date:20060308

Good grief! I posted the message below at 1:17pm... and it appeared on the list 
after 8pm. 
Nice

-Original Message- 
From: Douglas Garstang 
Sent: Tue 3/7/2006 1:17 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: RE: [Asterisk-Users] res_mysql.conf  DNS SRV lookup



Yay!

-Original Message-
From: Olle E Johansson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 07, 2006 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] res_mysql.conf  DNS SRV lookup



7 mar 2006 kl. 19.03 skrev Douglas Garstang:

 My bad. SRV lookups work, but Asterisk only uses the first entry 
 right? This means there's no redundancy.

That is correct. That is what we try to fix.

/O
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RE: [Asterisk-Users] Re: ON DEMAND call Recording

2006-03-08 Thread Tomislav Parcina

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Martin Joseph
 Sent: 7. ozujak 2006 18:40
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Re: ON DEMAND call Recording
 
 
 On Mar 7, 2006, at 2:38 AM, Giridhar Bandi wrote:
 
  ya i found it it *1 to start recording from the caller end
 
 Also pushing *1 again stops recording.

Do you know how to send that recording to e-mail address that is specified in 
voicemail.conf? That will be a real cool option.


--
Tomislav Parcina
tparcina#lama.hr
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[Asterisk-Users] MeetMe 'i' option not working correctly?

2006-03-08 Thread Jon Webster

I'm running 2.4.5 and app_meetme never plays conf-hasleft or
conf-hasjoined with user names. I looked at app_meetme.c, but couldn't
determine the cause. Any suggestions are greatly appreciated.

exten = 600,1,MeetMe(600|i) I get the following:

  -- Executing MeetMe(SIP/jon-21f8, 600|aciMps) in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
Mar  8 06:13:53 WARNING[5197]: channel.c:2535 ast_request: No channel
type registered for 'zap'
Mar  8 06:13:53 WARNING[5197]: app_meetme.c:461 build_conf: Unable to
open pseudo channel - trying device
-- Created MeetMe conference 1023 for conference '600'
-- Recording
-- Playing 'vm-rec-name' (language 'en')
-- Playing 'beep' (language 'en')
-- x=0, open writing:
/var/spool/asterisk/meetme/meetme-username-600-1 format: sln, 0x81a9278
-- User ended message by pressing #
-- Playing 'auth-thankyou' (language 'en')
-- Playing 'vm-review' (language 'en')
-- Playing 'vm-msgsaved' (language 'en')
-- Playing 'conf-onlyperson' (language 'en')
-- Started music on hold, class 'default', on channel 'SIP/jon-21f8'
-- Stopped music on hold on SIP/jon-21f8

-- Executing MeetMe(SIP/jon-0d36, 600|scpi) in new stack
-- Recording
-- Playing 'vm-rec-name' (language 'en')
-- Playing 'beep' (language 'en')
-- x=0, open writing:
/var/spool/asterisk/meetme/meetme-username-600-3 format: sln, 0x81a7ae8
-- User ended message by pressing #
-- Playing 'auth-thankyou' (language 'en')
-- Playing 'vm-review' (language 'en')
-- Playing 'vm-msgsaved' (language 'en')
-- Playing 'conf-thereare' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'conf-otherinparty' (language 'en')
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[Asterisk-Users] Called number not recognised

2006-03-08 Thread Jason Frisch

I have 10 different numbers that can come into my asterisk box, but
they all seem to end up as the same extension in my dialing plan.

As far as I can tell the reson is that the INVITE line is always the
same number; but t: shows the correct number. Is there a variable
that I need to check for this value?

Header;

INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
v: SIP/2.0/UDP x.x.x.x:5060;branch=something
v: SIP/2.0/UDP x.x.x.x:5060
f: sip:[EMAIL PROTECTED];user=phone;tag=e2078e10
t: sip:[EMAIL PROTECTED]:5060;user=phone
i: [EMAIL PROTECTED]
Cseq: 1 INVITE
m: sip:[EMAIL PROTECTED]:5060;user=phone
c: application/sdp
l: 129
Max-Forwards: 5
Proxy-Require: privacy
k: privacy,timer


Best Regards,

Jason Frisch

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Re: [Asterisk-Users] Polycom 501

2006-03-08 Thread Mojo with Horan Company, LLC

Or TRANSFER - BLIND - NUMBER - SEND, for a blind one.  Works for me, 
no special phone configs.


Moj

[EMAIL PROTECTED] wrote:
Ummm - from memory the sequence is TRANSFER - NUMBER - SEND - chat to 
other person - TRANSFER.
 
PaulH


- Original Message -
*From:* MBIT Technologies mailto:[EMAIL PROTECTED]
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
mailto:asterisk-users@lists.digium.com
*Sent:* Thursday, March 02, 2006 11:02 PM
*Subject:* RE: [Asterisk-Users] Polycom 501

AMP is being run but it seems the transfer needs to be configured in
the phone somewhere so when you press the transfer button its like
hitting #.

 

 


-Original Message-
*From:* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Mark
Aufflick
*Sent:* Thursday, 2 March 2006 10:51 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [Asterisk-Users] Polycom 501

 


One thing to keep in mind when someone says Asterisk does that by
default is that a lot of people have AMP installed, and an AMP
installation includes extra configuration and features as well as
the web interface. It may be that there is phone-specific config
installed with AMP that is not installed in a base Asterisk
installation.

Cheers,

Mark.

-- 
Mark Aufflick

e: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
w: www.pumptheory.com http://www.pumptheory.com (business)
w: mark.aufflick.com http://mark.aufflick.com  (personal)
p: +61 438 700 647
f: +61 2 9436 4737

On 3/2/06, *MBIT Technologies* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

I guess it doesn't work by default on my phone. You still need to
press hash to transfer calls. The transfer button doesn't work.
Where do I set it?

 

 


Regards

 

 


Mark Brooker

T: 02 4959 8670

M: 0415 846 865

F: 02 9882 0947

E: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]

W: http://www.mbit.com.au http://www.mbit.com.au

 


-Original Message-
*From:* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]] *On Behalf Of
*Anton Krall
*Sent:* Thursday, 2 March 2006 3:47 PM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'

*Subject:* RE: [Asterisk-Users] Polycom 501

 


Those buyttons do work with asterisk by default... what kind of
problems are you having?

 




*From:* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]] *On Behalf Of
*MBIT Technologies
*Sent:* Wednesday, March 01, 2006 7:56 PM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* [Asterisk-Users] Polycom 501

Hi Guys

 


Just a quick question regarding on the 501, has anyone been able
to configure the transfer button and messaging buttons to work
with asterisk?

 


Can you share a configuration to do this?

 


Thanks in advance.



iBurst Wireless Broadband from $34.95/month - Platform Networks
http://iburst.platformnetworks.net/emailfooter/

Spam  Virus Filtering by Mail Security
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--
Mojo [EMAIL PROTECTED]
Office Manger, Horan  Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] Echo Cancelation on TE110P

Kerry Garrison wrote:
 On a 55 station install onto a Cox PRI with a TE110P (Polycom 501 phones) a
 few users are complaiining about echo. According to the users, the echo
 seems to be phone number dependant. They claim that certain phone numbers
 have echo while others dont. Are there any tuning parametes like there is
 for a TDM400 card? 

You can either run the software echo can or use a hardware one.

You will have to enable it for all calls however.

The echo will be coming from the remote system.

-- 
Cheers,

Matt Riddell
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[Asterisk-Users] Mitel SIP firmware

2006-03-08 Thread Bromont



 Just in case anyone is interested, there is new Mitel SIP firmware out 
today. Version 5.00.00.16


http://sipdnld.mitel.com/


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Re: [Asterisk-Users] Call Monitor

2006-03-08 Thread Dan Littlejohn

On 1/16/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote:
 Simon Faulkner wrote:
  Does anyone know of a web based live call monitor for *?
 
  I would have thought this was an ideal application for Ajax?

 There's the flash operator panel but nothing much using Ajax.  We're
 doing some chat room stuff but other than than I haven't seen much.
 Sounds like a fun project :)

 --
 Cheers,

 Matt Riddell
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I saw this post about a week ago when I was trying to see if anyone
else was trying out AJAX with Asterisk Applications.

Happy to announce, that ARI is now AJAX enabled and that the voicemail
and call monitor pages self update.

You can take a look at it here
  www.littlejohnconsulting.com/ari

and it has been checked into FreePBX svn.

Dan
www.littlejohnconsulting.com
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[Asterisk-Users]Hangup with error

2006-03-08 Thread asterisk183

I used quadBri Junghanns card and I config
zaptel.conf:
ZAPTEL.CONF
loadzone=it
defaultzone=it
span=1,1,3,ccs,ami
span=2,2,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami

bchan=1,2
dchan=3
bchan=4,5
dchan=6
bchan=7,8
dchan=9
bchan=10,11
dchan=12

ZAPATA.CONF
[channels]
language=it
musiconhold=default
switchtype = euroisdn

; p2mp TE mode (for connecting ISDN lines in
point-to-multipoint mode)
signalling = bri_cpe_ptmp
; p2p TE mode (for connecting ISDN lines in
point-to-point mode)
;signalling = bri_cpe
; p2mp NT mode (for connecting ISDN phones in
point-to-multipoint mode)
;signalling = bri_net_ptmp
; p2p NT mode (for connecting an ISDN pbx in
point-to-point mode)
;signalling = bri_net

pridialplan = local
prilocaldialplan = local
nationalprefix = 0
internationalprefix = 00

echocancel = yes

context=isdn_incoming
group = 1
channel = 1-2

group = 2
channel = 4-5

group = 3
channel = 7-8

group = 4
channel = 10-11

But when I hangup the channel, Asterisk show this
message:

Mar  6 17:31:20 WARNING[1437]: chan_zap.c:6570
handle_init_event: Detected alarm on channel 1: Red
Alarm
Mar  6 17:31:20 WARNING[1437]: chan_zap.c:1593
zt_disable_ec: Unable to disable echo cancellation on
channel 1
Mar  6 17:31:20 WARNING[1437]: chan_zap.c:6570
handle_init_event: Detected alarm on channel 2: Red
Alarm
Mar  6 17:31:20 WARNING[1437]: chan_zap.c:1593
zt_disable_ec: Unable to disable echo cancellation on
channel 2
Mar  6 17:31:20 NOTICE[1433]: chan_zap.c:8511
pri_dchannel: PRI got event: Alarm (4) on Primary
D-channel of span 1
Mar  6 17:31:20 NOTICE[1433]: chan_zap.c:8518
pri_dchannel: pri_shutdown
Mar  6 17:31:20 NOTICE[1437]: chan_zap.c:6565
handle_init_event: Alarm cleared on channel 1
Mar  6 17:31:20 NOTICE[1437]: chan_zap.c:6565
handle_init_event: Alarm cleared on channel 2
Mar  6 17:31:20 NOTICE[1433]: chan_zap.c:8511
pri_dchannel: PRI got event: No more alarm (5) on
Primary D-channel of span 1

Why? And What i can doing for solve this problem?

Thanks






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[Asterisk-Users] sending text to display of sip phones

2006-03-08 Thread Alejandro Vargas

I red that it is possible to send instant messages to the displays of
sip phones. How can I do it using Asterisk?
--
Alejandro Vargas
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Re: [Asterisk-Users] PBX-VPN-SIP-Asterisk trouble

2006-03-08 Thread artifex maximus

Hello!

On 3/7/06, Martin Joseph [EMAIL PROTECTED] wrote:

 On Mar 7, 2006, at 7:02 AM, artifex maximus wrote:

  I have the following setup:
  Phone lines - traditional PBX - Welltech 3802
  - VPN -
  Asterisk - Linksys PAP2/Welltech ATA-151 - phone
 
  There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2)
  PBX extensions. Asterisk is a proxy here. Each device successfully
  register itself. I tried the setup above with Linksys and Welltech
  devices as well.
 
  I setup Asterisk as a local PBX and phones can call each others on
  Asterisk side and possible transfering calls. I setup Welltech 3802
  with hotline mode so if someone call the public number from outside
  the call transferred through VPN and phone rings in front of me.
  Great. It's still possible transfer call within Asterisk side.
  Excellent.
 
  The problem comes when I want to call extension on PBX side or
  transfer incoming call to the PBX side. I got the line sound when I
  press flash, the caller hear the MOH and when I call extensions on PBX
  side I got only busy tone.
 If I am not mistaken, then I think your question is regarding the flash
 event on the welltech hardware and getting that to work correctly for
 switching in a call waiting situation?  Although your setup is MUCH
 more complex then mine,  I have this problem with the welltech 3701a as
 well.
You're right. There is some problem with Welltech flash event (don't
or can't send to PBX) I think as well but I hope I was wrong. I tried
my config without Asterisk and directly connect ATA-151 (bureau mode)
with 3802 (hotline mode). Same problem. Then Asterisk came in picture
because we use 5 of them successfully (just massive memory leaks in
interactive mode and need reboot every 2-3 hours).

