date:20050430

Anybody has some command line examples on how to run sipp against asterisk?
.. 

I tried using sipp -sn uac 127.0.0.1 and I get

Apr 30 02:14:14 NOTICE[3619]: chan_sip.c:8361 handle_request_invite: Failed
to authenticate user sipp sip:[EMAIL PROTECTED]:5061;tag=62


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Re: [Asterisk-Users] Caller-ID Block

2005-04-30 Thread Time Bandit

 Question: how can I block someone from calling us?
 
 Sometimes we get crank calls into our office. We'd like to build a list
 of callers to be blocked. When they call, they should hear busy and
 then we hang up. We have about 100 DIDs routed to different contexts
 and I wouldn't want to have to manually edit all contexts. Is there a
 way to do something global to create something like a black list of
 caller IDs to block?

I'm working on something that could be usefull for this. It's a PHP
AGI that route the call based on the caller ID. When a call comes in,
you call this AGI which query a MySQL table and look for this caller
ID. If it finds it, it execute the defined steps one after the other.

The table as 6 fields (for now) : call_id_num, step, action, option1,
option2, option3

Example 1 : Someone you don`t want to answer to. You put something
like this in the table
555,1,Hangup

Example 2 : Your mother in law, send her to voicemail :)
555,1,Answer
555,2,Voicemail,u7001

Example 3 : Someone you want to know that you don't want him to call you
555,1,Answer
555,2,Playback,please-dont-call-here-again
555,3,Hangup

Example 4 : Your mistress, ring your phone directly
555,1,Answer
555,2,Dial,IAX,7001

The action I have in the script so far are these :
- V : send to voicemail 'option1'
- D : dial option1/option2
- P : playback sound 'option1'
- H : hangup
- A : Answer
- G : Goto option1,option2,option3

The next step will be to make a PHP page that will be used to build
the rules with simple choice from dropbox.

It's still in Alpha stage, but if it can help you get rid of those
annoying calls, I'll be glad to share it with you.

I think this could be extended to make pretty smart CallerID based routing.
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[Asterisk-Users] Avaya 4610SW IP phone?

2005-04-30 Thread Walt Reed

From what I've read, this is a H.323 phone only. Only the 4602 has SIP
images. Has anyone gotten a 4610 H.323 working with *?

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[Asterisk-Users] Dynamic phone groups.

2005-04-30 Thread Joris Vandalon

Hi,

I am looking for a way to dynamicly put phones in a group so if someone
calls an extentions everyone's phone who's member of the group will
ring.
Queues are not an options because as soon a call comes in to a queue
there is no getting out.
I want to let the phones ring and after a period of time stop trying and
continue to voicemail for example.
Can someone provide me with some hints or examples getting this done?

Regards,
Joris

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Re: RE: [Asterisk-Users] Problems with TDM400P card

2005-04-30 Thread Kim Culhan

On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said:
 I would also be interested in alternatives to the Tdm400p. I have had endless
 problems with a tdm400p card not being able to get the zttest numbers above
 99.975 and as a result not being able eliminate an intermitent but consistent 
 echo.
 I have tried to date 4 different motherboard and hardware combinations as
 well as different linux versions to no avial.I would welcome some feedback on 
 this.

Since there appear to be several combinations of hardware and operating system
which don't work well, here is a combination which appears to work fairly well:

Intel 925XCV mb

P-4 560 (3.6 gHz)

wcfxs0: Wildcard TDM400P REV E/F

FreeBSD 5.4-STABLE

zttest -v
Opened pseudo zap interface, measuring accuracy...

8192 samples in 8192 sample intervals 100.00% 
8192 samples in 8192 sample intervals 100.00% 
8192 samples in 8192 sample intervals 100.00% 
8192 samples in 8192 sample intervals 100.00% 
8192 samples in 8192 sample intervals 100.00% 
8192 samples in 8192 sample intervals 100.00% 
8192 samples in 8192 sample intervals 100.00% 
8192 samples in 8192 sample intervals 100.00% 
8192 samples in 8192 sample intervals 100.00% 
8192 samples in 8192 sample intervals 100.00% ^C
--- Results after 10 passes ---
Best: 100.00 -- Worst: 100.00 -- Average: 100.00

hope this helps

-kim

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Re: RE: [Asterisk-Users] Problems with TDM400P card

 On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said:
  I would also be interested in alternatives to the Tdm400p. I have had 
  endless
  problems with a tdm400p card not being able to get the zttest numbers above
  99.975 and as a result not being able eliminate an intermitent but 
  consistent echo.
  I have tried to date 4 different motherboard and hardware combinations as
  well as different linux versions to no avial.I would welcome some feedback 
  on this.
 
 Since there appear to be several combinations of hardware and operating system
 which don't work well, here is a combination which appears to work fairly 
 well:
 
 Intel 925XCV mb
 
 P-4 560 (3.6 gHz)
 
 wcfxs0: Wildcard TDM400P REV E/F
 
 FreeBSD 5.4-STABLE
 
 zttest -v
 Opened pseudo zap interface, measuring accuracy...
 
 8192 samples in 8192 sample intervals 100.00% 
 8192 samples in 8192 sample intervals 100.00% 
 8192 samples in 8192 sample intervals 100.00% 
 8192 samples in 8192 sample intervals 100.00% 
 8192 samples in 8192 sample intervals 100.00% 
 8192 samples in 8192 sample intervals 100.00% 
 8192 samples in 8192 sample intervals 100.00% 
 8192 samples in 8192 sample intervals 100.00% 
 8192 samples in 8192 sample intervals 100.00% 
 8192 samples in 8192 sample intervals 100.00% ^C
 --- Results after 10 passes ---
 Best: 100.00 -- Worst: 100.00 -- Average: 100.00
 
 hope this helps

Kim, that is helpful. I'm not a FreeBSD user, but does it have a
vmstat utility?

If so, what do see if you run 'vmstat 1' and let it run for about
twenty seconds?  Do you see the cpu utilization going to about
100% every five or six seconds?

Rich


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RE: [Asterisk-Users] Can't get incoming calls with IAX trunks (FWDTeliax)

2005-04-30 Thread Patrick M. Gray, Jr.

It looks like it registers:

asterisk1*CLI iax2 show registry
Host  UsernamePerceived Refresh  State
208.139.204.228:4569  memy.external.ip:154660
Registered
65.39.205.121:4569me  my.external.ip:154660  Registered

But the logs (I didn't know that trick, or at least didn't have enough v's
when looking at the CLI) show incoming calls are being rejected:

Fwd inbound:
Apr 30 09:55:06 NOTICE[1439]: Rejected connect attempt from 65.39.205.121,
request '[EMAIL PROTECTED]' does not exist

Teliax inbound:
Apr 30 09:55:34 NOTICE[1439]: Rejected connect attempt from 208.139.204.228,
request '[EMAIL PROTECTED]' does not exist

I followed the FWD and Teliax instructions exactly, but clearly I'm missing
something.

Here is my iax_additional.conf:

register=me:[EMAIL PROTECTED]
register=me:[EMAIL PROTECTED]

[fwd]
username=me
type=peer
secret=very
host=iax2.fwdnet.net
dtmf=inband
allow=ulaw

[iaxfwd]
type=user
inkeys=freeworlddialup
disallow=all
context=from-pstn
auth=rsa
allow=ulaw

[teliax]
username=me
type=friend
secret=very
host=voip.teliax.com
disallow=all
context=from-pstn
auth=md5
allow=gsm


I have a from-pstn context in extensions.conf as well:

[from-trunk]; just an
alias since VoIP shouldn't be called PSTN
include = from-pstn

[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did
include = from-pstn-timecheck  ; this has to be included otherwise
it overrides ext-did

[from-pstn-timecheck]
exten = .,1,Goto(s,1)  ; catch-all matching for calls that have DID
info (if a DID route hasn't matched them)
exten = s,1,GotoIf($[${IN_OVERRIDE} =
forcereghours]?from-pstn-reghours,s,1:)
exten = s,2,GotoIf($[${IN_OVERRIDE} =
forceafthours]?from-pstn-afthours,s,1:)
exten = s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:)
exten = s,4,Goto(from-pstn-afthours,s,1)

[from-pstn-reghours]
exten = s,1,GotoIf($[${FAX_RX} = disabled]?from-pstn-reghours-nofax,s,1:2)
; if fax detection is disabled, then jump to from-pstn-nofax - else continue
exten = s,2,Answer 
exten = s,3,Wait(1)
exten = s,4,SetVar(intype=${INCOMING})
exten = s,5,Cut(intype=intype,-,1) 
exten = s,6,GotoIf($[${intype} = EXT]?7:9) ; If INCOMING starts
with EXT, then assume its an extension
exten = s,7,Wait(3)
;wait 3 more second to make sure this isn't a fax before dialing someone
exten = s,8,Goto(ext-local,${INCOMING:4},1)
exten = s,9,GotoIf($[${intype} = GRP]?10:12)   ; If INCOMING starts with
GRP, then assume its a ring group
exten = s,10,Wait(3)
exten = s,11,Goto(ext-group,${INCOMING:4},1)
exten = s,12,GotoIf($[${intype} = QUE]?13:15)
exten = s,13,Wait(3)
exten = s,14,Goto(ext-queues,${INCOMING:4},1)
exten = s,15,Goto(${INCOMING},s,1) ; not EXT or GR1 -
it's an auto attendant
exten = fax,1,Goto(ext-fax,in_fax,1)
exten = h,1,Hangup


Thanks so much for any help!  I've been stumped by this one and feel like
I'm missing some piece of the puzzle.

Thanks again!

Pat

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Levine
Sent: Saturday, 30 April, 2005 00:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Can't get incoming calls with IAX trunks
(FWDTeliax)

Are you sure it's registering?

- 
Dan Levine 
CYTEXONE | Your Technology Specialists 
t: 877.CYTEXONE x 810 
l: 212.477.0990 x 810 
e: [EMAIL PROTECTED] 
http://www.cytexone.com 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Gray, Jr.
Sent: Saturday, April 30, 2005 12:04 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Can't get incoming calls with IAX trunks (FWD
Teliax)

I have [EMAIL PROTECTED] 0.9 running, and everything seems
to work well EXCEPT incoming calls.

I have an FWD and Teliax trunk (both using IAX), and a
Cisco 7960 SIP phone connected to Asterisk. 
Everything tests fine:

-   Can call from softphone to Cisco and vice versa
-   Asterisk inbound simulation works like a champ
-   Voicemail works fine
-   Outbound calls to both trunks works fine

However, when I call into my system on the FWD or
Teliax trunks, nothing happens.  Nothing appears on
the asterisk console so I'm not even sure where to
start.  I'm suspecting network problems, but don't
know what to look for.  My asterisk box sits on my
LAN, behind an IPCop-based NAT router.  I've forwarded
port 4569 UDP and TCP to the asterisk box, but still
no joy.  I've googled and checked voip-info, but
everything that mentions NAT as a potential problem
points to IAX as the solution.  Trunk-wise, I'm pure
IAX (only SIP is the 7960, and it's on the same
network as the asterisk box).

I'm pretty new to asterisk, so if you can dumb down
any debugging advice I'd appreciate it.

Thanks a ton!

Pat

RE: [Asterisk-Users] Can't get incoming calls with IAX trunks(FWDTeliax)

2005-04-30 Thread Joshua Colp

Hello,

They are being rejected because the extensions (your DIDs) do not exist in
the context from-pstn. How did I know? I read the error ;)

- Joshua Colp.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick M.
Gray, Jr.
Sent: Saturday, April 30, 2005 11:03 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Can't get incoming calls with IAX
trunks(FWDTeliax)

It looks like it registers:

asterisk1*CLI iax2 show registry
Host  UsernamePerceived Refresh  State
208.139.204.228:4569  memy.external.ip:154660
Registered
65.39.205.121:4569me  my.external.ip:154660  Registered

But the logs (I didn't know that trick, or at least didn't have enough v's
when looking at the CLI) show incoming calls are being rejected:

Fwd inbound:
Apr 30 09:55:06 NOTICE[1439]: Rejected connect attempt from 65.39.205.121,
request '[EMAIL PROTECTED]' does not exist

Teliax inbound:
Apr 30 09:55:34 NOTICE[1439]: Rejected connect attempt from 208.139.204.228,
request '[EMAIL PROTECTED]' does not exist

I followed the FWD and Teliax instructions exactly, but clearly I'm missing
something.

Here is my iax_additional.conf:

register=me:[EMAIL PROTECTED]
register=me:[EMAIL PROTECTED]

[fwd]
username=me
type=peer
secret=very
host=iax2.fwdnet.net
dtmf=inband
allow=ulaw

[iaxfwd]
type=user
inkeys=freeworlddialup
disallow=all
context=from-pstn
auth=rsa
allow=ulaw

[teliax]
username=me
type=friend
secret=very
host=voip.teliax.com
disallow=all
context=from-pstn
auth=md5
allow=gsm


I have a from-pstn context in extensions.conf as well:

[from-trunk]; just an
alias since VoIP shouldn't be called PSTN
include = from-pstn

[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did
include = from-pstn-timecheck  ; this has to be included otherwise
it overrides ext-did

[from-pstn-timecheck]
exten = .,1,Goto(s,1)  ; catch-all matching for calls that have DID
info (if a DID route hasn't matched them)
exten = s,1,GotoIf($[${IN_OVERRIDE} =
forcereghours]?from-pstn-reghours,s,1:)
exten = s,2,GotoIf($[${IN_OVERRIDE} =
forceafthours]?from-pstn-afthours,s,1:)
exten = s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:)
exten = s,4,Goto(from-pstn-afthours,s,1)

[from-pstn-reghours]
exten = s,1,GotoIf($[${FAX_RX} = disabled]?from-pstn-reghours-nofax,s,1:2)
; if fax detection is disabled, then jump to from-pstn-nofax - else continue
exten = s,2,Answer 
exten = s,3,Wait(1)
exten = s,4,SetVar(intype=${INCOMING})
exten = s,5,Cut(intype=intype,-,1) 
exten = s,6,GotoIf($[${intype} = EXT]?7:9) ; If INCOMING starts
with EXT, then assume its an extension
exten = s,7,Wait(3)
;wait 3 more second to make sure this isn't a fax before dialing someone
exten = s,8,Goto(ext-local,${INCOMING:4},1)
exten = s,9,GotoIf($[${intype} = GRP]?10:12)   ; If INCOMING starts with
GRP, then assume its a ring group
exten = s,10,Wait(3)
exten = s,11,Goto(ext-group,${INCOMING:4},1)
exten = s,12,GotoIf($[${intype} = QUE]?13:15)
exten = s,13,Wait(3)
exten = s,14,Goto(ext-queues,${INCOMING:4},1)
exten = s,15,Goto(${INCOMING},s,1) ; not EXT or GR1 -
it's an auto attendant
exten = fax,1,Goto(ext-fax,in_fax,1)
exten = h,1,Hangup


Thanks so much for any help!  I've been stumped by this one and feel like
I'm missing some piece of the puzzle.

Thanks again!

Pat

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Levine
Sent: Saturday, 30 April, 2005 00:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Can't get incoming calls with IAX trunks
(FWDTeliax)

Are you sure it's registering?

- 
Dan Levine 
CYTEXONE | Your Technology Specialists 
t: 877.CYTEXONE x 810 
l: 212.477.0990 x 810 
e: [EMAIL PROTECTED] 
http://www.cytexone.com 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Gray, Jr.
Sent: Saturday, April 30, 2005 12:04 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Can't get incoming calls with IAX trunks (FWD
Teliax)

I have [EMAIL PROTECTED] 0.9 running, and everything seems
to work well EXCEPT incoming calls.

