date:20040211

 bam == bam  [EMAIL PROTECTED] writes:

bam Is there a way to allow a caller to enter an extension
bam number that is more than one digit long in a voice menu?

In addition to what the other replies say, I'd note that it
is usually a good idea to not use the initial digit of the
extensions as one of the single-digit options in the menu.

By keeping them separate, you make *'s job of detecting end
of entry significantly easier:  timeouts only have to apply
to the multi-digit choices and the single digit choices can
jump immediately on key detection.

-JimC

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[Asterisk-Users] Re: Residential Plans for Asterisk Users

 Steve == Steve Rodgers [EMAIL PROTECTED] writes:

Steve BTW: If you are a low volume user, it seems to make more sense
Steve to go with one of per-minute plans offering IAX connectivity.

Low volume in this case is quite large.  USD 20 per month will
net you around 675 to 690 minutes; USD 30 around 1015 to 1035,
depending on vendor.

How many households make that many minutes of toll calls?

The fixed rate plans offered for biz accounts usually cost more
than that for just 1000 minutes.

Outside of the US, of course, things are probably completely
different

-JimC

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[Asterisk-Users] Multiple switch staments

2004-02-11 Thread Matteo Brancaleoni

Hi.

Does anybody ever had the need to use multiple
switch staments in one context?
like N slave asterisk servers, switching
to one master which has in one context
N switches to the slaves.

so the master only holds a switching table.

Any idea?

(I know that can be done with a proper dialplan
without switches, but making asterisk browse
between multiple sw can be useful)

matteo.
-- 
Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web   : http://www.espia.it
Phone : +39 02 70633354  - ext 201
IAX(2): [EMAIL PROTECTED] - ext 201
Iaxtel: 1-700-56-62458   - ext 201

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[Asterisk-Users] Noise and scratches when there are two concurrent CAPI calls

2004-02-11 Thread Costa Tsaousis

Hi,

When I have two concurrent CAPI calls, * produces a lot of noises and
scratches on both CAPI channels.

I am using SIP phones; it appears on all phones, even if two separate SIP
devices are connected to the two CAPI channels.

The problem does not appear with any number of concurrent calls using SIP
end-to-end (with * as a media gateway).

I am using FritzCard! DSL (the ISDN part of it) and kernel 2.4.24 vanilla.

Any help is appreciated.

Costa

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Re: [Asterisk-Users] Ghost Calls

If you are going to be lazy, be really lazy. Don't screw up threading by
replying to a message that is totally unrelated to your message. Just
click on the mailing list address in the message. This eliminates all
the deleting of the old message and will create the proper threading. 

On Wed, 2004-02-11 at 04:02, Brian Pollack wrote:
 Next, when someone calls in and hangs up the call still rings through to my
 timeout event which is the default operator extension.   Her phone rings and
 nobdoy is there.   I've been reading that I need something like Remote
 Disconnect Supervision but I can't locate it.

Under asterisk, Remote Disconnect Supervision is  known as Kewlstart. In
the Adtran it may be known just as disconnect supervision.  Unless it is
provided by the PSTN provider it doesn't help really. 

You may want to look into the dial plan and see if you are waiting
before answering. If so, you are possibly into the dial plan when the
disconnect comes through and is missed.  

-- 
Steven Critchfield [EMAIL PROTECTED]

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[Asterisk-Users] Asterisk ACD - Avaya ACD

2004-02-11 Thread Joelson S. Apon



Hello Everyone..

I have successfully setup an Asterisk PBX here. Also, I do have it interface
with our existing PBX system which is the Avaya Definity. I'm now on the
process of migrating all our system setup, from Avaya to Asterisk and in
this case, the ACD functionality. Below is the ACD script of our Avaya PBX.
As you could see, I queue a certain skill/hunt group, then caller hears
music for a specific period of time, example 60 seconds wait time, then
after which, caller would hear a playback announcement telling him to stay
on hold, then wait time again, then an option to leave a message on a
mailbox, and so and so forth. I won't ask if this is possible with Asterisk
cause I know Asterisk can do all the modern PBX functionalities..Just wonder
if someone have already implement this and hoping you could share to me your
scripts..

Thank in advance for the help...

Regards  :-)

Joelson


  CALL VECTOR

Number: 32   Name: Ispbrand ISP

 Basic? y   EAS? y   G3V4 Enhanced? y   ANI/II-Digits? y   ASAI Routing?
n
 Prompting? y   LAI? n  G3V4 Adv Route? y   CINFO? y   BSR? y   Holidays? y

01 wait-time2   secs hearing ringback
02 announcement 5405
03 queue-to skill 12  pri t
04 wait-time60  secs hearing music
05 announcement 5422
06 wait-time60  secs hearing music
07 checkskill 91  pri t if unconditionally
08 wait-time60  secs hearing music
09 checkskill 90  pri t if unconditionally
10 collect  1digits after announcement 5406
11 route-to number 1308 with cov y if digit   =  1
12 wait-time120 secs hearing music
13 announcement 5417
14 wait-time120 secs hearing music
15 announcement 5403
16 wait-time3   mins hearing music
17 announcement 5404
18 wait-time3   mins hearing music
19 announcement 5404
20 wait-time3   mins hearing music
21 messagingskill 99  for extension 1308
22 stop

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RE: [Asterisk-Users] Cisco 7960 - how to enable messages key

2004-02-11 Thread Paul Mahler

Thanks!  I looked for this SIP option in the cisco docs, but couldn't find
it.

Paul

 
Paul Mahler 
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Low, Adam
Sent: Wednesday, February 11, 2004 1:33 AM
To: '[EMAIL PROTECTED] '
Subject: RE: [Asterisk-Users] Cisco 7960 - how to enable messages key

Hmmm did you read any of the docs on cisco.com ?

You need to set the 'message_uri' option to the extension that you run
VoiceMailMain on into the configuration file (SIP000XXX.cnf) for the
phone.

-Original Message-
From: John Fraizer
To: [EMAIL PROTECTED]
Sent: 11-2-04 6:22
Subject: Re: [Asterisk-Users] Cisco 7960 - how to enable messages key


Um, tell it what to do?  I don't remember exactly what I did but, it was

intuitive enough that when I got my 7960 a week ago, it only took one
try to 
get it right.

Paul Mahler wrote:
 Does anyone know how to make the 7960 messages key dial voicemail?
SIP
 6.0.
 

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* DISCLAIMER * 

This message and any attachment are confidential and may be privileged or
otherwise protected from disclosure and may include proprietary information.
If you are not the intended recipient, please telephone or email the sender
and delete this message and any attachment from your system. If you are not
the intended recipient you must not copy this message or attachment or
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RE: [Asterisk-Users] Cisco 7960 - how to enable messages key

2004-02-11 Thread Brian Pollack

Paul,  this is how I made it work:

SIPDefault.cnf:messages_uri: 8500

extensions.conf:exten = 8500,1,VoicemailMain,s${CALLERIDNAME:-3}
extensions.conf:exten = 8500,2,VoicemailMain
extensions.conf:exten = 8500,3,Hangup

Sip.conf for each user I have
callerid=Brian 300 xxx-xxx-
callerid=John 310 xxx-xxx-

I'm sure there is a better way to work with callerid but I use this for
other things as well.
Brian

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler
Sent: Wednesday, February 11, 2004 7:45 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Cisco 7960 - how to enable messages key

Thanks!  I looked for this SIP option in the cisco docs, but couldn't find
it.

Paul

 
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Low, Adam
Sent: Wednesday, February 11, 2004 1:33 AM
To: '[EMAIL PROTECTED] '
Subject: RE: [Asterisk-Users] Cisco 7960 - how to enable messages key

Hmmm did you read any of the docs on cisco.com ?

You need to set the 'message_uri' option to the extension that you run
VoiceMailMain on into the configuration file (SIP000XXX.cnf) for the
phone.

-Original Message-
From: John Fraizer
To: [EMAIL PROTECTED]
Sent: 11-2-04 6:22
Subject: Re: [Asterisk-Users] Cisco 7960 - how to enable messages key


Um, tell it what to do?  I don't remember exactly what I did but, it was

intuitive enough that when I got my 7960 a week ago, it only took one try to
get it right.

Paul Mahler wrote:
 Does anyone know how to make the 7960 messages key dial voicemail?
SIP
 6.0.
 

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* DISCLAIMER * 

This message and any attachment are confidential and may be privileged or
otherwise protected from disclosure and may include proprietary information.
If you are not the intended recipient, please telephone or email the sender
and delete this message and any attachment from your system. If you are not
the intended recipient you must not copy this message or attachment or
disclose the contents to any other person 


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[Asterisk-Users] I need patch for musiconhold-multiful format

2004-02-11 Thread Çã¿µ







Hello

I need patch for musiconhold of multiful file format (.wav,.gsm etc)


Pls help me !

Have a nice day !

Young

[Asterisk-Users] Cisco ATA 186

2004-02-11 Thread Dawid Mielnik


Cisco ATAs come in two types

ATA186-I1 with 600 ohm impedance
and
ATA186-I2 with complex impedance (270 ohm in series with 750 ohm and 150 NF
in parallel)

What is the difference between the two ? Which one is suitable for Europe ?