  How could I tell that Asterisk send back the flashDTMF on the same
  PBX extension where call comes from? I think this is important for PBX
  to connect lines inside right. How could I route outcoming calls on
  a port of Welltech 3802?
 The outgoing calls on the welltech (at least for the SIP setup) are
 routed via the welltechs setup screens which include a totally cryptic
 routing setup or some such.  Try searching this list for the last
 couple of days for welltech I think someone explained this nicely.
  snip
 Sorry I am not helpful on this, but looking for a similar answer?
Thanks for your answer! I happy with every answer because I don't know
where could I go. I hate Welltech GUI because a lot description/label
is bad and might missleading and there are some parameters you could
adjust only in terminal mode. Documentation is bad as well. But I need
to work with this hardware. I'll do a search on past mails.

Do you know where should I find a list with Asterisk compatible USB
modems/chipsets? We plan putting Asterisk on the PBX side and with USB
modems as Zap channels we could make flash to PBX but I don't find any
list about compatible USB modems/chipsets.

Bye,
Zsolt
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Re: [Asterisk-Users] IAXy (S101) echo?

2006-03-08 Thread Anthony Rodgers


Hi Bradley,

Yes, I experienced quite a lot of echo with my IAXy, until I switched 
analog handsets - in my case, it was severe acoustic coupling in a 
cheap handset.


Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On Mar 7, 2006, at 11:38 AM, Bradley M. Kuhn wrote:

I just purchased an IAXy (S101) for a home setup; I've become a 
de-facto

expert on Asterisk for work.

Everything is working great, but I notice a substantial echo on calls
connected through the IAXy to POTS telephones.


Has anyone encountered something similar and found a solution?  I found
some posts about this in the past few years, but never any replies.  
The
Wiki on voip-info.org doesn't seem to have anything about it; I'd be 
happy

to condense any replies I receive to information to put up there.

Thanks!


   -- bkuhn


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[Asterisk-Users] OT: Polycom Registration Weirdness

This is a SER/Polycom question, but I hoped we may have some SER guru's here...

I have a series of Polycom phones that are tying to register with OpenSER. The 
phone sends a REGISTER message, and OpenSER replies with Unauthorised (all 
normal). The phone re-sends the REGISTER with the credentials, and OpenSER 
sends Ok.

Here's where it goes downhill. The polycom's appearance display does not change 
from an unregistered to a registered state, ie it does not change from an empty 
phone to a filled in one. It doesn't think it's registered eventhought it's 
gotten an OK. Then, a regular intervals it keeps trying to register again, 
because it still thinks it wasn't successful.

I have some other Polycom phones that are not doing this. All have the same SIP 
software version, and all essentially have the same xml config files, with 
minor variations. Happening with OpenSER 1.0.0 and 1.0.1

I have pasted ngrep output of one of these below. Anyone got any ideas?

#
U 216.187.128.72:5060 - 216.187.140.233:5060
REGISTER sip:ipt.oneeighty.com SIP/2.0.
Via: SIP/2.0/UDP 216.187.128.72;branch=z9hG4bKca28b8d3BC755D46.
From: Sandy Sauvageau sip:[EMAIL PROTECTED];tag=2A2425B5-B64A4132.
To: sip:[EMAIL PROTECTED].
CSeq: 1 REGISTER.
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED];methods=INVITE, ACK, BYE, CANCEL, OPTIONS, 
INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067.
Max-Forwards: 70.
Expires: 3600.
Content-Length: 0.
.

#
U 216.187.140.233:5060 - 216.187.128.72:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 216.187.128.72;branch=z9hG4bKca28b8d3BC755D46.
From: Sandy Sauvageau sip:[EMAIL PROTECTED];tag=2A2425B5-B64A4132.
To: sip:[EMAIL PROTECTED];tag=136c3bb27674cf7e44f7b05275ffaecc.0629.
CSeq: 1 REGISTER.
Call-ID: [EMAIL PROTECTED]
WWW-Authenticate: Digest realm=ipt.oneeighty.com, 
nonce=440e4b3f113243b90ba483b6a2f243ea51377e2d.
Server: OpenSer (1.0.0 (i386/linux)).
Content-Length: 0.
.

#
U 216.187.128.72:5060 - 216.187.140.233:5060
REGISTER sip:ipt.oneeighty.com SIP/2.0.
Via: SIP/2.0/UDP 216.187.128.72;branch=z9hG4bKff7cb008AF06A32B.
From: Sandy Sauvageau sip:[EMAIL PROTECTED];tag=2A2425B5-B64A4132.
To: sip:[EMAIL PROTECTED].
CSeq: 2 REGISTER.
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED];methods=INVITE, ACK, BYE, CANCEL, OPTIONS, 
INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067.
Authorization: Digest username=2944029, realm=ipt.oneeighty.com, 
nonce=440e4b3f113243b90ba483b6a2f243ea51377e2d, uri=sip:ipt.oneeighty.com, 
response=9d8b4708296f3fb88d5cfd453121860d, algorithm=MD5.
Max-Forwards: 70.
Expires: 3600.
Content-Length: 0.
.

#
U 216.187.140.233:5060 - 216.187.128.72:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 216.187.128.72;branch=z9hG4bKff7cb008AF06A32B.
From: Sandy Sauvageau sip:[EMAIL PROTECTED];tag=2A2425B5-B64A4132.
To: sip:[EMAIL PROTECTED];tag=136c3bb27674cf7e44f7b05275ffaecc.32b4.
CSeq: 2 REGISTER.
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED];expires=3600.
Server: OpenSer (1.0.0 (i386/linux)).
Content-Length: 0.
.

 
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Re: [Asterisk-Users] Question from a newbie on finding digium hosts

2006-03-08 Thread Joseph Tanner

What exactly do you need?  A digium card could be anything from one
pstn line, to multiple t1 lines, to who knows what else.  And serial
number authentication...what's this for?  Does a user dial in, enter
in a serial, then get access to something?  Like a calling card, or
something completely different?

If all you need is rack space, I'm sure there's some people here who
could help you out.  Even I have rack space available, and I'm not
exactly a big host.  Maybe you could ask this on the biz list?  If all
you need is an internet connection (don't need a voice T1 line), then
just about anybody who can colocate a server will do.  Might even be
cheaper to lease a server (seems odd, but leasing a server can be
cheaper than just renting space for a server you own). 
WebHostingTalk.com is a good place to look for a host, but first we
need to know exactly what you need, then we can steer you in the right
direction.

Joseph Tanner

On 3/7/06, Gene Expression [EMAIL PROTECTED] wrote:
 Hey all,

 I have a client whose previous programmer ditched.  I'm his webmaster,
 and now he wants me to have an asterisk system set up for serial
 number authentication...and I have a digium card from the previous
 guy.  Are there hosts that will set this up for me?  ie, rack space
 somwhere?  Are there guides online I can look at?

 Thanks
 Razib
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Re: [Asterisk-Users] Unable to start Asterisk 1.2.5 with Asterisk-Addons 1.2.1

2006-03-08 Thread Sharath Chandra

Thanks Moj. 
But i need to connect to MySQL. Could this be a problemwith C libraries that i am using.

Regards,
Sharath 


On 3/8/06, Mojo with Horan  Company, LLC [EMAIL PROTECTED] wrote:
This may not be the applicable solution, but if you're not using themysql config capabilities, add noload = res_config_mysql.so to
modules.confMojSharath Chandra wrote: Hi all, I installed the Asterisk 1.2.5 and asterisk-addons 1.2.1 of a new Red Hat linux box( Linux version 2.4.20-8smp). I was able to compile both
 the software but when i start Asterisk, it exits with the following dump. Error Text Start= [res_crypto.so] = (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel'
 -- Loaded PUBLIC key 'freeworlddialup'[res_config_mysql.so]Mar6 05:18:23 WARNING[12779]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: __stack_chk_fail
 Mar6 05:18:23 WARNING[12779]: loader.c:554 load_modules: Loading module res_config_mysql.so failed! End=== Can someone suggest a solution.
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Re: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

2006-03-08 Thread Brian Capouch


Douglas Garstang wrote:

Pardon my candour, but for a product Digium calls 'enterprise grade' it sure 
seems to be missing a few features.



You can't resist digging at Digium every time something doesn't work 
just the way you expect it to, can you?


Someday you'll be bleating in the ether all to yourself, ingrate.

B.
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RE: [Asterisk-Users] Send One Touch Record to mail

2006-03-08 Thread Tomislav Parčina




Hi Joe!
Thank you for your mail. The thing is that I have never 
program anything so it will take a lot of my time, which I don't have right now. 
Hopefully, when I finish started projects I'll be able to play with this 
stuff.

In the meantime if anybody solves this problem, please let 
the group know.



--Tomislav 
Parcinatparcina#lama.hr



  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Joe 
  PukepailSent: 7. ožujak 2006 20:41To: Asterisk Users 
  Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] Send One Touch Record to mail
  
  As far as I know, you will need to do this yourself with some creative 
  scripting. There was some talk on the list awhile ago to move the 
  recording tovoicemail, but I dont' know if anyone has made a patch to do 
  it yet.
  
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[Asterisk-Users] icmp 36: 192.168.30.32 udp port 5004 unreachable

2006-03-08 Thread Todd Vinson

Hello all,

I am having an issue with a BT-101 and * . When dialing a number from the
BT-101, upon the remote side answering, the call is established but no
audio is passed in either direction.  I have tcpdump'd this session and
found this:

(192.168.30.1 is * - 192.168.30.32 is BT-101)

22:41:47.899462 IP 192.168.30.32  192.168.30.1: icmp 36: 192.168.30.32
udp port 5004 unreachable
22:41:47.899553 IP 192.168.30.32  192.168.30.1: icmp 36: 192.168.30.32
udp port 5004 unreachable
22:41:47.930185 IP 192.168.30.31.8000  192.168.30.1.10020: UDP, length 172
22:41:47.930260 IP 192.168.30.1.10082  192.168.30.32.5004: UDP, length 172
22:41:47.930637 IP 192.168.30.32  192.168.30.1: icmp 36: 192.168.30.32
udp port 5004 unreachable
22:41:47.961942 IP 192.168.30.31.8000  192.168.30.1.10020: UDP, length 172
22:41:47.961983 IP 192.168.30.31.8000  192.168.30.1.10020: UDP, length 172
22:41:47.962019 IP 192.168.30.1.10082  192.168.30.32.5004: UDP, length 172
22:41:47.962043 IP 192.168.30.1.10082  192.168.30.32.5004: UDP, length 172
22:41:47.962495 IP 192.168.30.32  192.168.30.1: icmp 36: 192.168.30.32
udp port 5004 unreachable
22:41:47.962582 IP 192.168.30.32  192.168.30.1: icmp 36: 192.168.30.32
udp port 5004 unreachable
22:41:47.992671 IP 192.168.30.31.8000  192.168.30.1.10020: UDP, length 172
22:41:47.992729 IP 192.168.30.1.10082  192.168.30.32.5004: UDP, length 172
22:41:47.993111 IP 192.168.30.32  192.168.30.1: icmp 36: 192.168.30.32
udp port 5004 unreachable
22:41:48.003760 IP 192.168.30.30.5060  192.168.30.1.5060: UDP, length 2
22:41:48.023907 IP 192.168.30.31.8000  192.168.30.1.10020: UDP, length 172
22:41:48.023925 IP 192.168.30.31.8000  192.168.30.1.10020: UDP, length 172
22:41:48.023984 IP 192.168.30.1.10082  192.168.30.32.5004: UDP, length 172
22:41:48.024011 IP 192.168.30.1.10082  192.168.30.32.5004: UDP, length 172

I have tested the phone with its current config at a friends house running
the same version of * , and the phone works fine.  I plug it into my
network, and the issues remain.

How do I make * listen on udp/5004 so that the communications will work? 
I can't find anything anywhere on this, but have noticed a handful of
people reporting this same behavior.  I am starting to suspect an issue
with the NIC.

Thanks for the help!

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Re: [Asterisk-Users] MWI, SER and asterisk

2006-03-08 Thread Christian B

Hi Sharon!

