 
 useless...

Some of these features are in the 1.0.1.9 version that was released last 
week. Missed and dialed numgers are available, although not in a very 
good interface (press the left and right arrows while off-hook). They do 
have separate memories per configured account.

Grandstream clains thay will address the speakerphone problems in an 
upcoming release. I think they need a more advanced echo canceler since 
the speaker and microphone are acoustically strongly coupled.

Also expected in the near term is suppor for Subscribe/Notify.

 phone have nice backlight display and in-line power :-)
 but if you like features, grandstream is not for you...

On the other hand Grandstream seem to care about what their users want, at 
least for minor features. Everything we asked for was included in the 
current release. They seem serious in their attempt to break into the 
higher end market.

Peter

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Re: [Asterisk-Users] GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?

[Peter Svensson]

On Tue, 21 Jun 2005, Leandro Morgado wrote:

 Steve Underwood wrote:
  Robert Rozman wrote:
  I'm getting unreliable dtmf recognition (it works fine for 4-5
  digits, errors (duplicates) on more), when transferred inband from
  gsm gateway to NT port of quadbri under bristuffed Asterisk.

We get these quite often. If there is any line noise asterisk will 
interpret it as the end of a digit and then detect the same digit again. 
We are connected to the pstn via isdn. The problem is with calls where the 
dtmf tones are a bit unclean, i.e. too much energy is in the overtones. 
Clean dtmf tones seem to be much more resistant to line noise.

Out other systems are more accepting of slightly off-spec dtmf tones.

Peter



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[Asterisk-Users] is sip:%2321 valid invite?

[Domjan Attila]

Hi,

I tried to cable #21 with a thomson cable modem mta:

-- SIP read from 192.168.153.100:5060:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.153.100;branch=z9hG4bK1aa77a586
Max-Forwards: 70
Content-Length: 258
To: #21 sip:[EMAIL PROTECTED]:5060
From: sip:[EMAIL PROTECTED]:5060;tag=da42eb89613306c
Call-ID: [EMAIL PROTECTED]
CSeq: 1084359157 INVITE
Supported: timer
Allow: NOTIFY
Allow: REFER
Allow: OPTIONS
Allow: INVITE
Allow: ACK
Allow: CANCEL
Allow: BYE
Content-Type: application/sdp
Contact: sip:[EMAIL PROTECTED]
Supported: replaces
User-Agent: Brcm Callctrl/1.5.1.2 MxSF/v3.2.6.26

and asterisk doesn't translate %23 to #.

the grandstrem phones send it in this case:

-- SIP read from 192.168.50.224:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.50.224;branch=z9hG4bKba695b72a347ad40
From: sip:[EMAIL PROTECTED];tag=33c0bbf7cfe3e083
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Supported: replaces
Call-ID: [EMAIL PROTECTED]
CSeq: 4050 INVITE
User-Agent: Grandstream BT100 1.0.6.6
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 336

Anybody knows what says about it rfc?
Is it bug of mta or asterisk?

Regards,
Attila Domjan



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[Asterisk-Users] Is this server sufficient?

[Francesco Peeters]

I've tried to find some details on the wiki, but was unable to get a
satisfactory result, so I am asking here:

I have a Linux (FC3) box with these specs:

vendor_id   : AuthenticAMD
cpu family  : 6
model   : 3
model name  : AMD Duron(tm) Processor
stepping: 1
cpu MHz : 797.388
cache size  : 64 KB

MEM: currently 256, looking to upgrade to 512/768 (depending on available
sticks)

HDD: 80 GB

It is currently doing File/Printer serving.

Ideally I'd want it to do Asterisk (2 ISDN BRI  8 phones), File/Printer
server on a home network (3 clients) and some light SMTP ( 100 emails a
day)

Is this machine sufficient for the task? (Ignoring the fact it needs
either a multi-BRI card or 2 single BRI cards to be able to connect to the
PSTN G)

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] call divert to TRUNK ,if one number is unregistered?

[Mohamed A. Gombolaty]

Hi Erdem,
Can you try to put another dial command that points to the trunk afetr
the dial command to the SIP?
fro example:
exten => XXX,1, dial(sip/,20,r)
exten => XXX,2,dial(zap/) ->
note here that I am not sure if the order number should be 2 or 102 but
if this didn't work try the other one.
Thx
MAG




Erdem HAKÝ wrote:


I have a question.




I have two numbers
on Asterisk like 902121234567 and 902123645789 and i want to divert first
number's call to Trunk if second number is unregistered. Is it possible?
Ýf yes, how?




Flow Diagram:



*Two
numbers are registered on Asterisk



902121234567
registered to Asterisk



902123645789
registered to Asterisk



*One
number is registered, other one is not registered



902121234567
registered to Asterisk



902123645789-x
not registered to Asterisk



*So
first number want to make a call second one (desired situation)



902121234567>
Asteriskà
Trunk





Thanks
for your interest.



Erdem
HAKI - [EMAIL PROTECTED]


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--
Thx
MAG



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RE: [Asterisk-Users] Is this server sufficient?

[Dean Collins]

As an asterisk server it is more than fine but asterisk prefers to be a
standalone machine.

You would have a lot less issues if you had 2 machines, one handling
file serving, SMTP and one dedicate machine for asterisk.

Voice isn't very tolerant of interrupts.


Cheers,
Dean
 

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Francesco Peeters
 Sent: Wednesday, 22 June 2005 7:28 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Is this server sufficient?
 
 I've tried to find some details on the wiki, but was unable to get a
 satisfactory result, so I am asking here:
 
 I have a Linux (FC3) box with these specs:
 
 vendor_id   : AuthenticAMD
 cpu family  : 6
 model   : 3
 model name  : AMD Duron(tm) Processor
 stepping: 1
 cpu MHz : 797.388
 cache size  : 64 KB
 
 MEM: currently 256, looking to upgrade to 512/768 (depending on
available
 sticks)
 
 HDD: 80 GB
 
 It is currently doing File/Printer serving.
 
 Ideally I'd want it to do Asterisk (2 ISDN BRI  8 phones),
File/Printer
 server on a home network (3 clients) and some light SMTP ( 100 emails
a
 day)
 
 Is this machine sufficient for the task? (Ignoring the fact it needs
 either a multi-BRI card or 2 single BRI cards to be able to connect to
the
 PSTN G)
 
 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] Is this server sufficient?

[Francesco Peeters]

On Wed, June 22, 2005 13:39, Dean Collins said:
 As an asterisk server it is more than fine but asterisk prefers to be a
 standalone machine.

 You would have a lot less issues if you had 2 machines, one handling
 file serving, SMTP and one dedicate machine for asterisk.

 Voice isn't very tolerant of interrupts.


 Cheers,
 Dean



I am aware of that, but the server is doing nada 99.9% of the time right
now, so I'd rather give up the other functionality and have it to * rather
than the other way round!  ;-)

I thought I'd give it a try with * and see whether we have issues when the
rare SMTP/SMB access occurs (and deal with it then!)

I just wanted to be sure that the machine is sufficient to do * and then
some...  ;-)

I think I'll use the upcoming vacation period to go play with it then!  :-D

Cheers!

-- 
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] Is this server sufficient?

[Andrew Latham]

This will work fine I have a traveling 700mhz duron that does
great. CUPS should not be running but test it your self.

Rule of thumb, if you are transcoding many channels then you need a
bigger machine. If you are just switching then you can use a smaller
machine.


On 6/22/05, Francesco Peeters [EMAIL PROTECTED] wrote:
 On Wed, June 22, 2005 13:39, Dean Collins said:
  As an asterisk server it is more than fine but asterisk prefers to be a
  standalone machine.
 
  You would have a lot less issues if you had 2 machines, one handling
  file serving, SMTP and one dedicate machine for asterisk.
 
  Voice isn't very tolerant of interrupts.
 
 
  Cheers,
  Dean
 
 
 
 I am aware of that, but the server is doing nada 99.9% of the time right
 now, so I'd rather give up the other functionality and have it to * rather
 than the other way round!  ;-)
 
 I thought I'd give it a try with * and see whether we have issues when the
 rare SMTP/SMB access occurs (and deal with it then!)
 
 I just wanted to be sure that the machine is sufficient to do * and then
 some...  ;-)
 
 I think I'll use the upcoming vacation period to go play with it then!  :-D
 
 Cheers!
 
 --
 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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[Asterisk-Users] meetme problem

[Felix Skwarczynski]

Hello,

I seem to have a strange problem, which appeared out of nowhere.
Did anyone see something like this?

2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Illegal seek

2005-06-03 18:44:50 DEBUG[10721] res_agi.c: Zap/65-1 hungup
2005-06-03 18:44:50 DEBUG[10721] channel.c: Avoiding deadlock for 'Zap/63-1'
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame 
to channel: Success


Once these messages start showing, I must stop my asterisk (stop now) 
because the load goes sky high.

I'm using an Asterisk CVS-HEAD.

Looking forward for your answer.


Felix

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RE: [Asterisk-Users] Is this server sufficient?

[Simone Cittadini]

Il giorno mer, 22/06/2005 alle 07.39 -0400, Dean Collins ha scritto:
 As an asterisk server it is more than fine but asterisk prefers to be a
 standalone machine.
 
 You would have a lot less issues if you had 2 machines, one handling
 file serving, SMTP and one dedicate machine for asterisk.
 
 Voice isn't very tolerant of interrupts.
 
 

In my case, before switching to dedicated hw, asterisk was running on a
file/printer/ldap/web server, since we have an userbase of 10 people
dialing out on 2 isdn lines the glitches in audio weren't a real problem
because the probability of having an active call while the server is
used for some other intensive task is really low with few users.

If you're on budget imho you can start by implementing * on the hw you
have, and eventually switch to a dedicated machine, just be sure X isn't
running on the server.

(of course the idea of putting * directly on a production machine and
not on a test one isn't a good one, but this is another topic)

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Re: [Asterisk-Users] Is this server sufficient?

[Lists]

On Wednesday 22 June 2005 07:27, Francesco Peeters wrote:
 It is currently doing File/Printer serving.

 Ideally I'd want it to do Asterisk (2 ISDN BRI  8 phones), File/Printer
 server on a home network (3 clients) and some light SMTP ( 100 emails a
 day)

 Is this machine sufficient for the task? (Ignoring the fact it needs
 either a multi-BRI card or 2 single BRI cards to be able to connect to the
 PSTN G)

I don't see a reason why it won't work. I have done the same on a 600MHz 
machine and as long as I did not have anything major going on it worked fine. 
Of course you will find that you can have potential problems with all sorts 
of h/w once you put some load on Asterisk. 

Another box I tried was a top of the line Intel w dual Xeon (also 600MHz) that 
once cost over $10,000. It did not work too well, even on single phone calls. 
Then I tried an IBM 600MHz single processor which does just fine.

You will notice rather easily when your h/w cannot keep up. Ideally all you do 
on an Asterisk box is run Asterisk, but as I said you'll notice when it's not 
up to the task. When you use cpu intensive codecs you'll see when your 
machine is not up to it. Just dive in and try it.

I think someone here got a 233MHz machine to handle single calls...

As long as you use Linux you won't have any problems installing Asterisk just 
to test it. Unlike other O/S's it won't mess anything up.