Thanks,

Dave

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Re: [Asterisk-Users] I need patch for musiconhold-multiful format

On Wed, 2004-02-11 at 09:31,  wrote:
 Hello
 I need patch for musiconhold of multiful file format (.wav,.gsm etc)
 Pls help me !

Dude, you exhibit the second reason HTML email is soo bad. Why would
anyone on this list need to confirm they viewed your message? Why the
hell do you think it is important for us to do so. 

For those who didn't notice the actual URL for the image displayed, here
it is purposely broken up to avoid more uses of it.
http://mail.dacommi.com/no_auth/confirm_img.php?id=3D1076513462=
:[EMAIL PROTECTED][EMAIL PROTECTED]
-- 
Steven Critchfield  [EMAIL PROTECTED]

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[Asterisk-Users] Re: Asterisk-GS and codec selection

2004-02-11 Thread Bill Michaelson





 Regarding codec selection, I see a minor difference between the FWD 
 and the local * box test cases, but I know nothing about the 
 negotiation protocol...

 With FWD, the OK message lists 3 Media Formats:

 

Bingo...GS chokes with GSM...just disallow it in your sip.conf:
disallow=all
allow=alaw
allow=ulaw


Thank you, very much. That got it working. Actually, I used disallow=gsm
as suggested by someone else.
 
 Please forgive my ignorance, but this leaves open questions which are nagging
me...

I expected that the SIP dialog would be a negotiation such that the devices
agree on a mutually acceptable encoding. And I think it's obvious (correct
me if I'm missing any key points) that such a negotiation would involve selecting
one of the encoding formats which appears in both lists presented by each
side. It doesn't seem reasonable that the GS should just "flake out" as
it seems to do, simply because it is offered an option it can't accept amongst
ones that it can. Is this indeed what I am seeing, or am I mischaracterizing
it?

Also, as I noted earlier, shouldn't * wait for the ACK before spewing the
audio stream? It appears to be missing the ACK because it retransmits the
OK shortly after it begins sending the RTP data.

These loose ends make me very uncomfortable.

[Asterisk-Users] Can't connect KPhone to asterisk

2004-02-11 Thread Maciek Kaminski

Anyone managed to make KPhone work with Asterisk?

For me it looks as if KPhone does not ACK transactions, i.e.:

KPhone --INVITE-- Asterisk
Asterisk --Trying -- KPhone
Asterisk --OK -- KPhone
KPhone doest not acknowlege. Asterisk keeps resending OKs, KPhone 
INVITES. Both timeouts.

By the way: KPhone offers PCMU, GSM, iLBC in INVITE, Asterisk answers 
with PCMU and PCMA with doest not seem to be correct as it should answer 
with subset of codecs offered(as far as I understood SIP RFC). Another 
issue that bothers me is that Asterisk seems to start media transmission 
as soon as it send OK not after it received ACK. Begining of 
conversation may lost this way, isn't it?

Asterisk and KPhone logs below:

-
Asterisk log:
-
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: sip:[EMAIL PROTECTED]
Content-Type: application/sdp
From: Maciek Kaminski sip:[EMAIL PROTECTED];tag=B62B188
Call-ID: [EMAIL PROTECTED]
Subject: sip:[EMAIL PROTECTED]
Content-Length: 183
User-Agent: kphone/4.0
Contact: Maciek Kaminski sip:[EMAIL PROTECTED];transport=udp
v=0
o=username 0 0 IN IP4 192.168.0.3
s=The Funky Flow
c=IN IP4 192.168.0.3
t=0 0
m=audio 32778 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
11 headers, 9 lines
Using latest request as basis request
Sending to 192.168.0.3 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format GSM
Found description format iLBC
Capabilities: us - 12, them - 1030/0, combined - 4
Non-codec capabilities: us - 1, them - 0, combined - 0
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:4186 check_user: Setting NAT on 
RTP to 0
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:5277 handle_request: Check for 
res for maciejka
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:1128 find_user: Call from user 
'maciejka' is 1 out of 0
Looking for 700 in default
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:3572 build_route: build_route: 
Contact hop: Maciek Kaminski sip:[EMAIL PROTECTED];transport=udp
list_route: hop: sip:[EMAIL PROTECTED];transport=udp
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.3;rport
From: Maciek Kaminski sip:[EMAIL PROTECTED];tag=B62B188
To: sip:[EMAIL PROTECTED];tag=as3b0a9ff0
Call-ID: [EMAIL PROTECTED]
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0

to 192.168.0.3:5060
   -- Executing Answer(SIP/maciejka-b4b6, ) in new stack
We're at 192.168.0.2 port 15200
Answering with preferred capability 4
Answering with preferred capability 8
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: Maciek Kaminski sip:[EMAIL PROTECTED];tag=B62B188
To: sip:[EMAIL PROTECTED];tag=as3b0a9ff0
Call-ID: [EMAIL PROTECTED]
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Type: application/sdp
Content-Length: 153
v=0
o=root 3363 3363 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
to 192.168.0.3:5060
   -- Executing Festival(SIP/maciejka-b4b6, Press 1 to heaven press 
2 to go to hell press 3 to disconnect.) in new stack
 == Parsing '/etc/asterisk/festival.conf': Found
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:318 festival_exec: Text 
passed to festival server : Press 1 to heaven press 2 to go to hell 
press 3 to disconnect.
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:395 festival_exec: Passing 
text to festival...
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:414 festival_exec: Passing 
data to channel...
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:424 festival_exec: 
Festival WV command

Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: sip:[EMAIL PROTECTED]
Content-Type: application/sdp
From: Maciek Kaminski sip:[EMAIL PROTECTED];tag=B62B188
Call-ID: [EMAIL PROTECTED]
Subject: sip:[EMAIL PROTECTED]
Content-Length: 183
User-Agent: kphone/4.0
Contact: Maciek Kaminski sip:[EMAIL PROTECTED];transport=udp
v=0
o=username 0 0 IN IP4 192.168.0.3
s=The Funky Flow
c=IN IP4 192.168.0.3
t=0 0
m=audio 32778 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
11 headers, 9 lines
Ignoring this request
We're at 192.168.0.2 port 15200
Answering with preferred capability 4
Answering with preferred capability 8
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: Maciek Kaminski sip:[EMAIL PROTECTED];tag=B62B188
To: sip:[EMAIL PROTECTED];tag=as3b0a9ff0
Call-ID: [EMAIL PROTECTED]
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

[Asterisk-Users] OT: Cisco 7940 Smartnet in the UK

2004-02-11 Thread Jason Ross

This is slightly off topic so sorry for the intrusion.

I've got a couple of 7940 phones I'd like to put on Smartnet but I'm
looking for what I need to order, what it roughly costs and finally a
reseller in the UK who is easy to deal with.

Preferably I'd like someone I can deal with online.

Thanks,

Jason
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RE: [Asterisk-Users] Can't connect KPhone to asterisk

2004-02-11 Thread Regovich, Timothy

Not ACK'ing an invite can be problematic for the statemachine.  Without the
ACK, the Dialog is not in  acorrect state.

As for the SDP goes, the KPHONE is offering what it can accept, and asterisk
is doing the same.  There is no restriction that they must match.  You can
change your offer in the ACK, or with a re-INVITE.

As for the immediate transmission : yeah, it does seem a little strange
doesn't it?  But that is the way that I have seen almost all UAs work.  The
implication is that your offer must be a valid, not a conditional offer :
when you say you accept GSM on port 8000, you better have a listener on 800
ready to go. 

Tim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Maciek Kaminski
Sent: Wednesday, February 11, 2004 11:39 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Can't connect KPhone to asterisk


Anyone managed to make KPhone work with Asterisk?

For me it looks as if KPhone does not ACK transactions, i.e.:

KPhone --INVITE-- Asterisk
Asterisk --Trying -- KPhone
Asterisk --OK -- KPhone
KPhone doest not acknowlege. Asterisk keeps resending OKs, KPhone 
INVITES. Both timeouts.

By the way: KPhone offers PCMU, GSM, iLBC in INVITE, Asterisk answers 
with PCMU and PCMA with doest not seem to be correct as it should answer 
with subset of codecs offered(as far as I understood SIP RFC). Another 
issue that bothers me is that Asterisk seems to start media transmission 
as soon as it send OK not after it received ACK. Begining of 
conversation may lost this way, isn't it?