This is pretty difficult, i was not able to implement it so far(though
my ser-skills are pretty basic).
At http://www.voip-info.org/wiki-Asterisk+at+large you'll find some
howto's, method 2 seems to be the most promising to me...

regards
christian

On Tue, 7 Mar 2006 15:36:57 -0600
Sharon [EMAIL PROTECTED] wrote:

 I have my peers registered to SER.asterisk seems to be sending mwi for
 the peers seen in the sip show peers CLI command. i have my ser server
 registered with asterisk as a type=friend and all clients register to
 ser.how do i get mwi to work for these clients registered to SER.
 
 Thank you,
 -AA
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RE: [Asterisk-Users] new beta Grandstream firmware HT488_496_386

2006-03-08 Thread Andre Rodrigues $Cheyenne$

Hi Phil.

So, you are just like I am...

You can´t make a conclusion about this problem, and nobody has a clue about
what could be the problem...

I will change more than 20 to sipurar 2002... A pay for this

I will never buy again grandstreeam hardware,And I´ve bought more than 100
phones(bt and Gxp)...

I couldn´t believe so bad these hardware could be...


Regards.
Andre Rodrigues

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil Blundell
Sent: terça-feira, 7 de Março de 2006 22:26
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] new beta Grandstream firmware HT488_496_386

Experiences with the HT386s seem to be pretty variable: they work OK for
some folks, and are virtually unusable for others.  

A couple of months back, I installed two HT386s side by side.  One of
them would lock up on an almost daily basis; I replaced it with an
SPA2002, which has been much better.  The other one, interestingly, has
been much more reliable: it's crashed once or twice in the 8 or 10 weeks
since it's been there, but no more than that.  Both of them were running
the same firmware, so the difference in behaviour must be due to either
hardware revisions or some quirk of the use patterns that the two units
were seeing.

p.

On Tue, 2006-03-07 at 08:40 -0500, Steve Jones wrote:
 I'm almost afraid to ask, but is the HT 386 known for having a lot of
troubles?  I just installed one at home about 2 weeks ago, and knock on
wood, it's only locked up once, and this was when I was still in the process
of tweaking the config to work optimally w/ [EMAIL PROTECTED]  I can't say
I'm entirely pleased with the slight echo and buzz I'm detecting, but so far
it's at least worked..  This isn't the consensus though, huh?!
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
 Sent: sábado, 4 de Março de 2006 0:02
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] new beta Grandstream firmware HT488_496_386
 
 They promised me this for my POS 386 adapters that need to be rebooted
 every few days from lockups about 4 months ago.  Gee I wonder if this
 will work.  Probably not.
 
 On 3/3/06, Martin Joseph [EMAIL PROTECTED] wrote:
  http://grandstream.com/BETATEST/HT488_496_386/
 
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[Asterisk-Users] REGISTER headers changed

2006-03-08 Thread Jason Frisch


Can someone help me with upgrading to the lastest version. I am using the
same sip.conf file, but the headers have changed and registration fails.
Has something change in the conf file that would cause this?

Notice 1.2.5 has no Authoization at all...

Regards,

Jason


Version 1.0.9
---
REGISTER sip:voip-ca35323.ocn.ne.jp SIP/2.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK035070d8
From: sip:[EMAIL PROTECTED];tag=as1cdfeadf
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username=someusername,
realm=nc01.ipp.biglobe.ne.jp, algorithm=MD5, uri=sip:2
10.227.109.232, nonce=1141805370,
response=016070a49b3caa88a3fb76e8b7a91aa1, opaque=
Expires: 120
Contact: sip:[EMAIL PROTECTED]
Event: registration
Content-Length: 0

(no NAT) to 210.227.109.232:5060
denwa*CLI

Sip read:
SIP/2.0 200 OK
v: SIP/2.0/UDP xxx:5060;branch=z9hG4bK035070d8
From: sip:[EMAIL PROTECTED];tag=as1cdfeadf
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Expires: 7200
m: sip:[EMAIL PROTECTED]:5060;expires=7200
Event: registration
Content-Length: 0
Date: Wed, 08 Mar 2006 08:55:11 GMT


--
Version 1.2.5
--
REGISTER sip:voip-ca35323.ocn.ne.jp SIP/2.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK08b2597f;rport
From: sip:[EMAIL PROTECTED];tag=as6d23ff7d
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: sip:[EMAIL PROTECTED]
Event: registration
Content-Length: 0

-- SIP read from :5060:
SIP/2.0 400 Bad Request
v: SIP/2.0/UDP :5060;branch=z9hG4bK08b2597f
From: sip:[EMAIL PROTECTED];tag=as6d23ff7d
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Event: registration
l: 0



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RE: [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3

2006-03-08 Thread Azfhasterisk

We had the same issue but we found that it was really the MS proxy server
that the phone was going though. Set it up to use a different route out to
the server and everything worked fine.

Had to prove it to the admin at the location too, that was fun!

Rick

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio
Sent: Tuesday, March 07, 2006 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3

HTTP's nice, but FTP does the job.  Check the docs for supported FTP
servers -- many of the stock Linux FTP servers will give the exact problem
you discussed, below.  I should know -- took me almost a week before
trying proftpd, and WHAMMO, worked like a champ.

-Ken

On Tue, March 7, 2006 12:37 pm, William M Conlon wrote:
 I spent a weekend battling similar issues with 501s, using FC4/
 proftpd.  I finally switched from FTP to HTTP.


 On Mar 7, 2006, at 9:53 AM, Kristian Kielhofner wrote:


 Hello everyone,


 Please forgive the exclamation points but I have been battling
 this one off and on for about four days now.  Sorry for the cross post.

 It all started with a box of IP 501s.  I contacted my reseller and
 obtained the latest BootRom and SIP firmware.  Unzipped, configured,
 copied over to my FTP server (running AstLinux, of course).  The phone
 booted, so far so good.  Updated bootrom, nice.  Rebooted again. Updated
 sip firmware.  Also nice.

 Upon the next reboot, the wheels started falling off.  The phones
 would not get changes I made to any of the .cfg files.  Several phones
 would take 20 minutes or more to boot, only to display a 0x4000 config
 file error.  What happened?

 I have been using various Polycom's with AstLinux (and vsftpd
 2.0.3 that I include with it) for quite some time, with no problems
 whatsoever.  Until now.

 I had been running bootrom 3.0.1 and various versions of the SIP
 image at several other sites with no problem.  At this point I was still
 unable to accept the fact that I might not be able to run this latest
 bootrom.  After many trial and tribulations, I finally rsync'ed (with
 -avr) the FTP directory from the AstLinux machine to
 my laptop running CentOS 4.  I configured the vsftpd daemon (version
 2.0.1) IDENTICALLY (with the exception of PAM and TCP
 wrappers) and crossed my fingers...

 After re-configuring the IP 501 to use my laptop, imagine my
 surprise when the most problematic of them booted right away without
 problems. Again and again, everything was fine.

 So now I just had to break out ethereal and see what was going on.
 While I have not completely finished my analysis, it appears that
 Polycom firmware 3.1.3 bombs out when transferring files with
 vsftpd 2.0.3.  The symptom appears to be repeated TCP SYNs from the
 Polycom to the ftp daemon on port 20.  The Polycom will keep
 retrying and increment its source port number by one every few minutes.
 Like I said, I need to dig into this more, but I figured
 I'd report what I know and see if anyone out there can fill in the
 holes.

 Here's what I did.  It appears that BootRom 3.1.3 works with
 vsftpd 2.0.1, so I placed bootrom 3.0.1 (which I know works with vsftpd
 2.0.3) on my CentOS server and downgraded the phone to
 3.0.1.  I then placed 3.0.1 and SIP app 1.6.5 (which I was using
 the whole time, btw) on my AstLinux server running vsftpd 2.0.3.

 All was good.  So now I am successfully running with the following:


 Polycom IP 501
 Bootrom 3.0.1
 SIP 1.6.5
 AstLinux 0.3.7
 vsftpd 2.0.3

 I will also try to fix (or workaround) this by trying the following:


 upgrading AstLinux to include vsftpd 2.0.4 trying an intermediate
 BootRom release between 3.0.1 and 3.1.3
 (find out exactly where/when it broke)
 trying an even newer Polycom BootRom when it becomes available upgrading
 the kernel in AstLinux (I doubt that's it) fiddling with iptables rules
 in AstLinux (iptables was loaded, but obviously 3.0.1 doesn't have a
 problem with it)

 This also might be related to the problems described here:


 http://forums.digium.com/viewtopic.php?
 p=14847sid=6e70577c37bd345cfc164a01e64e113a


 Any thoughts?  Comments?  Suggestions?


 P.S. - I will be updating the Polycom config files at http://
 www.krisk.org/asterisk/pcom/ to reflect some new changes in this firmware
 release.  I just need to get my phones working first :)!

 --
 Kristian Kielhofner
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 Bill


 William M. Conlon, P.E., Ph.D.
 To the Point
 345 California Avenue Suite 2
 Palo Alto, CA 94306
 vox:  650.327.2175 (direct)
 fax:  650.329.8335
 mobile:  650.906.9929
 e-mail:  mailto:[EMAIL PROTECTED]
 web:  http://www.tothept.com

Re: [Asterisk-Users] indications SIP

2006-03-08 Thread Can2002

On Tue, 7 Mar 2006 18:49:58 +0100, Olle E Johansson [EMAIL PROTECTED]
said:
  With SIP phones, the phone, not Asterisk, generates all the indications.
  Check with Aastra.
  
  In some cases, like during a call transfer, Asterisk may generate a  
  tone.

Hi Olle,

Thanks for clarifying things for me!

Chris
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Re: [Asterisk-Users] Unable to start Asterisk 1.2.5 with Asterisk-Addons 1.2.1

2006-03-08 Thread Andrew D Kirch

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Mojo with Horan  Company, LLC wrote:
 This may not be the applicable solution, but if you're not using the
 mysql config capabilities, add noload = res_config_mysql.so to
 modules.conf
 
 Moj
 
 Sharath Chandra wrote:
 
 Hi all,
  
 I installed the Asterisk 1.2.5 and asterisk-addons 1.2.1 of a new Red
 Hat linux box( Linux version 2.4.20-8smp). I was able to compile both
 the software but when i start Asterisk, it exits with the following dump.
 Error Text Start=
 [res_crypto.so] = (Cryptographic Digital Signatures)
 -- Loaded PUBLIC key 'iaxtel'
 -- Loaded PUBLIC key 'freeworlddialup'
  [res_config_mysql.so]Mar  6 05:18:23 WARNING[12779]: loader.c:325
 __load_resource: /usr/lib/asterisk/modules/res_config_mysql.so:
 undefined symbol: __stack_chk_fail
 Mar  6 05:18:23 WARNING[12779]: loader.c:554 load_modules: Loading
 module res_config_mysql.so failed!
 End===
  
 Can someone suggest a solution.
  
 Regards,
 Sharath Chandra


 

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You may want to update/rebuild mysql-addons, if this still occurs file a
bug.

- --
Andrew D Kirch  |   Abusive Hosts Blocking List  | www.ahbl.org
Security Admin  |  Summit Open Source Development Group  | www.sosdg.org
Key fingerprint = 4106 3338 1F17 1E6F 8FB2  8DFA 1331 7E25 C406 C8D2
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

iD8DBQFEDmo+EzF+JcQGyNIRAuauAJ9ITVC+TVbF7UQLo5vaedYA4clvTwCeOo39
UfwgH91CafwZE2ZESjfrfWE=
=KzDV
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Re: [Asterisk-Users] can i get the script

pali ismail wrote:
 i have do some touch tones registration system in asterisk .
 
 know i hae some problem i my extensions.conf ,,,because the script there
 cannot run yet
 
 so i hope some budy have codeing in check password plase give to me
 
 i can check that my code its right or wrong

:)

You will need to show us the code.  Is there any reason you are not
simply using readdtmf or authenticate applications in the dialplan?

-- 
Cheers,

Matt Riddell
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[Asterisk-Users] HOWTO volume per (7960) phone

2006-03-08 Thread support

I may be mistaken (the first time) but I thought someone once told me it
was possible to set the volume on a per phone basis. All users have
7960's running 7.4 or 7.5; server running 1.2.5;  one tdm04b.  Is this
indeed possible?, and, if so, would someone be so generous as to point
me at a URL.

Thanks very much.

Cheers.

Jason SJOBECK
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RE: [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3

Docs? Polycom has docs? Where would one find this fabled land of... err I mean 
Polycom does stating what ftp servers are supported?

Doug.

-Original Message-
From: Ken D'Ambrosio [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 07, 2006 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3

HTTP's nice, but FTP does the job.  Check the docs for supported FTP
servers -- many of the stock Linux FTP servers will give the exact problem
you discussed, below.  I should know -- took me almost a week before
trying proftpd, and WHAMMO, worked like a champ.