I have an FWD and Teliax trunk (both using IAX), and a
Cisco 7960 SIP phone connected to Asterisk. 
Everything tests fine:

-   Can call from softphone to Cisco and vice versa
-   Asterisk inbound simulation works like a champ
-   Voicemail works fine
-   Outbound calls to both trunks works fine

However, when I call into my system on the FWD or
Teliax trunks, nothing happens.  Nothing appears on
the asterisk console so I'm not even sure where to
start.  I'm suspecting network problems, but don't
know what to look for.  My asterisk box sits on my

Re: RE: [Asterisk-Users] Problems with TDM400P card

2005-04-30 Thread Kim Culhan

On Sat, April 30, 2005 10:52 am, Rich Adamson said:
 On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said:
 
 I would also be interested in alternatives to the Tdm400p. I have
 had endless problems with a tdm400p card not being able to get
 the zttest numbers above 99.975 and as a result not being able
 eliminate an intermitent but consistent echo.

 Kim, that is helpful. I'm not a FreeBSD user, but does it have a
 vmstat utility?

Has vmstat, you might like FreeBSD..  :)

 If so, what do see if you run 'vmstat 1' and let it run for about
 twenty seconds?  Do you see the cpu utilization going to about 100%
 every five or six seconds?

Negative:

vmstat 1
 procs memory   page disk   faultscpu
 r b w   avm  freflt  re  pi  po  fr  sr ad4   in 
sy  cs  us sy id
 1 2 0   61684 9662607   0   0   0   6   0   0 1326  392 482  0  0 99
 0 2 0   61684 9662601   0   0   0   1   0   0 1337  501 494  0  1 99
 0 2 0   61684 9662600   0   0   0   0   0   0 1345  486 490  0  0 100
 0 2 0   61684 9662600   0   0   0   0   0   6 1350  492 509  0  2 98
 0 2 0   61684 9662600   0   0   0   0   0   0 1344  488 490  1  0 99
 0 2 0   61684 9662600   0   0   0   0   0   0 1344  492 489  0  0 100
 0 2 0   61684 9662600   0   0   0   0   0   0 1345  494 488  0  0 100
 0 2 0   61684 9662600   0   0   0   0   0   0 1344  492 493  0  0 100
 0 2 0   61684 9662600   0   0   0   0   0   0 1344  488 490  0  0 100
 0 2 0   61684 9662600   0   0   0   0   0   0 1344  492 490  0  0 100
 0 2 0   61684 9662600   0   0   0   0   0   0 1345  486 487  0  1 99
 0 2 0   61684 9662600   0   0   0   0   0   0 1344  513 494  0  0 100
 0 2 0   61684 9662600   0   0   0   0   0   0 1344  494 494  0  0 100
 0 2 0   61684 9662600   0   0   0   0   0   0 1345  492 492  0  0 100
 0 2 0   61684 9662600   0   0   0   0   0   0 1344  486 487  0  0 100
 0 2 0   61684 9662600   0   0   0   0   0   0 1344  492 490  0  0 100
 0 2 0   61684 9662600   0   0   0   0   0   0 1344  496 491  0  0 100
 0 2 0   61684 9662600   0   0   0   0   0   0 1345  492 491  0  0 100
 0 2 0   61684 9662600   0   0   0   0   0   1 1345  486 491  0  0 100
 0 2 0   61684 9662600   0   0   0   0   0   0 1344  492 490  0  0 100
^C

-kim

--
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RE: [Asterisk-Users] Can't get incoming calls with IAXtrunks(FWDTeliax)

2005-04-30 Thread Patrick M. Gray, Jr.

D-oh!!!  I told you I was missing one simple piece of the puzzle.  Light
finally dawns on marble head.

Thanks for your help (with the obvious).

Pat

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp
Sent: Saturday, 30 April, 2005 10:05
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Can't get incoming calls with
IAXtrunks(FWDTeliax)

Hello,

They are being rejected because the extensions (your DIDs) do not exist in
the context from-pstn. How did I know? I read the error ;)

- Joshua Colp.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick M.
Gray, Jr.
Sent: Saturday, April 30, 2005 11:03 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Can't get incoming calls with IAX
trunks(FWDTeliax)

It looks like it registers:

asterisk1*CLI iax2 show registry
Host  UsernamePerceived Refresh  State
208.139.204.228:4569  memy.external.ip:154660
Registered
65.39.205.121:4569me  my.external.ip:154660  Registered

But the logs (I didn't know that trick, or at least didn't have enough v's
when looking at the CLI) show incoming calls are being rejected:

Fwd inbound:
Apr 30 09:55:06 NOTICE[1439]: Rejected connect attempt from 65.39.205.121,
request '[EMAIL PROTECTED]' does not exist

Teliax inbound:
Apr 30 09:55:34 NOTICE[1439]: Rejected connect attempt from 208.139.204.228,
request '[EMAIL PROTECTED]' does not exist

I followed the FWD and Teliax instructions exactly, but clearly I'm missing
something.

Here is my iax_additional.conf:

register=me:[EMAIL PROTECTED]
register=me:[EMAIL PROTECTED]

[fwd]
username=me
type=peer
secret=very
host=iax2.fwdnet.net
dtmf=inband
allow=ulaw

[iaxfwd]
type=user
inkeys=freeworlddialup
disallow=all
context=from-pstn
auth=rsa
allow=ulaw

[teliax]
username=me
type=friend
secret=very
host=voip.teliax.com
disallow=all
context=from-pstn
auth=md5
allow=gsm


I have a from-pstn context in extensions.conf as well:

[from-trunk]; just an
alias since VoIP shouldn't be called PSTN
include = from-pstn

[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did
include = from-pstn-timecheck  ; this has to be included otherwise
it overrides ext-did

[from-pstn-timecheck]
exten = .,1,Goto(s,1)  ; catch-all matching for calls that have DID
info (if a DID route hasn't matched them)
exten = s,1,GotoIf($[${IN_OVERRIDE} =
forcereghours]?from-pstn-reghours,s,1:)
exten = s,2,GotoIf($[${IN_OVERRIDE} =
forceafthours]?from-pstn-afthours,s,1:)
exten = s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:)
exten = s,4,Goto(from-pstn-afthours,s,1)

[from-pstn-reghours]
exten = s,1,GotoIf($[${FAX_RX} = disabled]?from-pstn-reghours-nofax,s,1:2)
; if fax detection is disabled, then jump to from-pstn-nofax - else continue
exten = s,2,Answer 
exten = s,3,Wait(1)
exten = s,4,SetVar(intype=${INCOMING})
exten = s,5,Cut(intype=intype,-,1) 
exten = s,6,GotoIf($[${intype} = EXT]?7:9) ; If INCOMING starts
with EXT, then assume its an extension
exten = s,7,Wait(3)
;wait 3 more second to make sure this isn't a fax before dialing someone
exten = s,8,Goto(ext-local,${INCOMING:4},1)
exten = s,9,GotoIf($[${intype} = GRP]?10:12)   ; If INCOMING starts with
GRP, then assume its a ring group
exten = s,10,Wait(3)
exten = s,11,Goto(ext-group,${INCOMING:4},1)
exten = s,12,GotoIf($[${intype} = QUE]?13:15)
exten = s,13,Wait(3)
exten = s,14,Goto(ext-queues,${INCOMING:4},1)
exten = s,15,Goto(${INCOMING},s,1) ; not EXT or GR1 -
it's an auto attendant
exten = fax,1,Goto(ext-fax,in_fax,1)
exten = h,1,Hangup


Thanks so much for any help!  I've been stumped by this one and feel like
I'm missing some piece of the puzzle.

Thanks again!

Pat

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Levine
Sent: Saturday, 30 April, 2005 00:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Can't get incoming calls with IAX trunks
(FWDTeliax)

Are you sure it's registering?

- 
Dan Levine 
CYTEXONE | Your Technology Specialists 
t: 877.CYTEXONE x 810 
l: 212.477.0990 x 810 
e: [EMAIL PROTECTED] 
http://www.cytexone.com 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Gray, Jr.
Sent: Saturday, April 30, 2005 12:04 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Can't get incoming calls with IAX trunks (FWD
Teliax)

I have [EMAIL PROTECTED] 0.9 running, and everything seems
to work well EXCEPT incoming calls.

I have an FWD and Teliax trunk (both using IAX), and a
Cisco 7960 SIP phone connected to Asterisk. 
Everything tests fine:

-

RE: [Asterisk-Users] Can't get incoming calls with IAX trunks (FWDTeliax)

Inline

 It looks like it registers:
 
 asterisk1*CLI iax2 show registry
 Host  UsernamePerceived Refresh  State
 208.139.204.228:4569  me  my.external.ip:154660
 Registered
 65.39.205.121:4569me  my.external.ip:154660  Registered
 
 But the logs (I didn't know that trick, or at least didn't have enough v's
 when looking at the CLI) show incoming calls are being rejected:
 
 Fwd inbound:
 Apr 30 09:55:06 NOTICE[1439]: Rejected connect attempt from 65.39.205.121,
 request '[EMAIL PROTECTED]' does not exist

The above says you do not have an entry like:
[from-pstn]
exten = 632254 dowhatever

The s exten does _not_ match everything. You're incoming call is to a
specific exten and you need to have a specific entry to handle it. That
might include a Goto for some other context, etc.

 
 Teliax inbound:
 Apr 30 09:55:34 NOTICE[1439]: Rejected connect attempt from 208.139.204.228,
 request '[EMAIL PROTECTED]' does not exist

Same exact problem.

 I followed the FWD and Teliax instructions exactly, but clearly I'm missing
 something.
 
 Here is my iax_additional.conf:
 
 register=me:[EMAIL PROTECTED]
 register=me:[EMAIL PROTECTED]
 
 [fwd]
 username=me
 type=peer
 secret=very
 host=iax2.fwdnet.net
 dtmf=inband
 allow=ulaw
 
 [iaxfwd]
 type=user
 inkeys=freeworlddialup
 disallow=all
 context=from-pstn
 auth=rsa
 allow=ulaw
 
 [teliax]
 username=me
 type=friend
 secret=very
 host=voip.teliax.com
 disallow=all
 context=from-pstn
 auth=md5
 allow=gsm
 
 
 I have a from-pstn context in extensions.conf as well:
 
 [from-trunk]  ; just an
 alias since VoIP shouldn't be called PSTN
 include = from-pstn
 
 [from-pstn]
 include = from-pstn-custom   ; create this context in
 extensions_custom.conf to include customizations
 include = ext-did
 include = from-pstn-timecheck; this has to be included 
 otherwise
 it overrides ext-did
 
 [from-pstn-timecheck]
 exten = .,1,Goto(s,1); catch-all matching for calls that 
 have DID
 info (if a DID route hasn't matched them)
 exten = s,1,GotoIf($[${IN_OVERRIDE} =
 forcereghours]?from-pstn-reghours,s,1:)
 exten = s,2,GotoIf($[${IN_OVERRIDE} =
 forceafthours]?from-pstn-afthours,s,1:)
 exten = s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:)
 exten = s,4,Goto(from-pstn-afthours,s,1)
 
 [from-pstn-reghours]
 exten = s,1,GotoIf($[${FAX_RX} = disabled]?from-pstn-reghours-nofax,s,1:2)
 ; if fax detection is disabled, then jump to from-pstn-nofax - else continue
 exten = s,2,Answer   
 exten = s,3,Wait(1)
 exten = s,4,SetVar(intype=${INCOMING})
 exten = s,5,Cut(intype=intype,-,1) 
 exten = s,6,GotoIf($[${intype} = EXT]?7:9)   ; If INCOMING starts
 with EXT, then assume its an extension
 exten = s,7,Wait(3)
 ;wait 3 more second to make sure this isn't a fax before dialing someone
 exten = s,8,Goto(ext-local,${INCOMING:4},1)
 exten = s,9,GotoIf($[${intype} = GRP]?10:12) ; If INCOMING starts 
 with
 GRP, then assume its a ring group
 exten = s,10,Wait(3)
 exten = s,11,Goto(ext-group,${INCOMING:4},1)
 exten = s,12,GotoIf($[${intype} = QUE]?13:15)
 exten = s,13,Wait(3)
 exten = s,14,Goto(ext-queues,${INCOMING:4},1)
 exten = s,15,Goto(${INCOMING},s,1)   ; not EXT or GR1 -
 it's an auto attendant
 exten = fax,1,Goto(ext-fax,in_fax,1)
 exten = h,1,Hangup
 
 
 Thanks so much for any help!  I've been stumped by this one and feel like
 I'm missing some piece of the puzzle.

I typically use something like this for incoming calls:
[teliax-incoming]
include = bus-ivr-main
exten = 303222,1,Goto(bus-ivr-main|s|1)

[bus-ivr-main]
exten = s,1,Wait,1
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,15
exten = s,5,Background(npi-greeting)  ; Thanks for calling press 1 for  



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Re: [Asterisk-Users] Dynamic phone groups.

2005-04-30 Thread Robert P. McKenzie

Joris Vandalon wrote:
Hi,
I am looking for a way to dynamicly put phones in a group so if someone
calls an extentions everyone's phone who's member of the group will
ring.
Queues are not an options because as soon a call comes in to a queue
there is no getting out.
I want to let the phones ring and after a period of time stop trying and
continue to voicemail for example.
Can someone provide me with some hints or examples getting this done?
It may not be exactly what you are after but I do something like this:
extensions.conf
HOUSEPHONES=SIP/somepcSIP/anotherpcIAX2/desktopIAX2/someotherdesktopSIP/sipuraline1SIP/sipuraline2
; London Number - SIP Inbound provider
exten = 1438645,1,Answer
exten = 1438645,2,Dial(${HOUSEPHONES}|60|t)
exten = 1438645,3,Voicemail(u50)
Each phone listed above also has it's own extention, but the voicemail all goes to 50. That way I can call any
extension from anyother inside the house. But calling 50 directly will make every phone ring. Any phone not logged in
will just be ignored and skipped. The first phone to pick up gets it. Call parking is on so if there is a need to
transfer calls from one phone to another it can be done using parking.

--
Robert P. McKenzie | GammaRay Technical Services Ltd
[EMAIL PROTECTED] | [EMAIL PROTECTED]
http://www.uk-experience.com | http://www.gammaray-tech.com
Ecademy Profile: http://www.ecademy.com/account.php?op=viewid=64014
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RE: [Asterisk-Users] Dynamic phone groups.

2005-04-30 Thread Joel Duffield

I would think what you would need to look at is how to do this with the *
Data Base. I haven't done this, but it would seem that there is a way to
make it work with that.

Joel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert P.
McKenzie
Sent: Saturday, April 30, 2005 10:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dynamic phone groups.


Joris Vandalon wrote:
 Hi,

 I am looking for a way to dynamicly put phones in a group so if someone
 calls an extentions everyone's phone who's member of the group will
 ring.
 Queues are not an options because as soon a call comes in to a queue
 there is no getting out.
 I want to let the phones ring and after a period of time stop trying and
 continue to voicemail for example.
 Can someone provide me with some hints or examples getting this done?

It may not be exactly what you are after but I do something like this:

extensions.conf

HOUSEPHONES=SIP/somepcSIP/anotherpcIAX2/desktopIAX2/someotherdesktopSIP/
sipuraline1SIP/sipuraline2

; London Number - SIP Inbound provider
exten = 1438645,1,Answer
exten = 1438645,2,Dial(${HOUSEPHONES}|60|t)
exten = 1438645,3,Voicemail(u50)


Each phone listed above also has it's own extention, but the voicemail all
goes to 50.  That way I can call any
extension from anyother inside the house.  But calling 50 directly will make
every phone ring.  Any phone not logged in
will just be ignored and skipped.  The first phone to pick up gets it.  Call
parking is on so if there is a need to
transfer calls from one phone to another it can be done using parking.