Asterisk and KPhone logs below:


-
Asterisk log:

-
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: sip:[EMAIL PROTECTED]
Content-Type: application/sdp
From: Maciek Kaminski sip:[EMAIL PROTECTED];tag=B62B188
Call-ID: [EMAIL PROTECTED]
Subject: sip:[EMAIL PROTECTED]
Content-Length: 183
User-Agent: kphone/4.0
Contact: Maciek Kaminski sip:[EMAIL PROTECTED];transport=udp

v=0
o=username 0 0 IN IP4 192.168.0.3
s=The Funky Flow
c=IN IP4 192.168.0.3
t=0 0
m=audio 32778 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000

11 headers, 9 lines
Using latest request as basis request
Sending to 192.168.0.3 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format GSM
Found description format iLBC
Capabilities: us - 12, them - 1030/0, combined - 4
Non-codec capabilities: us - 1, them - 0, combined - 0
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:4186 check_user: Setting NAT on 
RTP to 0
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:5277 handle_request: Check for 
res for maciejka
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:1128 find_user: Call from user 
'maciejka' is 1 out of 0
Looking for 700 in default
Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:3572 build_route: build_route: 
Contact hop: Maciek Kaminski sip:[EMAIL PROTECTED];transport=udp
list_route: hop: sip:[EMAIL PROTECTED];transport=udp
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.3;rport
From: Maciek Kaminski sip:[EMAIL PROTECTED];tag=B62B188
To: sip:[EMAIL PROTECTED];tag=as3b0a9ff0
Call-ID: [EMAIL PROTECTED]
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


 to 192.168.0.3:5060
-- Executing Answer(SIP/maciejka-b4b6, ) in new stack
We're at 192.168.0.2 port 15200
Answering with preferred capability 4
Answering with preferred capability 8
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3;rport
From: Maciek Kaminski sip:[EMAIL PROTECTED];tag=B62B188
To: sip:[EMAIL PROTECTED];tag=as3b0a9ff0
Call-ID: [EMAIL PROTECTED]
CSeq: 1974 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Type: application/sdp
Content-Length: 153

v=0
o=root 3363 3363 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 15200 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

 to 192.168.0.3:5060
-- Executing Festival(SIP/maciejka-b4b6, Press 1 to heaven press 
2 to go to hell press 3 to disconnect.) in new stack
  == Parsing '/etc/asterisk/festival.conf': Found
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:318 festival_exec: Text 
passed to festival server : Press 1 to heaven press 2 to go to hell 
press 3 to disconnect.
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:395 festival_exec: Passing 
text to festival...
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:414 festival_exec: Passing 
data to channel...
Feb 11 17:12:36 DEBUG[180236]: app_festival.c:424 festival_exec: 
Festival WV command


Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: sip:[EMAIL

Re: [Asterisk-Users] Can't connect KPhone to asterisk

2004-02-11 Thread Maciek Kaminski

Regovich, Timothy wrote:

Not ACK'ing an invite can be problematic for the statemachine.  Without the
ACK, the Dialog is not in  acorrect state.
As for the SDP goes, the KPHONE is offering what it can accept, and asterisk
is doing the same.  There is no restriction that they must match.  You can
change your offer in the ACK, or with a re-INVITE.
 

Well, they must intersect:
For streams marked as sendrecv in the answer, the m= line MUST 
contain at least one codec the answerer is willing to both send and 
receive, from amongst those listed in the offer. The stream MAY indicate 
additional media formats, not listed in the corresponding stream in the 
offer, that the answerer is willing to send or receive (of course, it 
will not be able to send them at this time, since it was not listed in 
the offer).

As for the immediate transmission : yeah, it does seem a little strange
doesn't it?  But that is the way that I have seen almost all UAs work.  The
implication is that your offer must be a valid, not a conditional offer :
when you say you accept GSM on port 8000, you better have a listener on 800
ready to go. 
 

Optimistic strategy...

Maciek Kaminski

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Re: [Asterisk-Users] Can't connect KPhone to asterisk

2004-02-11 Thread Clif Jones

A typical response from the SIP UAS if no intersecting media types are 
found is:
415 Unsupported Media Type
Some user agents also add a warning header to tell you that it couldn't 
find a
usable CODEC.

Maciek Kaminski wrote:

Regovich, Timothy wrote:

Not ACK'ing an invite can be problematic for the statemachine.  
Without the
ACK, the Dialog is not in  acorrect state.

As for the SDP goes, the KPHONE is offering what it can accept, and 
asterisk
is doing the same.  There is no restriction that they must match.  
You can
change your offer in the ACK, or with a re-INVITE.
 

Well, they must intersect:
For streams marked as sendrecv in the answer, the m= line MUST 
contain at least one codec the answerer is willing to both send and 
receive, from amongst those listed in the offer. The stream MAY 
indicate additional media formats, not listed in the corresponding 
stream in the offer, that the answerer is willing to send or receive 
(of course, it will not be able to send them at this time, since it 
was not listed in the offer).

As for the immediate transmission : yeah, it does seem a little strange
doesn't it?  But that is the way that I have seen almost all UAs 
work.  The
implication is that your offer must be a valid, not a conditional 
offer :
when you say you accept GSM on port 8000, you better have a listener 
on 800
ready to go.  

Optimistic strategy...

Maciek Kaminski

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[Asterisk-Users] TDM card loses Dial tone

2004-02-11 Thread Bob Bevins









Hi,



I have a redhat 9 asterisk server with tdm400p three ports, and a x100p installed at
home. I am not running X or framebuffers. Every so often like once a month, I lose dial tone on my channels.
Asterisk still responds at the cli. I dont see
any log entries pertaining to this. If I restart asterisk it does not change. I
have to reboot the computer, which I would think would be a hardware problem,
or an OS issue. I cant seem to make it happen when I want so
troubleshooting is an issue. The irqs are ok
as seen below. I am not doing smp, or multithreading
as some posts would reveal that as a problem.



These are brand new cards from Digium.
The tdm is a new card with the power connected. I
tested the power supply and it is supplied the correct voltages.




CPU0

 0: 9298672
XT-PIC
timer

 1:
4
XT-PIC
keyboard

 2:
0
XT-PIC
cascade

 8:
1
XT-PIC rtc

 9: 251135
XT-PIC usb-uhci, eth0

10: 92378687
XT-PIC wcfxs

11: 92395359
XT-PIC wcfxo

12:
20
XT-PIC PS/2
Mouse

14: 75560
XT-PIC
ide0

15:
0
XT-PIC
ide1

NMI:
0

ERR:
0





Please respond if someone is aware of these types of
problems.



Thanks in advance,



Bob

Re: [Asterisk-Users] TDM card loses Dial tone

2004-02-11 Thread Glenn Dalgliesh

I have had similar issues with mine TDM400 w/4
modules. I get both no dial tone and sometime a large level of static on the
port and although sometimes manually unloading and reloading the drivers will
correct the problem most of the time I have to reboot the system. Also, do you
get messages like below in your messages log?

Feb 10 10:16:07 localhost kernel: Ouch, part reset,
quickly restoring reality (2)Feb 10 10:16:07 localhost kernel: Ouch, part
reset, quickly restoring reality (3)Feb 10 10:16:07 localhost kernel: Power
alarm on module 1, resetting!Feb 10 10:16:07 localhost kernel: Power alarm
on module 2, resetting!

Digium has replace my card once and I have seen the
same results in to different Computers. They have verified my zap.conf and
zapata.conf configurations and I am now having to reboot my machine every night
via crontab to keep the system running effectively. So, far for about a week the
reboot once a day has keep it running with out incident but I don't no if the
usage increase on the TDM400 if would start failing btw reboots.

Sorry no answer but it seems we may be having
similar problems.

- Original Message -
From:
Bob Bevins
To: [EMAIL PROTECTED]

Sent: Wednesday, February 11, 2004 12:24
PM
Subject: [Asterisk-Users] TDM card loses
Dial tone

Hi,

I have a redhat 9 asterisk server with
tdm400p three ports, and a x100p installed at home. I am not running X or
framebuffers. Every so often like
once a month, I lose dial tone on my channels. Asterisk still responds
at the cli. I dont see any log entries pertaining
to this. If I restart asterisk it does not change. I have to reboot the
computer, which I would think would be a hardware problem, or an OS issue. I
cant seem to make it happen when I want so troubleshooting is an issue. The
irqs are ok as seen below. I am not doing smp, or multithreading as some posts would reveal that as
a problem.

These are brand new cards from
Digium. The tdm is a new
card with the power connected. I tested the power supply and it is supplied
the correct voltages.

CPU0
0: 9298672
XT-PIC
timer
1:
4
XT-PIC
keyboard
2:
0
XT-PIC
cascade
8:
1
XT-PIC
rtc
9: 251135
XT-PIC
usb-uhci,
eth0
10: 92378687
XT-PIC
wcfxs
11: 92395359
XT-PIC
wcfxo
12:
20
XT-PIC
PS/2 Mouse
14: 75560
XT-PIC
ide0
15:
0
XT-PIC
ide1
NMI:
0
ERR:
0

Please respond if someone is aware
of these types of problems.

Thanks in
advance,

Bob

[Asterisk-Users] Stuck TE410P cards

2004-02-11 Thread Scott Stingel

Hello all-

I have 3 TE410P cards in service in the field.  Two of them have an regular
problem that they get stuck during a system reboot.  What I mean is that
they display no LED's during any part of the restart, and they are not seen
by the drivers during or after the reboot.

The only thing that brings them back to life is to power down and restart
the box they are in.  Even pressing the reset button on the processor does
not clear their state.

This sounds very much like a hardware problem with the cards, since one
would assume normally that a front panel reset would clear a stuck card.
Has anyone else experienced these symptoms?  This happens fairly regularly
on two of the three TE410P cards.  It does not happen with older cards such
as the E400P, of which I have several.

Thanks

Scott M. Stingel 
Emerging Voice Technology Inc.
Palo Alto, California and London, England
Email:  [EMAIL PROTECTED]  
URL:www.evtmedia.com  

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RE: [Asterisk-Users] Stuck TE410P cards

2004-02-11 Thread mattf

I had the same problem, Digium sent me a new card and now all is well.

MATT---

-Original Message-
From: Scott Stingel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 11, 2004 1:03 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Stuck TE410P cards

Hello all-

I have 3 TE410P cards in service in the field.  Two of them have an regular
problem that they get stuck during a system reboot.  What I mean is that
they display no LED's during any part of the restart, and they are not seen
by the drivers during or after the reboot.