Good Luck!
-- 

List Manager
Network Voice Comunications, Inc.
netwvcom.com
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[Asterisk-Users] Spanish doc

[Leonardo F. Bauchwitz]

Hi:
We have finished the translation of the FAQ of Digium to spanish.
They are already (in Spanish) available for download (in 
http://ourproject.org/projects/asterix/):


* FAQ  Frequently Asked Questions
* Features
* Hardware compatibility list
* Fast Installation Zaptel
All the documentation is available for download

Soon the following documents will be finished:
Volume one and Asterisk Gateway Interface (AGI)

Bye

Leonardo Federico Bauchwitz
Coordinator of Asterisk documentation in Spanish
https://ourproject.org/projects/asterix/
[EMAIL PROTECTED]






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Re: [Asterisk-Users] How can you check that eg TDM04B hardwareinstalled and drivers OK

[Angus Comber]

If I try dmesg - no mention of a Wildcard TDM400.

Sorry I am fairly new to Linux.  In Windows I suppose I would run some 
hardware program which came with the card to see if I could manually set 
IRQ's etc.  What should I be looking at now?


Please feel free to point me to a good book or whatever you feel is 
appropriate.  Could the card be faulty?


My motherboard is an Intel D865GLC.

I am running [EMAIL PROTECTED] version 1.0

Angus




- Original Message - 
From: Mike M [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Tuesday, June 21, 2005 3:14 PM
Subject: Re: [Asterisk-Users] How can you check that eg TDM04B 
hardwareinstalled and drivers OK




On Tue, Jun 21, 2005 at 07:09:46AM -0600, Rich Adamson wrote:


 I am struggling to get my TDM04B working.  Just to rule out a hardware 
 problem how can I check

that the hardware works?  How can I then
 check that the drivers are loaded correctly?


1. from the linux command line, type 'dmesg' and look for
 Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules)
if you see that, the TDM card is recognized by the OS.



Here's what I get on a working system:

[EMAIL PROTECTED] src]# modprobe wctdm
[EMAIL PROTECTED] src]# Jun 21 10:10:55 b2 kernel: PCI: Found IRQ 11 for device
01:0a.0
Jun 21 10:10:55 b2 kernel: Freshmaker version: 71
Jun 21 10:10:55 b2 kernel: Freshmaker passed register test
Jun 21 10:10:55 b2 kernel: Module 0: Installed -- AUTO FXS/DPO
Jun 21 10:10:55 b2 kernel: Module 1: Not installed
Jun 21 10:10:55 b2 kernel: Module 2: Not installed
Jun 21 10:10:55 b2 kernel: Module 3: Not installed
Jun 21 10:10:55 b2 kernel: Found a Wildcard TDM: Wildcard TDM400P REV
E/F (4 modules)
Jun 21 10:10:55 b2 kernel: Registered tone zone 0 (United States / North
America)

[EMAIL PROTECTED] src]# cat /proc/interrupts
  CPU0
 0:   17893766  XT-PIC  timer
 1:  2  XT-PIC  keyboard
 2:  0  XT-PIC  cascade
 4:  357411641  XT-PIC  eth0, wanpipe1
 8:  1  XT-PIC  rtc
10:3381408  XT-PIC  Intel ICH2
11:  178236906  XT-PIC  wctdm
14:  50492  XT-PIC  ide0
15:  0  XT-PIC  ide1
NMI:  0
ERR:  0
[EMAIL PROTECTED] src]# cat /proc/interrupts
  CPU0
 0:   17894203  XT-PIC  timer
 1:  2  XT-PIC  keyboard
 2:  0  XT-PIC  cascade
 4:  357419974  XT-PIC  eth0, wanpipe1
 8:  1  XT-PIC  rtc
10:3381408  XT-PIC  Intel ICH2
11:  178241275  XT-PIC  wctdm
14:  50494  XT-PIC  ide0
15:  0  XT-PIC  ide1
NMI:  0
ERR:  0

--
Mike
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[Asterisk-Users] meetme mute status

[bdz]

hi,

is there any way to figure out what the mute status
is of the meetme conference participants?

i personally can no see any difference on the output:
kamikaze*CLI meetme
Conf Num   PartiesMarked Activity  Creation
5000   0002   N/A00:00:40  Static  
* Total number of MeetMe users: 2
kamikaze*CLI 
kamikaze*CLI meetme list 5000
User #: 1  Channel: SIP/fizik-c4eb  
User #: 2  Channel: H323/ip$192.168.42.10:10659/14231  
kamikaze*CLI 
kamikaze*CLI meetme mute 5000 1
kamikaze*CLI meetme list 5000
User #: 1  Channel: SIP/fizik-c4eb  
User #: 2  Channel: H323/ip$192.168.42.10:10659/14231  
kamikaze*CLI 

i also can not see any mute/unmute event on the manager
interface only the join/leave events come.

any idea?
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[Asterisk-Users] Re: [Serusers] ASTERISK+SER+MWI

[harry gaillac]

What's wrong with ARA (asterisk realtime architecture)

from voip-info:
Asterisk, SER and MWI
http://mail.iptel.org/pipermail/serusers/2004-December/013727.html
Actually I wrote a patch for this and it supports
ast_data too. What you do is tell asterisk that all of
your phones IP addresses are your SER machine. Then
when a message gets left Asterisk sends the NOTIFY to
username at seripaddress:serport. SER gets it and
since it knows what's up, it relays it to the phone.
Tada! I gave my patch to Rob (who leads up the
ast_data patch. If you need it let me know. Its for
Asterisk 1.0.2 with the 1.0.2 ast_data patch. 
--- Iqbal [EMAIL PROTECTED] a écrit :

 Realtime with asterisk, and voicemail, #if you get i
 working let me 
 know, I got voicemail from flatfile all working, but
 realtime, just 
 doesnt seem to pull the info out of the database for
 some reason.
 
 MWI, if your phone is registered at SER, you will
 need to pass message 
 back with externotify back to ser, and get ser to
 send the message to 
 the phone, Paul(aka Java) had a working setup, but
 its not 
 straightforward, alot of little scripts to implement
 in cron etc.
 
 Also add to this different ways in which the IP
 phones work, and you may 
 as well not do it :-)
 
 Iqbal
 







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Re: [Asterisk-Users] DID not working? + sendmail problems

[Pascal Mosimann]

 to work when I used fxs_ks signaling on them. However I changed to
 fxs_ls signaling because the ks wasn’t detecting when people were
 hanging up properly. Could changing the signaling be whats causing the

I added the following to my zapata.conf in the [channels] section:
busydetect=yes
busycount=4

It helped to reduce the time to detect that the line was hang up.

HTH
Pascal.

Rick a écrit :
I have a box running [EMAIL PROTECTED] 1.0 and I noticed today when I went to 
change some settings to do with the DID’s that it is no longer detecting 
the different lines.


 

I have a Digium 4 port line card and Im pretty sure that the DID’s used 
to work when I used fxs_ks signaling on them. However I changed to 
fxs_ls signaling because the ks wasn’t detecting when people were 
hanging up properly. Could changing the signaling be whats causing the 
DIDs not to function properly?


 

 


My other problem is getting sendmail to work. I get the following error.

 


   - Transcript of session follows -

... while talking to my smtp server, details removed:


DATA


 550-Verification failed for [EMAIL PROTECTED]

 550-unrouteable mail domain asterisk1.local

 550 Sender verify failed

550 5.1.1 users email address, details removed... User unknown

 503 valid RCPT command must precede DATA

 

I cant find anywhere to change the login/pass info to change it to auth 
with a valid account from the mail server.


 


Any help would be greatly appreciated.




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[Asterisk-Users] Re: [Serusers] ASTERISK+SER+MWI

[Iqbal]

nothing wrong with it, and ast_data and realtime are different unless I 
have misunderstood.


Asterisk realtime and pulling Voicemail from a DB I have had problems 
with, if it works great if someone could let me know, ast_data does I 
believe work, so you could try that out, I just wanted to goto realtime, 
however maybe I'll give ast_data a go.


Iqbal

harry gaillac wrote:


What's wrong with ARA (asterisk realtime architecture)

from voip-info:
Asterisk, SER and MWI
http://mail.iptel.org/pipermail/serusers/2004-December/013727.html
Actually I wrote a patch for this and it supports
ast_data too. What you do is tell asterisk that all of
your phones IP addresses are your SER machine. Then
when a message gets left Asterisk sends the NOTIFY to
username at seripaddress:serport. SER gets it and
since it knows what's up, it relays it to the phone.
Tada! I gave my patch to Rob (who leads up the
ast_data patch. If you need it let me know. Its for
Asterisk 1.0.2 with the 1.0.2 ast_data patch. 
--- Iqbal [EMAIL PROTECTED] a écrit :


 


Realtime with asterisk, and voicemail, #if you get i
working let me 
know, I got voicemail from flatfile all working, but
realtime, just 
doesnt seem to pull the info out of the database for

some reason.

MWI, if your phone is registered at SER, you will
need to pass message 
back with externotify back to ser, and get ser to
send the message to 
the phone, Paul(aka Java) had a working setup, but
its not 
straightforward, alot of little scripts to implement

in cron etc.

Also add to this different ways in which the IP
phones work, and you may 
as well not do it :-)


Iqbal

   









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.

 


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Re: [Asterisk-Users] How can you check that eg TDM04B hardwareinstalled and drivers OK

[John Novack]

Probably means that your perfectly good motherboard can't see the TDM card.
There are many motherboards that this card doesn't seem to work with, 
Digium doesn't seem willing to address the issue or even acknowledge 
that is the case, and usually answers  try another motherboard rather 
than 'fess up that there is a design problem with the PCI interface and 
correct it.
PCI 2.2 is a stated requirement, but there is certainly more to the 
story than that.


In addition, when the board CAN be seen, report rev E/F when  the 
silkscreen reads Rev H, someone mentioned there is now a Rev I ( good 
luck getting an exchange ) and Digium 's answer is  if we can see it 
through remote access then there is no reason to replace it, and if we 
can't, try another MB.


Overall, if it works, lucky you, if not, Too bad.
Hard to support Digium and suggest others purchase such a product.
Best you look for other interfaces to Asterisk.

John Novack




Angus Comber wrote:


If I try dmesg - no mention of a Wildcard TDM400.

Sorry I am fairly new to Linux.  In Windows I suppose I would run some 
hardware program which came with the card to see if I could manually 
set IRQ's etc.  What should I be looking at now?


Please feel free to point me to a good book or whatever you feel is 
appropriate.  Could the card be faulty?


My motherboard is an Intel D865GLC.

I am running [EMAIL PROTECTED] version 1.0

Angus




- Original Message - From: Mike M 
[EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Tuesday, June 21, 2005 3:14 PM
Subject: Re: [Asterisk-Users] How can you check that eg TDM04B 
hardwareinstalled and drivers OK




On Tue, Jun 21, 2005 at 07:09:46AM -0600, Rich Adamson wrote:



 I am struggling to get my TDM04B working.  Just to rule out a 
hardware  problem how can I check

that the hardware works?  How can I then
 check that the drivers are loaded correctly?


1. from the linux command line, type 'dmesg' and look for
 Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules)
if you see that, the TDM card is recognized by the OS.