--
Robert P. McKenzie |   GammaRay Technical Services Ltd
[EMAIL PROTECTED] | [EMAIL PROTECTED]
http://www.uk-experience.com   |  http://www.gammaray-tech.com

Ecademy Profile:   http://www.ecademy.com/account.php?op=viewid=64014
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Re: [Asterisk-Users] Caller-ID Block

2005-04-30 Thread Tim Litwiller

If it references anything that is not in the default asterisk - then it 
came from asterisk at home.  I just looked again - it uses the festival 
text to speech engine to say the words - record those messages and then 
use playback(filename) instead of AGI(festival-script.pl|words to say)

for example:
record a file that says phone number and save it in your asterisk 
sounds directory called phone_number.wav
change
AGI(festival-script.pl|phone number)
playback(phone_number)

for the next one record has been blacklisted at level and save as 
filename blacklisted_at_level.wav

AGI(festival-script.pl|has been blacklisted at level)
playback(blacklisted_at_level)
or if you don't want to use your voice find one of the web sites that 
lets you test text to speech and download as a wav file and save those.



Daniel Salama wrote:
Tim,
This certainly looks interesting. I just have a question about the  
recipe: it makes reference to some AGI perl scripts. Is the source  
available? Or may be it's irrelevant.

Thanks,
Daniel
On Apr 29, 2005, at 9:10 PM, Tim Litwiller wrote:
Daniel Salama wrote:
Question: how can I block someone from calling us?
Sometimes we get crank calls into our office. We'd like to build a  
list of callers to be blocked. When they call, they should hear busy  
and then we hang up. We have about 100 DIDs routed to different  
contexts and I wouldn't want to have to manually edit all contexts.  
Is there a way to do something global to create something like a  
black list of caller IDs to block?
Thanks,
Daniel

I used bits and pieces that I got from this list and from the wiki 
and  made this up - I'm using it on aah - but it should be usable 
with  slight modification on any asterisk install.

If you just want busy you'll have to edit the blacklisted1 macro  
currently it plays the SIT sound and then the this number is no 
longer  in service message

It usually gets me taken off their call lists pretty quickly  :)
http://www.oinko.net/astrecipes/index.php?from=0q=astrecipes/ 
how+to+blacklist+unwanted+callerid

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RE: RE: [Asterisk-Users] Problems with TDM400P card

Hows does this look?

Opened pseudo zap interface, measuring accuracy...

8192 samples in 8192 sample intervals 100.00%
8192 samples in 8193 sample intervals 99.987793%
8192 samples in 8193 sample intervals 99.987793%
8192 samples in 8193 sample intervals 99.987793%
8192 samples in 8193 sample intervals 99.987793%
8192 samples in 8193 sample intervals 99.987793%
8192 samples in 8193 sample intervals 99.987793%
8192 samples in 8193 sample intervals 99.987793%
8192 samples in 8193 sample intervals 99.987793%
8192 samples in 8194 sample intervals 99.975586%
8192 samples in 8193 sample intervals 99.987793%
8192 samples in 8193 sample intervals 99.987793%
8192 samples in 8193 sample intervals 99.987793%
--- Results after 13 passes ---
Best: 100.00 -- Worst: 99.975586 -- Average: 99.987793

Good enough and what do I need to check in order to make 100%? What does the
test actually measure?

 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kim Culhan
|Sent: Sábado, 30 de Abril de 2005 08:45 a.m.
|To: asterisk-users@lists.digium.com
|Subject: Re: RE: [Asterisk-Users] Problems with TDM400P card
|
|On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said:
| I would also be interested in alternatives to the Tdm400p. I 
|have had 
| endless problems with a tdm400p card not being able to get 
|the zttest 
| numbers above
| 99.975 and as a result not being able eliminate an 
|intermitent but consistent echo.
| I have tried to date 4 different motherboard and hardware 
|combinations 
| as well as different linux versions to no avial.I would 
|welcome some feedback on this.
|
|Since there appear to be several combinations of hardware and 
|operating system which don't work well, here is a combination 
|which appears to work fairly well:
|
|Intel 925XCV mb
|
|P-4 560 (3.6 gHz)
|
|wcfxs0: Wildcard TDM400P REV E/F
|
|FreeBSD 5.4-STABLE
|
|zttest -v
|Opened pseudo zap interface, measuring accuracy...
|
|8192 samples in 8192 sample intervals 100.00%
|8192 samples in 8192 sample intervals 100.00%
|8192 samples in 8192 sample intervals 100.00%
|8192 samples in 8192 sample intervals 100.00%
|8192 samples in 8192 sample intervals 100.00%
|8192 samples in 8192 sample intervals 100.00%
|8192 samples in 8192 sample intervals 100.00%
|8192 samples in 8192 sample intervals 100.00%
|8192 samples in 8192 sample intervals 100.00%
|8192 samples in 8192 sample intervals 100.00% ^C
|--- Results after 10 passes ---
|Best: 100.00 -- Worst: 100.00 -- Average: 100.00
|
|hope this helps
|
|-kim
|
|--
|[EMAIL PROTECTED]
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|

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[Asterisk-Users] Polycom IP500 Forward problem codec issue

2005-04-30 Thread Scott Herrick

Polycom IP500 Forward problem codec issue
All,
Im running the Polycom IP500 phones at several sites.   My * server is 
at a collocation site and I have complete control of the T1s running to 
the remote sites with the IP500 phones.  Connectivity to the PSTN is 
through a Cisco 2600 with a PRI card.   During initial testing I ran 
G711/ulaw but have added G729 licenses to the system.

Problem:  When the forwarding function on the Polycom phones is enabled 
the forward/transfer does work but the caller does not hear any ringing. 
 During the time that the caller should hear ringing the * console 
produces pages of errors.
snip
..
Apr 30 08:41:03 NOTICE[2813]: channel.c:1314 ast_read: Dropping 
incompatible voice frame on Local/[EMAIL PROTECTED],2 of format g729 
since our native format has changed to ulaw
Apr 30 08:41:03 NOTICE[2813]: channel.c:1314 ast_read: Dropping 
incompatible voice frame on Local/[EMAIL PROTECTED],2 of format g729 
since our native format has changed to ulaw
..
/snip

I have tested this with the phones behind a PIX firewall with NAT, 
behind a PIX firewall without NAT, and without a firewall at all.  Nat 
is not the problem.

In the SIP.conf canreinvite=no so all traffic should be passing through 
the * server.

The problem seems to be in the translation of the G729 packets from the 
phone to the G711 packets to the router.   Only during the forwarding 
process is this a problem.

Here is a snip from the console when it worked.
(Note: it worked because I was ringing two phones with this line in my 
extensions.conf
(exten = --6081,1,Dial(SIP/--6081SIP/--6091,20)

=SNIP
 -- Executing Goto(SIP/---..241.35-40400490, TPN|--6081|1) 
in new stack
 -- Goto (TPN,--6081,1)
  -- Executing Dial(SIP/---.---.241.35-40400490, 
SIP/--6081SIP/--6091|20) in new stack
  -- Called --6081
  -- Called --6091
  -- Got SIP response 302 Moved Temporarily back from --.92.27
 -- Now forwarding SIP/---.---.---.35-40400490 to 
'Local/[EMAIL PROTECTED]' (thanks toSIP/--6091-6268)
 -- Executing Dial(Local/[EMAIL PROTECTED],2, 
SIP/[EMAIL PROTECTED]) in new stack
 -- Called [EMAIL PROTECTED]
 -- SIP/--6081-e558 is ringing
 -- SIP/---.---.241.35-f522 is making progress passing it to 
Local/[EMAIL PROTECTED],2
 -- Local/[EMAIL PROTECTED],1 is making progress passing it to 
SIP/---.---.241.35-40400490
 -- SIP/---.---.241.35-f522 answered Local/[EMAIL PROTECTED],2
 -- Local/[EMAIL PROTECTED],1 answered SIP/---.---.---.35-40400490
 == Spawn extension (TPN, --6081, 1) exited non-zero on 
'Local/[EMAIL PROTECTED],2ZOMBIE'
 -- Attempting native bridge of SIP/---.---.241.35-40400490 and 
SIP/---.---.241.35-f522
==/SNIP

Now here is the console output with a single phone defined in the 
extensions.conf
(exten = --6081,1,Dial(SIP/--6091,20)

*SNIP
Asterisk-A*CLI
-- Executing Goto(SIP/---.---.241.35-40418730, Charity|--3263|1) 
in new stack
-- Goto (Charity,---263,1)
-- Executing Dial(SIP/---.---.241.35-40418730, SIP/--3263|18) in 
new stack
-- Called --3263
-- Got SIP response 302 Moved Temporarily back from ---.---.243.5
-- Now forwarding SIP/---.---.241.35-40418730 to 
'Local/[EMAIL PROTECTED]' (thanks to SIP/--3263-f670)
-- Executing Dial(Local/[EMAIL PROTECTED],2, 
SIP/[EMAIL PROTECTED]) in new stack
 -- Called [EMAIL PROTECTED]
 -- SIP/---.---.241.35-36ca is making progress passing it to 
Local/[EMAIL PROTECTED],2
 -- Local/[EMAIL PROTECTED],1 is making progress passing it to 
SIP/---.---.241.35-40418730
Apr 29 11:30:03 NOTICE[2197]: channel.c:1314 ast_read: Dropping 
incompatible voice frame on Local/[EMAIL PROTECTED],2 of format 
g729 since our native format has changed to ulaw

pages of the same error

Apr 29 11:19:18 NOTICE[2185]: channel.c:1314 ast_read: Dropping 
incompatible voice frame on Local/[EMAIL PROTECTED],2 of format 
g729 since our native format has changed to ulaw
-- SIP/---.---.241.35-4e1f answered Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 answered 
SIP/---.---.241.35-40400490
-- Attempting native bridge of SIP/---.---.241.35-40400490 and 
SIP/---.---.241.35-4e1f
== Spawn exten (Charity, ---0059, 1) exited non-zero on 
'Local/[EMAIL PROTECTED],2'

*/SNIP
Im sure I could change everything to ulaw G711 the problem would go 
away but I do not want to do that.

Any Ideas?
Thanks
Scott H
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RE: [Asterisk-Users] Cisco 7960s and skinny

Andy

How did the 7910 worked with skinny under *? Did all the keys on the phone
worked? Ive seen sometimes the forward key or something does not fully do
what you would excpect.

What are the drawbacks from using skinny vs sip under *? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of Paul
|Sent: Miércoles, 27 de Abril de 2005 06:38 p.m.
|To: 'Andy Hamilton'
|Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Cisco 7960s and skinny
|
|Do you still have that image for the 7960? I bought a 7940 on 
|ebay and it doesn't have the SIP firmware. I can't find it 
|anywhere but Cisco's website and they require that I have an 
|account with them. Did you happen to save that binary file?
|
|Paul
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Andy Hamilton
|Sent: Tuesday, April 12, 2005 16:38
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Cisco 7960s and skinny
|
|Simon:
|
|I have had Skinny going on a 7960 (which I then reimaged to 
|SIP). I currently run a 7910 on Skinny (using chan_sccp) and 
|use the aforementioned 7960 simultaneously.
|
|Since you mentioned that you will have 50 phones, I assume you 
|are using them in a business setting.  I would *highly* 
|recommend using SIP, as I have found that the skinny driver is 
|not as reliable as it could be (not criticizing Jan or Julien 
|at all, here).
|
|Reimaging the 50 of them should only take a while (depending 
|on what version of CCM they have at the moment). I reimaged 12 
|phones once for a business and it took less than 30 minutes 
|after I got it going (toying with the phones to get them to 
|take the image, exactly how the config files were to be set 
|up, etc...).
|
|I imagine you could easily get the whole thing done in less 
|than a day (reimaging and config files), then figure out your dialplan.
|
|Then there is the whole issue of writing the config 
|files...but you'd have to do those with Skinny, anyhow.  I 
|think with SIP you'll have much better reliability.
|
|-Andy
|FWD: 428725
|
|On Apr 12, 2005 12:48 PM, Morris, Simon [EMAIL PROTECTED] wrote:
|  
| 
| Hello,
|  
|  Does anyone else have * running with Cisco 7960 phones and skinny?
|  
|  All the advise I am reading so far is telling me to load the SIP 
| image on the phone but I'd like to know what I'm going to lose by 
| persisting with skinny
|  
|  (Not reimaging 50 phones is one benefit amongst others of skinny)
|  
|  Thanks for any comparisons you can provide
|  
|  Rgds
|  
|  ~sm
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Re: [Asterisk-Users] Asterisk and sendmail

2005-04-30 Thread Derek Whitten

webmin works good for configuring sendmail too if you are not that
familiar with the sendmail 'mc' files...

http://www.webmin.com



On Sat, 2005-04-30 at 00:37, Craig Guy wrote:
 Sendmail isn't really that hard to configure for simple stuff like this.
 Most Linux distros have /etc/mail/sendmail.mc, so set your smart relay host
 and the appropriate masquerading options - the options for these are spelt
 out in the sendmail.mc file.  If you want to receive bounces then also set
 it to listen on your network interface.
 
 Craig
 
 - Original Message - 
 From: Chuck Keeter [EMAIL PROTECTED]
 To: asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Saturday, April 30, 2005 11:16 AM
 Subject: [Asterisk-Users] Asterisk and sendmail
 
 
  Hi all,
 
  Can someone point me in the right direction to configuring sendmail to
 work
  with Asterisk voicemail and faxes?
 
  I did  a bit of research on the web but came up more confused that when I
  started.
 
  It's the basic setup I'm having trouble with, where to add the SMTP and
  login and user name to sendmail to use a smart host in getting v-mail and
  faxes to the people they are going to.
 
  Thanks in advance.
 
  Chuck Keeter
 
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-- 
Derek Whitten [EMAIL PROTECTED]
kFuQ Productions


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Re: [Asterisk-Users] Dynamic phone groups.

2005-04-30 Thread Peter Svensson

On Sat, 30 Apr 2005, Joris Vandalon wrote:

 I am looking for a way to dynamicly put phones in a group so if someone
 calls an extentions everyone's phone who's member of the group will
 ring.

One way is to place the logic in an agi script. It can then dial all the 
current members of the group using the Dial(chanchanchan...) syntax.

Peter


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RE: RE: [Asterisk-Users] Problems with TDM400P card

The way that zttest is written makes it a little difficult to
interpret, but it essentially means that zttest tried to read
8192 bytes from the TDM card, and it took more then 1 second to
do it (the objective is exactly 1.0 seconds, or 100%).
The 99.987 numbers says it took something like 1.02 seconds
to read the 8192 bytes instead. Because it took about 21,
microseconds too long, frame slips are going to be happening
approximately every 10 seconds. (That's why spandsp doesn't work
right.)
I'm not sure (as yet) what the source of the delays are, but that's
what some of us are trying to figure out.

What OS distro are you using?


 Hows does this look?
 
 Opened pseudo zap interface, measuring accuracy...
 
 8192 samples in 8192 sample intervals 100.00%
 8192 samples in 8193 sample intervals 99.987793%
 8192 samples in 8193 sample intervals 99.987793%
 8192 samples in 8193 sample intervals 99.987793%
 8192 samples in 8193 sample intervals 99.987793%
 8192 samples in 8193 sample intervals 99.987793%
 8192 samples in 8193 sample intervals 99.987793%
 8192 samples in 8193 sample intervals 99.987793%
 8192 samples in 8193 sample intervals 99.987793%
 8192 samples in 8194 sample intervals 99.975586%
 8192 samples in 8193 sample intervals 99.987793%
 8192 samples in 8193 sample intervals 99.987793%
 8192 samples in 8193 sample intervals 99.987793%
 --- Results after 13 passes ---
 Best: 100.00 -- Worst: 99.975586 -- Average: 99.987793
 
 Good enough and what do I need to check in order to make 100%? What does the
 test actually measure?
 