The only thing that brings them back to life is to power down and restart
the box they are in.  Even pressing the reset button on the processor does
not clear their state.

This sounds very much like a hardware problem with the cards, since one
would assume normally that a front panel reset would clear a stuck card.
Has anyone else experienced these symptoms?  This happens fairly regularly
on two of the three TE410P cards.  It does not happen with older cards such
as the E400P, of which I have several.

Thanks

Scott M. Stingel 
Emerging Voice Technology Inc.
Palo Alto, California and London, England
Email:  [EMAIL PROTECTED]  
URL:www.evtmedia.com  

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[Asterisk-Users] T1 PRI CallerID

I have a new T1 PRI circuit from Eschelon.  They're sending the caller 
name in the facility record.

Is it possible for * to capture this information?  I remember an old 
post where Mark said the facility record was vendor dependant and that 
they had some special code for facility.

Does anyone have further information?

Thanks,
Michael Welter
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[Asterisk-Users] asterisk-oh323 new update, v0.5.9

2004-02-11 Thread Michael Manousos

This new version contains a workaround to an Asterisk bug
(see http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001029).
This bug caused random segfaults in H.323/SIP calls.
Regards,
Michael.
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Re: [Asterisk-Users] T1 PRI CallerID

On Wed, 2004-02-11 at 12:13, Michael Welter wrote:
 I have a new T1 PRI circuit from Eschelon.  They're sending the caller 
 name in the facility record.
 
 Is it possible for * to capture this information?  I remember an old 
 post where Mark said the facility record was vendor dependant and that 
 they had some special code for facility.
 
 Does anyone have further information?

If you look in the CDR logs, you should see the names in there. Mark
mentioned a while back that the facility message comes usually after the
call setup and therefore if you want it available for your phones, you
may have to include a wait before answer to let this message show up.

-- 
Steven Critchfield  [EMAIL PROTECTED]

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[Asterisk-Users] High Density configuration for Voice Fax

2004-02-11 Thread Costa Tsaousis

Hi,

Are there any well known good H/W configurations for high density E1
setups supporting * and FAX? The response we got from digium about their
cards, as far as FAX is concerned, is: I would not depend upon them for
FAX. It does work, but it is not completely reliable..

The minimum we would need is 4 x E1 on each system but both Voice and FAX
at the same time.

Also, I saw somewhere on the asterisk sites, that * can support up to 100
concurrent calls on a dual Xeon 1.8GHz when doing media conversion.

We have a few projects that we will need PRI - GSM or G729 media
conversion (for WAN use). Do you know any cards that can take care of the
media conversion and free the CPUs (for use with * of course)?

Kind regards,

Costa

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Re: [Asterisk-Users] T1 PRI CallerID

Yes, I tried Wait(1) but still no joy.

Steven Critchfield wrote:
On Wed, 2004-02-11 at 12:13, Michael Welter wrote:

I have a new T1 PRI circuit from Eschelon.  They're sending the caller 
name in the facility record.

Is it possible for * to capture this information?  I remember an old 
post where Mark said the facility record was vendor dependant and that 
they had some special code for facility.

Does anyone have further information?


If you look in the CDR logs, you should see the names in there. Mark
mentioned a while back that the facility message comes usually after the
call setup and therefore if you want it available for your phones, you
may have to include a wait before answer to let this message show up.


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Re: [Asterisk-Users] T1 PRI CallerID

On Wed, 2004-02-11 at 12:25, Michael Welter wrote:
 Yes, I tried Wait(1) but still no joy.

Is it in your CDRs?

 Steven Critchfield wrote:
  On Wed, 2004-02-11 at 12:13, Michael Welter wrote:
  
 I have a new T1 PRI circuit from Eschelon.  They're sending the caller 
 name in the facility record.
 
 Is it possible for * to capture this information?  I remember an old 
 post where Mark said the facility record was vendor dependant and that 
 they had some special code for facility.
 
 Does anyone have further information?
  
  
  If you look in the CDR logs, you should see the names in there. Mark
  mentioned a while back that the facility message comes usually after the
  call setup and therefore if you want it available for your phones, you
  may have to include a wait before answer to let this message show up.
  
 
 
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-- 
Steven Critchfield  [EMAIL PROTECTED]

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Re: [Asterisk-Users] TDM card loses Dial tone

2004-02-11 Thread Robert Lawrence

I am have the exact same issue.

I have/had a problem where the TDM400P turns to static and stops
responding to Asterisk. I also notice the "Ouch, part reset. Quickly
restoring reality." error messages in the log and CLI console.

I originally thought it was an IRQ sharing issue and/or powersupply..
but everything is OK in those respects.

CPU0
0: 6900712 XT-PIC timer
1: 244 XT-PIC keyboard
2: 0 XT-PIC cascade
5: 68954455 XT-PIC wcfxo
7: 68951573 XT-PIC wcfxs
8: 1 XT-PIC rtc
10: 62575 XT-PIC eth1
11: 73783 XT-PIC ide2, ide3, eth0
12: 566 XT-PIC PS/2 Mouse
14: 102 XT-PIC ide0
15: 61 XT-PIC ide1
NMI: 0
ERR: 0

I have not contacted Digium yet. I have received some help from the
nice guys on IRC. After some IRQ re-arrangements yesterday there has
been no problem. I hope it continues to work properly.

Glenn Dalgliesh wrote:

I have had similar issues with mine
TDM400 w/4 modules. I get both no dial tone and sometime a large level
of static on the port and although sometimes manually unloading and
reloading the drivers will correct the problem most of the time I have
to reboot the system. Also, do you get messages like below in your
messages log?

Feb 10 10:16:07 localhost kernel:
Ouch, part reset, quickly restoring reality (2)
Feb 10 10:16:07 localhost kernel: Ouch, part reset, quickly restoring
reality (3)
Feb 10 10:16:07 localhost kernel: Power alarm on module 1, resetting!
Feb 10 10:16:07 localhost kernel: Power alarm on module 2, resetting!

Digium has replace my card once and
I have seen the same results in to different Computers. They have
verified my zap.conf and zapata.conf configurations and I am now having
to reboot my machine every night via crontab to keep the system running
effectively. So, far for about a week the reboot once a day has keep it
running with out incident but I don't no if the usage increase on the
TDM400 if would start failing btw reboots.

Sorry no answer but it seems we may
be having similar problems.

-
Original Message -
From:
Bob Bevins

To:
[EMAIL PROTECTED]

Sent:
Wednesday, February 11, 2004 12:24 PM
Subject:
[Asterisk-Users] TDM card loses Dial tone

Hi,

I have a redhat 9 asterisk server
with tdm400p three ports, and a x100p installed at home. I am not
running X or framebuffers. Every so often like once a month, I lose dial tone on my
channels. Asterisk still responds at the cli.
I dont see any log entries pertaining to this. If I restart asterisk
it does not change. I have to reboot the computer, which I would think
would be a hardware problem, or an OS issue. I cant seem to make it
happen when I want so troubleshooting is an issue. The irqs are ok as seen below. I am not doing smp, or multithreading as some posts would
reveal that as a problem.

These are brand new cards
from Digium. The tdm
is a new card with the power connected. I tested the power supply and
it is supplied the correct voltages.

CPU0
0: 9298672 XT-PIC timer
1: 4 XT-PIC keyboard
2: 0 XT-PIC cascade
8: 1 XT-PIC rtc
9: 251135 XT-PIC usb-uhci, eth0
10: 92378687 XT-PIC wcfxs
11: 92395359 XT-PIC wcfxo
12: 20 XT-PIC PS/2 Mouse
14: 75560 XT-PIC ide0
15: 0 XT-PIC ide1
NMI:
0
ERR:
0

Please respond if someone
is aware of these types of problems.

Thanks in advance,

Bob

RE: [Asterisk-Users] Cisco 7960 - how to enable messages key

2004-02-11 Thread Brian Pollack

Paul,  this might be a hack but the -3 takes the extension number from the
caller id name that I have set in the sip.conf file.  I'm using this info
for other logic.   In this case callerid=Brian GRC Development 300
xx  in the sip.conf in the [brian] section passes the caller id
name to 8500 which takes on the extension part to log right into voicemail
from the persons phone.I use other parts of that callerid in other
places.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler
Sent: Wednesday, February 11, 2004 11:35 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Cisco 7960 - how to enable messages key

Why do you have the :-3 do in CALLERIDNAME:-3? 

Thanks!

 
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Pollack
Sent: Wednesday, February 11, 2004 7:25 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Cisco 7960 - how to enable messages key

Paul,  this is how I made it work:

SIPDefault.cnf:messages_uri: 8500

extensions.conf:exten = 8500,1,VoicemailMain,s${CALLERIDNAME:-3}
extensions.conf:exten = 8500,2,VoicemailMain extensions.conf:exten =
8500,3,Hangup

Sip.conf for each user I have
callerid=Brian 300 xxx-xxx-
callerid=John 310 xxx-xxx-

I'm sure there is a better way to work with callerid but I use this for
other things as well.
Brian

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler
Sent: Wednesday, February 11, 2004 7:45 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Cisco 7960 - how to enable messages key

Thanks!  I looked for this SIP option in the cisco docs, but couldn't find
it.