Here's what I get on a working system:

[EMAIL PROTECTED] src]# modprobe wctdm
[EMAIL PROTECTED] src]# Jun 21 10:10:55 b2 kernel: PCI: Found IRQ 11 for device
01:0a.0
Jun 21 10:10:55 b2 kernel: Freshmaker version: 71
Jun 21 10:10:55 b2 kernel: Freshmaker passed register test
Jun 21 10:10:55 b2 kernel: Module 0: Installed -- AUTO FXS/DPO
Jun 21 10:10:55 b2 kernel: Module 1: Not installed
Jun 21 10:10:55 b2 kernel: Module 2: Not installed
Jun 21 10:10:55 b2 kernel: Module 3: Not installed
Jun 21 10:10:55 b2 kernel: Found a Wildcard TDM: Wildcard TDM400P REV
E/F (4 modules)
Jun 21 10:10:55 b2 kernel: Registered tone zone 0 (United States / North
America)

[EMAIL PROTECTED] src]# cat /proc/interrupts
  CPU0
 0:   17893766  XT-PIC  timer
 1:  2  XT-PIC  keyboard
 2:  0  XT-PIC  cascade
 4:  357411641  XT-PIC  eth0, wanpipe1
 8:  1  XT-PIC  rtc
10:3381408  XT-PIC  Intel ICH2
11:  178236906  XT-PIC  wctdm
14:  50492  XT-PIC  ide0
15:  0  XT-PIC  ide1
NMI:  0
ERR:  0
[EMAIL PROTECTED] src]# cat /proc/interrupts
  CPU0
 0:   17894203  XT-PIC  timer
 1:  2  XT-PIC  keyboard
 2:  0  XT-PIC  cascade
 4:  357419974  XT-PIC  eth0, wanpipe1
 8:  1  XT-PIC  rtc
10:3381408  XT-PIC  Intel ICH2
11:  178241275  XT-PIC  wctdm
14:  50494  XT-PIC  ide0
15:  0  XT-PIC  ide1
NMI:  0
ERR:  0

--
Mike




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Re: [Asterisk-Users] 403 forbidden on SIP register

[Rich Adamson]

Someone else just reported what appears to be the same problem and he
narrowed it down to upper-lower case problem with sip digest. Don't
know if a fix has been applied to cvs-head for that as yet. Might
try an update.


  md5 instead of plaintext? 
 
 Doesn't asterisk take care of this automatically with SIP?  I have
 other providers that use md5 and they all respond with a 401 challenge
 and then asterisk generates the md5 and uses the realm given it in the
 401.
 
  Also, I think I just seen a change in the last day or two that had
  something to do with 403's, and if I recall correctly, it also addressed
  upper/lower case something or another.
  
  Can you use ethereal or sip debug to determine the exact item that was
  sent that might be causing the 403? Either one should at least provide
  a hint.
 
 Here is a sip debug.  All I get back is an immediate 403 forbidden. 
 This is also what I get back on other providers if I had the wrong
 password.
 
 This is also cvs HEAD from yesterday, although jumping back to a
 version from  a month ago didn't make any difference.
 
 Chris
 
 
 --- (10 headers 0 lines)---
 Jun 21 15:42:52 NOTICE[34281]: chan_sip.c:4671 sip_reregister:--
 Re-registration for  [EMAIL PROTECTED]
 REGISTER 11 headers, 0 lines
 REGISTER attempt 1 to [EMAIL PROTECTED]
 Reliably Transmitting (no NAT) to 208.139.204.228:5060:
 REGISTER sip:voip-co1.teliax.com SIP/2.0
 Via: SIP/2.0/UDP 69.25.136.31:5060;branch=z9hG4bK027fe0e3
 From: sip:[EMAIL PROTECTED];tag=as737624ba
 To: sip:[EMAIL PROTECTED]
 Call-ID: [EMAIL PROTECTED]
 CSeq: 102 REGISTER
 User-Agent: Asterisk PBX
 Expires: 120
 Contact: sip:[EMAIL PROTECTED]
 Event: registration
 Content-Length: 0
 
 
 -- SIP read from 208.139.204.228:5060: 
 SIP/2.0 403 Forbidden
 Via: SIP/2.0/UDP 69.25.136.31:5060;branch=z9hG4bK027fe0e3
 From: sip:[EMAIL PROTECTED];tag=as737624ba
 To: sip:[EMAIL PROTECTED];tag=as3201eb44
 Call-ID: [EMAIL PROTECTED]
 CSeq: 102 REGISTER
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
 Contact: sip:[EMAIL PROTECTED]
 Content-Length: 0
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---End of Original Message-


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[Asterisk-Users] Re: meetme mute status

[Tony Mountifield]

In article [EMAIL PROTECTED], bdz [EMAIL PROTECTED] wrote:
 hi,
 
 is there any way to figure out what the mute status
 is of the meetme conference participants?
 
 i personally can no see any difference on the output:
 kamikaze*CLI meetme
 Conf Num   PartiesMarked Activity  Creation
 5000   0002 N/A00:00:40  Static  
 * Total number of MeetMe users: 2
 kamikaze*CLI 
 kamikaze*CLI meetme list 5000
 User #: 1  Channel: SIP/fizik-c4eb  
 User #: 2  Channel: H323/ip$192.168.42.10:10659/14231  
 kamikaze*CLI 
 kamikaze*CLI meetme mute 5000 1
 kamikaze*CLI meetme list 5000
 User #: 1  Channel: SIP/fizik-c4eb  
 User #: 2  Channel: H323/ip$192.168.42.10:10659/14231  
 kamikaze*CLI 
 
 i also can not see any mute/unmute event on the manager
 interface only the join/leave events come.

Neither of these features currently exists in MeetMe, although they are
both on my ToDo list unless someone else beats me to it.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] TDM400P and Dell Poweredge 1750

[Adam Robins]

I installed a new Digium TDM400P in a Dell 1750 server.  The system
would not recognize the card.  I took the FXS modules off of it and put
them on another TDM400P card I already had.  Old card worked fine with
new modules.  Old card is Rev. H and new card is Rev. I.  Anyone else
having any issues with TDM400P rev. I?

Adam

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Re: [Asterisk-Users] How can you check that eg TDM04B hardwareinstalled and drivers OK

[Angus Comber]

Hello

Here is what I find.

Any help would be greatly appreciated.

Angus


- Original Message - 
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Tuesday, June 21, 2005 2:09 PM
Subject: Re: [Asterisk-Users] How can you check that eg TDM04B 
hardwareinstalled and drivers OK





I am struggling to get my TDM04B working.  Just to rule out a hardware 
problem how can I check

that the hardware works?  How can I then

check that the drivers are loaded correctly?



You didn't mention which linux distro you're using, so translate the


** [EMAIL PROTECTED] version 1.0 on Centos OS.

following into whatever your system expects. Try the following items:

1. from the linux command line, type 'dmesg' and look for
Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules)
if you see that, the TDM card is recognized by the OS.

** Did not find TDM!


2. from the linux command line, type 'cat /proc/interrupts and look for
an entry with 'wctdm' in the list. If you don't see wctdm listed,
the module is not loaded as yet.

** no wctdm in list


3. in /etc/zaptel.conf, ensure you have an entry like:
fxsks=1-4

** OK - but think a hardware issue needs to be resolved first


4. if you're using a linux v2.6 kernel, read
/usr/src/zaptel/README.udev

5. with asterisk stopped and from the linux command line, try
sysconfig zaptel start

** Command not found



6. What do you see if you run 'zttool' from the linux command line?


**
** Zaptel Tool loads and I see this:

Zapata Telephony Interfaces
Alarms  Span

nothing else

If click on Select go to another screen:

Current Alarms: No Alarms
Sync Source: Internally clocked
IRQ Misses: 0
Bipolar Viol: 0
Tx/Rx Levels: 0/  0
Total/Conf/Act: 0/  0/  0







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Re: [Asterisk-Users] Dialplan Q: Dialing with Capi

[Patrik Schindler]

On Wed, 22 Jun 2005, jurczak wrote:

 You could try the new chan_capi-cm-0.5.1 in which you dont need to have any 
 msn defined in capi.conf

Thank you very much, this solved my problem!

Do you know a solution for non-capi but i4l devices? It's not an error but 
a warning only, so it's not a real problem.

:wq! PoC

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[Asterisk-Users] gsm gateway

[David Hajek]

Hi,

can you recommend cheap GSM Gateway which works with Asterisk? VoiceBlue 
solution is quite expensive.


Thanks,

--
-
David

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[Asterisk-Users] Asterisk ended with exit status 139

[Ronald Wiplinger]

What could be th reason for that?


/usr/sbin/safe_asterisk: line 41:  4590 Segmentation fault  (core 
dumped) asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}

Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.


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Re: [Asterisk-Users] Re: New JAVA application server for Asterisk - OrderlyCalls

[Tom Rymes]

On Jun 21, 2005, at 5:05 PM, Adam Megacz wrote:



Matt King [EMAIL PROTECTED] writes:


I am familiar with the OSI definitiion.  I've read it again, but I
can't work out exactly how asking for permission contravenes this
definition.




2)  OrderlyCalls MAY NOT be used to provide or augment call  
queuing without

the prior written permission of Orderly Software.



6. No Discrimination Against Fields of Endeavor

 The license must not restrict anyone from making use of the program
 in a specific field of endeavor.  For example, it may not restrict
 the program from being used in a business...

  http://www.opensource.org/docs/definition.php


Maybe I'm wrong here, but his restriction does not seem to be a  
restriction on a field of endeavor. His restriction is a restriction  
on using the software to implement a certain function. If he had said  
You may  not use this software for commercial purposes or  
Individuals engaged in agricultural activities may not use this  
software, or This software cannot be used by commercial software  
vendors, THEN it would be a restriction based on a specific field of  
endeavor.


Tom
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[Asterisk-Users] FXS interfaces

[Alessandro]

Hi all,

 Does somebody know why no load modules to FXS? I used zaptel-1.0.7 version driver. 