  
 
 |-Original Message-
 |From: [EMAIL PROTECTED] 
 |[mailto:[EMAIL PROTECTED] On Behalf Of 
 |Kim Culhan
 |Sent: Sábado, 30 de Abril de 2005 08:45 a.m.
 |To: asterisk-users@lists.digium.com
 |Subject: Re: RE: [Asterisk-Users] Problems with TDM400P card
 |
 |On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said:
 | I would also be interested in alternatives to the Tdm400p. I 
 |have had 
 | endless problems with a tdm400p card not being able to get 
 |the zttest 
 | numbers above
 | 99.975 and as a result not being able eliminate an 
 |intermitent but consistent echo.
 | I have tried to date 4 different motherboard and hardware 
 |combinations 
 | as well as different linux versions to no avial.I would 
 |welcome some feedback on this.
 |
 |Since there appear to be several combinations of hardware and 
 |operating system which don't work well, here is a combination 
 |which appears to work fairly well:
 |
 |Intel 925XCV mb
 |
 |P-4 560 (3.6 gHz)
 |
 |wcfxs0: Wildcard TDM400P REV E/F
 |
 |FreeBSD 5.4-STABLE
 |
 |zttest -v
 |Opened pseudo zap interface, measuring accuracy...
 |
 |8192 samples in 8192 sample intervals 100.00%
 |8192 samples in 8192 sample intervals 100.00%
 |8192 samples in 8192 sample intervals 100.00%
 |8192 samples in 8192 sample intervals 100.00%
 |8192 samples in 8192 sample intervals 100.00%
 |8192 samples in 8192 sample intervals 100.00%
 |8192 samples in 8192 sample intervals 100.00%
 |8192 samples in 8192 sample intervals 100.00%
 |8192 samples in 8192 sample intervals 100.00%
 |8192 samples in 8192 sample intervals 100.00% ^C
 |--- Results after 10 passes ---
 |Best: 100.00 -- Worst: 100.00 -- Average: 100.00
 |
 |hope this helps
 |
 |-kim
 |
 |--
 |[EMAIL PROTECTED]
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 |   http://lists.digium.com/mailman/listinfo/asterisk-users
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 |
 
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---End of Original Message-


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RE: [Asterisk-Users] Queues configuration

Mmhh let me try that. Thx! 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kevin P. Fleming
|Sent: Jueves, 28 de Abril de 2005 11:02 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Queues configuration
|
|Anton Krall wrote:
|
| How do you do it? I mean, if a caller is already on the queue and 
| suddenly all agents logoff.. How do you make the caller fall out of 
| the queue and into an IVR where he can leave a message?
|
|Have you read the sample queues.conf file? There is an option 
|there called 'leavewhenempty' that does exactly that.
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RE: [Asterisk-Users] Queues configuration

Weird..

I also have joinwhenempty=no and user can still go into the queue without
any agents logged in.

Any ideas? Im using cvs head 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kevin P. Fleming
|Sent: Jueves, 28 de Abril de 2005 11:02 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Queues configuration
|
|Anton Krall wrote:
|
| How do you do it? I mean, if a caller is already on the queue and 
| suddenly all agents logoff.. How do you make the caller fall out of 
| the queue and into an IVR where he can leave a message?
|
|Have you read the sample queues.conf file? There is an option 
|there called 'leavewhenempty' that does exactly that.
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Re: [Asterisk-Users] Dynamic phone groups.

2005-04-30 Thread Joris Vandalon

Citeren Robert P. McKenzie [EMAIL PROTECTED]:

 It may not be exactly what you are after but I do something like this:

 extensions.conf


HOUSEPHONES=SIP/somepcSIP/anotherpcIAX2/desktopIAX2/someotherdesktopSIP/sipuraline1SIP/sipuraline2

 ; London Number - SIP Inbound provider
 exten = 1438645,1,Answer
 exten = 1438645,2,Dial(${HOUSEPHONES}|60|t)
 exten = 1438645,3,Voicemail(u50)


 Each phone listed above also has it's own extention, but the voicemail all
 goes to 50.  That way I can call any
 extension from anyother inside the house.  But calling 50 directly will make
 every phone ring.  Any phone not logged in
 will just be ignored and skipped.  The first phone to pick up gets it.  Call
 parking is on so if there is a need to
 transfer calls from one phone to another it can be done using parking.
The thing is that i want to add and remove phones dynamicly from the group with
astdb or so.

Cheers,
Joris
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RE: RE: [Asterisk-Users] Problems with TDM400P card

Im using RH9 and celerom 1.7 with 256 Mb RAM

Can you give me the detailed math on your calculations? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Rich Adamson
|Sent: Sábado, 30 de Abril de 2005 11:07 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: RE: [Asterisk-Users] Problems with TDM400P card
|
|The way that zttest is written makes it a little difficult to 
|interpret, but it essentially means that zttest tried to read
|8192 bytes from the TDM card, and it took more then 1 second 
|to do it (the objective is exactly 1.0 seconds, or 100%).
|The 99.987 numbers says it took something like 1.02 
|seconds to read the 8192 bytes instead. Because it took about 
|21, microseconds too long, frame slips are going to be 
|happening approximately every 10 seconds. (That's why spandsp 
|doesn't work
|right.)
|I'm not sure (as yet) what the source of the delays are, but 
|that's what some of us are trying to figure out.
|
|What OS distro are you using?
|
|
| Hows does this look?
| 
| Opened pseudo zap interface, measuring accuracy...
| 
| 8192 samples in 8192 sample intervals 100.00%
| 8192 samples in 8193 sample intervals 99.987793%
| 8192 samples in 8193 sample intervals 99.987793%
| 8192 samples in 8193 sample intervals 99.987793%
| 8192 samples in 8193 sample intervals 99.987793%
| 8192 samples in 8193 sample intervals 99.987793%
| 8192 samples in 8193 sample intervals 99.987793%
| 8192 samples in 8193 sample intervals 99.987793%
| 8192 samples in 8193 sample intervals 99.987793%
| 8192 samples in 8194 sample intervals 99.975586%
| 8192 samples in 8193 sample intervals 99.987793%
| 8192 samples in 8193 sample intervals 99.987793%
| 8192 samples in 8193 sample intervals 99.987793%
| --- Results after 13 passes ---
| Best: 100.00 -- Worst: 99.975586 -- Average: 99.987793
| 
| Good enough and what do I need to check in order to make 100%? What 
| does the test actually measure?
| 
|  
| 
| |-Original Message-
| |From: [EMAIL PROTECTED]
| |[mailto:[EMAIL PROTECTED] On Behalf Of Kim 
| |Culhan
| |Sent: Sábado, 30 de Abril de 2005 08:45 a.m.
| |To: asterisk-users@lists.digium.com
| |Subject: Re: RE: [Asterisk-Users] Problems with TDM400P card
| |
| |On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said:
| | I would also be interested in alternatives to the Tdm400p. I
| |have had
| | endless problems with a tdm400p card not being able to get
| |the zttest
| | numbers above
| | 99.975 and as a result not being able eliminate an
| |intermitent but consistent echo.
| | I have tried to date 4 different motherboard and hardware
| |combinations
| | as well as different linux versions to no avial.I would
| |welcome some feedback on this.
| |
| |Since there appear to be several combinations of hardware and 
| |operating system which don't work well, here is a combination which 
| |appears to work fairly well:
| |
| |Intel 925XCV mb
| |
| |P-4 560 (3.6 gHz)
| |
| |wcfxs0: Wildcard TDM400P REV E/F
| |
| |FreeBSD 5.4-STABLE
| |
| |zttest -v
| |Opened pseudo zap interface, measuring accuracy...
| |
| |8192 samples in 8192 sample intervals 100.00%
| |8192 samples in 8192 sample intervals 100.00%
| |8192 samples in 8192 sample intervals 100.00%
| |8192 samples in 8192 sample intervals 100.00%
| |8192 samples in 8192 sample intervals 100.00%
| |8192 samples in 8192 sample intervals 100.00%
| |8192 samples in 8192 sample intervals 100.00%
| |8192 samples in 8192 sample intervals 100.00%
| |8192 samples in 8192 sample intervals 100.00%
| |8192 samples in 8192 sample intervals 100.00% ^C
| |--- Results after 10 passes ---
| |Best: 100.00 -- Worst: 100.00 -- Average: 100.00
| |
| |hope this helps
| |
| |-kim
| |
| |--
| |[EMAIL PROTECTED]
| |___
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| |
| 
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|
|---End of Original Message-
|
|
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RE: RE: [Asterisk-Users] Problems with TDM400P card

Look at the ms = statements in the code. I'm trying to rewrite
the code right now to provide something more useful.


 Im using RH9 and celerom 1.7 with 256 Mb RAM
 
 Can you give me the detailed math on your calculations? 
 
 |-Original Message-
 |From: [EMAIL PROTECTED] 
 |[mailto:[EMAIL PROTECTED] On Behalf Of 
 |Rich Adamson
 |Sent: Sábado, 30 de Abril de 2005 11:07 a.m.
 |To: Asterisk Users Mailing List - Non-Commercial Discussion
 |Subject: RE: RE: [Asterisk-Users] Problems with TDM400P card
 |
 |The way that zttest is written makes it a little difficult to 
 |interpret, but it essentially means that zttest tried to read
 |8192 bytes from the TDM card, and it took more then 1 second 
 |to do it (the objective is exactly 1.0 seconds, or 100%).
 |The 99.987 numbers says it took something like 1.02 
 |seconds to read the 8192 bytes instead. Because it took about 
 |21, microseconds too long, frame slips are going to be 
 |happening approximately every 10 seconds. (That's why spandsp 
 |doesn't work
 |right.)
 |I'm not sure (as yet) what the source of the delays are, but 
 |that's what some of us are trying to figure out.
 |
 |What OS distro are you using?
 |
 |
 | Hows does this look?
 | 
 | Opened pseudo zap interface, measuring accuracy...
 | 
 | 8192 samples in 8192 sample intervals 100.00%
 | 8192 samples in 8193 sample intervals 99.987793%
 | 8192 samples in 8193 sample intervals 99.987793%
 | 8192 samples in 8193 sample intervals 99.987793%
 | 8192 samples in 8193 sample intervals 99.987793%
 | 8192 samples in 8193 sample intervals 99.987793%
 | 8192 samples in 8193 sample intervals 99.987793%
 | 8192 samples in 8193 sample intervals 99.987793%
 | 8192 samples in 8193 sample intervals 99.987793%
 | 8192 samples in 8194 sample intervals 99.975586%
 | 8192 samples in 8193 sample intervals 99.987793%
 | 8192 samples in 8193 sample intervals 99.987793%
 | 8192 samples in 8193 sample intervals 99.987793%
 | --- Results after 13 passes ---
 | Best: 100.00 -- Worst: 99.975586 -- Average: 99.987793
 | 
 | Good enough and what do I need to check in order to make 100%? What 
 | does the test actually measure?
 | 
 |  
 | 
 | |-Original Message-
 | |From: [EMAIL PROTECTED]
 | |[mailto:[EMAIL PROTECTED] On Behalf Of Kim 
 | |Culhan
 | |Sent: Sábado, 30 de Abril de 2005 08:45 a.m.
 | |To: asterisk-users@lists.digium.com
 | |Subject: Re: RE: [Asterisk-Users] Problems with TDM400P card
 | |
 | |On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said:
 | | I would also be interested in alternatives to the Tdm400p. I
 | |have had
 | | endless problems with a tdm400p card not being able to get
 | |the zttest
 | | numbers above
 | | 99.975 and as a result not being able eliminate an
 | |intermitent but consistent echo.
 | | I have tried to date 4 different motherboard and hardware
 | |combinations
 | | as well as different linux versions to no avial.I would
 | |welcome some feedback on this.
 | |
 | |Since there appear to be several combinations of hardware and 
 | |operating system which don't work well, here is a combination which 
 | |appears to work fairly well:
 | |
 | |Intel 925XCV mb
 | |
 | |P-4 560 (3.6 gHz)
 | |
 | |wcfxs0: Wildcard TDM400P REV E/F
 | |
 | |FreeBSD 5.4-STABLE
 | |
 | |zttest -v
 | |Opened pseudo zap interface, measuring accuracy...
 | |
 | |8192 samples in 8192 sample intervals 100.00%
 | |8192 samples in 8192 sample intervals 100.00%
 | |8192 samples in 8192 sample intervals 100.00%
 | |8192 samples in 8192 sample intervals 100.00%
 | |8192 samples in 8192 sample intervals 100.00%
 | |8192 samples in 8192 sample intervals 100.00%
 | |8192 samples in 8192 sample intervals 100.00%
 | |8192 samples in 8192 sample intervals 100.00%
 | |8192 samples in 8192 sample intervals 100.00%
 | |8192 samples in 8192 sample intervals 100.00% ^C
 | |--- Results after 10 passes ---
 | |Best: 100.00 -- Worst: 100.00 -- Average: 100.00
 | |
 | |hope this helps
 | |
 | |-kim
 | |
 | |--
 | |[EMAIL PROTECTED]
 | |___
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 | |Asterisk-Users@lists.digium.com
 | |http://lists.digium.com/mailman/listinfo/asterisk-users
 | |To UNSUBSCRIBE or update options visit:
 | |   http://lists.digium.com/mailman/listinfo/asterisk-users
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 | |
 | 
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 |
 |---End of Original Message-
 |
 |
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 |

[Asterisk-Users] IPSwitchBoard version 0.111 released

2005-04-30 Thread Thorben Jensen

Version 0.111 - 30. April 2005.

* Security added, you can now specify what the user of IPS is allowed to do
such as start different programs, hang-up calls etc. 
* Many bug fixes 

Download: http://ipswitchboard.thorben.dk


___
IPSwitchBoard is a FREE Windows.Net application that will allow you to: 

Unattended/attended transfers. 
Park calls and retrieve/forward them again. 
Organize all your Zap, SIP and IAX extensions (automatically retrieved from
Asterisk). 
Hotel/Call Shop Billing module 
Monitor all extensions. 
Monitor all queues. 
Monitor Agents. 
Monitor Parked Calls. 
Dynamically log extensions in and out of queues. 
Integration with CRM software on the web. 
Record conversations. 
Browse Call Records 
Drop any active call. 
Set Do Not Disturb on Extensions and give a reason. 
Speed Dialling. 
User selectable ring tones for IPSwitchBoard. 
User selectable button colors.

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[Asterisk-Users] 7910 and Skinny

2005-04-30 Thread Mark Johnson

I just had a very successful installation of Asterisk and have a 
question.  On my 7910's using the Skinny protocol, the user does not 
hear ringing when they make another call.  I found a patch that makes 
the ringing work, but something is still wrong with it.  If I use the 
7910 to make internal Skinny to other internal Skinny or SIP phones, the 
ringing works.  Once they make an outside call, they can not hear 
ringing again until I shutdown Asterisk and start it back up.  I'm using 
1.0.7.  Anyone have any ideas?  I also tried chan_sccp and that was a 
real disaster.  Asterisk kept crashing after a period of about 30 
minutes.  It was like when the phones reregistered so many times, it 
started claiming that some of the phones were dead and that others 
couldn't be registered because they already were, then it crashed.  
Anyone have any ideas?  Below is the patch code I found.

Mark
/@@ -1715,14 +1756,17 @@
   }
   switch(ind) {
   case AST_CONTROL_RINGING:
-   if (ast-_state == AST_STATE_RINGING) {
+   ast_verbose(VERBOSE_PREFIX_3 State AST_CONTROL_RINGINGn);
+   // if (ast-_state == AST_STATE_RINGING) {
+   ast_verbose(VERBOSE_PREFIX_3 State AST_STATE_RINGINGn);
   if (!sub-progress) {   
   transmit_tone(s, SKINNY_ALERT);
   transmit_callstate(s, l-instance, SKINNY_RINGOU
T, sub-callid);
   sub-ringing = 1;
+   ast_verbose(VERBOSE_PREFIX_3 Started Ringingn
);
   break;
   }
-   }
+   // }
/

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RE: [Asterisk-Users] Bouncing DTMF?

2005-04-30 Thread Jan Johansson

What type of phone SIP or analog? What is your DTMF type set for?

It's a system phone, via PBX to a PRI to Operator to my SIP Provider to my
Asterisk box.