Paul

 
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Low, Adam
Sent: Wednesday, February 11, 2004 1:33 AM
To: '[EMAIL PROTECTED] '
Subject: RE: [Asterisk-Users] Cisco 7960 - how to enable messages key

Hmmm did you read any of the docs on cisco.com ?

You need to set the 'message_uri' option to the extension that you run
VoiceMailMain on into the configuration file (SIP000XXX.cnf) for the
phone.

-Original Message-
From: John Fraizer
To: [EMAIL PROTECTED]
Sent: 11-2-04 6:22
Subject: Re: [Asterisk-Users] Cisco 7960 - how to enable messages key


Um, tell it what to do?  I don't remember exactly what I did but, it was

intuitive enough that when I got my 7960 a week ago, it only took one try to
get it right.

Paul Mahler wrote:
 Does anyone know how to make the 7960 messages key dial voicemail?
SIP
 6.0.
 

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RE: [Asterisk-Users] Can't connect KPhone to asterisk

2004-02-11 Thread Regovich, Timothy

Where is that quote from?
Are rtpmaps marked as sendrecv or recvonly?
There is nothing really that says that I couldn't receive mpeg audio, but
only be able to send ulaw.

If you don't want to start listening until you send the ACK, then don't send
an SDP in the INVITE.  Wait until the ACK to send it.

Tim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Maciek Kaminski
Sent: Wednesday, February 11, 2004 12:11 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Can't connect KPhone to asterisk


Regovich, Timothy wrote:

Not ACK'ing an invite can be problematic for the statemachine.  Without the
ACK, the Dialog is not in  acorrect state.

As for the SDP goes, the KPHONE is offering what it can accept, and
asterisk
is doing the same.  There is no restriction that they must match.  You can
change your offer in the ACK, or with a re-INVITE.
  

Well, they must intersect:
For streams marked as sendrecv in the answer, the m= line MUST 
contain at least one codec the answerer is willing to both send and 
receive, from amongst those listed in the offer. The stream MAY indicate 
additional media formats, not listed in the corresponding stream in the 
offer, that the answerer is willing to send or receive (of course, it 
will not be able to send them at this time, since it was not listed in 
the offer).

As for the immediate transmission : yeah, it does seem a little strange
doesn't it?  But that is the way that I have seen almost all UAs work.  The
implication is that your offer must be a valid, not a conditional offer :
when you say you accept GSM on port 8000, you better have a listener on 800
ready to go. 
  

Optimistic strategy...

Maciek Kaminski

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Re: [Asterisk-Users] Stuck TE410P cards

2004-02-11 Thread Bob Knight

Scott Stingel wrote:

Hello all-

I have 3 TE410P cards in service in the field.  Two of them have an regular
problem that they get stuck during a system reboot.  What I mean is that
they display no LED's during any part of the restart, and they are not seen
by the drivers during or after the reboot.
The only thing that brings them back to life is to power down and restart
the box they are in.  Even pressing the reset button on the processor does
not clear their state.
This sounds very much like a hardware problem with the cards, since one
would assume normally that a front panel reset would clear a stuck card.
Has anyone else experienced these symptoms?  This happens fairly regularly
on two of the three TE410P cards.  It does not happen with older cards such
as the E400P, of which I have several.
Do pci read cycles show anything in the slot?
Does pci id come back as all 1's or 0's or just some invalid number?
Gee, the price on those sip gateways don't seem quite so high now.
have fun, bk.
--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163
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Re: [Asterisk-Users] T1 PRI CallerID

Yes, it is in the CDR.  I'll put PRI in debug and try to determine just 
when the facility record arrives.

Thanks,
Mike
Steven Critchfield wrote:

On Wed, 2004-02-11 at 12:25, Michael Welter wrote:

Yes, I tried Wait(1) but still no joy.


Is it in your CDRs?


Steven Critchfield wrote:

On Wed, 2004-02-11 at 12:13, Michael Welter wrote:


I have a new T1 PRI circuit from Eschelon.  They're sending the caller 
name in the facility record.

Is it possible for * to capture this information?  I remember an old 
post where Mark said the facility record was vendor dependant and that 
they had some special code for facility.

Does anyone have further information?


If you look in the CDR logs, you should see the names in there. Mark
mentioned a while back that the facility message comes usually after the
call setup and therefore if you want it available for your phones, you
may have to include a wait before answer to let this message show up.


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[Asterisk-Users] speex with VoicePulse

2004-02-11 Thread Warren H. Prince

Has anyone been able to get the speex codec to work with VoicePulse?  
When we force * to use speex for the connection, VoicePulse responds 
that there are no lines available.   When we change it back to 
another codec, it works fine... 

VoicePulse has not responded to our support request, so I'm hoping 
someone here could help.
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[Asterisk-Users] Constant crashes with Asterisk 0.7.2

2004-02-11 Thread John Fraizer

I recently upgraded to Asterisk 0.7.2 from Asterisk 0.5.0.  The server 
crashes constantly now for some reason.  Simply issuing a reload will 
cause it to die.

I am not sure what the cause is but, it is definitely frustrating.  Has 
anyone else experienced this when upgrading?

John

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Re: [Asterisk-Users] Can't connect KPhone to asterisk

2004-02-11 Thread Maciek Kaminski

Regovich, Timothy wrote:

Where is that quote from?
 

RFC - 3264 An Offer/Answer Model with the Session Description Protocol 
(SDP) chapter 6.

Maciek Kaminski

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RE: [Asterisk-Users] Loading module chan_capi.so failed! still some problems ...

2004-02-11 Thread Bodo Hahnke

Hi again,

this solved my first problem ... thanks for the help, some more changes
were necessary but this was also needed to have asterisk start up.
At 09:30 11.02.2004, you wrote:
 Asterisk Dynamic Loader Starting:
== Parsing '/etc/asterisk/modules.conf': Found
 [chan_capi.so] Feb 11 03:35:57 WARNING[1024]: loader.c:239
 ast_load_resource:
 /usr/lib/asterisk/modules/chan_capi.so: undefined symbol:
 ast_get_group Feb 11 03:35:57 WARNING[1024]: loader.c:358
 load_modules: Loading module chan_capi.so failed!
From a message dated February 6, by Klaus-Peter:

oh yes...

i added callgroup support for chan_capi. That's why you have to load
res_parking.so before chan_capi.so. So in modules.conf you need.
load = res_parking.so
load = chan_capi.so
[global]
chan_capi.so=yes


Now, here comes the next problem ... regarding all instructions about
installing asterisk with a fritzcard and capidriver I should now be able to
reach the demo context when I call the right number. When I call the
asterisk box it rings two times and then hangs up ... this are the mess-
ages from asterisk:
asterisk:~# asterisk -vvvc
  == Parsing '/etc/asterisk/asterisk.conf': Found
Asterisk CVS-02/10/04-18:37:59, Copyright (C) 1999-2001 Linux Support 
Services, Inc.
Written by Mark Spencer [EMAIL PROTECTED]
=
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
  == Manager registered action Ping
  == Manager registered action Logoff
  == Manager registered action Hangup
  == Manager registered action Status
  == Manager registered action Redirect
  == Manager registered action Originate
  == Manager registered action MailboxStatus
  == Manager registered action Command
  == Manager registered action ExtensionState
  == Manager registered action AbsoluteTimeout
  == Manager registered action MailboxCount
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 1 - 2
Asterisk PBX Core Initializing
Registering builtin applications:
 [AbsoluteTimeout]
  == Registered application 'AbsoluteTimeout'
 [Answer]
  == Registered application 'Answer'
 [BackGround]
  == Registered application 'BackGround'
 [Busy]
  == Registered application 'Busy'
 [Congestion]
  == Registered application 'Congestion'
 [DigitTimeout]
  == Registered application 'DigitTimeout'
 [Goto]
  == Registered application 'Goto'
 [GotoIf]
  == Registered application 'GotoIf'
 [GotoIfTime]
  == Registered application 'GotoIfTime'
 [Hangup]
  == Registered application 'Hangup'
 [NoOp]
  == Registered application 'NoOp'
 [Prefix]
  == Registered application 'Prefix'
 [ResetCDR]
  == Registered application 'ResetCDR'
 [ResponseTimeout]
  == Registered application 'ResponseTimeout'
 [Ringing]
  == Registered application 'Ringing'
 [SayNumber]
  == Registered application 'SayNumber'
 [SayDigits]
  == Registered application 'SayDigits'
 [SetAccount]
  == Registered application 'SetAccount'
 [SetGlobalVar]
  == Registered application 'SetGlobalVar'
 [SetLanguage]
  == Registered application 'SetLanguage'
 [SetVar]
  == Registered application 'SetVar'
 [StripMSD]
  == Registered application 'StripMSD'
 [Suffix]
  == Registered application 'Suffix'
 [Wait]
  == Registered application 'Wait'
Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
 [res_parking.so] = (Call Parking Resource)
  == Parsing '/etc/asterisk/parking.conf': Found
-- Registered extension context 'parkedcalls'
-- Added extension '701' priority 1 to parkedcalls
-- Added extension '702' priority 1 to parkedcalls
-- Added extension '703' priority 1 to parkedcalls
-- Added extension '704' priority 1 to parkedcalls
-- Added extension '705' priority 1 to parkedcalls
-- Added extension '706' priority 1 to parkedcalls
-- Added extension '707' priority 1 to parkedcalls
-- Added extension '708' priority 1 to parkedcalls
-- Added extension '709' priority 1 to parkedcalls
-- Added extension '710' priority 1 to parkedcalls
-- Added extension '711' priority 1 to parkedcalls
-- Added extension '712' priority 1 to parkedcalls
-- Added extension '713' priority 1 to parkedcalls
-- Added extension '714' priority 1 to parkedcalls
-- Added extension '715' priority 1 to parkedcalls
-- Added extension '716' priority 1 to parkedcalls
-- Added extension '717' priority 1 to parkedcalls
-- Added extension '718' priority 1 to parkedcalls
-- Added extension '719' priority 1 to parkedcalls
-- Added extension '720' priority 1 to parkedcalls
  == Registered application 'ParkedCall'
  == Manager registered action ParkedCalls
 [chan_capi.so] = (Common ISDN API for Asterisk)
  == Parsing '/etc/asterisk/capi.conf': Found
Feb 11 19:54:48 NOTICE[1024]: chan_capi.c:2338 mkif: 
ast_capi_pvt(43910906,43910906,demo,0x2,2) (1,2,64)

Re: [Asterisk-Users] System freeze

Side question:  should all of us on RH9 do the LD_ASSUME_KERNEL=2.4.1 ?