[EMAIL PROTECTED] zaptel-1.0.7]# modprobe wctdm
ZT_CHANCONFIG failed on channel 3: Invalid argument (22)
Did you forget that FXS interfaces are configured with FXO signalling
and that FXO interfaces use FXS signalling?
/lib/modules/2.4.21-27.EL/misc/wcfxs.o: post-install wcfxs failed
/lib/modules/2.4.21-27.EL/misc/wcfxs.o: insmod wctdm failed
You have new mail in /var/spool/mail/root


/var/log/messages:
###
Jun 21 19:06:15 darthvaden kernel: Zapata Telephony Interface Registered on major 196
Jun 21 19:06:16 darthvaden kernel: PCI: Found IRQ 9 for device 02:09.0
Jun 21 19:06:16 darthvaden kernel: PCI: Sharing IRQ 9 with 00:1f.5
Jun 21 19:06:16 darthvaden kernel: Freshmaker version: 71
Jun 21 19:06:16 darthvaden kernel: Freshmaker passed register test
Jun 21 19:06:16 darthvaden kernel: ProSLIC 3210 version 2 is too old
Jun 21 19:06:16 darthvaden kernel: Module 0: Not installed
Jun 21 19:06:16 darthvaden kernel: ProSLIC 3210 version 2 is too old
Jun 21 19:06:16 darthvaden kernel: Module 1: Not installed
Jun 21 19:06:16 darthvaden kernel: Module 2: Installed -- AUTO FXO (FCC mode)
Jun 21 19:06:16 darthvaden kernel: Module 3: Installed -- AUTO FXO (FCC mode)
Jun 21 19:06:16 darthvaden kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)


It's TDM22B device.
- 
http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TDM400Ptab=details


See below zaptel.conf: 

# 
# Zaptel Configuration File 
# 
# This file is parsed by the Zaptel Configurator, ztcfg 
# 
# 
# First come the span definitions, in the format 
# span=span num,timing,line build out 
(LBO),framing,coding[,yellow] 
# 
# The timing parameter determines the selection of primary, secondary, and 
# so on sync sources. If this span should be considered a primary sync 
# source, then give it a value of 1. For a secondary, use 2, and so on. 
# To not use this as a sync source, just use 0 
# 
# The line build-out (or LBO) is an integer, from the following table: 
# 0: 0 db (CSU) / 0-133 feet (DSX-1) 
# 1: 133-266 feet (DSX-1) 
# 2: 266-399 feet (DSX-1) 
# 3: 399-533 feet (DSX-1) 
# 4: 533-655 feet (DSX-1) 
# 5: -7.5db (CSU) 
# 6: -15db (CSU) 
# 7: -22.5db (CSU) 
# 
# The framing is one of d4 or esf for T1 or cas or ccs for E1 
# 
# Note: d4 could be referred to as sf or superframe 
# 
# The coding is one of ami or b8zs for T1 or ami or hdb3 for E1 
# 
# E1's may have the additional keyword crc4 to enable CRC4 checking 
# 
# If the keyword yellow follows, yellow alarm is transmitted when no 
# channels are open. 
# 
#span=1,0,0,esf,b8zs 
#span=2,1,0,esf,b8zs 
#span=3,0,0,ccs,hdb3,crc4 
# 
# Next come the dynamic span definitions, in the form: 
# dynamic=driver,address,numchans,timing 
# 
# Where driver is the name of the driver (e.g. eth), address is the 
# driver specific address (like a MAC for eth), numchans is the number 
# of channels, and timing is a timing priority, like for a normal span. 
# use 0 to not use this as a timing source, or prioritize them as 
# primary, secondard, etc. Note that you MUST have a REAL zaptel device 
# if you are not using external timing. 
# 
# dynamic=eth,eth0/00:02:b3:35:43:9c,24,0 
# 
# Next come the definitions for using the channels. The format is: 
# device=channel list 
# 
# Valid devices are: 
# 
# em : Channel(s) are signalled using EM signalling (specific 
# implementation, such as Immediate, Wink, or Feature Group D 
# are handled by the userspace library). 
# fxsls : Channel(s) are signalled using FXS Loopstart protocol. 
# fxsgs : Channel(s) are signalled using FXS Groundstart protocol. 
# fxsks : Channel(s) are signalled using FXS Koolstart protocol. 
# fxols : Channel(s) are signalled using FXO Loopstart protocol. 
# fxogs : Channel(s) are signalled using FXO Groundstart protocol. 
# fxoks : Channel(s) are signalled using FXO Koolstart protocol. 
# sf : Channel(s) are signalled using in-band single freq tone. 
#Syntax as follows: 
# channel# = sf:rxfreq,rxbw,rxflag,txfreq,txlevel,txflag 
#rxfreq is rx tone freq in hz, rxbw is rx notch (and decode) 
#bandwith in hz (typically 10.0), rxflag is either 'normal' or 
#'inverted', txfreq is tx tone freq in hz, txlevel is tx tone 
#level in dbm, txflag is either 'normal' or 'inverted'. Set 
#rxfreq or txfreq to 0.0 if that tone is not desired. 
# unused : No signalling is performed, each channel in the list remains 
idle 
# clear : Channel(s) are bundled into a single span. No conversion or 
# signalling is performed, and raw data is available on the 
master. 
# indclear: Like clear except all channels are treated individually and 
# are not bundled. bchan is an alias for this. 
# rawhdlc : The zaptel driver performs HDLC encoding and decoding on the 
# bundle, and the resulting data is communicated via the master 
# device. 
# fcshdlc : The zapdel driver performs HDLC encoding and 

RE: [Asterisk-Users] call divert to TRUNK , if one number is unregistered?

[Erdem HAKİ]

Yes it works :) but i need to add
voicemail option, how can i do this?



I want a configuration like this



exten = XXX,1, Dial(sip/,20,r) 
exten = XXX,2,Dial(zap/)

exten = XXX,3,Voicemail(u)

exten = XXX,103,Voicemail(b) -what should order
number be?

exten = XXX,104,Hangup













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohamed A. Gombolaty
Sent: Wednesday, June 22, 2005
2:27 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call
divert to TRUNK ,if one number is unregistered?





Hi Erdem, 

Can you
try to put another dial command that points to the trunk afetr the dial command
to the SIP? 

fro
example: 

exten
= XXX,1, dial(sip/,20,r) 
exten = XXX,2,dial(zap/) - note
here that I am not sure if the order number should be 2 or 102 but if this
didn't work try the other one. 

Thx 
MAG 
 
 
 
 

Erdem
HAK] wrote: 



 





I have a question.



I have two
numbers on Asterisk like 902121234567 and 902123645789 and i want to divert
first number's call to Trunk if second number is unregistered. Is it possible?
]f yes, how? 
 

 

Flow
Diagram: 

*Two
numbers are registered on Asterisk 

902121234567
registered to Asterisk 

902123645789
registered to Asterisk 

*One
number is registered, other one is not registered 

902121234567
registered to Asterisk 

902123645789-x
not registered to Asterisk 

*So
first number want to make a call second one (desired situation)


902121234567
Asterisk` Trunk 

Thanks
for your interest. 

Erdem
HAKI - [EMAIL PROTECTED]







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--ThxMAG

 






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RE: [Asterisk-Users] Grandstream 100 pricing question

[barton-lists]

[EMAIL PROTECTED] wrote:
 Thanks for the info... I am considering getting these to
 experiment with,
 so I can do some testing *before* I actually get in to the
 real thing. The
 cheaper the test period, the better, so 2 of these (which later can
be
 reused in less used area's) look pretty interesting...

I have several Grandstream BT101's that are slightly used and I'm
looking to sell, if you're interested.

Barton


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[Asterisk-Users] Telrad + EM T1 Trunk

[Karthik Natarajan]

Hi,

Please refer to my post here regarding a similar issue as you are trying to
achieve,
I am not sure how far you have gotten with your attempt, but I would
appreciate if you can share your experience with me and help me get mine
working.

http://lists.digium.com/pipermail/asterisk-users/2005-June/113085.html

thanks in advance,

Karthik Natarajan
InfoPro Corporation
732-283-2589 x 241
[EMAIL PROTECTED]

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RE: [Asterisk-Users] DID not working? + sendmail problems

[Syed Akbar]

Digium's analog cards do not support DID lines.

Syed Akbar

Alico Systems Inc
www.alicosystems.com
Tel: 562-436-1510 
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pascal
Mosimann
Sent: Wednesday, June 22, 2005 8:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DID not working? + sendmail problems

  to work when I used fxs_ks signaling on them. However I changed to
  fxs_ls signaling because the ks wasn’t detecting when people were
  hanging up properly. Could changing the signaling be whats causing the

I added the following to my zapata.conf in the [channels] section:
busydetect=yes
busycount=4

It helped to reduce the time to detect that the line was hang up.

HTH
Pascal.

Rick a écrit :
 I have a box running [EMAIL PROTECTED] 1.0 and I noticed today when I went to 
 change some settings to do with the DID’s that it is no longer detecting 
 the different lines.
 
  
 
 I have a Digium 4 port line card and Im pretty sure that the DID’s used 
 to work when I used fxs_ks signaling on them. However I changed to 
 fxs_ls signaling because the ks wasn’t detecting when people were 
 hanging up properly. Could changing the signaling be whats causing the 
 DIDs not to function properly?
 
  
 
  
 
 My other problem is getting sendmail to work. I get the following error..
 
  
 
- Transcript of session follows -
 
 ... while talking to my smtp server, details removed:
 
 DATA
 
  550-Verification failed for [EMAIL PROTECTED]
 
  550-unrouteable mail domain asterisk1.local
 
  550 Sender verify failed
 
 550 5.1.1 users email address, details removed... User unknown
 
  503 valid RCPT command must precede DATA
 
  
 
 I cant find anywhere to change the login/pass info to change it to auth 
 with a valid account from the mail server.
 
  
 
 Any help would be greatly appreciated.
 
 
 
 
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[Asterisk-Users] Problem with Connecting PBX to Asterisk

[Karthik Natarajan]

As suggested by you, I have tried the timing parameter with 0 an 1 both but
to no avail.

Signalling I have tried are both em_w and featd (suggested by digium tech
support).

Zap show channels shows all 24 channels to be ok with no alarms.

Zttool also confirms that.

The problem is that the pbx (telrad) does not even seem to sense the T1 that
is plugged in. No yellow and no green. Only the RED LED glows. But when I
try a T1 loopback plugged into that the card turns green on telrad
suggesting that the card is fine. (T1 loopback build using a small connector
with 1 to 4 and 2 to 5 connected).

Also regarding timing, I do have 2 X100Ps installed so I wonder if this
needs to be primary source of timing?

As I mentioned earlier I am getting a GREEN on asterisk side (TE405P card)
while a red on the pbx side (TELRAD)


Any thoughts?

Clock source will be important here.  For phase one, you should probably
set asterisk to time from the PBX since the PBX is likely timing from the
T1 circuit.  At phase two, you will likely want to reverse this having
your PBX clock from the Asterisk system and having Asterisk clock from the
telco T1.  This of course assumes that there is only 1 Telco T1 involved. 
Timing is the first most important consideration.  After that, you want to
verify that both ends are using the same signalling type (it appears that
you are using CAS signalling). Check that your PBX is using CAS and find
out exactly what type.  Zapata will need to be configured to use the same
type of signalling.  Check both sides looking for any red alarms etc that
might indicate a cable problem.  From the CLI, 'zap show channels' or
using zttool
from the command line should help you determine the status of the link. 
Red alarms usually indicate that this end sees an out of frame condition
while a yellow means that the opposite side sees and out of frame. A
yellow on one side is a red on the other.



Karthik Natarajan
InfoPro Corporation
732-283-2589 x 241
[EMAIL PROTECTED]

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[Asterisk-Users] FOP related questions

[Daniel ANDRE]

Hello,

I have downloaded and installed Flash Operator Panel. version 0.21. It 
works pretty well and I have some questions about it.


1. The text label of the buttons are partially hidden by their icons. Is 
there a way to adjust right margin for the buttons?
2. I would like to have the fop brought in the front of screen whenever 
and extension rings. Sort of crm feature but with fop and not another 
url. Is there a way to do that?
3. This question is notre directly related to fop but you may have the 
answer. I would like to have fop panel in tis own windows (no toolbar, 
menu, title, ...) either with FireFox and Internet Explorer. Any Idea?