Sip.conf is

[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=alaw
allow=gsm
;allow=ilbc
realm=asterisk
;
register=085000:[EMAIL PROTECTED]/1000

[rix]
type=peer
nat=yes
username=0850007696
fromuser=0850007696
secret=6ARiAnME
host=82.96.24.7
fromdomain=82.96.24.7
insecure=very



smime.p7s
Description: S/MIME cryptographic signature
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Re: [Asterisk-Users] IPSwitchBoard version 0.111 released

2005-04-30 Thread TC

what is the url for the version of the framework it wants now ?
the dot net auto installer is busted ?
- Original Message -
From: Thorben Jensen [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Saturday, April 30, 2005 8:39 AM
Subject: [Asterisk-Users] IPSwitchBoard version 0.111 released

 Version 0.111 - 30. April 2005.

 * Security added, you can now specify what the user of IPS is allowed to
do
 such as start different programs, hang-up calls etc.
 * Many bug fixes

 Download: http://ipswitchboard.thorben.dk

___
 IPSwitchBoard is a FREE Windows.Net application that will allow you to:

 Unattended/attended transfers.
 Park calls and retrieve/forward them again.
 Organize all your Zap, SIP and IAX extensions (automatically retrieved
from
 Asterisk).
 Hotel/Call Shop Billing module
 Monitor all extensions.
 Monitor all queues.
 Monitor Agents.
 Monitor Parked Calls.
 Dynamically log extensions in and out of queues.
 Integration with CRM software on the web.
 Record conversations.
 Browse Call Records
 Drop any active call.
 Set Do Not Disturb on Extensions and give a reason.
 Speed Dialling.
 User selectable ring tones for IPSwitchBoard.
 User selectable button colors.

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[Asterisk-Users] Re: chan_capi crashes asterisk

2005-04-30 Thread Stefan Tichy

On Fri, Apr 29, 2005 at 01:44:24AM +0200, Sebastian Voitzsch wrote:
 I can?t get chan_capi to work with any version of asterisk. I tried several 
 versions, all with the same effect: the phone rings, as soon as the call gets 
 answerd, asterisk crashes.

Certainly it is chan_capi 0.3.5, but which kernel and libcapi
(capi4linux) versions are used?


-- 
Stefan Tichy   [EMAIL PROTECTED]
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[Asterisk-Users] Re: IPSwitchBoard version 0.111 released

2005-04-30 Thread tgj

You will find the URL on my download page

thorben

TC [EMAIL PROTECTED] skrev i en meddelelse 
news:[EMAIL PROTECTED]
 what is the url for the version of the framework it wants now ?
 the dot net auto installer is busted ?
 - Original Message -
 From: Thorben Jensen [EMAIL PROTECTED]
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 asterisk-users@lists.digium.com
 Sent: Saturday, April 30, 2005 8:39 AM
 Subject: [Asterisk-Users] IPSwitchBoard version 0.111 released


 Version 0.111 - 30. April 2005.

 * Security added, you can now specify what the user of IPS is allowed to
 do
 such as start different programs, hang-up calls etc.
 * Many bug fixes

 Download: http://ipswitchboard.thorben.dk



 ___
 IPSwitchBoard is a FREE Windows.Net application that will allow you to:

 Unattended/attended transfers.
 Park calls and retrieve/forward them again.
 Organize all your Zap, SIP and IAX extensions (automatically retrieved
 from
 Asterisk).
 Hotel/Call Shop Billing module
 Monitor all extensions.
 Monitor all queues.
 Monitor Agents.
 Monitor Parked Calls.
 Dynamically log extensions in and out of queues.
 Integration with CRM software on the web.
 Record conversations.
 Browse Call Records
 Drop any active call.
 Set Do Not Disturb on Extensions and give a reason.
 Speed Dialling.
 User selectable ring tones for IPSwitchBoard.
 User selectable button colors.

 ___
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[Asterisk-Users] Grandstream Budgetone 100 Caller ID shows extension, not incoming Caller ID

2005-04-30 Thread Amit Sharma

Hi,
Is this a known problem with Grandstream Budgetone 100, I could see 
several people complaining about this but no answers.

Details
--
1) I am simply trying to go from one SIP extension to another, so the 
zapata.conf and zaptel.conf entries are irrelavant.
2) I added a NoOp(CALLERID=${CALLERID}), to my dial plan cand could see 
the Caller ID on the console, so asterisk is aware of the caller ID.
3) All ID's are simply numbers no fancy alphanumeric strings.

I have been looking for a solution for a quite some time and seem to 
have hit a wall, any pointers would be greatly appreciated.

Thanks,
Amit
sip.conf

[116]  ; Extension 1
type = friend context = sip-phone   username = 
116 fromuser = 116callerid = 116 116
host = 10.0.1.116  nat = no   
canreinvite=yes   dtmfmode = rfc-2833
[EMAIL PROTECTED] disallow=all   
allow=ulaw allow=alaw
  

  

[117]
type = friend  ; extension 2
context = sip-phoneusername = 117 fromuser = 
117 callerid = 117 117
host = 10.0.1.117 nat = no   
canreinvite=yes   dtmfmode = rfc-2833   
[EMAIL PROTECTED]   disallow=all 
allow=ulawallow=alaw
 extensions.conf
--
[macro-exten]
exten = s,1,NoOp(CALLERID=${CALLERID})
exten = s,2,Dial(SIP/${ARG1},20)
exten = s,3,Voicemail(u${ARG1})
exten = s,4,Hangup

[default]
exten = 116,1,Macro(exten,116)
exten = 117,1,Macro(exten,117)

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[Asterisk-Users] Chan_modem_*

2005-04-30 Thread Marco Supino

Hi,
I was looking for solutions for simple FXO cards, and came across the 
two modem channels in the asterisk channels/ dir, i assume they are 
there becuase someone made these two types of modems work as FXO (or are 
they there for other purpose ?),

does anyone have any info on these channels ? anyone has them working 
with any type of modem ? (aopen or bestdata).

Marco.
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[Asterisk-Users] Intel 536EP

2005-04-30 Thread Jeff

Forgive me if this has been asked before, I wasn't able to find any
clear answers in the archives.
Will the Intel 536EP function as a FXO? And if so, do I need to use a
different version of the Zaptel driver?
Any assistance would be great.
PS - that's 536EP, not 537EP.
Thanks!
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[Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line

2005-04-30 Thread Thomas Miller

I was wondering if there was a way to have incoming
calls to my PSTN line be transferred to a voip line?

I would like to make it so that as soon as the pstn
call is recieved it will switch the call to the voip
line, thus freeing up the pstn line to get more calls.
Kind of like roaming.

Tom

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Re: [Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line

2005-04-30 Thread Time Bandit

 I was wondering if there was a way to have incoming
 calls to my PSTN line be transferred to a voip line?
 
 I would like to make it so that as soon as the pstn
 call is recieved it will switch the call to the voip
 line, thus freeing up the pstn line to get more calls.
 Kind of like roaming.
If you have call transfer on your line, you can do it with somethin like this
(from the top of my head)
exten = s,1,Answer()
exten = s,2,Flash()
exten = s,3,SendDTMF(${MYVOIPNUM})
exten = s,4,Hangup()

Basically, you hook-flash the line (giving you a dialtone), compose
the number where you want the calls to be forwarded, then hangup the
line. The calling party will be connected to the destination and your
line will be free

hth
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[Asterisk-Users] help with compiling addons for cdr

2005-04-30 Thread forums









Im running Asterisk 1.0.7. Ive checked
out using cvscheckoutasterisk-addons.




When I make install I get the following
errors:



app_addon_sql_mysql.c:162:36: macro
AST_LIST_REMOVE requires 4 arguments, but only 3 given



Im using the default FC3 mysql:



mysql-server-3.23.58-16.FC3.1

perl-DBD-MySQL-2.9003-5

mysql-3.23.58-16.FC3.1

mysql-devel-3.23.58-16.FC3.1

php-mysql-4.3.11-2.4

libdbi-dbd-mysql-0.6.5-9

MySQL-python-0.9.2-4



Ive search wiki, etal and have
found a couple of references with a proposed patch file, but the patch file
fails too. 



I would appreciate any assistance.










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[Asterisk-Users] Amp extensions script

2005-04-30 Thread Michael D Schelin

Hi, Is there a script in amp for adding the extensions?  And can it be 
modified?  When adding a new extension it rewrites all of the 
information it the context blowing out my additions.
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RE: [Asterisk-Users] Amp extensions script

2005-04-30 Thread Robert Webb


 Hi, Is there a script in amp for adding the extensions?  And can it be
 modified?  When adding a new extension it rewrites all of the
 information it the context blowing out my additions.

You my want to try the AMP forum. Since they are the producers of AMP,
they may have a little better info.

http://sourceforge.net/forum/?group_id=121515



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RE: [Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line

2005-04-30 Thread Robert Webb


 I was wondering if there was a way to have incoming
 calls to my PSTN line be transferred to a voip line?

 I would like to make it so that as soon as the pstn
 call is recieved it will switch the call to the voip
 line, thus freeing up the pstn line to get more calls.
 Kind of like roaming.

 Tom


Why not just call forward everything to your Voip line and then run it
through *. Most all providers allow for at least two incoming calls at a
time. You would then have your PSTN line free for outgoing only and tie
it into a group with your Voip and save some outgoing VoIP minutes.

Robert

P.S. - This does work very well. It is what I am using at home with my
PSTN and myphonecompany.com



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Re: [Asterisk-Users] Cisco 7960s and skinny

2005-04-30 Thread Andy Hamilton

Anton:

I'll be able to get back to you Sunday night about specifics; the
phone is not where I am right now. Using chan_sccp, (I think November
2004 or so CVS Head) I know I can receive calls, place calls, etc. It
is a rather low volume phone, so I don't know off hand about specific
keys; I'll check those later.
Additionally, I have not yet tried a new copy from CVS.

Occasionally, I think the chan_sccp driver blips out in Asterisk (it
may be the phone; I've had it apart several times because the on/off
hook switch membrane is a little sketchy). I have dealt with this by
restarting Asterisk. The only other thing I can say right now about
the 7910 is that it and my Cisco FastHub don't get along. At all. I
have the 7910 plugged into my 7960.

Overall, I would say that if you have a non-critical system and would
like to use a 7910, chan_sccp should be able to handle it fine. 
However, if you budget permits, the 7960 and 7940 phones are quite
nice (use SIP with those -- it's far more reliable. I must say,
though, that my 7960 has frozen/crashed a handful of time when running
the SIP image. That was the phone itself, Asterisk was fine.) I have
yet to purchase a 7905 or 7912, but I've played around with some
7912's on a CCM system -- they seem quite nice and I think they take
SIP. The 7920 is also nice because it's wireless. However, I don't
think Cisco has anything but a Skinny image for it [yet].

I would stick with SIP wherever you can.

-Andy



On 4/30/05, Anton Krall [EMAIL PROTECTED] wrote:
 Andy
 
 How did the 7910 worked with skinny under *? Did all the keys on the phone
 worked? Ive seen sometimes the forward key or something does not fully do
 what you would excpect.
 
 What are the drawbacks from using skinny vs sip under *?

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Re: [Asterisk-Users] Group/Broadcast Voicemail

2005-04-30 Thread Eric Wieling aka ManxPower

in app_voicemail.c in the function vm_exec set the tmp[256] to be 
tmp[4096]

Chris Stinson wrote:
I have one with 33. but I can't get the voicemail to copy to more than 
20 mailboxes.

Eric Wieling aka ManxPower wrote:
Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes?
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line

2005-04-30 Thread Joseph

On Sat, 2005-04-30 at 13:23 -0400, Time Bandit wrote:
  I was wondering if there was a way to have incoming
  calls to my PSTN line be transferred to a voip line?
  
  I would like to make it so that as soon as the pstn
  call is recieved it will switch the call to the voip
  line, thus freeing up the pstn line to get more calls.
  Kind of like roaming.
 If you have call transfer on your line, you can do it with somethin like this
 (from the top of my head)
 exten = s,1,Answer()
 exten = s,2,Flash()
 exten = s,3,SendDTMF(${MYVOIPNUM})
 exten = s,4,Hangup()
 
 Basically, you hook-flash the line (giving you a dialtone), compose
 the number where you want the calls to be forwarded, then hangup the
 line. The calling party will be connected to the destination and your
 line will be free

Correct me anybody if I'm wrong; but I think this way he will only free
his internal extension not the PSTN line.
I think the only way of doing this is to order call forward feature form
his PSTN service provider, so when the call comes and his number is busy
it will automatically redirect it to let say DID number over IP (or any
other number). 

-- 
#Joseph
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Re: [Asterisk-Users] help with compiling addons for cdr

2005-04-30 Thread Matthew Boehm

Well for some reason, you decided to use the stable version of asterisk but
also decided not to use the stable version of addons. Hmm...interesting
decisions.

rm -rf asterisk-addons/
cvs co addons -r v1.0.7

Then it will work.

-Matthew

 From: forums [EMAIL PROTECTED]
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Sat, 30 Apr 2005 10:24:24 -0700
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] help with compiling addons for cdr
 
 I'm running Asterisk 1.0.7.  I've checked out using cvs checkout
 asterisk-addons.
 
  
 
 When I make install I get the following errors:
 
  
 
 app_addon_sql_mysql.c:162:36: macro AST_LIST_REMOVE requires 4 arguments,
 but only 3 given
 
  
 
 I'm using the default FC3 mysql:
 
  
 
 mysql-server-3.23.58-16.FC3.1
 
 perl-DBD-MySQL-2.9003-5
 
 mysql-3.23.58-16.FC3.1
 
 mysql-devel-3.23.58-16.FC3.1
 
 php-mysql-4.3.11-2.4
 
 libdbi-dbd-mysql-0.6.5-9
 
 MySQL-python-0.9.2-4
 
  
 
 I've search wiki, etal and have found a couple of references with a proposed
 patch file, but the patch file fails too.
 
  
 
 I would appreciate any assistance.
 
  
 
  
 
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[Asterisk-Users] ANNOUNCEMENT: Asterisk-java 0.1 released

2005-04-30 Thread Stefan Reuter

Asterisk-java 0.1 a Java control for the Asterisk PBX has been released.

The Asterisk-java package consists of a set of Java classes that allow
you to easily build Java applications that interact with an Asterisk
PBX Server. Asterisk-java supports both interfaces that Asterisk
provides for this scenario: The FastAGI  protocol and the Manager API.

The FastAGI implementation supports all commands currently available
from Asterisk.

The Manager API implementation supports receiving events from the
Asterisk server (e.g. call progess, registered peers, channel state)
and sending actions to Asterisk (e.g. originate call,
agent login/logoff, start/stop voice recording).

Asterisk-java is available under Apache 2.0 license at
http://asterisk-java.sourceforge.net



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Re: [Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line

2005-04-30 Thread Eric Wieling aka ManxPower

Joseph wrote:
On Sat, 2005-04-30 at 13:23 -0400, Time Bandit wrote:
I was wondering if there was a way to have incoming
calls to my PSTN line be transferred to a voip line?
I would like to make it so that as soon as the pstn
call is recieved it will switch the call to the voip
line, thus freeing up the pstn line to get more calls.
Kind of like roaming.
If you have call transfer on your line, you can do it with somethin like 
this
(from the top of my head)
exten = s,1,Answer()
exten = s,2,Flash()
exten = s,3,SendDTMF(${MYVOIPNUM})
exten = s,4,Hangup()
Basically, you hook-flash the line (giving you a dialtone), compose
the number where you want the calls to be forwarded, then hangup the
line. The calling party will be connected to the destination and your
line will be free

Correct me anybody if I'm wrong; but I think this way he will only free
his internal extension not the PSTN line.
I think the only way of doing this is to order call forward feature form
his PSTN service provider, so when the call comes and his number is busy
it will automatically redirect it to let say DID number over IP (or any
other number). 

The better way would be to order Call Forward Busy Line from their 
telco.  In the example above when the hangup() happens both legs of 
the call will be disconneced.  The person should order 
Conference/Drop/Transfer service from his/her telco, rather than the 
traditional Three-Way Calling service if they really want to transfer 
the call.