TC wrote:

-do you use hyperthreading
-do you use the LD_ASSUME_KERNEL=2.4.1 b4 loading asterisk
-have you compiled zaptel with the SMP flag on
Can anybody site some real hardcore technical facts
about SMP  hyperthreading support in the RH9 kernel rpm images
I hear what i would call 'old wives tales' about turning off ht support on
2.4 kernels
are there any valid tech ref as to why to not use ht on dial xeon systems
with *.
-does anybody know about what the issue is with back port of the 2.6 POSIX
thread model
 into RH9, an the siggestion to turn it off with the LD_ASSUME_KERNEL
setting


- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 09, 2004 11:16 AM
Subject: RE: [Asterisk-Users] System freeze


We've had 2 unexplainable system freezes.
is this deadlock ??, can you attach to the main asterisk PID
 follow this http://www.voip-info.org/wiki-Asterisk+debugging

We have SMP and Redhat 9 2.4.20-20.9. There has been no evidence
anywhere of why our system crashed.
Tan


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Re: [Asterisk-Users] Stuck TE410P cards

2004-02-11 Thread TC

Yup
we see that somes times on dell2650, need to power cycle to come again


on a Dell 1650
do you also get no interrupts
cat /proc/interrupts

i had a bug note on it here
http://bugs.digium.com/bug_view_page.php?bug_id=707
http://bugs.digium.com/bug_view_page.php?bug_id=708
we sent that card back ..

- Original Message -
From: Scott Stingel [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, February 11, 2004 10:03 AM
Subject: [Asterisk-Users] Stuck TE410P cards


 Hello all-

 I have 3 TE410P cards in service in the field.  Two of them have an
regular
 problem that they get stuck during a system reboot.  What I mean is that
 they display no LED's during any part of the restart, and they are not
seen
 by the drivers during or after the reboot.

 The only thing that brings them back to life is to power down and restart
 the box they are in.  Even pressing the reset button on the processor does
 not clear their state.

 This sounds very much like a hardware problem with the cards, since one
 would assume normally that a front panel reset would clear a stuck card.
 Has anyone else experienced these symptoms?  This happens fairly regularly
 on two of the three TE410P cards.  It does not happen with older cards
such
 as the E400P, of which I have several.

 Thanks

 Scott M. Stingel
 Emerging Voice Technology Inc.
 Palo Alto, California and London, England
 Email:  [EMAIL PROTECTED]
 URL:www.evtmedia.com

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[Asterisk-Users] Asterisk Critical Mass: Thursday, Miami, 9:00 PM

2004-02-11 Thread John Todd

So, there may be a dense group (in both interpretations of dense) 
of Asterisk users in the Miami area tomorrow (Thursday, February 12, 
2004) due to the Internet Telephony Expo.

Current attendees:
  - Marcelo Rodriguez (voxilla)
  - Mark Spencer (digium)
  - John Todd (myob)
  - you?
Please drop me a line directly via email if you plan to attend.  We 
will be meeting at 9:00 PM at the Hyatt on 2nd street, on the lower 
level near the registration booth (in front of the Tuttle room - 
that's Tuttle, not Buttle)  You don't need to be a conference 
attendee to show up for the get-together.

I expect we will find a local bar or something like we did in Boston 
and hang out for an hour or two.

JT
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[Asterisk-Users] Please Explain newchan-pvt-pvt

2004-02-11 Thread Matt Lawson

I'm into my 4th or 5th day of working on bug #981.  I know that part of 
the problem is that the fixup routine is called in chan_sip.c.  Well in 
there is a line that says p=newchan-pvt-pvt.  Problem is, that doesn't 
exist in this case.

I see pvt described as private lock but that doesn't mean I have any 
clue what the ramifications are in this instance.  Of course I can 
easily put a check into fixup that says if p is NULL then return 
success.  But what does it *mean* if newchan-pvt-pvt is NULL?  What 
should be done in this case?

This situation happens (sometimes) when the dual redirect is used and 
it's in the process of transferring the original receipient.   i.e. A 
calls B, then you do the dual redirect.  Regardless of which parameter 
goes into Channel or ExtraChannel, B is the one that will cause 
the crash, as it's going through the masquerade process.

Yes it is updated to cvs.



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[Asterisk-Users] Re: speex with VoicePulse

2004-02-11 Thread Matt Lawson

Ours are setup to allow GSM or Speex, and I see that using VoicePulse it 
chooses GSM.  Don't know the official policy.

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Re: [Asterisk-Users] Cisco ATA 186

2004-02-11 Thread Iain Stevenson

Search the list - there's a detailed answer on it.

I have two of the I1 version (at least that's what they say they are - 
ProductId: ATA186I1) and they work with UK spec phones.  All you need to 
watch for is that UK phones are three wire and US phones are 2 wire. 
Maplin sells an adapter to sort this out (Part no. VD36P).

 Iain

--On Wednesday, February 11, 2004 4:54 pm +0100 Dawid Mielnik 
[EMAIL PROTECTED] wrote:

Cisco ATAs come in two types

ATA186-I1 with 600 ohm impedance
and
ATA186-I2 with complex impedance (270 ohm in series with 750 ohm and 150
NF in parallel)
What is the difference between the two ? Which one is suitable for Europe
?
Thanks,

Dave

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Re: [Asterisk-Users] OT: Cisco 7940 Smartnet in the UK

2004-02-11 Thread stan

On Wed, Feb 11, 2004 at 04:34:46PM -, Jason Ross wrote:
 This is slightly off topic so sorry for the intrusion.
 
 I've got a couple of 7940 phones I'd like to put on Smartnet but I'm
 looking for what I need to order, what it roughly costs and finally a
 reseller in the UK who is easy to deal with.
 
 Preferably I'd like someone I can deal with online.
 

outside the UK there is
http://www.ams.net/products/product_info.cfm?Product_ID=10891
at $6.90 - dont know if they'd deal with people outside the US.

However, elsewhere that part number doesn't seem to mean much.
A 7940 is a cateogory 1 device so product codes that seem to mean
something to others are:

CON-SW-VPKG1 (insight.com/uk - £39.99, tradeprice.co.uk - £31.99)
telephone and web support only (search with smartnet on tradeprice, 
their search doesn't seem to like part numbers)

CON-AR-VPKG1 - advanced replacement - normally cheaper than above but
didn't provide web support so I didn't look into it

CON-SNT-VPKG1 - 8x5xNBD version (8am-5pm, next business day) which I
believe is when you can expect support and get hardware replacement
respectively
(insight.com/uk - £63.99, tradeprice.co.uk - £45.35)

you can pay more for 24/7 support, 4hr response time or onsite engineer

I ended up going with insight coz I didn't find tradeprice till after,
so dont know what they're like

Also, it seems to take ages for the contract to get registered (2+
weeks) with cisco once u finally get one

HTH

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Re: [Asterisk-Users] asterisk-oh323 new update, v0.5.9

2004-02-11 Thread Brian West

Was this bug fixed or was it really a bug.  I'm reading the bug notes and
it doesn't appear to be a bug in asterisk from what Mark said on the
notes.

bkw

On Wed, 11 Feb 2004, Michael Manousos wrote:


 This new version contains a workaround to an Asterisk bug
 (see http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001029).
 This bug caused random segfaults in H.323/SIP calls.

 Regards,
 Michael.


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[Asterisk-Users] Asterisk and Wildcard T100P

2004-02-11 Thread Mike Fryer




Hey guys,

I have an Asterisk system here and have it on a 
single-span T1 card. Got everything on the T1 side squared away, no 
warning lights on the smart jack and the carrier is able to see the 
D-Channel. However, when you call the numbers associated with the T1 all 
you get is a busy tone from the Asterisk. I've tried coding a small 
auto-attendant script so the Asterisk can answer the outside lines, but it still 
gives me busy rather than running the script. This is what it looks 
like

[default]
ResponseTimeout=10
exten = s,1,Answer
exten = s,2,Background(greeting)
.
exten = t,1,Hangup
exten = i,1,Background(invalid)
exten = i,2,Goto(default|s|2)

This is just the basics of the script but Asterisk 
never even answers. I've even included default in the users context but 
still nothing. I know there are no problems with my script because I can 
connect a SiP phone, and put an extension inthat just has a goto statement 
that points to the above script and it works perfectly.