-Ken

On Tue, March 7, 2006 12:37 pm, William M Conlon wrote:
 I spent a weekend battling similar issues with 501s, using FC4/
 proftpd.  I finally switched from FTP to HTTP.

 On Mar 7, 2006, at 9:53 AM, Kristian Kielhofner wrote:

 Hello everyone,

 Please forgive the exclamation points but I have been battling
 this one off and on for about four days now.  Sorry for the cross post.

 It all started with a box of IP 501s.  I contacted my reseller and
 obtained the latest BootRom and SIP firmware.  Unzipped, configured,
 copied over to my FTP server (running AstLinux, of course).  The phone
 booted, so far so good.  Updated bootrom, nice.  Rebooted again. Updated
 sip firmware.  Also nice.

 Upon the next reboot, the wheels started falling off.  The phones
 would not get changes I made to any of the .cfg files.  Several phones
 would take 20 minutes or more to boot, only to display a 0x4000 config
 file error.  What happened?

 I have been using various Polycom's with AstLinux (and vsftpd
 2.0.3 that I include with it) for quite some time, with no problems
 whatsoever.  Until now.

 I had been running bootrom 3.0.1 and various versions of the SIP
 image at several other sites with no problem.  At this point I was still
 unable to accept the fact that I might not be able to run this latest
 bootrom.  After many trial and tribulations, I finally rsync'ed (with
 -avr) the FTP directory from the AstLinux machine to
 my laptop running CentOS 4.  I configured the vsftpd daemon (version
 2.0.1) IDENTICALLY (with the exception of PAM and TCP
 wrappers) and crossed my fingers...

 After re-configuring the IP 501 to use my laptop, imagine my
 surprise when the most problematic of them booted right away without
 problems. Again and again, everything was fine.

 So now I just had to break out ethereal and see what was going on.
 While I have not completely finished my analysis, it appears that
 Polycom firmware 3.1.3 bombs out when transferring files with
 vsftpd 2.0.3.  The symptom appears to be repeated TCP SYNs from the
 Polycom to the ftp daemon on port 20.  The Polycom will keep
 retrying and increment its source port number by one every few minutes.
 Like I said, I need to dig into this more, but I figured
 I'd report what I know and see if anyone out there can fill in the
 holes.

 Here's what I did.  It appears that BootRom 3.1.3 works with
 vsftpd 2.0.1, so I placed bootrom 3.0.1 (which I know works with vsftpd
 2.0.3) on my CentOS server and downgraded the phone to
 3.0.1.  I then placed 3.0.1 and SIP app 1.6.5 (which I was using
 the whole time, btw) on my AstLinux server running vsftpd 2.0.3.

 All was good.  So now I am successfully running with the following:

 Polycom IP 501
 Bootrom 3.0.1
 SIP 1.6.5
 AstLinux 0.3.7
 vsftpd 2.0.3

 I will also try to fix (or workaround) this by trying the following:

 upgrading AstLinux to include vsftpd 2.0.4 trying an intermediate
 BootRom release between 3.0.1 and 3.1.3
 (find out exactly where/when it broke)
 trying an even newer Polycom BootRom when it becomes available upgrading
 the kernel in AstLinux (I doubt that's it) fiddling with iptables rules
 in AstLinux (iptables was loaded, but obviously 3.0.1 doesn't have a
 problem with it)

 This also might be related to the problems described here:

 http://forums.digium.com/viewtopic.php?
 p=14847sid=6e70577c37bd345cfc164a01e64e113a

 Any thoughts?  Comments?  Suggestions?

 P.S. - I will be updating the Polycom config files at http://
 www.krisk.org/asterisk/pcom/ to reflect some new changes in this firmware
 release.  I just need to get my phones working first :)!

 --
 Kristian Kielhofner
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 Bill

 William M. Conlon, P.E., Ph.D.
 To the Point
 345 California Avenue Suite 2
 Palo Alto, CA 94306
 vox:  650.327.2175 (direct)
 fax:  650.329.8335
 mobile:  650.906.9929
 e-mail:  mailto:[EMAIL PROTECTED]
 web:  http://www.tothept.com

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 To

Re: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

Douglas Garstang wrote:
 Pardon my candour, but for a product Digium calls 'enterprise grade' it sure 
 seems to be missing a few features.

Um...it's Open Source.  Why don't you add the features you require
yourself or pay someone to add them for you...

This is your third similar post in as many days.

-- 
Cheers,

Matt Riddell
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[Asterisk-Users] Dial command

2006-03-08 Thread Ronald Wiplinger

I have an ZAP extension number 222 which is connected instead to a phone 
to a FXS/FXO converter and from there to a CDMA gateway.


To dial my mobile phone I use:

222 (wait 2 seconds) 09123456789

I cannot figure out how to write this into the dialplan as a default number!

222 as above I will use for dialing any other number, but I want to add 
this phone as an extension which rings if 601 is not picking up within 
20 seconds.


How to write this?


Some parts of my existing dial plan:
[Globals]
PHONE_222=ZAP/2r1; transfer to mobile phone ===  
hier I want to add the mobile phone number


[incoming]
...
exten = 
s,7,Dial(${PHONE_601}${PHONE_621}${PHONE_603}${PHONE_610},30,tr)  ; 
ring phone_601, 621  603 for 30 seconds

exten = s,8,Dial(${PHONE_222},30,tr)  ; ring phone_222
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[Asterisk-Users] System Design

2006-03-08 Thread Jason Adams




Hey 
Everyone,

We are in the works 
of planning a new * installation for our company. We have 20 users in our 
main office and 5 users in a remote office a couple of states away. Our 
call volume for the main office will be anywhere from 5-10 concurrent 
calls. The remote office will have about 3 heavy users with two users 
making calls occasionally.

Right now we have an 
existing PBX. We have a T-1/PRI coming into the main office and a DSL 
connection at the remote office. We have a Cisco 2610/PIX 501 at the main 
office a cheesy linksys router at the remote site.

We are planning on 
purchasing new Cisco IP phones for everyone.

My main question is 
this: What type of hardware/network design would be best for this 
situation? Would a full T-1 at the remote site work with a VPN between the 
offices? Or would a higher bandwidth DSL work with a VPN? Or should 
we move to a Point-to-Point connection? What type of hardware would be 
best for the end-to-end communication in regards to QoS? I know the PIX 
501 doesn't support it.
Would it be best to 
have two * servers in each office or for that call volume at the remote office 
does it make sense? I was thinking of a Dell Power Edge server with 4GB of 
ram and a dual processor.. is that enough?

Sorry for all the 
questions!

- Jason
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RE: [Asterisk-Users] [EMAIL PROTECTED] and H323

2006-03-08 Thread leonimar cape

Hi,

You have get the openh323 and pwlib release supported
by asterisk 1.2.1. You can check it on the README file
located at the /path/of/asterisk/channels/h323/



--- Viktor Tatianin [EMAIL PROTECTED] wrote:

 Hi
 I use Asterisk 1.2.1
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED]
 Behalf Of leonimar cape
 Sent: Tuesday, March 07, 2006 12:29 PM
 To: Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] and H323
 
 
 Hi Viktor,
 
 What is the version of the asterisk you are using?
 You
 should use the right version of Openh323 and pwlib
 to
 be able to compile chan_oh323 successfully.
 Currently using asterisk 1.2.4 used
 openh323-Mimas_patch2-src-tar.gz and
 pwlib-Mimas_patch2-src-tar.gz for compiling
 chan_oh323.
 
 Hope this help.
 
 
 --- Viktor Tatianin [EMAIL PROTECTED] wrote:
 
 
  Hello
 
  I attempt installing H323 at my [EMAIL PROTECTED] for
  this  use
  asteriskathome-h323-1.0.zip but have next problem
 
  chan_oh323.c:37:34: asterisk/channel_pvt.h: No
 such
  file or directory
  chan_oh323.c: In function `oh323_show_channels':
 
 
  Please help for resolve this problem
 
 
  Viktor Tatianin
 
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Re: [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud

Brian Roy wrote:
 On 3/3/06, Gary Richardson [EMAIL PROTECTED] wrote:
 I'm running 1.2.4 and just about every call is cut short. I'm using Cisco
 IP phones as end points. All the outbound calls are routed via SIP through a
 PRI line attached to a Cisco 2811..

 
 
 I'm running 1.2.1 and most of mine get cut short too. I posted this on the
 list a few months ago and nobody had any suggestions. BJ said I should
 probably post a bug on it but I haven't had time to continue to troubleshoot
 it. I will go to 1.2.4 (now 5 probably) and see if mine goes away. I've been
 watching change logs and hadn't seen anything surrounding mixmonitor so I've
 let it go.
 
 Please continue to update us if anyone gets some resolution. I'm glad to
 know there are lots of us experiencing this. That should be the catalyst to
 get it fixed.

The only catalyst to getting it fixed will be if someone posts a bug
entry with full details on bugs.digium.com

If you do, post again here with the ID and discussion and testing can
continue there.

-- 
Cheers,

Matt Riddell
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[Asterisk-Users] Reverse group in zapata.conf

2006-03-08 Thread Sean Kennedy


Hey all,

I have a situation where I have 8 lines from the phone company in a hunt 
group coming in to my asterisk box.  These are the same lines I'm using 
for outgoing calls ( named g0 ). 

The problem arises when someone dials our number at the same time 
asterisk tries to put a call out on one of the zap channels in the g0 
group.  This has happened twice that I know of so far, once to myself.  
Asterisk opens the line before it's answered, and tries to dial.  This 
has the effect of connecting the outside caller to the dialing party, 
which is the problem.


My rather messy solution would be to have a reverse 'group' command in 
my zapata.conf file.  So if I try dialing out on g1 ( my reverse group, 
24-17 ), it starts at the top and works it's way down.  Meanwhile, my 
external hunt group would still ring normally ( 17-24 ), thus minimizing 
the potential for conflict to a level that I'm comfortable with.


Is this possible?  If it isn't, I plan to reverse the order in which the 
lines are connected to my * box, having the same effect ( with no 
configuration changes.  :) ).  Anybody have any advice why I shouldn't 
do this either?  Any other suggestions?


Thanks

Sean Kennedy
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Re: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

2006-03-08 Thread C F

Right, Avaya can do that. Use Avaya.

On 3/7/06, Douglas Garstang [EMAIL PROTECTED] wrote:
 Pardon my candour, but for a product Digium calls 'enterprise grade' it sure 
 seems to be missing a few features.

 -Original Message-
 From: Alexander Lopez [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, March 07, 2006 9:39 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Oh this is bad bindaddr and rtp
 traffic

 Asterisk does not like multiple interfaces in the way you are configured. You 
 can either:

 A) use the bindaddr in the sip.conf to limit where the packsge come and go.

 B) use an outside traffic manager

 Look up the archives, kpf explained why this would not work, as asterisk 
 can't do load balancing at this time

 -Original Message-
 From: Robert Webb [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: 3/7/06 11:27 AM
 Subject: Re: [Asterisk-Users] Oh this is bad bindaddr and rtp traffic

 On Tue, 7 Mar 2006 09:12:25 -0700
   Douglas Garstang [EMAIL PROTECTED] wrote:
  I have a configuration where RTP traffic is going out
 interface pub0, and coming back into through pub1.
  I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an
 shows:

  udp0788 0.0.0.0:50600.0.0.0:*

  which means that Asterisk is listening on all addresses
 (on all interfaces?).

  Anyway, when the RTP traffic comes back in on interface
 pub0, Asterisk does nothing with it. A 'rtp debug' shows
 it's receiving the RTP packets, it just seems it does
 nothing with them.

  Anyone seen this?

  Doug.

 I thought all RTP was controlled through rtp.conf and only
 the SIP traffic was controlled through SIP.conf. I am not
 sure what settings, beside the RTP port range, you can out
 into the rtp.conf though.

 Robert
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[Asterisk-Users] Agents and agent counts

2006-03-08 Thread Kevin Smith


Hey everyone,

I have noticed a few questions close to the issue I am having but I 
haven't seen any that quite match the problem I am seeing.


I have 3 queues. Some members share one queue and some are completely 
separate. Some members have a higher penalty then others. I am using 
addqueuememeber and removequeuemember for the login and log out and I 
verify members with their password for voicemail (that all seems to work 
just fine). The problem I am having is that if a member is in a queue on 
their own, everything works fine, a call can go into the queue. However, 
if 2 members with different penalties are logged in on the same queue, 
the test for the number of members in a queue fails. Below is the code 
that is failing.