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] Grandstream Budgetone 100 Caller ID shows extension, not incoming Caller ID

2005-04-30 Thread Brian Capouch

Amit Sharma wrote:
Hi,
Is this a known problem with Grandstream Budgetone 100, I could see 
several people complaining about this but no answers.

Details
--
1) I am simply trying to go from one SIP extension to another, so the 
zapata.conf and zaptel.conf entries are irrelavant.
2) I added a NoOp(CALLERID=${CALLERID}), to my dial plan cand could see 
the Caller ID on the console, so asterisk is aware of the caller ID.
3) All ID's are simply numbers no fancy alphanumeric strings.

I have been looking for a solution for a quite some time and seem to 
have hit a wall, any pointers would be greatly appreciated.

The BT101 doesn't display alpha characters.
B.
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[Asterisk-Users] Zaptel and Boostringer

2005-04-30 Thread Ian Pattison

Hi All,

For some time now I've had issues with ringing voltages on my TDM400P. Numerous 
folks have told me that using modprobe wcfxs boostringer=1  when loading the 
module will force the driver to use boosted ring voltage. For some strange 
reason this has never worked for me. Today I got creative... another way to do 
it is to edit wcfxs.c (in the zaptel CVS) and find the following block of 
declarations:

static int debug = 0;
static int robust = 0;
static int timingonly = 0;
static int lowpower = 0;
static int boostringer = 0;
static int _opermode = 0;
static char *opermode = FCC;
static int fxshonormode = 0;

set boostringer=1 instead of 0 and recompile Zaptel. The FXS ports will be 
forced to generate 89V ring signals from now on.

Now if I can just stop the FXO ports from dropping calls

Thanks,

Ian

Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: [EMAIL PROTECTED] 
WWW: http://www.technologyassociates.ca

BEGIN:VCARD
VERSION:2.1
FN:Ian Pattison
EMAIL;WORK;PREF:[EMAIL PROTECTED]
TEL;WORK:416-657-2464 ext. 204
N:Pattison;Ian
TITLE:Senior Analyst
ADR;INTL;WORK;PARCEL;POSTAL:;;9052 Creditview Rd.;Brampton;Ontario;L6V 
1A1;Canada
LABEL;INTL;WORK;PARCEL;POSTAL;ENCODING=QUOTED-PRINTABLE:Ian Pattison=0A=
9052 Creditview Rd.=0A=
Brampton, Ontario  L6V 1A1=0A=
Canada
LABEL;DOM;WORK;PARCEL;POSTAL;ENCODING=QUOTED-PRINTABLE:Ian Pattison=0A=
9052 Creditview Rd.=0A=
Brampton, Ontario  L6V 1A1
TEL;CELL:416-568-6548
TEL;PREF:416-657-2464 ext. 204
TEL;WORK:905-459-2100 ext. 204
ORG:Technology Associates Inc.
END:VCARD

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[Asterisk-Users] X-lite and * behind Firewalls

2005-04-30 Thread Julio Zavala









Hi All, i´m new in this list.



I have an Asterisk behind a firewall with forwarded ports
and my SIP clients is X-Lite.



In local connection we dont have problem, and the
same with VPN connection. All work fine.



But when I try to connect to * from Internet or from others
LANs, the connection some time is successfully but the audio from * dont
work.



ASTERISK  Firewall  Internet (SIP Client)
 Firewall  Other LAN (SIP Client)



I have NAT=Yes in each configuration of SIP Phone in SIP.CONF

I Have externip = asterisk.mydomain.com and internalip = 172.16.10.40



May be I have missed somes firewalls ports,
or my setup in asterisk isnt complete



Ports Forwarded in my firewall.



Tipo   Internal Server  Type  External/Internal
Port

SIP   172.16.10.40     *   
5060/5060

MGCP  172.16.10.40     UDP     2727/2727

X-Lite    172.16.10.40     UDP     3478/3478

xlite  172.16.10.40     UDP     8000/8000 


xlite 2   172.16.10.40     UDP     8001/8001 


SIP 2    172.16.10.40     UDP     5061/5061 


sip 3     172.16.10.40     *   631/631



Ports Opened in my firewall  

Allow UDP ASTERISK Wan,* LAN,172.16.10.40  FROM
WAN/LAN, 1-2



Cordialmente




 
  
  Julio Zavala A.
  
  
  
  
  Servicios Triactivos
  
 






_
Servicios Triactivos Limitada - www.triactivos.cl - Proyectos - Ingeniería - Servicios - Telefonía IP - Redes



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[Asterisk-Users] A good SIP receptionist phone

2005-04-30 Thread Jason Brown

I have a problem. The average person is too freaking stupid to use a VOIP 
phone. My experience has so far been that if it doesn't have 20 buttons with 
little red LED's on it, the user cannot comprehend call parking, attended 
transfer, blind transfer, DND, and navigating through a voicemail menu.

I need a good receptionist phone that works with Asterisk. It basically needs 
to act like an avaya partner phone, I don't need 20 buttons with little red 
LED's...what I do need is for the phone to register multiple extensions to my 
asterisk server and act like each SIP extension is a line, so if the idiot 
receptionist has a call ringing in on line 1, she can pick it up, look at the 
buttons, see a call ringing in on line 2 (and the phone ringer rings), put call 
1 on hold without hanging the caller up, and hit the little I am an idiot and 
need a line 2 button to pick up line 2, so on and so forth.

I love VOIP systems and all the functionality they bring and features I get. 
Unfortunately, the average person in this country anymore is apparently 
completely stupid and cannot understand how to juggle calls without hanging up 
on people.

/rant

So seriously does anyone have a recommendation for a good receptionist phone? I 
tried the Snom today and I can't get the programmable buttons to do this, even 
by following the manual. So please, any suggestions would be great, before I 
get fired at my dayjob for everyone else's idiocy.
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Re: [Asterisk-Users] A good SIP receptionist phone

2005-04-30 Thread Eric Wieling aka ManxPower

Jason Brown wrote:
So seriously does anyone have a recommendation for a good receptionist phone? I tried the Snom today and I can't get the programmable buttons to do this, even by following the manual. So please, any suggestions would be great, before I get fired at my dayjob for everyone else's idiocy.
My suggestion is to get a good receptionist.  The receptionists at my 
customers are consistantly more technology oriented than other 
employees.

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] A good SIP receptionist phone

2005-04-30 Thread Mike Dent

On 4/30/05, Jason Brown [EMAIL PROTECTED] wrote:
 I have a problem. The average person is too freaking stupid to use a VOIP 
 phone. My experience has so far been that if it doesn't have 20 buttons with 
 little red LED's on it, the user cannot comprehend call parking, attended 
 transfer, blind transfer, DND, and navigating through a voicemail menu.
 
 I need a good receptionist phone that works with Asterisk. It basically needs 
 to act like an avaya partner phone, I don't need 20 buttons with little red 
 LED's...what I do need is for the phone to register multiple extensions to my 
 asterisk server and act like each SIP extension is a line, so if the idiot 
 receptionist has a call ringing in on line 1, she can pick it up, look at the 
 buttons, see a call ringing in on line 2 (and the phone ringer rings), put 
 call 1 on hold without hanging the caller up, and hit the little I am an 
 idiot and need a line 2 button to pick up line 2, so on and so forth.
 
 I love VOIP systems and all the functionality they bring and features I get. 
 Unfortunately, the average person in this country anymore is apparently 
 completely stupid and cannot understand how to juggle calls without hanging 
 up on people.
 
 /rant
 
 So seriously does anyone have a recommendation for a good receptionist phone? 
 I tried the Snom today and I can't get the programmable buttons to do this, 
 even by following the manual. So please, any suggestions would be great, 
 before I get fired at my dayjob for everyone else's idiocy.
 
 

Hi,
the Cisco 7960 (6 SIP lines) or the 7940 (2 lines) does what you wan, i think!
I have one here which is registered with 6 different extensions on my * box.
I can switch between calls on different buttons.

Mike

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Re: [Asterisk-Users] A good SIP receptionist phone

2005-04-30 Thread Ron Wellsted

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Jason Brown wrote:
 I have a problem. The average person is too freaking stupid to use a VOIP 
 phone. My experience has so far been that if it doesn't have 20 buttons with 
 little red LED's on it, the user cannot comprehend call parking, attended 
 transfer, blind transfer, DND, and navigating through a voicemail menu.
 
 I need a good receptionist phone that works with Asterisk. It basically needs 
 to act like an avaya partner phone, I don't need 20 buttons with little red 
 LED's...what I do need is for the phone to register multiple extensions to my 
 asterisk server and act like each SIP extension is a line, so if the idiot 
 receptionist has a call ringing in on line 1, she can pick it up, look at the 
 buttons, see a call ringing in on line 2 (and the phone ringer rings), put 
 call 1 on hold without hanging the caller up, and hit the little I am an 
 idiot and need a line 2 button to pick up line 2, so on and so forth.
 
 I love VOIP systems and all the functionality they bring and features I get. 
 Unfortunately, the average person in this country anymore is apparently 
 completely stupid and cannot understand how to juggle calls without hanging 
 up on people.
 
 /rant
 
 So seriously does anyone have a recommendation for a good receptionist phone? 
 I tried the Snom today and I can't get the programmable buttons to do this, 
 even by following the manual. So please, any suggestions would be great, 
 before I get fired at my dayjob for everyone else's idiocy.

How many lines do you need?

The Cisco 7960 gives you 6, with call waiting you can get 2 calls on
each line.

You have to trade off between incoming lines and speed dials, unless you
can train the monkey^w receptionist (sorry, unfair to simians there) to
use the directories.

Seriously, you may need to look deeper here on the human side. Could
this be a people problem in that the receptionist does not want to
learn/is a friend/relative of a PBX supplier who is being usurped?  Have
you made an enemy of this person?

We have just switched over to Asterisk with 7960s. We have had a few
little problems but have not lost a call yet.  OK, we have left a few
callers on hold a bit longer than we intended, once or twice ;)

- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
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Re: [Asterisk-Users] A good SIP receptionist phone

2005-04-30 Thread Mike Clark

Jason Brown wrote:
I have a problem. The average person is too freaking stupid to use a VOIP
phone. My experience has so far been that if it doesn't have 20 buttons with
little red LED's on it, the user cannot comprehend call parking, attended
transfer, blind transfer, DND, and navigating through a voicemail menu.
I need a good receptionist phone that works with Asterisk. It basically needs to act like
an avaya partner phone, I don't need 20 buttons with little red LED's...what I do need is
for the phone to register multiple extensions to my asterisk server and act like each SIP
extension is a line, so if the idiot receptionist has a call ringing in on line 1, she
can pick it up, look at the buttons, see a call ringing in on line 2 (and the phone
ringer rings), put call 1 on hold without hanging the caller up, and hit the little
I am an idiot and need a line 2 button to pick up line 2, so on and so forth.
I love VOIP systems and all the functionality they bring and features I get.
Unfortunately, the average person in this country anymore is apparently
completely stupid and cannot understand how to juggle calls without hanging up
on people.
/rant
So seriously does anyone have a recommendation for a good receptionist phone? I tried the Snom today and I can't get the programmable buttons to do this, even by following the manual. So please, any suggestions would be great, before I get fired at my dayjob for everyone else's idiocy.

Take a look at the Polycom IP 600
Mike Clark
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Re: [Asterisk-Users] A good SIP receptionist phone

2005-04-30 Thread Michael Welter

So seriously does anyone have a recommendation for a good receptionist phone? I tried the Snom today and I can't get the programmable buttons to do this, even by following the manual. So please, any suggestions would be great, before I get fired at my dayjob for everyone else's idiocy.

How many lines do you need?
The Cisco 7960 gives you 6, with call waiting you can get 2 calls on
each line.
You have to trade off between incoming lines and speed dials, unless you
can train the monkey^w receptionist (sorry, unfair to simians there) to
use the directories.
Seriously, you may need to look deeper here on the human side. Could
this be a people problem in that the receptionist does not want to
learn/is a friend/relative of a PBX supplier who is being usurped?  Have
you made an enemy of this person?
We have just switched over to Asterisk with 7960s. We have had a few
little problems but have not lost a call yet.  OK, we have left a few
callers on hold a bit longer than we intended, once or twice ;)
In a multi-tenant environment, is there a way to display, on the phone, 
which DID (which tenant) is being called?

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Re: [Asterisk-Users] A good SIP receptionist phone

2005-04-30 Thread Florian Overkamp

Hi,

Citeren Jason Brown [EMAIL PROTECTED]:

 I need a good receptionist phone that works with Asterisk. It basically needs
 to act like an avaya partner phone, I don't need 20 buttons with little red
 LED's...what I do need is for the phone to register multiple extensions to my
 asterisk server and act like each SIP extension is a line, so if the idiot
 receptionist has a call ringing in on line 1, she can pick it up, look at the
 buttons, see a call ringing in on line 2 (and the phone ringer rings), put
 call 1 on hold without hanging the caller up, and hit the little I am an
 idiot and need a line 2 button to pick up line 2, so on and so forth.

Take a look at the SNOM220 phone. They come with an optional side panel to add
line or speed dial keys.

Florian
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Re: [Asterisk-Users] A good SIP receptionist phone

2005-04-30 Thread Florian Overkamp

Hi,

Citeren Michael Welter [EMAIL PROTECTED]:

 In a multi-tenant environment, is there a way to display, on the phone,
 which DID (which tenant) is being called?

We use the callerID name for that purpose.

Florian
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Re: [Asterisk-Users] Pattern Matching

2005-04-30 Thread Mojo Jojo

Not sure what you mean exactly... Can you give me a hint?
Private Label Wholesale Internet Access!
http://www.YourOwnISP.com
- Original Message - 
From: Michael D Schelin [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Friday, April 29, 2005 10:10 PM
Subject: Re: [Asterisk-Users] Pattern Matching

Hey Mojo, I'm thinking you might try using priorty 's to set some kind 
routing. just a thought..

Mojo Jojo wrote:
We recently had our PRI installed, we currently have 100 toll-free's 
pointing to it.

I have almost everything working great but..
I have setup the first few numbers we want to use coming in from the PRI 
and they work great, but..

What I want to do is setup an extension with pattern matching to answer 
for any numbers called that are pointed to our system and PRI but not yet 
in use/configured.

I have been successful at setting up pattern matching as a catch all for 
98 or so numbers not in use yet and I have been successful setting up the 
2 numbers I want to make use of for now.

Problem is, I can't use both at the same time!
If I turn on the pattern matching then my greeting plays for the 
configured number, then the message plays for the invalid number 
(basically executing the extension with the pattern matching).

I have read about sorting with pattern matching by using an include, I 
did this but it's not really helping.

I have set a response timeout after the first extension plays it's 
greeting, I would think it should wait until it times out but it doesn't, 
it just immediately moves to the pattern matched extension.

I must be missing something big here..
Any help is appreciated..
--
Private Label Wholesale Internet Access!
http://www.YourOwnISP.com
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[Asterisk-Users] Send DTMF AFTER channels are bridged

2005-04-30 Thread Shady

Someone to know how can I send a DTMF after the channels are bridged?
I need something like the D option of the Dial application, but this
option sends the DTMF before the channels are bridged. In fact I want the
caller and the callee to receive the DTMF. Please help :)


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RE: [Asterisk-Users] Cisco 7960s and skinny

Thank you for the detailed description Andy.

Please let me know how about the specs when you can.

My client has legacy 7610 but I am trying to suggest swithcing to native sip
phones like grandstream or better in order to make everything 100% asterisk
compliant.

Plus, Cisco charging for the sip images and such ($150) doesnt look good,
for that price you can get some SIP phones.