Is there something special I have to do to make 
Asterisk know how to answeran incomingT1 line? I was thinking 
that since there are DINS digits coming across the PRI that Asterisk is seeing 
them asextensions, but the i rule should take care of that if we don't 
have them defined as extensions, right?

This is how my zapata.conf and zapetl.conf files 
are setup.

zapata.conf
--
[channels]
context = users
signalling = pri_cpe
switchype = 5ess
group = 1
channel = 1-23
pridialplan = national

zaptel.conf
--
span = 1,1,0,esf,b8zs
bchan = 1-23
dchan = 24

Thanks,

Mike Fryer

Re: [Asterisk-Users] Asterisk and Wildcard T100P

From your zapata.conf file below, I see you have configured for a PRI.
PRI by default is treated like a DID. You MUST define a extension entry
for every incoming call. If you had looked at the console error messages
this would have been fairly easy to diagnose. Most likely you will need
the full 10 digit number if you are in the USA to make it work.

On Wed, 2004-02-11 at 17:04, Mike Fryer wrote:
 Hey guys,
  
 I have an Asterisk system here and have it on a single-span T1 card. 
 Got everything on the T1 side squared away, no warning lights on the
 smart jack and the carrier is able to see the D-Channel.  However,
 when you call the numbers associated with the T1 all you get is a busy
 tone from the Asterisk.  I've tried coding a small auto-attendant
 script so the Asterisk can answer the outside lines, but it still
 gives me busy rather than running the script.  This is what it looks
 like
  
 [default]
 ResponseTimeout=10
 exten = s,1,Answer
 exten = s,2,Background(greeting)
 .
 exten = t,1,Hangup
 exten = i,1,Background(invalid)
 exten = i,2,Goto(default|s|2)
  
 This is just the basics of the script but Asterisk never even
 answers.  I've even included default in the users context but still
 nothing.  I know there are no problems with my script because I can
 connect a SiP phone, and put an extension in that just has a goto
 statement that points to the above script and it works perfectly.
  
 Is there something special I have to do to make Asterisk know how to
 answer an incoming T1 line?  I was thinking that since there are DINS
 digits coming across the PRI that Asterisk is seeing them
 as extensions, but the i rule should take care of that if we don't
 have them defined as extensions, right?
  
 This is how my zapata.conf and zapetl.conf files are setup.
  
 zapata.conf
 --
 [channels]
 context = users
 signalling = pri_cpe
 switchype = 5ess
 group = 1
 channel = 1-23
 pridialplan = national
  
 zaptel.conf
 --
 span = 1,1,0,esf,b8zs
 bchan = 1-23
 dchan = 24
  
 Thanks,
  
 Mike Fryer
-- 
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[Asterisk-Users] Mediatrix 1204 sip g/w now working

2004-02-11 Thread Rich Adamson


For those that might have the Mediatrix 1204 4-port FXO sip gateway or
for those that might have an interest, finally got it to work the way
one would expect when interconnecting to analog pstn lines.

Configuring the box for incoming calls was rather easy and worked 
shortly after installing the box.

Configuring it for outgoing pstn calls has been at least a two week effort
interacting with the reseller multiple times. The issues:

Port Selection:
---
The 1204 does not provide any documented method to select which of the
four ports will be used for outgoing calls. The manufacturer assumes all
four ports are the equivalent of a trunk group.
Fix: 
In extensions.conf, add something like:
 exten = _6X.,1,SETCIDNUM() 
 exten = _6X.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
 exten = _6X.,3,Congestion
and in the 1204:
 set gatewayPort1NetToPstnSourceFilter = 
Since the callerid that is set in asterisk never gets forward out the pstn
line, the above mechanism works fine for selecting port 1. (Use , ,
 for the remaining ports.)

Outbound calls dropping first digit:

The 1204 automatically drops the 1 when calling any long distance call
such as 1-800-555-1212.
Fix:
on the 1204, set countryCountryCode = 2
This is an undocumented item, but essentially stops the 1204 from stripping
leading digits.

Summary:

The limited testing conducted thus far indicates the 1204 is working very
well. There is no noticeable echo at any time. Seems to work very well with
canreinvite=yes although I've not tried it with a remote nat phone.

One of the nice things about the box is you can locate it at your demarc
and not have to provide 2-wire pstn connections to the asterisk system.

Rich


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Re: [Asterisk-Users] Asterisk and Wildcard T100P

On my Eschelon T1, all I get are the last four digits.

Steven Critchfield wrote:
From your zapata.conf file below, I see you have configured for a PRI.
PRI by default is treated like a DID. You MUST define a extension entry
for every incoming call. If you had looked at the console error messages
this would have been fairly easy to diagnose. Most likely you will need
the full 10 digit number if you are in the USA to make it work.
On Wed, 2004-02-11 at 17:04, Mike Fryer wrote:

Hey guys,

I have an Asterisk system here and have it on a single-span T1 card. 
Got everything on the T1 side squared away, no warning lights on the
smart jack and the carrier is able to see the D-Channel.  However,
when you call the numbers associated with the T1 all you get is a busy
tone from the Asterisk.  I've tried coding a small auto-attendant
script so the Asterisk can answer the outside lines, but it still
gives me busy rather than running the script.  This is what it looks
like

[default]
ResponseTimeout=10
exten = s,1,Answer
exten = s,2,Background(greeting)
.
exten = t,1,Hangup
exten = i,1,Background(invalid)
exten = i,2,Goto(default|s|2)
This is just the basics of the script but Asterisk never even
answers.  I've even included default in the users context but still
nothing.  I know there are no problems with my script because I can
connect a SiP phone, and put an extension in that just has a goto
statement that points to the above script and it works perfectly.
Is there something special I have to do to make Asterisk know how to
answer an incoming T1 line?  I was thinking that since there are DINS
digits coming across the PRI that Asterisk is seeing them
as extensions, but the i rule should take care of that if we don't
have them defined as extensions, right?
This is how my zapata.conf and zapetl.conf files are setup.

zapata.conf
--
[channels]
context = users
signalling = pri_cpe
switchype = 5ess
group = 1
channel = 1-23
pridialplan = national
zaptel.conf
--
span = 1,1,0,esf,b8zs
bchan = 1-23
dchan = 24
Thanks,

Mike Fryer


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Re: [Asterisk-Users] Asterisk and Wildcard T100P

On Wed, 2004-02-11 at 17:26, Michael Welter wrote:
 On my Eschelon T1, all I get are the last four digits.

I bet yours is a channelized T1 not a PRI. 2 and 4 digits are expected
on a channelized T1 sine they are sent in DTMF or MF, or even pulse and
that is added on to the call setup time. The relative difference between
4 and 10 digits sent in binary on the D channel is so little a human
wouldn't notice it.

 Steven Critchfield wrote:
 From your zapata.conf file below, I see you have configured for a PRI.
  PRI by default is treated like a DID. You MUST define a extension entry
  for every incoming call. If you had looked at the console error messages
  this would have been fairly easy to diagnose. Most likely you will need
  the full 10 digit number if you are in the USA to make it work.
  
  On Wed, 2004-02-11 at 17:04, Mike Fryer wrote:
  
 Hey guys,
  
 I have an Asterisk system here and have it on a single-span T1 card. 
 Got everything on the T1 side squared away, no warning lights on the
 smart jack and the carrier is able to see the D-Channel.  However,
 when you call the numbers associated with the T1 all you get is a busy
 tone from the Asterisk.  I've tried coding a small auto-attendant
 script so the Asterisk can answer the outside lines, but it still
 gives me busy rather than running the script.  This is what it looks
 like
  
 [default]
 ResponseTimeout=10
 exten = s,1,Answer
 exten = s,2,Background(greeting)
 .
 exten = t,1,Hangup
 exten = i,1,Background(invalid)
 exten = i,2,Goto(default|s|2)
  
 This is just the basics of the script but Asterisk never even
 answers.  I've even included default in the users context but still
 nothing.  I know there are no problems with my script because I can
 connect a SiP phone, and put an extension in that just has a goto
 statement that points to the above script and it works perfectly.
  
 Is there something special I have to do to make Asterisk know how to
 answer an incoming T1 line?  I was thinking that since there are DINS
 digits coming across the PRI that Asterisk is seeing them
 as extensions, but the i rule should take care of that if we don't
 have them defined as extensions, right?
  
 This is how my zapata.conf and zapetl.conf files are setup.
  
 zapata.conf
 --
 [channels]
 context = users
 signalling = pri_cpe
 switchype = 5ess
 group = 1
 channel = 1-23
 pridialplan = national
  
 zaptel.conf
 --
 span = 1,1,0,esf,b8zs
 bchan = 1-23
 dchan = 24
  
 Thanks,
  
 Mike Fryer
 
 
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Re: [Asterisk-Users] Mediatrix 1204 sip g/w now working

2004-02-11 Thread Christian Hecimovic

I've had one of these things working for ages, although I never set it up to 
select which port to use on outgoing lines. I overcame the first-digit 
stripping by telling the 1204 to prefix outgoing calls not in my area code. I 
seem to remember it stripping leading zeroes (as in 011 for international 
calls). I should try your undocumented feature.