852,5,Set(Queue_Count_Switch=${IF(${QUEUEAGENTCOUNT(sales)}?7:100)})   
;Checks to see if there are active agents
exten = 
852,6,Goto(Mercury-Sales,852,${Queue_Count_Switch}) 
   ;Sends to closed if there are none
exten = 852,7,Queue(sales|tT|||) 

Here is what the CLI shows for queue members (note: NUMBER1 and NUMBER2 
represent phone numbers that are real. they are different however and 
typed in correctly)


saleshas 0 calls (max unlimited) in 'leastrecent' strategy (0s 
holdtime), W:0, C:0, A:0, SL:0.0% within 0s

  Members:
 SIP/NUMBER1queue with penalty 3 (dynamic) (Not in use) has taken 
no calls yet
 SIP/NUMBER2queue with penalty 2 (dynamic) (Not in use) has taken 
no calls yet

  No Callers

And here is the CLI output.

- Executing Answer(Zap/1-1, ) in new stack
   -- Executing Wait(Zap/1-1, 2) in new stack
   -- Executing Playback(Zap/1-1, mercury-prompts/Sales-welcome) in 
new stack

   -- Playing 'mercury-prompts/Sales-welcome' (language 'en')
   -- Executing Wait(Zap/1-1, 1) in new stack
   -- Executing Set(Zap/1-1, Queue_Count_Switch=100) in new stack
   -- Executing Goto(Zap/1-1, Mercury-Sales|852|100) in new stack
   -- Goto (Mercury-Sales,852,100)
   -- Executing Wait(Zap/1-1, 2) in new stack
   -- Executing Playback(Zap/1-1, mercury-prompts/Sales-afterhours) 
in new stack

   -- Playing 'mercury-prompts/Sales-afterhours' (language 'en')
   -- Channel 0/1, span 1 got hangup request
 == Spawn extension (Mercury-Sales, 852, 101) exited non-zeroexten =
   -- Hungup 'Zap/1-1'

Now what really confuses me is that when only 1 member say, NUMBER1, is 
in the sales queue, it works fine. And vice-versa, but as soon as the 
other member is in, then it stops working. Now even if they are both at 
the same penalty then it still it fails saying we are closed (which is 
exten 852,100). I am at a loss as to what could be causing it. Anyone 
have any ideas or see if something that may be going wrong? Does the IF 
statement return true for anything but 0 and -1 or is it only 1?


Thanks,
Kevin


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Re: [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3

2006-03-08 Thread Kristian Kielhofner


Ken D'Ambrosio wrote:

HTTP's nice, but FTP does the job.  Check the docs for supported FTP
servers -- many of the stock Linux FTP servers will give the exact problem
you discussed, below.  I should know -- took me almost a week before
trying proftpd, and WHAMMO, worked like a champ.

-Ken



Ken,

Proftpd has had more problems than vsftpd:

http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,5082,00.pdf

	Anyways, vsftpd has always worked flawlessly for me and many, many 
others that use AstLinux for this purpose (until Polycom bootrom 3.1.3). 
 I'm actually very interested to find out SPECIFICALLY what happened 
here.  I'll be sure to keep everyone updated.


--
Kristian Kielhofner
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Re: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel

2006-03-08 Thread John Daragon

Sina, hi;

Let's just do a little recap.

You've downloaded zaptel-1.2.4 and done the

make linux26
make install
make config

thing on it.  If you don't uncomment anything, the builds complete
without error and modules are installed in

/lib/modules/`uname -r`/extra.

You've performed the 2.6 kernel udev configuration :


edit /etc/udev/rules.d/50-udev.rules

and insert the lines :

KERNEL=zapctl,NAME=zap/ctl
KERNEL=zapchannel,NAME=zap/channel
KERNEL=zaptimer,  NAME=zap/timer
KERNEL=zappseudo, NAME=zap/pseudo
KERNEL=zap[0-9]*, NAME=zap/%n


Assuming you're using a user called asterisk...

edit

/etc/udev/permissions.d/50-udev.permissions


and insert :

zap/* asterisk:asterisk:660


If running

/etc/init.d/zaptel start

still fails, then  run

/etc/init.d/zaptel stop

and then

sh -x /etc/init.d/zaptel start

You should be able to work out what's failing from the output here. If
you can't, post the output to the list or email it to me.

If, for example, modprobe is failing on ztdummy.ko, then run

strace modprobe ztdummy

and look at the output. This will identify problems like the modules
being in a directory that modprobe isn't looking at, c c.

Again, if the cause isn't clear either post the last (say) 20 lines of
the strace err... trace her or email them to me.

Let's put this one to bed, huh ?

jd
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[Asterisk-Users] Clock is runing too fast, [EMAIL PROTECTED] Ztdummy and VMware workstation

2006-03-08 Thread Marco Mouta

Hi all,

I've [EMAIL PROTECTED] with Ztdummy running on VMWare, and i've adjust
already three times the date and it seems to me it is running clock
faster... After a while Asterisk clock greater than my windows clock
time


Isn't this strange?
I'm just waiting for a Digium card to change this to a real Linux System.

Does any one could help me understanding what is going on?

Best regards,
Marco Mouta
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[Asterisk-Users] Size'ing/performance

2006-03-08 Thread John Jensen

Hi,
Anybody got an idea of how many SIP calls I can run through asterisk on
a Dual 3.6 Xeon (Dell 2850) 
if:
- It doesn't perform any transcoding
- Calls are g.729a
- It doesn't have an interface to reg phone network (MGW in an other
box) ie. everything is SIP to SIP.
- Re-Invite is not allowed (I need the CDR's)
- Uses Realtime (sip extensensions) on db on an other box
- Exports CDR to db on an other box
- Asterisk 1.2.5


How about if I move to a quad 3.6 Xeon (Dell 6850)


Anybody's got a clue ?


Cheers,

John
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[Asterisk-Users] can't call some numbers/providers , ani code missing in sip header (found the problem, not the solution)

2006-03-08 Thread Simone Cittadini

With the help of one of the providers we terminate on, I've found the 
source of the problem of getting busy even when the called isn't really 
busy in the absence of ANI codes in sip headers generated by asterisk.


If I put a NoOp(${CALLINGANI2}) in the dialplan before the dial I can 
see it holds the value '0', but seems that value won't find the way to 
the sip header.


Is this an error for asterisk to not put the code or a misconfiguration 
of the remote switches to drop calls without it ?

(Have I to open a bug or to request a feature ?)


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RE: [Asterisk-Users] pap2 Dial plan

2006-03-08 Thread S McGowan




Your long pause complaint is the 
timeout on the PAP2 before it thinks you're done dialing. The voicemail issue 
sounds like the dialplan on the PAP2, what do you use to connect? if it's a 
star-code (*), you need *#. in the plan to pass *+any 
numbers

SKM


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Giridhar 
  BandiSent: Tuesday, March 07, 2006 12:47 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] pap2 Dial 
  plan
  Hi i am using pap2 phone adaptors as clients to connect to 
  asterisk server i am able to make calls but i cannot access voice mail 
  using phone or start recording while call is in progress and when 
  i place a call to local sip extension there is a long pause ( 15 sec ) 
  before the call gets dialled i assume that the problem would be 
  due to the dial plan in PAP2 if so please help me changing it 
  thanks Giridhar Bandi 
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[Asterisk-Users] Conference room owner Changing his room password? [EMAIL PROTECTED]

2006-03-08 Thread Marco Mouta

Hi all,

I didn't find yet any info about this. Is there any way for a
Conference Room Owner to change his own password? A kind of Menu like
calling his conference room:

example:8200

And an IVR option to change password.


That seems to me interesting, because i may not want the same users
entering two diferent days on my conference room... Also I don't think
it is a good choice to contact Administrator to change my Meetme
password.

Best regards,
Marco Mouta
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Re: [Asterisk-Users] Receiving Multiple calls on asterisk at home

2006-03-08 Thread Sean Cook

To add to the other post... aah or amp actually has a DB that contains
call waiting information.  It may have the default setup such that call
waiting is disabled.  You should be able to dial *70 and enable it.

Sean

On Tue, 2006-03-07 at 11:33 -0700, Rolf Brusletto wrote:
 All - I've been muddling around with this for a few days now.. and I'm 
 trying to figure out why I am not receiving more than one phone call on 
 each polycom 501 phone. I can make more than one phone call out, but not 
 receive another one in, while on a call. Has anybody seen this behaivior 
 before, or is there something simple in the config i'm missing, like.. 
 maxcalls.. or something.
 
 Thanks!
 
 Rolf Brusletto
 
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RE: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970

2006-03-08 Thread Darren Wright

ThanksI've got the SEPMAC files that I use successfully with SCCP.

-D


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Oliver
Sent: Tuesday, March 07, 2006 4:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco
7970

On Mon, 2006-03-06 at 22:11 -0500, Darren Wright wrote:
 InterestingI've upgraded the 7970 to SIP, but it is still saying
 unprovisioned.  I've got a SIPMAC file, but it is still looking for
the
 SEPMAC file...
 

That's correct - the CCM5 loads only look for SEP files.  Even when you
give it one, it will not register with Asterisk.  If you need a fully
formatted SEPxml file, I will email you one off line for a 70.

 
 Anyone got this working yet?

Nope :(

 -D
 
 
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[Asterisk-Users] Mitel SX-2000 and Asterisk integration

2006-03-08 Thread Richard OSS

Hello,Somebody has managed to make Mitel SX-2000 and Asterisk integration work.  http://www.voip-info.org/wiki-Asterisk+legacy+integrationCan you please post your zaptel.conf and zapata.conf for T1/PRI config?I will be configuring a TE210P to connect to an SX-2000 PBX.Thanks.richard  ___
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[Asterisk-Users] [Slightly OT] Does TE110P (a 32-bit PCI) fit into PCIe x8 slot?

2006-03-08 Thread Josip Gracin


Hello!

Does TE110P (a 32-bit PCI) fit into PCI Express x8 slot?  I'm thinking 
of buying a Sun X2100 and it has a PCI Express x8 slot.


Or perhaps, does Digium produce PCI Express E1 cards?

Thanks in advance!
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[Asterisk-Users] parking slot lights - testers wanted

2006-03-08 Thread Dr. Michael J. Chudobiak


Hi all,

The metermaid patch allows you to use the programmable buttons and 
LEDs on phones (like the Snom 2xx or 3xx) to view the status of parking 
slots and transfer to them. This should be really useful for 
small-office environments.


Anyway, the patch seems to work with Snom phones (and hopefully others) 
now! The curious are encouraged to download the metermaid-v3.txt patch 
against v1.2.4 for testing and feedback! See 
http://bugs.digium.com/view.php?id=5779 for details.


- Mike

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[Asterisk-Users] Asterisk sip and radius authentication

2006-03-08 Thread Sergio Iñigo Ibáñez









Hello all,



I am new in asterisk configuration. I want to configure a Radius server
to authenticate the sip users of asterisk. I have trying to use the next
document:



http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html





Can you help me?



Regards,



Sergio








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[Asterisk-Users] setmusiconhold doesn't work between 2 SIP phones

2006-03-08 Thread Joseph Rothstein

I have the exact same problem. SetMusicOnHold between two sip phones always
returns the default class.

Any ideas?

Joe


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Re: [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?

2006-03-08 Thread Wireless

Does anyone have this working on 1800MHz eg TMobile or Orange in the UK

and does CLID work or not?

THanks

Harvey
- Original Message - 
From: Conrad Wood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, February 17, 2006 1:01 AM
Subject: Re: [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom
?

 On Thu, 2006-02-16 at 23:39 +0100, adibar wrote:
  Hi List

  Is someone out there using one or more GSMgateway(s) from CyberTelecom ?
  Me and some friends are interested in buying some of them, but before
  we would like to ask, how the experiences are others have made.

  e.g.
  How easy to setup ?
  How reliable ?
  How's the voice quality ?
  etc.

 I use the analogue version.
 it was very easy to setup, essentially plug sim in and go. voice quality
 is good. Delivery was prompt.
 one caveat I found though: It doesn't seem to work with T-Mobile in UK.
 Linus Surgus on asterisk-biz suggested it might only be working on
 900Mhz instead of - as advertised - 1800 Mhz and 900 Mhz. I am going to
 try an orange simcard next, because orange also uses 1800 (like
 t-mobile). Because of this I cannot comment on the reliabilty yet.