 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Andy Hamilton
|Sent: Sábado, 30 de Abril de 2005 01:00 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Cisco 7960s and skinny
|
|Anton:
|
|I'll be able to get back to you Sunday night about specifics; 
|the phone is not where I am right now. Using chan_sccp, (I 
|think November
|2004 or so CVS Head) I know I can receive calls, place calls, 
|etc. It is a rather low volume phone, so I don't know off hand 
|about specific keys; I'll check those later.
|Additionally, I have not yet tried a new copy from CVS.
|
|Occasionally, I think the chan_sccp driver blips out in 
|Asterisk (it may be the phone; I've had it apart several times 
|because the on/off hook switch membrane is a little sketchy). 
|I have dealt with this by restarting Asterisk. The only other 
|thing I can say right now about the 7910 is that it and my 
|Cisco FastHub don't get along. At all. I have the 7910 plugged 
|into my 7960.
|
|Overall, I would say that if you have a non-critical system 
|and would like to use a 7910, chan_sccp should be able to 
|handle it fine. 
|However, if you budget permits, the 7960 and 7940 phones are 
|quite nice (use SIP with those -- it's far more reliable. I 
|must say, though, that my 7960 has frozen/crashed a handful of 
|time when running the SIP image. That was the phone itself, 
|Asterisk was fine.) I have yet to purchase a 7905 or 7912, but 
|I've played around with some 7912's on a CCM system -- they 
|seem quite nice and I think they take SIP. The 7920 is also 
|nice because it's wireless. However, I don't think Cisco has 
|anything but a Skinny image for it [yet].
|
|I would stick with SIP wherever you can.
|
|-Andy
|
|
|
|On 4/30/05, Anton Krall [EMAIL PROTECTED] wrote:
| Andy
| 
| How did the 7910 worked with skinny under *? Did all the keys on the 
| phone worked? Ive seen sometimes the forward key or 
|something does not 
| fully do what you would excpect.
| 
| What are the drawbacks from using skinny vs sip under *?
|
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RE: [Asterisk-Users] help with compiling addons for cdr

2005-04-30 Thread Jim Sturtevant

:-) ok... so I feel foolish... Mathew thanks a lot, worked like a charm.

Jim


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Saturday, April 30, 2005 11:16 AM
To: Asterisk Users
Subject: Re: [Asterisk-Users] help with compiling addons for cdr

Well for some reason, you decided to use the stable version of asterisk but
also decided not to use the stable version of addons. Hmm...interesting
decisions.

rm -rf asterisk-addons/
cvs co addons -r v1.0.7

Then it will work.

-Matthew

 From: forums [EMAIL PROTECTED]
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Sat, 30 Apr 2005 10:24:24 -0700
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] help with compiling addons for cdr
 
 I'm running Asterisk 1.0.7.  I've checked out using cvs checkout
 asterisk-addons.
 
  
 
 When I make install I get the following errors:
 
  
 
 app_addon_sql_mysql.c:162:36: macro AST_LIST_REMOVE requires 4
arguments,
 but only 3 given
 
  
 
 I'm using the default FC3 mysql:
 
  
 
 mysql-server-3.23.58-16.FC3.1
 
 perl-DBD-MySQL-2.9003-5
 
 mysql-3.23.58-16.FC3.1
 
 mysql-devel-3.23.58-16.FC3.1
 
 php-mysql-4.3.11-2.4
 
 libdbi-dbd-mysql-0.6.5-9
 
 MySQL-python-0.9.2-4
 
  
 
 I've search wiki, etal and have found a couple of references with a
proposed
 patch file, but the patch file fails too.
 
  
 
 I would appreciate any assistance.
 
  
 
  
 
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[Asterisk-Users] Install Asterisk on CCM MCS-7835 Server

2005-04-30 Thread Walid Azab


Hi All,

I am replacing Cisco Call Manager with Asterisk. As you know CCM
is on a MCS 7835 Server which comes with a custom version of
Windows. Does any one know how to install Linux on that H/W. My
guess is that someone must have tried the same thing before. I
know how to install Linux however I cannot get passed the H/W
limitation. 



Walid
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[Asterisk-Users] Hotel CDR Software

Guys.

Anybody knows of asterisk compliant cdr software for Hotel that will let you
enter diff. rates, checkin and out that will create the extension and setup
voicemail for the room, etc?


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[Asterisk-Users] Call-park timeouts..









 If a person parks a call, the call hits the
timeout exten for that context after the park expires.. Is there any way to
make it ring back to the person who parked the call instead of using the
timeout?








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RE: [Asterisk-Users] A good SIP receptionist phone

Can you show us an example of using the callerID for this purpose?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Florian
Overkamp
Sent: Saturday, April 30, 2005 3:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] A good SIP receptionist phone

Hi,

Citeren Michael Welter [EMAIL PROTECTED]:

 In a multi-tenant environment, is there a way to display, on the phone,
 which DID (which tenant) is being called?

We use the callerID name for that purpose.

Florian
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RE: [Asterisk-Users] Pattern Matching









Like this:



[dids]

Exten = 2145550001,1,dial(SIP/6001)

Exten = 2145550002,1,dial(SIP/6002)

Exten = 2145550003,1,dial(SIP/6003)

Include = default-did



[default-did]

Exten = _.,1,dial(SIP/6000)





Seems pretty simple. I used this method of least/highest cost routing
to choose my LD carrier. Should work the same though.





http://www.voip-info.org/tiki-index.php?page=Asterisk%20least%20cost%20routing%20using%20broadvoice







-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo Jojo
Sent: Saturday, April 30, 2005 3:08 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Pattern Matching



Not sure what you mean exactly... Can you give me a hint?





Private Label Wholesale Internet Access!

http://www.YourOwnISP.com



- Original Message - 

From: Michael D Schelin [EMAIL PROTECTED]

To: Asterisk Users Mailing List - Non-Commercial Discussion


asterisk-users@lists.digium.com

Sent: Friday, April 29, 2005 10:10 PM

Subject: Re: [Asterisk-Users] Pattern Matching





 Hey Mojo, I'm thinking you might try using priorty 's to set some
kind 

 routing. just a thought..







 Mojo Jojo wrote:



 We recently had our PRI installed, we currently have 100
toll-free's 

 pointing to it.



 I have almost everything working great but..



 I have setup the first few numbers we want to use coming in
from the PRI 

 and they work great, but..



 What I want to do is setup an extension with pattern matching
to answer 

 for any numbers called that are pointed to our system and PRI
but not yet 

 in use/configured.



 I have been successful at setting up pattern matching as a
catch all for 

 98 or so numbers not in use yet and I have been successful
setting up the 

 2 numbers I want to make use of for now.



 Problem is, I can't use both at the same time!



 If I turn on the pattern matching then my greeting plays for
the 

 configured number, then the message plays for the invalid
number 

 (basically executing the extension with the pattern matching).



 I have read about sorting with pattern matching by using an
include, I 

 did this but it's not really helping.



 I have set a response timeout after the first extension plays
it's 

 greeting, I would think it should wait until it times out but
it doesn't, 

 it just immediately moves to the pattern matched extension.



 I must be missing something big here..



 Any help is appreciated..





 -- 

 Private Label Wholesale Internet Access!

 http://www.YourOwnISP.com



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RE: [Asterisk-Users] A good SIP receptionist phone

2005-04-30 Thread Chris Mason (Lists)

 Take a look at the Polycom IP 600

I just added one to my desk as a test unit, I can't image you would need
anything more. We have Mitel 4015/4025/Superset for the office pbx I will be
replacing with *, and the Polycom 600 is a much better unit by far.

Chris Mason
www.anguillaguide.com
 

 
 

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[Asterisk-Users] Asterisk on Radio Tonight

2005-04-30 Thread Kerry Garrison

Kerry Garrison from The Geek Gazette (http://geekgazette.com)  will be
interviewed tonight on Mick Mick Williams' Cyber Line radio program at
9:00PM PST. The show is broadcast on the USA Radio network. If you do not
have a channel in your area, you can listen listen live online
http://www.usaradio.com/listen_live.htm. The show will cover the basic of
what the Asterisk PBX is all about and what it takes to implement a system.

-Kerry


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[Asterisk-Users] Problem with PSTN

2005-04-30 Thread Salina Jain

Hi,
I am a new user of Linux and Asterisk. I bought Digium TDM400P card and now 
want to setup my dial plan. With some help from the suggestions given online 
I have been able to configure the two SIP phones to interact with each 
other.

I want to use this to call on to a Telecom line(PSTN) and vice versa. I read 
somewhere that we need to use some provider for it like FWD or iconnect, do 
we need to use them to make outgoing and incoming calls to PSTN lines or we 
can do it without them. I can post my .conf files if anybody needs them to 
help me out with this. I don't know what should I put in the .conf files so 
that it enables these calls.

Any amount of help or suggestions would really be appreciated.
Thanx,
Salina
_
News, views and gossip. http://www.msn.co.in/Cinema/ Get it all at MSN 
Cinema!

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[Asterisk-Users] Problem with Sangoma/Adtran 600 installation

2005-04-30 Thread Chris Mason (Lists)

I have installed Asterisk on a CentOS4 box and then installed Asterisk from
CVS.
I installed a Sangoma A101 and connected it to a Adtran 600 using a T1
Crossover cable. The 600 has 12 x FXS, 12 x FXO interfaces.

I ran through the wanpipe install instructions and configured it, now I can
run 

[EMAIL PROTECTED] asterisk]#  wanrouter hwprobe

---
| Wanpipe Hardware Probe Info |
---
1 . AFT-A101u  : SLOT=1 : BUS=1 : IRQ=209 : CPU=A : PORT=PRI

Card Cnt: S508=0  S514X=0  S518=0  A101-2=1  A104=0  A300=0 

So I know the card is there OK.

My /etc/zaptel.conf looks like:
span=1,1,0,esf,b8zs
loadzone = us
defaultzone=us
fxsls=1-12

I am only trying to get half to load for now to make it simple.

[EMAIL PROTECTED] asterisk]# ztcfg
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

When I run service zaptel restart I get:

Waiting for zap to come online...Error: missing /dev/zap!

Wha am I doing wrong?


Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (646)722-0001 Fax: (815)301-9759 
(305) 704-7249
Yahoo IM: [EMAIL PROTECTED] 
Skype ID: netconcepts

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RE: [Asterisk-Users] A good SIP receptionist phone

2005-04-30 Thread Rusty Shackleford


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Michael Welter
 Sent: Saturday, April 30, 2005 12:53 PM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
 Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] A good SIP receptionist phone
 
 In a multi-tenant environment, is there a way to display, on 
 the phone, 
 which DID (which tenant) is being called?


Yes. We've done this by simply prepending a meaningful string onto the
front of the CIDName. It's a total kludge, but it works.

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.0 - Release Date: 04/29/2005
 

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[Asterisk-Users] SIP over IAX2

2005-04-30 Thread Daniel Salama

I have two asterisk boxes. I'm running an IVR script in one of them and 
I have agents registered on the second box.

I wish to create an extension on the * box where the agents are 
registered, so that when dialed, it will connect the agent to the IVR 
script on the other * box. However, I'd like for the connection to be 
done using SIP instead of IAX. Can anyone help me, if at all possible, 
write this configuration?

Thanks,
Daniel
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[Asterisk-Users] Kernel 2.4 or 2.6

2005-04-30 Thread Daniel Salama

I was reading on the wiki about the supported kernels and I __THINK__ 
the main issues with the kernel versions have more to do with Zaptel 
driver and not necessarily Asterisk itself. Is this correct?

Thanks,
Daniel
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[Asterisk-Users] budgetphone

2005-04-30 Thread Bert Haverkamp

Dear all,

I'm trying to get asterisk to register to budgetphone.nl. In several threads
 I saw people who got this to work:
http://www.voip-info.org/wiki-Talkin2ya
http://lists.digium.com/pipermail/asterisk-users/2005-March/092850.html

But I've spent a whole saturday on it now and didn't get any further.
I also have a granstrema handytone 486. This thing manages to register.

I've tried to look into the differences between the sip messages with
ethereal. In ethereal I see the following sip conversation for the handytone:
192.168.0.60-81.23.228.150   SIP Request: REGISTER sip:budgetphone.nl
81.23.228.150-192.168.0.60  SIP Status: 401 Unauthorized(0 bindings)
192.168.0.60-81.23.228.150  SIP Request: REGISTER sip:budgetphone.nl
81.23.228.150-192.168.0.60  SIP Status: 200 OK(1 bindings)

In the first message the handytone tries to register, but it gets a request
for authentication (second packet) with a challenge. The third packet is a
retry to register, but this time with the response to the challenge. The
fourth packet is then the confirmation that all went well.

When I do the same with asterisk I get the following
192.168.0.35-81.23.228.150   SIP Request: REGISTER sip:budgetphone.nl
81.23.228.150-192.168.0.35  SIP Status: 401 Unauthorized(0 bindings)
192.168.0.35-81.23.228.150  SIP Request: REGISTER sip:budgetphone.nl
192.168.0.35-81.23.228.150  SIP Request: REGISTER sip:budgetphone.nl
192.168.0.35-81.23.228.150  SIP Request: REGISTER sip:budgetphone.nl
...
Asterisk gives a response to the challenge, but never gets an answer back.
What is going wrong?
Hope someone can shed some light here..

; SIP Configuration for Asterisk
;
[general]
context=default ; Default context for incoming calls
recordhistory=yes ; Record SIP history by default
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
language=en ; Default language setting for all users/peers
nat=no
defaultexpirey=1200
disallow=all
allow=g729
allow=gsm
allow=ulaw
allow=alaw

register = 31437110310:[EMAIL PROTECTED]/31437110310

[31437110310]
type=friend
context=from-budgetphone
host=budgetphone.nl
callerid=John Doe
fromuser=31437110310
fromdomain=budgetphone.nl
username=31437110310
insecure=very
secret=PASSWD
qualify=no
canreinvite=no
nat=yes
port=5060

---
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Re: [Asterisk-Users] budgetphone

2005-04-30 Thread Michiel van Baak

On 01:11, Sun 01 May 05, Bert Haverkamp wrote:
 Dear all,
 
 I'm trying to get asterisk to register to budgetphone.nl. In several threads
  I saw people who got this to work:
 http://www.voip-info.org/wiki-Talkin2ya

Hi,

The directions on the page there are working like a charm.
The one who made this page helped me too. We did this setup
at the same time, and I was able to get 50% done without his
help and after his help all worked fine.

Did you setup the /etc/hosts file ?
That was the thing I needed to do to get it all working.
My config now is (ip's and telephone numbers replaced with
bogus values):

/etc/hosts:
# Host Database
81.23.228.150   budgetphone.nl

/etc/asterisk/sip.conf:
[general]
context=from-sip ; Default context for incoming calls
realm=vanbaak   ; Realm for digest authentication
port=5060   ; UDP Port to bind to (SIP standard port is 
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes   ; Enable DNS SRV lookups on outbound calls
maxexpirey=3600 ; Max length of incoming registration we allow
defaultexpirey=120  ; Default length of incoming/outoing 
registration
allow=all   ; This may also be set for individual 
users/peers
language=en ; Default language setting for all users/peers
relaxdtmf=yes   ; Relax dtmf handling
rtptimeout=60   ; Terminate call if 60 seconds of no RTP 
activity
rtpholdtimeout=300  ; Terminate call if 300 seconds of no RTP 
activity
useragent=Asterisk  ; Allows you to change the user agent string
nat=no  ; NAT settings 
externip=XXX.XXX.XXX.XXX
localnet=192.168.2.0/255.255.255.0
promiscredir = no  ; If yes, allows 302 or REDIR to non-local SIP address
register = 31X:[EMAIL PROTECTED]/31X

[budgetphone]
canreinvite=no
context=from-budgetphone
fromuser=31
fromdomain=budgetphone.nl
host=budgetphone.nl
insecure=very
nat=yes
;qualify=yes
secret=my_passwd
type=friend
username=31X

/etc/asterisk/extensions.conf:
[general]
static=yes
writeprotect=no
[globals]
VMBOX=michiel ; the VM box

[outgoing-budgetphone]
exten = _0X,1,SetAccount(outgoing-budgetphone)
exten = _0X,2,SetCallerID(31X)
exten = _0X,3,SetCIDName(Michiel en Nancy van Baak)
exten = _0X,4,SetCIDNum(31)
exten = _0X,5,Dial(SIP/budgetphone/${EXTEN},50,Tr)
exten = _0X,6,Congestion
exten = _0X,106,Busy

[from-budgetphone]
exten = 31X,1,SetCIdNum(0${CALLERIDNUM:2})
exten = 31X,2,LookupCIDName
exten = 31X,3,Macro(stdexten,michiel,SIP/michiel)


We are using this setup for 3 months now and the KPN line is
already history.