Best regards,

Daniel ANDRÉ

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Re: [Asterisk-Users] Polycom and CallerID

[Johann]

We have the following versions:

App Version: 1.4.1.0040 (SIP)
Bootrom: 2.6.1.0003

I also noticed that the polycom IP600 phones are Rev 3.

--johann

Chris Coulthurst wrote:


Which software pack to you have for the IP600? Sip.ld, bootrom, etc...

Chris Coulthurst
[EMAIL PROTECTED]

 


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RE: [Asterisk-Users] logged in agent make an outbound call?

[Damon Estep]

Yes, I know.

In this case the agent is logging in from a remote phone (pots line) and
staying logged in. If they used agentcallbacklogin they could make
outbound calls, but the long distance bill would hit their line, not the
* box...


 
 You could use Agentcallbacklogin instead - the queue will call them
when
 a call comes in, but they are free to make outbound calls in the
meantime.
 
 Julian.
 
 Damon Estep wrote:
  Is there a way for a logged in agent (hearing music on hold) to
initiate
  an outbound call without logging out of the queue?
 
 
 
  We want sales agents to be able to make outcalls when there is no
  callers in queue, but still be logged in to get new inbound calls if
  they come in.
 
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Re: [Asterisk-Users] meetme mute status

[Moises Silva]

In my little experience with Meetme, i have not found how to know if
certain user is muted or not, so im keeping track of the commands i
execute from the web interface, so i know if its muted or not. Its not
so hard to add a manager event, check manager.c to know how to add
events.

On 6/22/05, bdz [EMAIL PROTECTED] wrote:
 hi,
 
 is there any way to figure out what the mute status
 is of the meetme conference participants?
 
 i personally can no see any difference on the output:
 kamikaze*CLI meetme
 Conf Num   PartiesMarked Activity  Creation
 5000   0002   N/A00:00:40  Static
 * Total number of MeetMe users: 2
 kamikaze*CLI
 kamikaze*CLI meetme list 5000
 User #: 1  Channel: SIP/fizik-c4eb
 User #: 2  Channel: H323/ip$192.168.42.10:10659/14231
 kamikaze*CLI
 kamikaze*CLI meetme mute 5000 1
 kamikaze*CLI meetme list 5000
 User #: 1  Channel: SIP/fizik-c4eb
 User #: 2  Channel: H323/ip$192.168.42.10:10659/14231
 kamikaze*CLI
 
 i also can not see any mute/unmute event on the manager
 interface only the join/leave events come.
 
 any idea?
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-- 
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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Re: [Asterisk-Users] Spanish doc

[Moises Silva]

404 - Not Found

:(

On 6/22/05, Leonardo F. Bauchwitz [EMAIL PROTECTED] wrote:
 Hi:
 We have finished the translation of the FAQ of Digium to spanish.
 They are already (in Spanish) available for download (in
 http://ourproject.org/projects/asterix/):
 
 * FAQ  Frequently Asked Questions
 * Features
 * Hardware compatibility list
 * Fast Installation Zaptel
 All the documentation is available for download
 
 Soon the following documents will be finished:
 Volume one and Asterisk Gateway Interface (AGI)
 
 Bye
 
 Leonardo Federico Bauchwitz
 Coordinator of Asterisk documentation in Spanish
 https://ourproject.org/projects/asterix/
 [EMAIL PROTECTED]
 
 
 
 
 
 
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Re: [Asterisk-Users] Problem with Connecting PBX to Asterisk

[qrss]

What about framing?  ESF/B8ZS vs D4/AMI?

-Original Message-
From: Karthik Natarajan
Sent: Wed, June 22, 2005 10:01 am

As suggested by you, I have tried the timing parameter with 0 an 1 both
 but
to no avail.

Signalling I have tried are both em_w and featd (suggested by digium tech
support).

Zap show channels shows all 24 channels to be ok with no alarms.

Zttool also confirms that.

The problem is that the pbx (telrad) does not even seem to sense the T1
 that
is plugged in. No yellow and no green. Only the RED LED glows. But when I
try a T1 loopback plugged into that the card turns green on telrad
suggesting that the card is fine. (T1 loopback build using a small
 connector
with 1 to 4 and 2 to 5 connected).

Also regarding timing, I do have 2 X100Ps installed so I wonder if this
needs to be primary source of timing?

As I mentioned earlier I am getting a GREEN on asterisk side (TE405P card)
while a red on the pbx side (TELRAD)


Any thoughts?

Clock source will be important here.  For phase one, you should probably
set asterisk to time from the PBX since the PBX is likely timing from the
T1 circuit.  At phase two, you will likely want to reverse this having
your PBX clock from the Asterisk system and having Asterisk clock from
 the
telco T1.  This of course assumes that there is only 1 Telco T1 involved.
Timing is the first most important consideration.  After that, you want
 to
verify that both ends are using the same signalling type (it appears that
you are using CAS signalling). Check that your PBX is using CAS and find
out exactly what type.  Zapata will need to be configured to use the same
type of signalling.  Check both sides looking for any red alarms etc that
might indicate a cable problem.  From the CLI, 'zap show channels' or
using zttool
from the command line should help you determine the status of the link.
Red alarms usually indicate that this end sees an out of frame condition
while a yellow means that the opposite side sees and out of frame. A
yellow on one side is a red on the other.



Karthik Natarajan
InfoPro Corporation
732-283-2589 x 241
[EMAIL PROTECTED]

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RE: [Asterisk-Users] voip-info.org unreliable lately?

[Damon Estep]

 
 Damon Estep wrote:
  I assume the bandwidth is being donated or something, but surely
someone
  would be willing to donate reliable bandwidth as the knowledge
hosted on
  the site (which is also donated!) is worth way more than the
bandwidth.
 
 Sure it's the bandwidth? If the wiki is loaded, I see Server load on
 the bottom of the page, the numbers sometimes go as high as 80-100..?
 
 Not sure if it's a Linux (guess so? :p) but if that represents the
 system load.. 80 is a 'bit' high indeed.
 
 Cheers
 

80-100 might be a lot for the current environment, but given the number
of * users it is very small. Point is the server and bandwidth should be
able to handle a lot more users if we are all going to rely on it as the
(un)official repository for * guides.

I have seen many posts from users willing to pitch in, but still have no
idea where the site is now or what the arrangement is.
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Re: [Asterisk-Users] logged in agent make an outbound call?

[Asterisk]

What type of connection are you using to link their pots with * ?

For the inbound part, * would be calling them to connect the call. For 
their outbound, could you not use the same mechanism that you are 
currently using to login, but dial the outbound number instead (so it is 
* doing the dialling) ?


I would have thought that if they can call * to login, then they can 
call * to make an outbound call .


Julian.

Damon Estep wrote:


Yes, I know.

In this case the agent is logging in from a remote phone (pots line) and
staying logged in. If they used agentcallbacklogin they could make
outbound calls, but the long distance bill would hit their line, not the
* box...


 


You could use Agentcallbacklogin instead - the queue will call them
   


when
 


a call comes in, but they are free to make outbound calls in the
   


meantime.
 


Julian.

Damon Estep wrote:
   


Is there a way for a logged in agent (hearing music on hold) to
 


initiate
 


an outbound call without logging out of the queue?



We want sales agents to be able to make outcalls when there is no
callers in queue, but still be logged in to get new inbound calls if
they come in.

 


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[Asterisk-Users] Fwd:protocol TCP/UDP question

[[EMAIL PROTECTED]]

can you help me to configure lcs2005 with asterisk...
I use SER to resolve the problem that there is for communication protocol...LCS 
uses tcp, Asterik UDP.

Someone, knows how to do the configuration beetwen LCS and SER , SER and 
Asterisk? the function of asterisk is SIP-PSTN Gateway for the LCS PC-phone 
communicationù?

or there is a way to configure asterisk to accept tcp communication from a 
server like LCS?

thanks 




Navighi a 4 MEGA e i primi 3 mesi sono GRATIS. 
Scegli Libero Adsl Flat senza limiti su http://www.libero.it


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Re: [Asterisk-Users] Extension Configuration Best Practice

[Denis Galvão - iSolve]

On 21 de jun de 2005, at 17:20, Andrew Kohlsmith wrote:

How would you have asterisk know which IP to ring if nobody is 
registered

until the phone rings??


You're right Andrew. I didn't thought about the ring...

Honestly -- what's wrong with 
SIP/location1SIP/location2SIP/location3 ?


For me, nothing. I would use some AGIs to solve that, or the serial 
rings like you told.


Denis.

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[Asterisk-Users] Asterisk Manager Api

[gale81]

Hi
One question for you!
Which operation are allows by Asterisk Manager API?
Can I connect to Asterisk server, create a new Channel,add channel on Asterisk
server,specify  a Voip protocol like SIP and generate Sip signaling?
Thanks Ale

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Re: [Asterisk-Users] MeetMe Problems

[Waldo Rubinstein]

Absolutely. Here is the CLI output. I made two attempts. First, I  
dialed inbound into an extension and then tried using meetme room  
0201 from Server B, which didn't work. Then I dialed inbound into the  
same extension and then tried using meetme room 0215 which resides in  
Server A. Note that all inbound calls come into Server A, for it has  
the Digium card.


SERVER A
=

gateway0:~# aa
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk CVS-Nv1-0-7-06/01/05-01:27:25, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
 
=
Connected to Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 currently running  
on gateway0 (pid = 2653)

Verbosity is at least 10
-- Starting simple switch on 'Zap/1-1'
-- Executing NoOp(Zap/1-1, Inbound Call - 3211 - ) in new stack
-- Executing Dial(Zap/1-1, IAX2/corona/3211||r) in new stack
-- Called corona/3211
-- Call accepted by 10.0.10.13 (format ulaw)
-- Format for call is ulaw
-- IAX2/corona/16386 is ringing
-- IAX2/corona/16386 answered Zap/1-1
-- Hungup 'IAX2/corona/16386'
  == Spawn extension (inbound, 3211, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
-- Executing NoOp(Zap/1-1, Inbound Call - 3211 - ) in new stack
-- Executing Dial(Zap/1-1, IAX2/corona/3211||r) in new stack
-- Called corona/3211
-- Call accepted by 10.0.10.13 (format ulaw)
-- Format for call is ulaw
-- IAX2/corona/16388 is ringing
-- IAX2/corona/16388 answered Zap/1-1
  == Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '0215'
-- Started music on hold, class 'default', on IAX2/ 
[EMAIL PROTECTED]/16390

-- Hungup 'Zap/31-1'
-- Hungup 'IAX2/corona/16388'
  == Spawn extension (inbound, 3211, 2) exited non-zero on 'Zap/1-1'
-- Stopped music on hold on IAX2/[EMAIL PROTECTED]/16390
-- Hungup 'Zap/pseudo-1262753463'
  == Spawn extension (meetme, 0215, 1) exited non-zero on 'IAX2/ 
[EMAIL PROTECTED]/16390'

-- Hungup 'Zap/1-1'
-- Hungup 'IAX2/[EMAIL PROTECTED]/16390'

Here are the relevant sections in the .conf files:

meetme.conf:
[rooms]
conf = 0215

extensions.conf:
[meetme]
exten = 0215,1,MeetMe(0215|qM)
exten = 0215,2,Hangup


SERVER B
=

corona root # aa
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.7, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
 