 -- conrad

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[Asterisk-Users] Asterisk @ Home 2.6 Call hangs up

2006-03-08 Thread Jerry Rasmussen

I have installed asterisk @ home 2.6. I am using a 
Telasip VOIP account. When I make outbound or inbound calls the calls seem 
to connect and then get hung up. I was wondering if there was something 
that I am misisng. I have tried several different sip.conf 
configurations. Here is what they are currently.

telasip-gw
canreinvite=yescontext=telasip-indtmfmode=rfc2833fromuser=jrasxxxhost=gw4.telasip.cominsecure=verynat=yessecret=xyztype=peerusername=jrasxxx

551212
canreinvite=yescontext=from-pstndtmfmode=rfc2833host=gw4.telasip.cominsecure=verynat=yesqualify=yessecret=xyztype=peerusername=jrasxxx

The odd thing is it worked once or twice then stopped. If anyone 
could shed some light it would be greatly apperciated.


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RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

I can't be bothered looking for the link right now, but it's definitely stated 
somewhere on Digium's website.

-Original Message-
From: Alexander Lopez [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 07, 2006 3:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Oh this is bad bindaddr and rtp
traffic

To retort, Digium has ever to my knowledge, stamped an 'Enterprise
Grade' mark on the product.  If you are worried about a single point of
failure you may want to replace your toaster.

Asterisk is missing a 'few features' no doubt about it, but it is open
source, it will be a welcome addition if you would like to add
multi-homing support in, might as well do media multi-homing with call
diversity. This will definably be a non-trivial re-architecture of the
core.

The 'missing a few features' way of thinking is what has made Asterisk
what it is today.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Douglas Garstang
 Sent: Tuesday, March 07, 2006 11:46 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Oh this is bad bindaddr and rtp
traffic

 Pardon my candour, but for a product Digium calls 'enterprise grade'
it
 sure seems to be missing a few features.

 -Original Message-
 From: Alexander Lopez [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, March 07, 2006 9:39 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Oh this is bad bindaddr and rtp
 traffic

 Asterisk does not like multiple interfaces in the way you are
configured.
 You can either:

 A) use the bindaddr in the sip.conf to limit where the packsge come
and
 go.

 B) use an outside traffic manager

 Look up the archives, kpf explained why this would not work, as
asterisk
 can't do load balancing at this time

 -Original Message-
 From: Robert Webb [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-
 [EMAIL PROTECTED]
 Sent: 3/7/06 11:27 AM
 Subject: Re: [Asterisk-Users] Oh this is bad bindaddr and rtp
traffic

 On Tue, 7 Mar 2006 09:12:25 -0700
   Douglas Garstang [EMAIL PROTECTED] wrote:
  I have a configuration where RTP traffic is going out
 interface pub0, and coming back into through pub1.
  I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an
 shows:

  udp0788 0.0.0.0:50600.0.0.0:*

  which means that Asterisk is listening on all addresses
 (on all interfaces?).

  Anyway, when the RTP traffic comes back in on interface
 pub0, Asterisk does nothing with it. A 'rtp debug' shows
 it's receiving the RTP packets, it just seems it does
 nothing with them.

  Anyone seen this?

  Doug.

 I thought all RTP was controlled through rtp.conf and only
 the SIP traffic was controlled through SIP.conf. I am not
 sure what settings, beside the RTP port range, you can out
 into the rtp.conf though.

 Robert
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[Asterisk-Users] status on jitter buffer for SIP/RTP? (OT?)

2006-03-08 Thread Adam Moffett

This might be a better question for the dev list, but does anyone know 
the status of a jitter buffer for SIP channels?


I know they created a generic jitter buffer and implemented it for IAX 
channels.  Does it work yet for SIP?  Like is it there and disabled or 
not there at all?


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[Asterisk-Users] Chinaroby VOIP phones?

2006-03-08 Thread Darko Sundek




Hi all,

Do anyone have experience www.Chinaroby.com VOIP 
phones?
I am very interestedfor models:PY-60 and PB-35 
Phones.
Good or bad 
experience with sip and IAX2, please comment.

Regards

Darko 
Sundek
eLink 
Group
Kotor-Montenegro




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[Asterisk-Users] Real Time Asterisk

2006-03-08 Thread Fernando Lujan


Hi guys,

I want to setup a environment where asterisk load all information from a 
Postgresql database. So here goes my questions:


1) Is real time asterisk  stable enough?
2) Where can I found documentation about using it with Postgresql? ( 
including meet me conferences)


Thanks in advance.

Fernando Lujan
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Re: [Asterisk-Users] MWI, SER and asterisk

2006-03-08 Thread David Thomas

There is a patch to chan_sip on voip-info.org that I use. It seems to
work very well. I believe it is on the Astrisk at large page on the
voip-info.org wiki.

regards,
Darvid

On 3/7/06, Sharon [EMAIL PROTECTED] wrote:
 I have my peers registered to SER.asterisk seems to be sending mwi for
 the peers seen in the sip show peers CLI command. i have my ser server
 registered with asterisk as a type=friend and all clients register to
 ser.how do i get mwi to work for these clients registered to SER.

 Thank you,
 -AA
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[Asterisk-Users] Is everyone getting mails except me?

2006-03-08 Thread Ron McCarthy

I havent got any mails since 2:42 this morning..usually i get at least
the normal 10-15 a hour, if someone gets this can they reply?

Thanks!
Ron
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[Asterisk-Users] Cisco Call Manager SIP trunk + Asterisk

2006-03-08 Thread Chris HARIGA









Hi,



I setup a SIP trunk in a brand new Cisco Call Manager and I
try to place the calls using Asterisk but I get error:



-- SIP read from 192.168.11.10:5060:

SIP/2.0 400 Bad Request - 'Malformed/Missing URL'

Via: SIP/2.0/UDP
192.168.10.199:5060;branch=z9hG4bK2e7ca9c9;rport

From: asterisk
sip:[EMAIL PROTECTED];tag=as56c7728f

To: sip:192.168.11.10

Call-ID: [EMAIL PROTECTED]

CSeq: 102 OPTIONS

Content-Length: 0



Question: How I can setup asterisk to get the sip call
without authentication? I check on voip-info.org but I didnt find a
sip.conf sample L



Best regards,



Chris HARIGA








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[Asterisk-Users] pickup last ringing phone

2006-03-08 Thread erkan kolemen

Hello,I am using pickup, i can pickup an extension from outside of the queue, but i cannot pickup any call comes to queue.queue strategy=ringallWhat is the problem with queue?Is there anyway to pickup last ringing phone?-erkaN
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Re: [Asterisk-Users] grandstream handytone 286 sometimes dials out wrong number

2006-03-08 Thread Giorgio Incantalupo


Hi Martin,

I have 3 choices on my ATA webpage and I chose SIP INFO:
/Send DTMF: / in-audio via RTP (RFC2833) via SIP INFO

This is the only point I can make changes since it is connected to my 
asterisk box through a TDM400P:
asterisk box ---TDM400P -(telephone cable)- HT-288 --- LAN --- 
Internet --- Messagenet VoIP provider


We examined Messagenet provider logs and, I do not why, we lose 1 call 
on 30 made...our customer loses 1 call on 2 (50%).

We think it is the ATA sending bad DTMF sometime.
Seems strange anybody else but me hadn't had problems like this...I 
found nothing on internet...



TIA

Giorgio Incantalupo


Martin Joseph wrote:


On Mar 6, 2006, at 6:46 AM, Giorgio Incantalupo wrote:


Hi,
I have an asterisk 1.2.1 (on a debian sarge) box with a TDM400P card. 
I connected the TDM400P to a grandstream 286 to use a VoIP provider.
It seems all right except for a little problem: one call every 30 is 
made to a wrong number.

Is there anybody who had the same problem and solved it?

Usually this is DTMF issue? So make sure the extensions and the HT286 
have the correct DTMF config. I have some experience with the HT-488 
FXS and that needed to have dtmfmode=rfc2833 in the extensions and the 
configuration on the HT-488 set the same.


Hope this helps,
Marty

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[Asterisk-Users] Location of MeetMe Recordings

2006-03-08 Thread Gavin Adams

In Asterisk 1.2.4 is love being able to recording conferences. However,
using the default variables, the files are being written to
/var/lib/asterisk/sounds instead of /var/spool/asterisk/meetme.

If I change MEETME_RECORDINGFILE variable to something different in works,
bit I lose the ability to define CONFNO as part of the file name, which is
handy when sorting for users to review. I call meetme using (,r,) so the
conference number is not defined yet.

My /etc/asterisk/asterisk.conf file is set to point to /var/spool/asterisk
for recording related bits, and voicemail and general recordings are being
stored in the appropriate subdirectories. It's only meetme that is going to
a different place.


Regards,

--- Gavin Adams
VP Operations
PARC Inc.

E-mail: [EMAIL PROTECTED]
Office: +1 678.281.6402
   Fax: +1 678.281.6401
Mobile: +1 404.933.8183
 Skype: gadams999

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RES: [Asterisk-Users] pap2 Dial plan

2006-03-08 Thread Filipe Mordhorst









Youre almost right.

The PAP2 has some features
that are factory default. I dont remember the section in the web
interface, but heres what you going to do:



Find the section that
contains a lot of features name with values like this *56 or *78.

Erase all of them. Letting
this filled youll not be able to implement your asterisk features, cause
they are conflicting with the (factory defaults) PAP2 commands.



About the long time
waiting for start to call, the problem is that the PAP2 waits 10 or 15 (I dont
remember de default) seconds after a digit is pressed to start the send
procedure.

To change these settings,
go to Regional/Control Timer Values/ Interdigit Long Times and change the value
to any other (this is expressed in seconds).



Hope it helps.





Regards,





Filipe Mordhorst

Brazil-SC













De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Giridhar Bandi
Enviada em: terça-feira, 7 de
março de 2006 14:47
Para:
asterisk-users@lists.digium.com
Assunto: [Asterisk-Users] pap2
Dial plan





Hi 

i am using pap2 phone adaptors as clients to connect to asterisk server 
i am able to make calls but i cannot access voice mail using phone 
or start recording while call is in progress 

and when i place a call to local sip extension there is a long pause ( 15 sec )

before the call gets dialled 

i assume that the problem would be due to the dial plan in PAP2 

if so please help me changing it 

thanks 
Giridhar Bandi 








smime.p7s
Description: S/MIME cryptographic signature
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[Asterisk-Users] PAP2 won't make two g729 calls at the same time

2006-03-08 Thread Warren Burstein

I have a Linksys PAP2.  Identical setups for the two channels in both 
the unit and in Asterisk.  In particular, both channels enable g729 and 
set it as the preferred codec, and have disallow=all and allow=g729 in 
sip.conf.


If we make a call on one channel, it works (and uses g729), but if we 
make a call on the other channel when the first one is still connected, 
it fails.  We have three g729 licenses, and no others were in use at the 
times this happened, but even if we didn't have enough, how would the 
PAP2 know that?


Here's a good, and a bad INVITE message, from the log file with sip 
debug enabled.  Has anyone seen anything like this?


INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3e521aa
From: PAP 220 sip:[EMAIL PROTECTED];tag=6b66e68deef168b2o0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: PAP 220 sip:[EMAIL PROTECTED]:5060
Expires: 240
User-Agent: Linksys/PAP2-3.1.3(LS)
Content-Length: 246
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 261305180 261305180 IN IP4 192.168.254.44
s=-
c=IN IP4 192.168.254.44
t=0 0
m=audio 16392 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3fda1e15
From: PAP 220 sip:[EMAIL PROTECTED];tag=b8b86be991749af5o0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: PAP 220 sip:[EMAIL PROTECTED]:5060
Expires: 240
User-Agent: Linksys/PAP2-3.1.3(LS)
Content-Length: 267
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 261589835 261589835 IN IP4 192.168.254.44
s=-
c=IN IP4 192.168.254.44
t=0 0
m=audio 16400 RTP/AVP 0 8 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv




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[Asterisk-Users] Upgrading Asterisk witk G729 license installed

2006-03-08 Thread Álvaro Palma

I've an Asterisk 1.2.4 installation, where I've also installed the G729 
codec license. I'd like to upgrade that installation to 1.2.5, but I'm 
not sure if I'll lost the license in the process (and if I'll be able to 
recover it later!!!).


Is there any special consideration I've to keep in mind in this case, or 
should I just run the typical make + make install and it will take 
care of keeping the license information?


Thanks a lot for your attention.

--
Atly.
Alvaro Palma

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RES: [Asterisk-Users] Inserting access codes as prefixes to CID

2006-03-08 Thread Filipe Mordhorst









Theres the
SetCallerID cmd that you should read about.



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



It has others links to clarify
your ideas.



Tell us if you get
something.









Filipe Mordhorst
Brazil-SC











De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de AR Tarzi
Enviada em: domingo, 5 de março de
2006 12:58
Para: Asterisk Users Mailing List
- Non-Commercial Discussion
Assunto: [Asterisk-Users]
Inserting access codes as prefixes to CID







When I receive a call from fwd, I'd like to insert a prefix prior to
the caller ID - 1) to be able to look it up in a database ofidentified
numbers and 2) for the receiver to be able to dial it back.