-- 
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Two of the most famous products of Berkeley are LSD and BSD. I don't think 
that this is a coincidence.

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Re: [Asterisk-Users] Kernel 2.4 or 2.6

2005-04-30 Thread Ian Pattison

I believe it is Zaptel only that becomes a problem. I'm running asterisk on a 
2.6 kernel... the only concession I had to make was to use make linux26 when 
I compiled Zaptel.

Thanks,

Ian

 [EMAIL PROTECTED] 30/04/2005 19:10 
I was reading on the wiki about the supported kernels and I __THINK__ 
the main issues with the kernel versions have more to do with Zaptel 
driver and not necessarily Asterisk itself. Is this correct?

Thanks,
Daniel

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[Asterisk-Users] How to bridge 2 calls

Guys.

I have some dialing rules defined for my internal extensions but I am now
defning a call forward option that allow an extension to be forwarded to an
outside number, right now Im using Dial cmds but I was wondering if ther is
a way to do this but using the dialing rules that I have also defined for
the internal extensions? For exaple, like DISA does...

Any ideas?

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Re: [Asterisk-Users] SIP over IAX2

2005-04-30 Thread Daniel Salama

I understand and I guess I know how to do that within a single box.
If I have the following:
Asterisk Box 1 (no agents)
extensions.conf
[test-ivr]
exten = s,1,AGI(play_ivr)
exten = s,2,Hangup
Asterisk Box 2 (agents register)
extensions.conf
[agents-context]
exten = 1234,1,Dial(?)
exten = 1234,2,Hangup
Question is, when the agents dial 1234, how do I tell the application 
to connect to the agent with context test-ivr of Asterisk_1?

Thanks,
Daniel
On Apr 30, 2005, at 7:12 PM, Tim Connolly wrote:
Maybe I'm missing something, but as long as you have the entension 
defined
on the agent box to dial the extension on the IVR, you should be okay. 
Just
make sure the default SIP context on the IVR has that extension 
defined, or
define the IVR box as a SIP peer.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel 
Salama
Sent: Saturday, April 30, 2005 5:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP over IAX2

I have two asterisk boxes. I'm running an IVR script in one of them and
I have agents registered on the second box.
I wish to create an extension on the * box where the agents are
registered, so that when dialed, it will connect the agent to the IVR
script on the other * box. However, I'd like for the connection to be
done using SIP instead of IAX. Can anyone help me, if at all possible,
write this configuration?
Thanks,
Daniel
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[Asterisk-Users] Programing a call forward feature to cel phones

Guys.

I just programed a feature that allows any extension to be forwarded to any
outside number, for example, forward your extension 201 to any number
outside (via zap) so that if somebody calls your extension either from
inside out outside (using another zap we have) it gets directed.

Problem I have is that if somebody using a cel phone calls in and gets
directed to my extension which in turn is directed to my cel phone, the call
comes thru but after 2 seconds, the call gets all garbled and with a sound
like b and the caller cant be heard anymore.

Anybody has any idea why this is happening?


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RE: [Asterisk-Users] SIP over IAX2

Asterisk Box 2 (agents register)
extensions.conf
[agents-context]
exten = 1234,1,Dial(SIP/[EMAIL PROTECTED])
exten = 1234,2,Hangup

Asterisk Box 1
Sip.conf
[ab1]
type=friend
host=ip of ab2
context=incoming
canreinvite=yes
qualify=yes

extension.conf
[incoming]
Exten = 1234etc...

-Original Message-
From: Daniel Salama [mailto:[EMAIL PROTECTED] 
Sent: Saturday, April 30, 2005 6:50 PM
To: Tim Connolly
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP over IAX2

I understand and I guess I know how to do that within a single box.

If I have the following:

Asterisk Box 1 (no agents)
extensions.conf
[test-ivr]
exten = s,1,AGI(play_ivr)
exten = s,2,Hangup

Asterisk Box 2 (agents register)
extensions.conf
[agents-context]
exten = 1234,1,Dial(?)
exten = 1234,2,Hangup

Question is, when the agents dial 1234, how do I tell the application 
to connect to the agent with context test-ivr of Asterisk_1?

Thanks,
Daniel

On Apr 30, 2005, at 7:12 PM, Tim Connolly wrote:

 Maybe I'm missing something, but as long as you have the entension 
 defined
 on the agent box to dial the extension on the IVR, you should be okay. 
 Just
 make sure the default SIP context on the IVR has that extension 
 defined, or
 define the IVR box as a SIP peer.



 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Daniel 
 Salama
 Sent: Saturday, April 30, 2005 5:57 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] SIP over IAX2

 I have two asterisk boxes. I'm running an IVR script in one of them and
 I have agents registered on the second box.

 I wish to create an extension on the * box where the agents are
 registered, so that when dialed, it will connect the agent to the IVR
 script on the other * box. However, I'd like for the connection to be
 done using SIP instead of IAX. Can anyone help me, if at all possible,
 write this configuration?

 Thanks,
 Daniel

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Re: [Asterisk-Users] Zaptel and Boostringer

2005-04-30 Thread Andrew Kohlsmith

On April 30, 2005 02:56 pm, Ian Pattison wrote:
 For some time now I've had issues with ringing voltages on my TDM400P.
 Numerous folks have told me that using modprobe wcfxs boostringer=1  when
 loading the module will force the driver to use boosted ring voltage. For
 some strange reason this has never worked for me. Today I got creative...
 another way to do it is to edit wcfxs.c (in the zaptel CVS) and find the
 following block of declarations:

It would do you well to figure out WHY your system is not passing parameters 
properly; 'boostringer=1' is supposed to set that boostringer variable.  The 
fact that it isn't indicates a deeper seated problem with your kernel, 
modprobe, or your distribution.

-A.
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Re: [Asterisk-Users] Problems with TDM400P card

2005-04-30 Thread Andrew Kohlsmith

On April 30, 2005 10:23 am, Kim Culhan wrote:
  If so, what do see if you run 'vmstat 1' and let it run for about
  twenty seconds?  Do you see the cpu utilization going to about 100%
  every five or six seconds?

 Negative:

That's interesting; so that can potentially narrow the problematic code down 
to any bits specific to Linux and not BSD.  This is very helpful!

Thank you Richard for thinking to ask this, and thank you Kim for responding!

-A.
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Re: [Asterisk-Users] Send DTMF AFTER channels are bridged

2005-04-30 Thread Matt Riddell

Shady wrote:
Someone to know how can I send a DTMF after the channels are bridged?
I need something like the D option of the Dial application, but this
option sends the DTMF before the channels are bridged. In fact I want the
caller and the callee to receive the DTMF. Please help :)
If using a codec with inband DTMF, you could always use the option to 
play an audio file once connected, and just put the DTMF in there.

--
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Matt Riddell
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Re: [Asterisk-Users] Programing a call forward feature to cel phones

2005-04-30 Thread Matt Riddell

Anton Krall wrote:
Problem I have is that if somebody using a cel phone calls in and gets
directed to my extension which in turn is directed to my cel phone, the call
comes thru but after 2 seconds, the call gets all garbled and with a sound
like b and the caller cant be heard anymore.
Maybe the caller is cold?
:)
Sorry.
--
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Matt Riddell
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RE: [Asterisk-Users] Programing a call forward feature to cel phones

Jejejejeje I didnt know how to put the sound it does... Its like an
intermitent sound like when you are to lose a cel phone connection. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Matt Riddell
|Sent: Sábado, 30 de Abril de 2005 07:15 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|Anton Krall wrote:
| Problem I have is that if somebody using a cel phone calls 
|in and gets 
| directed to my extension which in turn is directed to my cel phone, 
| the call comes thru but after 2 seconds, the call gets all 
|garbled and 
| with a sound like b and the caller cant be heard anymore.
|
|Maybe the caller is cold?
|
|:)
|
|Sorry.
|
|--
|Cheers,
|
|Matt Riddell
|___
|
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|http://www.sineapps.com/rssfeed.php (Daily Asterisk News - 
|rss) ___
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|

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Re: [Asterisk-Users] A good SIP receptionist phone

2005-04-30 Thread Sean Kennedy

Jason Brown wrote:
I have a problem. The average person is too freaking stupid to use a VOIP 
phone. My experience has so far been that if it doesn't have 20 buttons with 
little red LED's on it, the user cannot comprehend call parking, attended 
transfer, blind transfer, DND, and navigating through a voicemail menu.
I need a good receptionist phone that works with Asterisk. It basically needs to act like 
an avaya partner phone, I don't need 20 buttons with little red LED's...what I do need is 
for the phone to register multiple extensions to my asterisk server and act like each SIP 
extension is a line, so if the idiot receptionist has a call ringing in on line 1, she 
can pick it up, look at the buttons, see a call ringing in on line 2 (and the phone 
ringer rings), put call 1 on hold without hanging the caller up, and hit the little 
I am an idiot and need a line 2 button to pick up line 2, so on and so forth.
I love VOIP systems and all the functionality they bring and features I get. 
Unfortunately, the average person in this country anymore is apparently 
completely stupid and cannot understand how to juggle calls without hanging up 
on people.
/rant
So seriously does anyone have a recommendation for a good receptionist phone? I 
tried the Snom today and I can't get the programmable buttons to do this, even 
by following the manual. So please, any suggestions would be great, before I 
get fired at my dayjob for everyone else's idiocy.
1) I suggest you learn to live and like those idiots.  I also suggest 
you tone down that attitude and adjust it.  Those idiots contribute to 
YOUR pay.

2) There isn't anything like what you want.  I know, I want the same 
thing.  There is no phone out there that will do this with any protocol 
that asterisk uses.  This is the one major failing of asterisk ( and 
voip in general.  I smell an oportunity for a phone manufacture ), and 
what keeps it out of a lot of places.

I can see this being implemented with a phone that speaks to *'s manager 
interface.  Who wants to talk to polycom or cisco about it?  :)

Sean
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Re: [Asterisk-Users] Programing a call forward feature to cel phones

2005-04-30 Thread Matt Riddell

Anton Krall wrote:
Jejejejeje I didnt know how to put the sound it does... Its like an
intermitent sound like when you are to lose a cel phone connection. 
Ah...and the cellphone has range?
Normally I would say that seeing as you normally hear in on a cell phone 
it would likely be that end, but maybe they are just playing back the 
last bit of audio data repeatedly until they get another one.

How are you connecting to the cell phone?
What do you see in the Asterisk console?
How is the call getting to your cellphone?
What happens if you dial straight from your Asterisk box to your Cellphone?
What happens if you dial another cellphone?
Do you have any problems when you dial in from you cellphone to Asterisk?
What happens if you send the call to another land line instead?
--
Cheers,
Matt Riddell
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RE: [Asterisk-Users] A good SIP receptionist phone

2005-04-30 Thread Gregory Wiktor - ADCom Corp.

Hello Sean,
I thought the Polycom's had some kind of BLF Feature don't they?

I am thinking of getting two of them, so it would be nice to know,
otherwise I would get 2 more 7960's. (which are great phones)

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
Kennedy
Sent: Saturday, April 30, 2005 9:03 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] A good SIP receptionist phone

Jason Brown wrote:

I have a problem. The average person is too freaking stupid to use a
VOIP phone. My experience has so far been that if it doesn't have 20
buttons with little red LED's on it, the user cannot comprehend call
parking, attended transfer, blind transfer, DND, and navigating through
a voicemail menu.

I need a good receptionist phone that works with Asterisk. It basically
needs to act like an avaya partner phone, I don't need 20 buttons with
little red LED's...what I do need is for the phone to register multiple
extensions to my asterisk server and act like each SIP extension is a
line, so if the idiot receptionist has a call ringing in on line 1, she
can pick it up, look at the buttons, see a call ringing in on line 2
(and the phone ringer rings), put call 1 on hold without hanging the
caller up, and hit the little I am an idiot and need a line 2 button
to pick up line 2, so on and so forth.

I love VOIP systems and all the functionality they bring and features I
get. Unfortunately, the average person in this country anymore is
apparently completely stupid and cannot understand how to juggle calls
without hanging up on people.

/rant

So seriously does anyone have a recommendation for a good receptionist
phone? I tried the Snom today and I can't get the programmable buttons
to do this, even by following the manual. So please, any suggestions
would be great, before I get fired at my dayjob for everyone else's
idiocy.


1) I suggest you learn to live and like those idiots.  I also suggest
you tone down that attitude and adjust it.  Those idiots contribute to
YOUR pay.

2) There isn't anything like what you want.  I know, I want the same
thing.  There is no phone out there that will do this with any protocol
that asterisk uses.  This is the one major failing of asterisk ( and
voip in general.  I smell an oportunity for a phone manufacture ), and
what keeps it out of a lot of places.

I can see this being implemented with a phone that speaks to *'s manager
interface.  Who wants to talk to polycom or cisco about it?  :)

Sean
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RE: [Asterisk-Users] Programing a call forward feature to cel phones

2005-04-30 Thread Gregory Wiktor - ADCom Corp.

I know I saw something about not using GSM codecs when on cell phones, could 
this be the case?
The 2 second delay, well unfortunately all cell's have about a .5 second delay 
on their own, so that may be what you are hearing.  You just need to learn how 
to talk like you are on an international call...

Greg 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, April 30, 2005 8:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Programing a call forward feature to cel phones

Jejejejeje I didn't know how to put the sound it does... Its like an 
intermitent sound like when you are to lose a cel phone connection. 

|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of Matt 
|Riddell
|Sent: Sábado, 30 de Abril de 2005 07:15 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Programing a call forward feature to cel 
|phones
|
|Anton Krall wrote:
| Problem I have is that if somebody using a cel phone calls
|in and gets
| directed to my extension which in turn is directed to my cel phone, 
| the call comes thru but after 2 seconds, the call gets all
|garbled and
| with a sound like b and the caller cant be heard anymore.
|
|Maybe the caller is cold?
|
|:)
|
|Sorry.
|
|--
|Cheers,
|
|Matt Riddell
|___
|
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|rss) ___
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RE: [Asterisk-Users] How to bridge 2 calls

2005-04-30 Thread Gregory Wiktor - ADCom Corp.

I just made an extension 390 that calls my cell, so people can hold,
then send to 390 and hangup.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, April 30, 2005 7:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] How to bridge 2 calls

Guys.

I have some dialing rules defined for my internal extensions but I am
now defning a call forward option that allow an extension to be
forwarded to an outside number, right now Im using Dial cmds but I was
wondering if ther is a way to do this but using the dialing rules that I
have also defined for the internal extensions? For exaple, like DISA
does...

Any ideas?

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RE: [Asterisk-Users] Programing a call forward feature to cel phones

2005-04-30 Thread Race Vanderdecken

Curious,

How did you do the forward? Was it a script or programming in C?

Any output from debug?

Race The Tyrant Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, April 30, 2005 8:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Programing a call forward feature to cel
phones

Guys.

I just programed a feature that allows any extension to be forwarded to
any
outside number, for example, forward your extension 201 to any number
outside (via zap) so that if somebody calls your extension either from
inside out outside (using another zap we have) it gets directed.

Problem I have is that if somebody using a cel phone calls in and gets
directed to my extension which in turn is directed to my cel phone, the
call
comes thru but after 2 seconds, the call gets all garbled and with a
sound
like b and the caller cant be heard anymore.

Anybody has any idea why this is happening?


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RE: [Asterisk-Users] Programing a call forward feature to cel phones