=

Connected to Asterisk 1.0.7 currently running on corona (pid = 5105)
Verbosity is at least 10
-- Remote UNIX connection
-- Call accepted by 10.0.10.9 (format ulaw)
-- Format for call is ulaw
-- Accepting unauthenticated call from 10.0.10.9, requested  
format = 4, actual format = 4

-- Executing Goto(IAX2/[EMAIL PROTECTED]/16395, 211|1) in new stack
-- Goto (client,211,1)
-- Executing Macro(IAX2/[EMAIL PROTECTED]/16395, stdexten|211|SIP/ 
3211) in new stack
-- Executing Dial(IAX2/[EMAIL PROTECTED]/16395, SIP/3211|20|t)  
in new stack

-- Called 3211
-- SIP/3211-3c74 is ringing
-- SIP/3211-3c74 answered IAX2/[EMAIL PROTECTED]/16395
-- Started music on hold, class 'default', on  
IAX2/[EMAIL PROTECTED]/16395

-- Executing MeetMe(SIP/3211-4ed5, 0201|qM) in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '0201'
-- Started music on hold, class 'default', on SIP/3211-4ed5
-- Stopped music on hold on SIP/3211-4ed5
-- Stopped music on hold on IAX2/[EMAIL PROTECTED]/16395
Jun 21 18:57:41 WARNING[8254]: app_meetme.c:667 conf_run: Error  
getting conference

-- Hungup 'Zap/pseudo-510190782'
  == Spawn extension (client_INT, 0201, 1) exited non-zero on  
'IAX2/[EMAIL PROTECTED]/16395'

-- Executing Hangup(IAX2/[EMAIL PROTECTED]/16395, ) in new stack
  == Spawn extension (client_INT, h, 1) exited non-zero on  
'IAX2/[EMAIL PROTECTED]/16395'
  == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/ 
3211-4ed5ZOMBIE' in macro 'stdexten'
  == Spawn extension (client, 211, 1) exited non-zero on 'SIP/ 
3211-4ed5ZOMBIE'

-- Executing Hangup(SIP/3211-4ed5ZOMBIE, ) in new stack
  == Spawn extension (client, h, 1) exited non-zero on 'SIP/ 
3211-4ed5ZOMBIE'

-- Hungup 'IAX2/[EMAIL PROTECTED]/16395'
-- Hungup 'IAX2/gateway0/16384'
-- Accepting unauthenticated call from 10.0.10.9, requested  
format = 4, actual format = 4

-- Executing Goto(IAX2/[EMAIL PROTECTED]/16386, 211|1) in new stack
-- Goto (client,211,1)
-- Executing Macro(IAX2/[EMAIL PROTECTED]/16386, stdexten|211|SIP/ 
3211) in new stack
-- Executing Dial(IAX2/[EMAIL PROTECTED]/16386, SIP/3211|20|t)  
in new stack

-- Called 

[Asterisk-Users] New Asterisk Implementation

[Don Brearley]

Hello,

I've been planning to replace my aging CENTREX switch with a new PBX and am 
seriously
considering Asterisk as my solution.

I work at a college, and we currently support just under 300 regular analog 
lines to
the offices and whatnot.   

I was wondering..  Is asterisk ready for such a job?  Would I be making a 
mistake
to deploy this across the campus at this time?  

It seems that Asterisk is extremely powerful and I absoultly love the way it 
appears
to integrate into almost any existing system, so it would be fantastic if I 
could 
deploy it successfully.

I do understand that I would need to replace all of my existing telephones with 
VoIP-capable
phones, and that I'll need to re-wire most of the campus telephone 
infrastructure (it's still
all cat-3) -- these arent problems.   

I just want to be sure that it's possible to do this, and that im not wasting 
my time.

Thanks for any insight provided!

- Don Brearley 
  HCC Computer Services

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[Asterisk-Users] TDM400P DevKit Problem

[Hamdy Elbarkouky]

I installed the TDM400P and installed it on a system running
on Fedora Core1, please refer to the steps below. I connected an analogue phone
to the FXS port. 



When I pick up the speaker I don't hear the dialtone although
when I press a number key I hear a DTMF generated. What could be the problem?

I appreciate your help



1- First I installed the TDM400P in the PCI slot and
connected the power socket, no problems found



2- I compiled zaptel 1.0.4:

cd /usr/src/zaptel/

make clean

make install



3- Configured etc/zaptel.conf

fxoks=1; this is the location of my fxs port

fxsks=4; this is the location of my fxo port

loadzone=us

defaultzone=us



4-Compiled asterisk 1.0.5

make clean 

make install



5- Created Sample configuration files

make samples



6- Modified zapata.conf, now it looks like this

language=en

context=incoming

switchtype=national

signalling=fxo_ks

channel = 1 ; this is the location of my fxs port

signalling=fxs_ks

channel = 4; this is the location of my fxo port



7- Modified extensions.conf

[general]

[incoming]

exten = s,1,Answer()

exten = s,2,Playback(goodbye)

exten = s,3,Hangup()







8- Loaded the Zaptel Drivers

modprobe zaptel



9- Loaded the xcfxs driver

modprobe wcfxs



THE RESULT IS:

Freshmaker version: 71

Freshmaker passed register test

Module 0: Installed -- AUTO FXS/DPO

Module 1: Not Installed

Module 2: Not Installed

Module 3: Installed -- AUTO FXO (FCC mode)

Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)



10- Configured the signalling using ztcfg

ztcfg -vv



THE RESULT IS:

Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)

Channel 04: FXS Kewlstart (Default) (Slaves: 04)

2 channels configured.



11- Start asterisk

asterisk -cvvv

cli*



Debarko






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[Asterisk-Users] TDM400P DevKit Problem

[Hamdy Elbarkouky]

I installed the TDM400P and installed it on a system running
on Fedora Core1, please refer to the steps below. I connected an analogue phone
to the FXS port. 



When I pick up the speaker I don't hear the dialtone
although when I press a number key I hear a DTMF generated. What could be the
problem?

I appreciate your help



1- First I installed the TDM400P in the PCI slot and
connected the power socket, no problems found



2- I compiled zaptel 1.0.4:

cd /usr/src/zaptel/

make clean

make install



3- Configured etc/zaptel.conf

fxoks=1; this is the location of my fxs port

fxsks=4; this is the location of my fxo port

loadzone=us

defaultzone=us



4-Compiled asterisk 1.0.5

make clean 

make install



5- Created Sample configuration files

make samples



6- Modified zapata.conf, now it looks like this

language=en

context=incoming

switchtype=national

signalling=fxo_ks

channel = 1 ; this is the location of my fxs port

signalling=fxs_ks

channel = 4; this is the location of my fxo port



7- Modified extensions.conf

[general]

[incoming]

exten = s,1,Answer()

exten = s,2,Playback(goodbye)

exten = s,3,Hangup()







8- Loaded the Zaptel Drivers

modprobe zaptel



9- Loaded the xcfxs driver

modprobe wcfxs



THE RESULT IS:

Freshmaker version: 71

Freshmaker passed register test

Module 0: Installed -- AUTO FXS/DPO

Module 1: Not Installed

Module 2: Not Installed

Module 3: Installed -- AUTO FXO (FCC mode)

Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)



10- Configured the signalling using ztcfg

ztcfg -vv



THE RESULT IS:

Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)

Channel 04: FXS Kewlstart (Default) (Slaves: 04)

2 channels configured.



11- Start asterisk

asterisk -cvvv

cli*



Debarko






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Re: [Asterisk-Users] Re: New JAVA application server for Asterisk - OrderlyCalls

[Adam Goryachev]

On Wed, 2005-06-22 at 09:36 -0400, Tom Rymes wrote:
 On Jun 21, 2005, at 5:05 PM, Adam Megacz wrote:
 
 
  Matt King [EMAIL PROTECTED] writes:
 
  I am familiar with the OSI definitiion.  I've read it again, but I
  can't work out exactly how asking for permission contravenes this
  definition.
  2)  OrderlyCalls MAY NOT be used to provide or augment call  
  queuing without
  the prior written permission of Orderly Software.
 Maybe I'm wrong here, but his restriction does not seem to be a  
 restriction on a field of endeavor. His restriction is a restriction  
 on using the software to implement a certain function. If he had said  
 You may  not use this software for commercial purposes or  
 Individuals engaged in agricultural activities may not use this  
 software, or This software cannot be used by commercial software  
 vendors, THEN it would be a restriction based on a specific field of  
 endeavor.

So what is different between these two:
Individuals engaged in agricultural activities may not use this
software

AND

Individuals or companies engaged in tele-marketing/cold calling
activities may not use this software

IMHO, that kind of exclusion does not allow this application to be
called 'free' since it restricts your freedom to use it however you
want... 

While I might agree with the philosophy, I don't agree with the
restriction being placed.

Also, I really don't agree with the other restriction saying that you
can't use this software in order to derive some other function (eg, the
equivalent of their other queue product). That definitely reeks of
non-free

Again, that might be their specific business model, but I don't think
the 'free' software community will be bothered with their applications
if they are so encumbered. Either they will be re-written (re-invented
if you like) or else they really aren't important to anyone anyway

Just my 0.02c worth

PS, why would you need to host it on sourceforge anyway, why not just
stick it on your own website ??

Regards,
Adam

Regards,
Adam


-- 
 -- 
Adam Goryachev
Website Managers
Ph:  +61 2 9345 4395[EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au

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[Asterisk-Users] (no subject)

[[EMAIL PROTECTED]]

can you help me to configure lcs2005 with asterisk...
I use SER to resolve the problem that there is for communication protocol...LCS 
uses tcp, Asterik UDP.

Someone, knows how to do the configuration beetwen LCS and SER , SER and 
Asterisk? the function of asterisk is SIP-PSTN Gateway for the LCS PC-phone 
communication

thanks






Navighi a 4 MEGA e i primi 3 mesi sono GRATIS. 
Scegli Libero Adsl Flat senza limiti su http://www.libero.it


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[Asterisk-Users] Asterisk Manager Api

[gale81]

Hi
One question for you!
Which operation are allows by Asterisk Manager API?
Can I connect to Asterisk server, create a new Channel,add channel on Asterisk
server,specify  a Voip protocol like SIP and generate Sip signaling?
Thanks Ale

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RE: [Asterisk-Users] Grandstream 100 pricing question

[Francesco Peeters]

On Wed, June 22, 2005 17:17, Francesco Peeters said:
 On Wed, June 22, 2005 15:48, [EMAIL PROTECTED] said:
 [EMAIL PROTECTED] wrote:
 Thanks for the info... I am considering getting these to
 experiment with,
 so I can do some testing *before* I actually get in to the
 real thing. The
 cheaper the test period, the better, so 2 of these (which later can
 be
 reused in less used area's) look pretty interesting...

 I have several Grandstream BT101's that are slightly used and I'm
 looking to sell, if you're interested.

 Barton



 Where are you located? What price? Shipping?
 (Always interested in deals!)  ;-)

 --
 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.


Ehrm... Perhaps you'd better respond off-list though!  ;-)

--FP
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Re: [Asterisk-Users] Asterisk to NEC NEAX

[Allen Niven]

although it has been announced, to my knowledge,
the NEC pure SIP fone has not yet been released as of this date

Ramkumar wrote:


Hi,

 

How can I make calls from Asterisk client to NEC NEAX 2400 traditional 
phone ?