So what I need is to identify the DID and based on that, insert the
prefix.











Any pointers to documentation would be appreciated
















smime.p7s
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[Asterisk-Users] What port mpg123 uses for MoH?

2006-03-08 Thread Zach A

Hi,

What port does mpg123 uses to play music on when it starts MoH after
asterisk has put called on hold?

Zach A

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[Asterisk-Users] Putting caller in queue and dialing an extension simultaneously

2006-03-08 Thread Zach A









Hi,



Is it possible to do this in extensions.conf to put a caller
in queue and dial an agents extension so that he knows that somebody is
in queue waiting to be answered. This agent will be a remote agent and
extension will dial his cell phone.



Thanks



Zach A.








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Re: [Asterisk-Users] Changing REINVITE status of the channel dynamically

2006-03-08 Thread Luki

 I'd like to know if it's possible to set the REINVITE on or off dynamically,
 based on the extension being dialed.

Define two peers in sip.conf, one with canreinvite=yes and the second
with canreinvite=no. Then you can route your calls with or without
reinvites depending on the dialed number. Like:

[provider-reinvite]
type=peer
host=external_sip_server.com
canreinvite=yes
...

[provider-noreinvite]
trype=peer
host=external_sip_server.com
canreinvite=no
...

exten = _1[0-4]X0.,1,Dial(SIP/{EXTEN:[EMAIL PROTECTED])
exten = _1[5-9]X0.,1,Dial(SIP/{EXTEN:[EMAIL PROTECTED])

--Luki
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RE: [Asterisk-Users] Receiving Multiple calls on asterisk at home

2006-03-08 Thread Jeff Herring

Do you have the phone specific config file for the polycom set to something
like this?

?xml version=1.0 encoding=UTF-8 standalone=yes?
phone1
  reg reg.1.displayName=default reg.1.address=27 reg.1.label=27
reg.1.type=private reg.1.auth.userId=27 reg.1.auth.password=
reg.1.lineKeys=3/
  msg msg.bypassInstantMessage=1
  mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact
msg.mwi.1.callBack=*97/
  /msg
/phone1


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rolf
Brusletto
Sent: Tuesday, March 07, 2006 1:34 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Receiving Multiple calls on asterisk at home


All - I've been muddling around with this for a few days now.. and I'm
trying to figure out why I am not receiving more than one phone call on
each polycom 501 phone. I can make more than one phone call out, but not
receive another one in, while on a call. Has anybody seen this behaivior
before, or is there something simple in the config i'm missing, like..
maxcalls.. or something.

Thanks!

Rolf Brusletto

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[Asterisk-Users] impact of qualify=yes

2006-03-08 Thread Damon Estep









Anyone have any information on the performance impact of
using qualify=yes for hundreds (500ish) of SIP UAs?



I have seen tidbits on qualifyspreading=yes, but not enough
to understand what it does. I assume lessens the peak load of qualify sip
options queries?



Thx!






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RE: [Asterisk-Users] Receiving Multiple calls on asterisk at home

2006-03-08 Thread Jeff Herring

Do you have call-limit parameter set to 3 in sip.conf or possibly
sip_additional.conf on AAH?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rolf
Brusletto
Sent: Tuesday, March 07, 2006 1:34 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Receiving Multiple calls on asterisk at home

All - I've been muddling around with this for a few days now.. and I'm
trying to figure out why I am not receiving more than one phone call on
each polycom 501 phone. I can make more than one phone call out, but not
receive another one in, while on a call. Has anybody seen this behaivior
before, or is there something simple in the config i'm missing, like..
maxcalls.. or something.

Thanks!

Rolf Brusletto

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[Asterisk-Users] Memory Problems

2006-03-08 Thread Dumpolid Exeplish

Hello,This is not a question directly related to asterisk.I am currently rinning ansterisk on a Debian server and i just upgraded my memory from 1GB to 2GB. However, my linux OS does not recognise the memory upgrade. The BIOS does, but the Debian Linux refuses to use the entier memory, currently, it registered only 900MB.
Can anyone tell me why thi is and a solution to this??My Debian version is Linux asterisk 2.6.12.3 #1 Mon Aug 1 19:33:51 WAT 2005 i686 GNU/LinuxThe server is currently routing calls from SIP internal users through an E1 card (TE410)
OUTPUT FROM dmesg command009dc00 (usable)BIOS-e820: 0009dc00 - 000a (reserved)BIOS-e820: 000f - 0010 (reserved)BIOS-e820: 0010 - 7fee (usable)
BIOS-e820: 7fee - 7fee3000 (ACPI NVS)BIOS-e820: 7fee3000 - 7fef (ACPI data)BIOS-e820: 7fef - 7ff0 (reserved)BIOS-e820: fec0 - 0001 (reserved)
Warning only 896MB will be used.Use a HIGHMEM enabled kernel.896MB LOWMEM available.found SMP MP-table at 000f5a20On node 0 totalpages: 229376 DMA zone: 4096 pages, LIFO batch:1
 Normal zone: 225280 pages, LIFO batch:31 HighMem zone: 0 pages, LIFO batch:1-END
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Re: [Asterisk-Users] Polling Asterisk for Life

2006-03-08 Thread Matt

  Are you guys perchance using Local/[EMAIL PROTECTED] in your installations?
 
  --
  Cheers,
 
  Matt Riddell
  ___
 

 Is there a known issue when using the Local/[EMAIL PROTECTED]

 thanks,

This is how I would read it.. but yes.. can someone give more
information on this apparently huge bug!
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[Asterisk-Users] No DTMF

Some one was on my server making changes to my sip.conf files. I am now having trouble with DTMF. No matter what I use (inband,auto,rfc2833) the dtmf tones seem to not come thru. I compared it to the wiki and all the configs seem to be in order.Here is my sip.conf[general]disallow=all;allow=g729 ; requires license for g729allow=ulawport = 5060nat=yescontext=from-sipbindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)maxexpirey=4800 ; Maximum expiration for registrationsdefaultexpirey=1800 ; Default expiration for registrationscanreinvite=no ; Allow clients to directly connect if set to yes. Set to no if behind NAT.tos=reliabilitysrvlookup=yes ; Enable DNS SRV lookups on outbound callsvideosupport=no ; Turn on support for SIP videodtmfmode=rfc2833 ;rfc2833 ;inband ;rfc2833 ; DTMF inband need to be set
 here.pedantic=noexternip=..XXX  ;Sip Mediaregister = XX:[EMAIL PROTECTED]/7322761368  [sipmedia6]type=frienduser=XX ;(Phone Number)username=XX ;(Phone Number)fromuser=XX ;(Phone Number)authname=XX ;(Phone Number)secret= ;(SIP Password)host=sip.sipmedia.com disallow=allallow=ulawcontext=ServerHighwayrealm=sip1.xchangetele.comfromdomain=sip.sipmedia.comdtmfmode=rfc2833canreinvite=no insecure=veryHere is my extensions.conf  [general]static=yeswriteprotect=yes  [ServerHighway];Play Server Highway IVR  Exten = s,1,Background(server-highway-ivr)Exten = s,2,Background(blank-file-10)  Exten = 1,1,Ringing()Exten =
 1,2,Wait(15)Exten = 1,3,Macro(stdexten,9511,9511)Exten = 2,1,Ringing()Exten = 2,2,Wait(15)Exten = 2,3,Macro(stdexten,9512,9512)Exten = 3,1,Ringing()Exten = 3,2,Wait(15)Exten = 3,3,Macro(stdexten,9513,9513)Exten = 4,1,Ringing()Exten = 4,2,Wait(15)Exten = 4,3,Macro(stdexten,9514,9514)Exten = i,1,Background(invalid)Exten = i,2,Goto(s,1)  Exten = t,1,Goto(s,1)  exten = 9,1,Goto(s,1);Extension To Record Main IVR Messageexten = 500,1,Authenticate(XXX)exten = 500,2,Record(ServerHighwayIvr:gsm)
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Re: [Asterisk-Users] Setting Vaaibles

Figured it out. It was simple had to add Answer and hangupDovid Bender [EMAIL PROTECTED] wrote: Helo List,First I would like to apologize for my bad spelling aswell as that I did not search the wiki first. I onlyhave email access at the moment.I am having trouble setting both variables and globalvariables thru an extension.I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4with an Xlite softphone. I have two xlite phones ondiffent computers. One logs in as xlite1 and the otheras SNOM.My dial plan is as followsExten = 200,1,Dial(${OnCall},10)Exten = 201,1,Set(>Exten = 202,1,Set(>(I have tried Set and SetGlobalVar).When I use Set I get the following in the CLI-- Executing Set("SIP/snom-a6
45",
">in new stack== Auto fallthrough, cahnnel 'SIP/snom\a645 status is'UNKNOWN'If I dial ext. 201 or 202 I get call failed: 603declined on the xlite phone. When I dail 200 I get anerrorIf I use SetGlobalVar the output from the CLI is-- Executing SetGlobalVar("SIP/snom-24f8"," in new stack= Setting global variable 'OnCall' to 'SIP/SNOM'== Auto fallthrough, channel 'SIP/snom-24f8' status is'UNKNOWN'When I use SetGlobalVar I get the same error in thexlite phone. However when I dial ext. 200 it works.I tried dialing 201 and 202 from both softphones and Igot the same errors.Thanks a lot.Dovid __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easyn
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[Asterisk-Users] System Design

2006-03-08 Thread Jason Adams





Hey 
Everyone,

We are in the works 
of planning a new * installation for our company. We have 20 users in our 
main office and 5 users in a remote office a couple of states away. Our 
call volume for the main office will be anywhere from 5-10 concurrent 
calls. The remote office will have about 3 heavy users with two users 
making calls occasionally.

Right now we have an 
existing PBX. We have a T-1/PRI coming into the main office and a DSL 
connection at the remote office. We have a Cisco 2610/PIX 501 at the main 
office a cheesy linksys router at the remote site.

We are planning on 
purchasing new Cisco IP phones for everyone.

My main question is 
this: What type of hardware/network design would be best for this 
situation? Would a full T-1 at the remote site work with a VPN between the 
offices? Or would a higher bandwidth DSL work with a VPN? Or should 
we move to a Point-to-Point connection? What type of hardware would be 
best for the end-to-end communication in regards to QoS? I know the PIX 
501 doesn't support it.
Would it be best to 
have two * servers in each office or for that call volume at the remote office 
does it make sense? I was thinking of a Dell Power Edge server with 4GB of 
ram and a dual processor.. is that enough?

Sorry for all the 
questions!

Jason 
AdamsSumo Systems 57 E. Wilson Bridge 
RdSuite 
200Worthington, 
OH 
43085 
Phone | 614.433.9906 
ext: 102Fax | 
614.433.9931 
E-mail | [EMAIL PROTECTED] 

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[Asterisk-Users] Softphone for Windows CE 3.0

2006-03-08 Thread Matt

Hi,
I've found several softphones for Windows Mobile 2003, but does anyone
know of a softphone (or older version of a current softphone) that
will run on Windows CE 3.0?

~ Matt
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[Asterisk-Users] Faxing with MFC/r2

2006-03-08 Thread Carlos Chavez

 I am having a problem when trying to send a receive faxes on an E1
running with unicall on an asterisk 1.2.4 x64 server.  The same server has a
TDM02 card and if I send or receive faxes through there there is usually no
problem.  I am afraid that my customer insists that he wants to use the DID on
the E1 for faxes so I need to fix this. 

 The fax is connected to a Linksys PAP2 adapter but I have also tried
rxfax and I get the same results when trying to use the E1 connection.  Is
there a setting or modification that can be done to unicall?

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

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Re: [Asterisk-Users] HW Echo Cancellers

2006-03-08 Thread Matt

  Tellabs looks a little too up-scale for what I need :). $1k for a
  single port orion unit might be worth considering for really stubborn
  installs though.
 

 Why? they go for around $100.00 on eBay.