The other thing I've had troubles with is provisioning it via DHCP. All of the 
DHCP key-value pairs are recognised except the one for outgoing proxy. It's 
very annoying, and seems to be a firmware bug. So I've configured the gateway 
to use a static IP.

Anyway, once set up, it seems to work okay, though it of course suffers from 
the same hangup detection problems that afflict all users of loop start.

Thanks for the config tip to manually select outgoing ports; that could be 
handy.

Christian

On Wednesday 11 February 2004 14:51, Rich Adamson wrote:
 For those that might have the Mediatrix 1204 4-port FXO sip gateway or
 for those that might have an interest, finally got it to work the way
 one would expect when interconnecting to analog pstn lines.

 Configuring the box for incoming calls was rather easy and worked
 shortly after installing the box.

 Configuring it for outgoing pstn calls has been at least a two week effort
 interacting with the reseller multiple times. The issues:

 Port Selection:
 ---
 The 1204 does not provide any documented method to select which of the
 four ports will be used for outgoing calls. The manufacturer assumes all
 four ports are the equivalent of a trunk group.
 Fix:
 In extensions.conf, add something like:
  exten = _6X.,1,SETCIDNUM()
  exten = _6X.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
  exten = _6X.,3,Congestion
 and in the 1204:
  set gatewayPort1NetToPstnSourceFilter = 
 Since the callerid that is set in asterisk never gets forward out the pstn
 line, the above mechanism works fine for selecting port 1. (Use , ,
  for the remaining ports.)

 Outbound calls dropping first digit:
 
 The 1204 automatically drops the 1 when calling any long distance call
 such as 1-800-555-1212.
 Fix:
 on the 1204, set countryCountryCode = 2
 This is an undocumented item, but essentially stops the 1204 from stripping
 leading digits.

 Summary:
 
 The limited testing conducted thus far indicates the 1204 is working very
 well. There is no noticeable echo at any time. Seems to work very well with
 canreinvite=yes although I've not tried it with a remote nat phone.

 One of the nice things about the box is you can locate it at your demarc
 and not have to provide 2-wire pstn connections to the asterisk system.

 Rich


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[Asterisk-Users] Re: High Density configuration for Voice Fax

 Costa == Costa Tsaousis [EMAIL PROTECTED] writes:

Costa Are there any well known good H/W configurations for high
Costa density E1 setups supporting * and FAX?

To do fax well still requires something on the board itself handling
the (de-)modulation.

Unfortunately, the current state of the art still uses one dsp per
ds0, rather than using a faster dsp and hard real-time scheduling
to process an entire span on one chip, so they re a lot more
expensive than digium's cards, and AFAIK are only available in
one (T1/E1) span per card configurations.

Given the expected adoption of IPP Fax by the multi-function device
vendors I don't expect there will be any multi-span fax-capable cards
coming out either.

Hylafax.org has pointers to a couple of good boards for fax.

-JimC

 http://pwg.org/qualdocs/

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Re: [Asterisk-Users] Re: High Density configuration for Voice Fax

2004-02-11 Thread Darren Nickerson


 To do fax well still requires something on the board itself handling
 the (de-)modulation.

 Unfortunately, the current state of the art still uses one dsp per
 ds0, rather than using a faster dsp and hard real-time scheduling
 to process an entire span on one chip, so they re a lot more
 expensive than digium's cards, and AFAIK are only available in
 one (T1/E1) span per card configurations.

 Given the expected adoption of IPP Fax by the multi-function device
 vendors I don't expect there will be any multi-span fax-capable cards
 coming out either.

 Hylafax.org has pointers to a couple of good boards for fax.

The HylaFAX.org website is a little lacking (and is in some cases so out of
date it's misleading) in terms of describing high-density (T1/E1) fax with
HylaFAX - the focus at hylafax.org is where more of the open-source
community plays ... with 1-2 line analog setups. That's where we come in ...
we specialize in larger stuff ;-)

We recommend Brooktrout or EICON intelligent fax boards. They do ECM error
correction, support 2D MMR compression and dynamic recompression of fax
image data, and they have robust implementations of V.34 (33.6 speed) fax
which can cut down on call setup times and duration, and therefore reduce
toll charges pretty significantly.

-Darren

-- 
Darren Nickerson
Senior Sales  Support Engineer
iFAX Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 ext 8106 office
+1.215.243.8335 fax

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[Asterisk-Users] Force SIP Phones to Register

2004-02-11 Thread Tom Green

Hi,

Is it possible to force a SIP phone to send a register
message to the PBX? I want to change a phone's
extension. By forcing that phone to send a register
msg, I can ensure that the phone is able to make or
receive calls without any delay.

Any pointers/help is appreciated.

TG

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[Asterisk-Users] Integrated T1 PRI (voice and data)

I presently have a T1 with eight voice channels and four data channels. 
  Channels 1-4 are data, 16-23 are voice, and 24 is the dchan.

The vendor plugs the T1 into a Vina Integrator 300 which splits the 
data out to a LAN jack.  This device is only capable of a half duplex 
LAN connection which isn't very good for VoIP.

I would like to plug the T1 circuit straight into the T100P (bypassing 
the Integrator 300).  As I understand it, this interface is capable of 
handling both voice and data.

My question is: how do I configure my IP address, netmask, gateway 
address and so forth?

Thanks,
Mike
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[Asterisk-Users] Asterisk hangs up when a call comes in

2004-02-11 Thread Bodo Hahnke

Hello,

I am trying to setup an asterisk box on a simple isdn line with a fritz card.
The Capi4Linux drivers are installed and seem to work correct as I can
connect to an ISP, have not tried it with ISDN4Linux yet as I read that
CAPI has many advantages over i4l ... but I think I will do this next.
Next I have compiled zaptel, libpri (are these really needed for a fritz card?)
and asterisk and finally did 'make samples' as these are my first expierience
with asterisk and wanted to try the demo context. To get it work with CAPI
i have then installed the chan_capi.so driver from junghanns.net ... here some
output from the cli.
[chan_capi.so] = (Common ISDN API for Asterisk)
  == Parsing '/etc/asterisk/capi.conf': Found
Feb 12 03:08:07 NOTICE[1024]: chan_capi.c:2338 mkif: 
ast_capi_pvt(43910906,43910906,demo,0x2,2) (1,2,64) (0)(0.80/0.80) 0
Feb 12 03:08:07 NOTICE[1024]: chan_capi.c:2338 mkif: 
ast_capi_pvt(43910906,43910906,demo,0x2,2) (1,2,64) (0)(0.80/0.80) 0
Feb 12 03:08:07 NOTICE[1024]: chan_capi.c:2694 load_module: this box has 1 
capi controller(s)
-- listening on contr1 CIPmask = 0x1fff03ff
-- CAPI[contr1] supports DTMF
-- CAPI[contr1] supports supplementary services
HOLD/RETRIEVE
TERMINAL PORTABILITY
ECT
3PTY
CF
CD
MCID
CCBS
MWI
CCNR
  == Registered channel type 'CAPI' (Common ISDN API Driver (0.3.0) aLaw)
 [res_musiconhold.so] = (Music On Hold Resource)
  == Parsing '/etc/asterisk/musiconhold.conf': Found
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold'
  == Registered application 'SetMusicOnHold'
  == Parsing '/etc/asterisk/enum.conf': Found
Asterisk Ready.
*CLI -- started pbx on channel (callgroup=0)!
Feb 12 03:08:19 WARNING[4101]: pbx.c:1778 ast_pbx_run: Channel 
'CAPI[contr1/43910906]/0' sent into invalid extension 's' in context 
'default', but no invalid handler
-- CAPI Hangingup
Feb 12 03:08:33 WARNING[5125]: pbx.c:1778 ast_pbx_run: Channel 
'CAPI[contr1/43910906]/1' sent into invalid extension 's' in context 
'default', but no invalid handler
-- CAPI Hangingup
-- started pbx on channel (callgroup=0)!
Feb 12 03:08:47 WARNING[6149]: pbx.c:1778 ast_pbx_run: Channel 
'CAPI[contr1/43910906]/2' sent into invalid extension 's' in context 
'default', but no invalid handler
-- CAPI Hangingup
-- started pbx on channel (callgroup=0)!
Feb 12 03:08:48 ERROR[3076]: chan_capi.c:1196 pipe_frame: wrote -1 bytes 
instead of 40



Any solution ??

bye

bodo

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[Asterisk-Users] New Zealand

2004-02-11 Thread Wayne Methorst




Can anyone point me in the direction of a Asterisk 
developer in New Zealand that we could contact???

Many thanks
Wayne Methorst
New Zealand
[EMAIL PROTECTED]

Re: [Asterisk-Users] Force SIP Phones to Register

2004-02-11 Thread Todd Lieberman

Tom Green wrote:

Hi,

Is it possible to force a SIP phone to send a register
message to the PBX? I want to change a phone's
extension. By forcing that phone to send a register
msg, I can ensure that the phone is able to make or
receive calls without any delay.
Any pointers/help is appreciated.

TG

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Just set the refresh rate low, say 3600 ms, or reboot the phone.  TL

--
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http://tlsolutions.net
mailto:[EMAIL PROTECTED]
215.500.6913
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[Asterisk-Users] Re: High Density configuration for Voice Fax