 

Is it possible to have a connection between Asterisk and NEC NEAX 
2400, since NEC-NEAX2400 is an IP-PBX and supports SIP.


 


Please help me to find a solution ;;;

 


Thanks  Regards

*Ram Kumar*

*Customer Support Engineer*

*Barcode Gulf LLC*

*Dubai** , UAE*

*Mobile** : + 971 50 5594178*

*Email  : [EMAIL PROTECTED]

 




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--
Allen Niven
GlobalFone
350 Fifth Avenue #6206
Empire State Building
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+1-212-678-4381 office
http://www.GlobalFone.biz


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RE: [Asterisk-Users] Grandstream 100 pricing question

[Francesco Peeters]

On Wed, June 22, 2005 15:48, [EMAIL PROTECTED] said:
 [EMAIL PROTECTED] wrote:
 Thanks for the info... I am considering getting these to
 experiment with,
 so I can do some testing *before* I actually get in to the
 real thing. The
 cheaper the test period, the better, so 2 of these (which later can
 be
 reused in less used area's) look pretty interesting...

 I have several Grandstream BT101's that are slightly used and I'm
 looking to sell, if you're interested.

 Barton



Where are you located? What price? Shipping?
(Always interested in deals!)  ;-)

-- 
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] FXS interfaces

[Mike M]

On Wed, Jun 22, 2005 at 10:41:18AM -0300, Alessandro wrote:
 
 Does somebody know why no load modules to FXS? I used zaptel-1.0.7
 version driver. 
 
 [EMAIL PROTECTED] zaptel-1.0.7]# modprobe wctdm

Here's a clue:

 ZT_CHANCONFIG failed on channel 3: Invalid argument (22)
 Did you forget that FXS interfaces are configured with FXO signalling
 and that FXO interfaces use FXS signalling?
 /lib/modules/2.4.21-27.EL/misc/wcfxs.o: post-install wcfxs failed
 /lib/modules/2.4.21-27.EL/misc/wcfxs.o: insmod wctdm failed
 You have new mail in /var/spool/mail/root

snip
 
 It's TDM22B device.

It's got 4 modules.  What color are the modules in positions 1, 2, 3, 4
on the TDM400P card?  Don't be confused by the 0-3 numbering, just add
1.

 See below zaptel.conf: 

Just the relevant sections next time :).  (The lines not starting with
'#'.)

Think opposite.  Green modules are fxs and should be handled with the
fxo signaling. Red modules are fxo and should be handled with fxs
signaling.

Note the red and green colors here:
http://www.digium.com/index.php?menu=fxsvfxo

 fxsks=1,2 
 fxoks=3,4 

For TDM22B:

fxsks=(position of red   module)(position of red   module)
fxoks=(position of green module)(position of green module)

_Don't_ plug a phone into a red module jack.  
_Don't_ plug a PSTN line into a green module jack.  

-- 
Mike
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SV: [Asterisk-Users] voip-info.org unreliable lately?

[Bjørn Ove Kristiansen]

I think he was referring to load measurements on the server, not the amount
of * users. A guideline of how high load should be can be taken from the
amount of CPUs on the server we're talking about. If the server runs on a
single CPU, then ideal load is up to 1, it it's two CPUs, then up to 2 etc.

Now, these are just guidelines. I am not saying that a server cannot handle
higher load, but if it exceeds these guidelines, then it'll slow the server
down. Therefore, unless voip-info.org is running on monster servers, a load
of 80 is extremely high.

Regards,
Bjorn 

-Opprinnelig melding-
Fra: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne av Damon Estep
Sendt: 22. juni 2005 16:28
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: RE: [Asterisk-Users] voip-info.org unreliable lately?

 
 Damon Estep wrote:
  I assume the bandwidth is being donated or something, but surely
someone
  would be willing to donate reliable bandwidth as the knowledge
hosted on
  the site (which is also donated!) is worth way more than the
bandwidth.
 
 Sure it's the bandwidth? If the wiki is loaded, I see Server load on
 the bottom of the page, the numbers sometimes go as high as 80-100..?
 
 Not sure if it's a Linux (guess so? :p) but if that represents the
 system load.. 80 is a 'bit' high indeed.
 
 Cheers
 

80-100 might be a lot for the current environment, but given the number
of * users it is very small. Point is the server and bandwidth should be
able to handle a lot more users if we are all going to rely on it as the
(un)official repository for * guides.

I have seen many posts from users willing to pitch in, but still have no
idea where the site is now or what the arrangement is.
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RE: [Asterisk-Users] logged in agent make an outbound call?

[Damon Estep]

Calls into the asterisk box, including non-VoIP remote agents, are via a
ISDN/PRI on a Digium T1 card.

It is the same PRI that inbound and outbound calls come in on and go out
through, there are no IP dial tone providers.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Asterisk
 Sent: Wednesday, June 22, 2005 8:41 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] logged in agent make an outbound call?
 
 What type of connection are you using to link their pots with * ?
 
 For the inbound part, * would be calling them to connect the call. For
 their outbound, could you not use the same mechanism that you are
 currently using to login, but dial the outbound number instead (so it
is
 * doing the dialling) ?
 
 I would have thought that if they can call * to login, then they can
 call * to make an outbound call .
 
 Julian.
 
 Damon Estep wrote:
 
 Yes, I know.
 
 In this case the agent is logging in from a remote phone (pots line)
and
 staying logged in. If they used agentcallbacklogin they could make
 outbound calls, but the long distance bill would hit their line, not
the
 * box...
 
 
 
 
 You could use Agentcallbacklogin instead - the queue will call them
 
 
 when
 
 
 a call comes in, but they are free to make outbound calls in the
 
 
 meantime.
 
 
 Julian.
 
 Damon Estep wrote:
 
 
 Is there a way for a logged in agent (hearing music on hold) to
 
 
 initiate
 
 
 an outbound call without logging out of the queue?
 
 
 
 We want sales agents to be able to make outcalls when there is no
 callers in queue, but still be logged in to get new inbound calls
if
 they come in.
 
 
 
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RE: [Asterisk-Users] Asterisk Manager Api

[Maxime Renaud]

asterisk2*CLI show manager commands
  Action   Privilege   Synopsis 
  AbsoluteTimeout  callSet Absolute Timeout 
  ChangeMonitorcallChange monitoring filename of a channel  
  Command  command Execute Command  
  Events   Contol Event Flow
  ExtensionState   callCheck Extension Status   
  Getvar   callGets a Channel Variable  
  Hangup   callHangup Channel   
  IAXpeers List IAX Peers   
  ListCommands List available manager commands  
  Logoff   Logoff Manager   
  MailboxCount callCheck Mailbox Message Count  
  MailboxStatuscallCheck Mailbox
  Monitor  callMonitor a channel
  OriginatecallOriginate Call   
  ParkedCalls  List parked calls
  Ping Ping 
  QueueAdd agent   Add interface to queue.  
  QueueRemove  agent   Remove interface from queue. 
  Queues   Queues   
  QueueStatus  Queue Status 
  Redirect callRedirect 
  SetCDRUserField  callSet the CDR UserField
  Setvar   callSet Channel Variable 
  Status   callStatus   
  StopMonitor  callStop monitoring a channel
  ZapDialOffhook   Dial over Zap channel while offhook  
  ZapDNDoffToggle Zap channel Do Not Disturb status OFF 
  ZapDNDon Toggle Zap channel Do Not Disturb status ON  
  ZapHangupHangup Zap Channel   
  ZapShowChannels  Show status zapata channels  
  ZapTransfer  Transfer Zap Channel 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, June 22, 2005 10:49 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk Manager Api

Hi
One question for you!
Which operation are allows by Asterisk Manager API?
Can I connect to Asterisk server, create a new Channel,add channel on
Asterisk server,specify  a Voip protocol like SIP and generate Sip
signaling?
Thanks Ale

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Re: [Asterisk-Users] Fwd:protocol TCP/UDP question

[Sergio Chersovani]

[EMAIL PROTECTED] ha scritto:

can you help me to configure lcs2005 with asterisk... 
I use SER to resolve the problem that there is for communication protocol...LCS uses tcp, Asterik UDP. 
 


prova http://www.winton.org.uk/zebedee/
It should proxy udp over tcp, but I didn't try it yet

Sergio

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RE: [Asterisk-Users] New Asterisk Implementation

[harry gaillac]

look at ser projects:
asterisk is limited to 250 channels 
You need cat 5e and manage qos if you setup ip phones
Harry
--- Don Brearley [EMAIL PROTECTED] a écrit :

 
 Hello,
 
 I've been planning to replace my aging CENTREX
 switch with a new PBX and am seriously
 considering Asterisk as my solution.
 
 I work at a college, and we currently support just
 under 300 regular analog lines to
 the offices and whatnot.   
 
 I was wondering..  Is asterisk ready for such a job?
  Would I be making a mistake
 to deploy this across the campus at this time?  
 
 It seems that Asterisk is extremely powerful and I
 absoultly love the way it appears
 to integrate into almost any existing system, so it
 would be fantastic if I could 
 deploy it successfully.
 
 I do understand that I would need to replace all of
 my existing telephones with VoIP-capable
 phones, and that I'll need to re-wire most of the
 campus telephone infrastructure (it's still
 all cat-3) -- these arent problems.   
 
 I just want to be sure that it's possible to do
 this, and that im not wasting my time.
 
 Thanks for any insight provided!
 
 - Don Brearley 
   HCC Computer Services
 
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[Asterisk-Users] volume fading in and out

[Asterisk]

I've had several users today inform me that whilst they were on a call, 
the volume kept fading in and out to such an extent that they thought 
the caller had hung up.


I would dismiss this if it were a single person mentioning it, but it 
isn't ..


Has anyone else seen anything like this ? We haven't, and we've been 
running for nearly a year ...


CVS head as of 3 days ago, TE410p ISDN32 EuroISDN

Julian.
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RE: [Asterisk-Users] New Asterisk Implementation

[Alexander Lopez]

 300 phones should not be a problem if you design the system correctly.

If they are all analog sets with no transcoding your should fine.


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Don Brearley
 Sent: Wednesday, June 22, 2005 10:51 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] New Asterisk Implementation
 
 
 Hello,
 
 I've been planning to replace my aging CENTREX switch with a 
 new PBX and am seriously considering Asterisk as my solution.
 
 I work at a college, and we currently support just under 300 
 regular analog lines to
 the offices and whatnot.   
 
 I was wondering..  Is asterisk ready for such a job?  Would I 
 be making a mistake to deploy this across the campus at this time?  
 
 It seems that Asterisk is extremely powerful and I absoultly 
 love the way it appears to integrate into almost any existing 
 system, so it would be fantastic if I could deploy it successfully.
 
 I do understand that I would need to replace all of my 
 existing telephones with VoIP-capable phones, and that I'll 
 need to re-wire most of the campus telephone infrastructure 
 (it's still
 all cat-3) -- these arent problems.   
 
 I just want to be sure that it's possible to do this, and 
 that im not wasting my time.
 
 Thanks for any insight provided!
 