What goes for $100 on eBay?  Tellabs?  or Orion?  I can't find any
Orion equipment on eBay.  What model Tellabs am I looking for?
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[Asterisk-Users] Zap not installing

2006-03-08 Thread Curt Shaffer









I have decided to move on from [EMAIL PROTECTED] and start
compiling asterisk myself now. I got a dedicated box and put my X100P in it. I
installed the server version of CentOS 4.2 (2.6.9-22.EL kernel) barebones. The
box is a dell GX270 workstation with 1GB of RAM. I got a fresh copy of
OReillys Asterisk the future of technology and begun. I
downloaded the zaptel-1.2.4.tar.gz, libpri-1.0.9, and asterisk-1.2.5. I started
compiling the zaptel (make  make install  make clean) when
I try to start zaptel - /etc/init.d/zaptel start I get the following error:



Loading zaptel framework: FATAL: Module zaptel not
found 

Unable to open /dev/zap/ctl: No such file or directory



Below are the only things I have declared in my
/etc/zaptel.conf



ks=1

loadzone=us

defaultzone=us

fxoks=1 ( I have tried fxsks=1 as well, because the book had
a section that read the following):



...a physical FXO port will be defined in
configuration with FXS signaling..an FXO card connects to a central office(CO),
which means it will need to behave like a station that use FXS signaling



I tried this both in /etc/udev/rules.d/50-udev.rules and
/etc/udev/rules.d/zaptel.rules (rebooting after each change)



Zaptel devices

KERNEL=zapctl,
NAME=zap/ctl

KERNEL=zaptimer,
NAME=zap/timer

KERNEL=zapchannel, NAME=zap/channel

KERNEL=zappseudo,
NAME=zap/pseudo

KERNEL=zap[0-9]*, NAME=zap/%n



When I run ztcfg I get the following error:



line 0: Unable to open master device '/dev/zap/ctl'



When I run zttool I get the following error:



Unable to open /dev/zap/ctl: No such file or directory



I have started from scratch multiple times and I get the
same result. 



I get no errors when compiling and the card can be removed
and put back in the old system and work properly. Also Linux does notice the
device when I install and boot into the OS.



Any help would be appreciated.



Curt


















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Re: [Asterisk-Users] Calls between Asterisk servers using SIP?What about IAX (got it working w/ IAX but I have questions)

2006-03-08 Thread Gabriel Afana




Does anybody have any experience with capabilities 
here? I need to know if IAX is able to handle more than that. I 
think I might just benchmark this with a bunch of .call files between servers to 
see how they are handled.

Any input?

- Gabriel Afana


  - Original Message - 
  From: 
  Umair Bari 
  
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Tuesday, March 07, 2006 3:30 
  AM
  Subject: Re: [Asterisk-Users] Calls 
  between Asterisk servers using SIP?What about IAX (got it working w/ IAX but I 
  have questions)
  
  Hello Gabriel,
  
  IMHO, using IAX between * servers is a good choice, I dont see any 
  problem in it. Actually I used it for sometime and never encounter any issue, 
  but i had max 5 concurrent connections.
  regards,
  
  Umair bari
  On 3/7/06, Gabriel 
  Afana [EMAIL PROTECTED] 
  wrote: 
  Hi 
everyone, I just spend the last two hours trying to get two 
asterisk boxes totransfer calls between eachother using 
SIP.I dont know why but I *could not* get the calls to 
authenticate!I think I got everything setup. 
There was Server A and Server B.I was trying to place a call 
from ausers registered on Server A to a user regsitered on Server 
B.I setup the registration info for Server A and even had 
Server A registeringsuccessfully to Server B.However, 
whenever I would hand off the calls fromserver A to Server B, it would 
*always* say it failed to authenticate (passwords did not 
match).Here was my setup:SERVER A:register = serga:[EMAIL PROTECTED][to_80]username=sergatype=friendsecret=test 
host=216.152.244.81disallow=allallow=ulawuser=phoneusereqphone=yescanreinvite=yesregseconds=0cancallforward=yesdtmfmode=rfc2833disallow=allallow=ulaw 
insecure=verytrunk=yesSERVER 
B:[serga]type=friendusername=sergatrunk=yesnotransfer=yessecret=testcontext=302host=dynamicqualify=yesDIALPLAN 
ON SERVER A: exten = 302,1,Dial(SIP/to_80/[EMAIL PROTECTED],30,r)It 
always says authentication failed.However I always noticed it 
showedthe user as [EMAIL PROTECTED].This is 
the extension of the phone I am calling from.It seems it is 
trying to authenticate the actual phone I amcalling from on Server A, 
and not Server A itself.Was I doing somethingwrong?I 
tried doing this with IAX and within 5 minutes I had it all 
working!!I feel it was too easy :-) However, 
this brings up a big question.IsIAX very reliable for 
this?I've heard from people that I should not useIAX under 
any condition because it really is not 
veryreliable/thourough/consistant...etc.I am trying to start 
a VOBB company and will obviosly need a reliable setup.I am 
thinking to have all phonesregister to the servers via SIP and maybe 
just have all the servers transfercalls between eachother via 
IAX.Does this sound like a correct setup? - 
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[Asterisk-Users] Cisco 7960 SIP - Displaying Time

2006-03-08 Thread Ben Blakely









Is there a way to display the time of the 7960 running
firmware 7.4? Im unable to find any information.



Thanks,



Ben Blakely






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[Asterisk-Users] More 7940 Questions

2006-03-08 Thread Aaron Daniel

Does anyone know why putting an outbound proxy in the SIPmac.cnf file 
causes the phone to not pull it's logo from logo_url?


Aaron
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[Asterisk-Users] Calls forwarding to numbers only in user's context

2006-03-08 Thread Bartosz Piec


Hello,

I'm trying to do call forwarding based on this: 
http://www.voip-info.org/wiki/view/Asterisk+call+forwarding


In the extensions.conf file I have several context defined (local, 
longdistance, mobile, international and so on). Each user can be 
associated with different context (so can make only i.e. local calls). 
How to set calls forwarding only to numbers that are available in user's 
context (so if he has only locals calls he cannot set calls forwarding 
for mobile phones)?


I'm using this for forwarding:

[forwarding] ; available for all users
; Unconditional Call Forward
exten = _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})
exten = _*21*X.,2,Background(auth-thankyou)
exten = _*21*X.,3,Hangup
exten = #21#,1,DBdel(CFIM/${CALLERIDNUM})
exten = #21#,2,Background(auth-thankyou)
exten = #21#,3,Hangup

; Call Forward on Busy or Unavailable
exten = _*61*X.,1,DBput(CFBS/${CALLERIDNUM}=${EXTEN:4})
exten = _*61*X.,2,Background(auth-thankyou)
exten = _*61*X.,3,Hangup
exten = #61#,1,DBdel(CFBS/${CALLERIDNUM})
exten = #61#,2,Background(auth-thankyou)
exten = #61#,3,Hangup

[macro-call-forwarding]
exten = s,1,Set(temp=${DB(CFIM/${ARG1})})
exten = s,n,GotoIf(${temp}?cfim:nocfim)
exten = s,n(cfim),Dial(SIP/[EMAIL PROTECTED])   ; Unconditional forward
exten = s,n(nocfim),NoOp

exten = s,n,Dial(SIP/${ARG1},20,tTwW)

exten = s,n,Set(temp=${DB(CFBS/${ARG1})})
exten = s,n,GotoIf(${temp}?cfbs:nocfbs)
exten = s,n(cfbs),Dial(SIP/[EMAIL PROTECTED]) ; Forward on busy or unavailable
exten = s,n(nocfbs),Goto(s-${DIALSTATUS},1) ; 
NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER


...

[incoming]
;
; Incoming calls.
;

exten = XYY,1,Macro(call-forwarding,YY)

--
Best regards,
Bartosz Piec
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[Asterisk-Users] List Problems

Is anyone with a yahoo account having problems
recieving emails from the list. I have not recieved
any emails in about 8 hours and I posted something
about 3 hours ago. If anyone knows please email to
asteriskdigium _AT_ yahoo.com

Thanks

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[Asterisk-Users] Any way to change dns timeout value? Asterisk hangs if internet unreachable

2006-03-08 Thread Joseph Tanner

I don't have the most reliable internet connection in the world. 
Whenever it goes out, I can't receive any incoming calls at all, not
even from pstn.  When it first goes out I can still make outgoing
calls through pstn, but eventually that fails too (as does voicemail,
everything's out).  Yes, asterisk and the local phones are all on the
same network and can communicate fine.

Ok, that's the symptom, and I believe I know what's causing it. 
Asterisk seems to be hanging on dns lookups.  After a while, it gets
so bad that it won't process anything at all.  The reason incoming
calls via pstn won't work is because I have a calleridname.agi script
that runs as soon as a call comes in.  Instead of trying for say, 5
seconds and then giving up, asterisk just sits there forever waiting
for it to resolve.  Once asterisk gives up, the caller has hung up
ages ago.  Obviously, I don't want pstn calls to be dependent on my
internet connection, kinda defeats having a pstn line at all.

Now, as soon as the internet connection craps out, I can still make
outgoing calls via pstn, access voicemail, etc.  If it's a long outage
(like this morning, some fiber cut and the whole county is without
internet, redundancy anyone?), eventually everything stops.  I think
it's because asterisk is re-trying to register with a host, before the
dns timed out, and the built-up dns queries just bring the whole thing
to a halt eventually.  This morning after I noticed the internet
connection was down, I tried to call the phone company (through the
pstn line) and could not.  When I watched the CLI, I noticed it try to
call a minute or two after I hung up, quite a delayed reaction.  Also
could not access voicemail.  When the connection came back up for a
minute and crapped back out again, I was suddenly able to access
voicemail and make a call.  Shortly after that, I'd dial a number and
it'd connect after 10 seconds or so.  After that, it wouldn't try to
connect until after the phone received a fast busy.

A workaround was to backup my sip.conf and iax.conf files, then edit
them taking out every single host reference that wasn't an ip address.
 If I left them in and tried to restart asterisk, it would hang on the
first host trying to resolve.  A minute or so later it'd give up and
move on to the second.  Obviously very bad news if you have several
hosts that it needs to resolve (side note, why can't asterisk try to
resolve multiple hosts at once; say one every 5 seconds, so it doesn't
flood your network with dns requests, but also if one host hangs it
can try resolving other hosts while waiting?).

I've looked in dns.c and dnsmgr.c and can't see where I can set a
timeout.  Perhaps it's somewhere else?  Maybe hiding in several files?
 Any ideas?  I'd like to set it to five seconds, this should give most
hosts that aren't down plenty of time to respond.  Perhaps even
better, I could cache dns results and save them to a file?  Run a
background application to query dns servers, if it hangs then asterisk
uses the last good values (and if it's not reachable, no big deal,
asterisk will just move on).

I promise I searched on google before posting here.  The closest thing
I could find is this:

http://bugs.digium.com/view.php?id=3946

Doesn't seem to have a real solution.

Joseph Tanner
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[Asterisk-Users] Problem ChanSpy

2006-03-08 Thread David Guarnido









Sorry, This is a mistake, sip.conf:



[302]canreinvite=no[301]canreinvite=noAny idea?Thanks  












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RE: [Asterisk-Users] res_mysql.conf DNS SRV lookup

2006-03-08 Thread Bob McDowell


Good to know I'm not the only one...

I thought perhaps I had been expelled from the list...


Bob McDowell

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, March 07, 2006 10:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] res_mysql.conf  DNS SRV lookup

Good grief! I posted the message below at 1:17pm... and it appeared on
the list after 8pm.
Nice

-Original Message-
From: Douglas Garstang
Sent: Tue 3/7/2006 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: RE: [Asterisk-Users] res_mysql.conf  DNS SRV lookup



Yay!

-Original Message-
From: Olle E Johansson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 07, 2006 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] res_mysql.conf  DNS SRV lookup



7 mar 2006 kl. 19.03 skrev Douglas Garstang:

 My bad. SRV lookups work, but Asterisk only uses the first
entry
 right? This means there's no redundancy.

That is correct. That is what we try to fix.

/O
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Re: [Asterisk-Users] Asterisk download file locations

we mirror all the files our selves so our scripts work
flawlessly. 

--- Alistair Cunningham [EMAIL PROTECTED]
wrote:

 This is a request to the website manager for
 asterisk.org.
 
 The build scripts for our ITSP product include the
 URLs to download the 
 Asterisk files, such as:
 
 wget

http://ftp.digium.com/pub/asterisk/asterisk-1.2.5.tar.gz;
 
 However, if a new version is released,
 asterisk-1.2.5.tar.gz is moved to 
 the old directory. This breaks our scripts until
 we can update them 
 and send them to our resellers.
 
 Would it be possible to have a fixed address for a
 particular asterisk 
 release that will never (or at least not for a long
 time) change? 
 Perhaps put all (except very old) versions in the
 same directory, with a 
   'latest' link to the latest one?
 
 -- 
 Alistair Cunningham,
 Integrics Ltd,
 +44 20 799 39 799
 sip:[EMAIL PROTECTED]
 http://integrics.com/
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Re: [Asterisk-Users] Asterisk Prepaid Card