 - Don Brearley
   HCC Computer Services
 
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Re: [Asterisk-Users] New Asterisk Implementation

[Mike M]

On Wed, Jun 22, 2005 at 09:50:42AM -0500, Don Brearley wrote:

 I've been planning to replace my aging CENTREX switch with a new PBX and am 
 seriously
 considering Asterisk as my solution.

You have a CENTREX switch?  I'm confused.

Do you have a PBX?
Do you pay for Centrex services from your phone company?
Do you have a Class 5 swith with Centrex features?

Why not take a cautious approach?

Leave what you have in place.  Install Asterisk and investigate the
various line interface options.  Enlist early adopters on your campus to
participate in the trial.  Connect Asterisk to your existing system with
PRI.  Gradually ramp up the Asterisk system and ramp down the existing
system.

-- 
Mike
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Re: [Asterisk-Users] How can you check that egTDM04B hardwareinstalled and drivers OK

[Angus Comber]

Digium logged in and fixed the problem.  It seems they had to fix the zaptel 
source code - so not really something I could easily have done.  something 
about adding the subvendors ID to the cards source.  So I assume a bug.


I personally feel a little indebted to Digium for sorting the problem and 
obviously making the Asterisk available.  But would like them even more if I 
didn't have to go through these problems ;)


Angus




- Original Message - 
From: John Novack [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Wednesday, June 22, 2005 2:08 PM
Subject: Re: [Asterisk-Users] How can you check that egTDM04B 
hardwareinstalled and drivers OK



Probably means that your perfectly good motherboard can't see the TDM 
card.
There are many motherboards that this card doesn't seem to work with, 
Digium doesn't seem willing to address the issue or even acknowledge that 
is the case, and usually answers  try another motherboard rather than 
'fess up that there is a design problem with the PCI interface and correct 
it.
PCI 2.2 is a stated requirement, but there is certainly more to the story 
than that.


In addition, when the board CAN be seen, report rev E/F when  the 
silkscreen reads Rev H, someone mentioned there is now a Rev I ( good luck 
getting an exchange ) and Digium 's answer is  if we can see it through 
remote access then there is no reason to replace it, and if we can't, try 
another MB.


Overall, if it works, lucky you, if not, Too bad.
Hard to support Digium and suggest others purchase such a product.
Best you look for other interfaces to Asterisk.

John Novack




Angus Comber wrote:


If I try dmesg - no mention of a Wildcard TDM400.

Sorry I am fairly new to Linux.  In Windows I suppose I would run some 
hardware program which came with the card to see if I could manually set 
IRQ's etc.  What should I be looking at now?


Please feel free to point me to a good book or whatever you feel is 
appropriate.  Could the card be faulty?


My motherboard is an Intel D865GLC.

I am running [EMAIL PROTECTED] version 1.0

Angus




- Original Message - From: Mike M 
[EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Tuesday, June 21, 2005 3:14 PM
Subject: Re: [Asterisk-Users] How can you check that eg TDM04B 
hardwareinstalled and drivers OK




On Tue, Jun 21, 2005 at 07:09:46AM -0600, Rich Adamson wrote:



 I am struggling to get my TDM04B working.  Just to rule out a
hardware  problem how can I check
that the hardware works?  How can I then
 check that the drivers are loaded correctly?


1. from the linux command line, type 'dmesg' and look for
 Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules)
if you see that, the TDM card is recognized by the OS.



Here's what I get on a working system:

[EMAIL PROTECTED] src]# modprobe wctdm
[EMAIL PROTECTED] src]# Jun 21 10:10:55 b2 kernel: PCI: Found IRQ 11 for device
01:0a.0
Jun 21 10:10:55 b2 kernel: Freshmaker version: 71
Jun 21 10:10:55 b2 kernel: Freshmaker passed register test
Jun 21 10:10:55 b2 kernel: Module 0: Installed -- AUTO FXS/DPO
Jun 21 10:10:55 b2 kernel: Module 1: Not installed
Jun 21 10:10:55 b2 kernel: Module 2: Not installed
Jun 21 10:10:55 b2 kernel: Module 3: Not installed
Jun 21 10:10:55 b2 kernel: Found a Wildcard TDM: Wildcard TDM400P REV
E/F (4 modules)
Jun 21 10:10:55 b2 kernel: Registered tone zone 0 (United States / North
America)

[EMAIL PROTECTED] src]# cat /proc/interrupts
  CPU0
 0:   17893766  XT-PIC  timer
 1:  2  XT-PIC  keyboard
 2:  0  XT-PIC  cascade
 4:  357411641  XT-PIC  eth0, wanpipe1
 8:  1  XT-PIC  rtc
10:3381408  XT-PIC  Intel ICH2
11:  178236906  XT-PIC  wctdm
14:  50492  XT-PIC  ide0
15:  0  XT-PIC  ide1
NMI:  0
ERR:  0
[EMAIL PROTECTED] src]# cat /proc/interrupts
  CPU0
 0:   17894203  XT-PIC  timer
 1:  2  XT-PIC  keyboard
 2:  0  XT-PIC  cascade
 4:  357419974  XT-PIC  eth0, wanpipe1
 8:  1  XT-PIC  rtc
10:3381408  XT-PIC  Intel ICH2
11:  178241275  XT-PIC  wctdm
14:  50494  XT-PIC  ide0
15:  0  XT-PIC  ide1
NMI:  0
ERR:  0

--
Mike




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[Asterisk-Users] TE110P Card

[MSEYE]

Hi everybody !
I'm trying to setup a TE110P card. But i'm facind this 
error... and * cannot start.

[chan_zap.so]Jun 22 11:50:33 WARNING[16384]: 
loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: 
undefined symbol: pri_dump_infoJun 22 11:50:33 WARNING[16384]: loader.c:429 
load_modules: Loading module chan_zap.so failed!

Does someone already experience this 
case...
Waiting for u'r help

THANX

MSEYE
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RE: [Asterisk-Users] Grandstream 100 pricing question

[barton-lists]

[EMAIL PROTECTED] wrote:
 On Wed, June 22, 2005 15:48, [EMAIL PROTECTED] said:
 [EMAIL PROTECTED] wrote:
 Thanks for the info... I am considering getting these to
experiment
 with, so I can do some testing *before* I actually get in to the
 real thing. The
 cheaper the test period, the better, so 2 of these (which later
can
 be reused in less used area's) look pretty interesting...
 
 I have several Grandstream BT101's that are slightly used and I'm
 looking to sell, if you're interested.
 
 Where are you located? What price? Shipping?
 (Always interested in deals!)  ;-)

I'm in Arkansas, USA.  $50 each + shipping?  I'd prefer to ship via
UPS ground if that's possible.  If that's too much, then make me an
offer.  I just counted, and I have 6 (4 black, 2 white).  They work
fine, we just replaced them recently.

I have some Grandstream HandyTone 286 ATAs for sale too
http://tinyurl.com/8srg5

Taking this off-list might be appropriate.  Email me at
[EMAIL PROTECTED]

Barton



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[Asterisk-Users] TDM400P Channel Group

[Adam Robins]

I installed a TDM400P with 4 FXO modules.  Before moving all of my
office phone lines to it, I decided to move only one for testing.  I
plugged it into port 4 on the card.

In zaptel.conf I have:
fxsks=1-4

And zapata.conf:
context=incoming
signalling=fxs_ks
busydetect=yes
callprogress=no
musiconhold=default
usecallerid=yes
callerid=asreceived
group=1
channel = 1-4

When I launch an outbound call as ZAP/g1/${EXTEN}, Asterisk goes to
Zap/1 and I hear dead air because there is no line attached to that
port.  Shouldn't it be smart enough to go to Zap/4 as the only available
port in the group?

Thanks,
Adam

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[Asterisk-Users] automated response

[Steve Hearst]

I will be out of the office until Friday June 24th. If this is an emergency 
please call our office and ask for customer service.

Thank you.

Steve Hearst
Accent Communication Services Inc.
740-548-7378
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[Asterisk-Users] ISDN (PRI) in the US and Redirect?

[John Millican]

Hello,
I have read about using redirect on a sip channel to get * to step out of the 
voice path.  Is this possible with ISDN or maybe a US T-1?  I would like to 
have the * box answer a call, do some niffty IVR/AGI stuff, and then redirect 
that call back to the originating switch to be passed on to a dialed number 
or straight to a dialed number and then step out of the voice path to free up 
bandwidth.   Is this possible or do I need to get involved in SS7 stuff? I 
have seen a lot of talk on the wiki about redirect but nothing that I 
understand as what I need.  I have seen the manager API redirect to place 2 
calls into a meetme but this is not what I need, also not looking for call 
deflect.
Thank you for any sugestions,
John M
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Re: [Asterisk-Users] OT: Asterisk and Mambo - help wanted

[JD Austin]

We do mambo projects all the time, contact me off lists and we can get 
you rolling.


JD

Kristian Kielhofner wrote:


Hello everyone,

So, this isn't exactly what it seems.  I am not looking to 
integrate Asterisk and Mambo.  I am the maintainer/creator of 
AstLinux, and I have recently decided that I should really have a 
better web site for it.  I would like to use Mambo so that I can do 
updates easily, from anywhere, without having to waste time learning 
PHP/HTML/etc.  Mambo CMS seems the best and most powerful way to do 
this.  It's not that easy, however, to go from the default Mambo site 
to a site suitable for an open source project such as AstLinux.  I 
just need some help to get a layout, theme, etc. going.  Updates and 
maintenance I can handle (probably).


So, what I am looking for is someone who is familiar with Mambo 
(and preferably Asterisk, too) and would be willing to help me jump 
start astlinux.org/.com.  Because AstLinux is an open source project, 
I will be unable to directly compensate anyone (monetarily) for their 
work at this time.  However, any people that help out are more than 
welcome to plug their own projects, companies, names, etc. on the site 
(within reason).


Interested?  Comments?  Questions?  Suggestions?  Drop me a line.

Thanks!



--
JD Austin
Twin Geckos Technology Services LLC
email: [EMAIL PROTECTED]
http://www.twingeckos.com
phone/fax: 480.288.8195 


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[Asterisk-Users] Garbled one-way audio only with ulaw

[rsenykoff]

For some reason a couple weeks ago users began experiencing garbled audio 
in one direction when dialing out via our VoIP provider. This happened at 
multiple sites simultaneously. The VoIP provider doesn't think it's their 
problem. If I switch to another codec so that Asterisk transcodes 
everything is fine. On conference calls (where Asterisk gets in the middle 
to relay ulaw to all channels) everything is fine. Calls between sites via 
ulaw are excellent.

We have plenty of bandwidth, and QoS in place. I've monitored and our QoS 
box never drops a VoIP packet (4569 UDP).

Phones are Polycom IP500s. We're running 1_0_7 stable. Plenty of CPU 
horsepower (P4 northwood, 2x256 dual channel DDR, 3ware RAID for 
mirroring) to handle maybe 6 calls simultaneously. This problem occurs 
even when there's only 1 call on the line. We're now running GSM with 
decent quality, but I would love to get them back to ulaw.

Any ideas?

TIA,
-Ron

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[Asterisk-Users] Performance Monitoring.

[someshwarak]

Hi All,

Does the T1 PCI cards supported by Digium support Performance Moinitoring.
If so are they in compliance with any of GR or ANSI standards of T1.

Thanks,
Somesh

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