date:20030717

Greetings,

I am attempting to configure an AS5300 to provide a SIP based gateway to the PSTN from 
Asterisk. I have been unable to identify through the docs how specifically this should 
be configured in Asterisk and have not been able to get things working through trial 
and error.

I am sure I am missing something fairly obvious here but any guidance (or example 
cfgs) would be much appreciated.

Rgds,
Adam


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intended recipient, please telephone or email the sender and delete this message and 
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Re: [Asterisk-Users] Segmentation fault with chan_oh323

2003-07-17 Thread Michael Manousos

Michael Ulitskiy wrote:
Hi,

I'm trying to interconnect sip and h323 endpoints using asterisk
and asterisk crashes with segmentation fault whenever h323 
connection needs to be established. It registers with gatekeeper ok though.
Here are the symptoms.
If the call initiated by SIP device, asterisk replies to it Trying and then
silently crashes (it launched as asterisk -cd).
In debug log I can see the following:
Jul 16 18:11:52 DEBUG[196621]: File pbx.c, Line 1123 (pbx_extension_helper): Launching 'Dial'
Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1393 (oh323_request): In oh323_request.
Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1394 (oh323_request): type=oh323, format=4, data=phone number.
Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1440 (oh323_request): Created new call structure 0 (2428 bytes).
That's it.
If the call initiated by H323 device, then I see
*CLI   
WrapH323Connection::WrapH323Connection: WrapH323Connection created.
Segmentation fault
and debug log shows:
Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2141 (init_h323_connection): In init_h323_connection...
Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2180 (init_h323_connection): Created new call structure 0 (2428 bytes).
Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1527 (copy_call_details): --- CALL DETAILS ---
Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1528 (copy_call_details): call_token = ip$192.168.0.227:5018/92
Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1529 (copy_call_details): call_source_alias = tnt [192.168.0.227]
Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1530 (copy_call_details): call_dest_alias = 12125551234  12125551234 ip$192.168.0.70:1720
Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1531 (copy_call_details): call_source_e164 = phone number
Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1532 (copy_call_details): call_dest_e164 = 12125551234
That's it. And gatekeeper log shows that after normal ARQ-ACF exchange originating device
immediately sent DRQ.
If anybody knows a reason for this (and the way to fix it of course ;)), I'd appreciate if you let me know.
If you need any additional info to troubleshoot it, let me know too.
First of all, in oh323.conf, set

wrapLibTraceLevel=3
libTraceLevel=3
libTraceFile=/tmp/trace.txt
Run Asterisk again, with -vvvcd, and make it crash.
Then send me (offlist) the trace file, the screen log
and a backtrace of the core file dumped.
Thank a lot.

Michael


Michael.



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Re: [Asterisk-Users] Cisco 7960g

2003-07-17 Thread Yifang Dai

On Wed, Jul 16, 2003 at 12:32:41PM +0200, Siggi Langauf wrote:
  Has anybody tried Cisco 7960G? Or 7940?
 
 sure, using them all the time here (the Skinny version, which requires
 Cisco CallManager which in turn connects to asterisk via H.323).
 
This is very interesting... Can you provide some more details on how you
connect Cisco CallManager with asterisk via H.323? Thanks!

-- 
Yifang Dai   |
eFax: (847)628-0255  |Debian GNU/Linux
[EMAIL PROTECTED] |



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[Asterisk-Users] Cisco 7960

2003-07-17 Thread William Carlson




I bought a 7960 it was running version 3.3 of the 
SIP software. It worked fine. Me being the idiot I am upgraded to 5.1. Now 
it downloads the configs and then reboots. if I unplug the ethernet it doesn't 
rebootor if I remove all the lines in the SIP config it won't reboot. 
Since this is used cisco won't give me any support. For now I am running the 
MGCP version but eh asterisk seems to have some issues with it.
 Thanks,
 
Will

Re: [Asterisk-Users] Asterisk - AS5300 SIP Interoperability

2003-07-17 Thread Daniel Concepcion

Hi Adam, 

I have an AS5300 working well with Asterisk. I have the following cfg: 

Asterisk Server: 10.11.0.3
AS5300: 10.11.0.2
in sip.conf

[sa1-voip]
context=sa1-voip
type=friend
host=10.11.0.2
dtmf=rfc2833

in extensions.conf


[sa1-voip]
;
; Llamadas Externas

exten = _0.,1,SetCallerID()
exten = _0.,2,SetCIDName(X)
exten = _0.,3,Dial(SIP/[EMAIL PROTECTED])

In the AS5300:

dial-peer voice 1000 voip
 application session
 destination-pattern .T
 voice-class codec 10
 session protocol sipv2
 session target ipv4:10.11.0.3
 session transport udp
 dtmf-relay rtp-nte
!
dial-peer voice 100 pots
 application session
 max-conn 30
 destination-pattern 0.
 translate-outgoing called 1
 no digit-strip
 direct-inward-dial
 port 0:D
 forward-digits all
!
sip-ua
 retry invite 4
 retry response 3
 retry bye 2
 retry cancel 2
 sip-server ipv4:10.11.0.3

regards,
Daniel


On Thursday 17 July 2003 11:33, Low, Adam wrote:
 Greetings,

 I am attempting to configure an AS5300 to provide a SIP based gateway to
 the PSTN from Asterisk. I have been unable to identify through the docs how
 specifically this should be configured in Asterisk and have not been able
 to get things working through trial and error.

 I am sure I am missing something fairly obvious here but any guidance (or
 example cfgs) would be much appreciated.

 Rgds,
 Adam


 * 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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Re: [Asterisk-Users] Cisco 7960

2003-07-17 Thread William Carlson




lol well I probaly should ask a question lol. Any 
idea what could be causing this? Also I cannot call from my pingtel phone to the 
7960 but I can call the other way around. any ideas on that?
 Thanks,
 Will

- Original Message - 

  From: 
  William Carlson 
  
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, July 17, 2003 7:34 
  AM
  Subject: [Asterisk-Users] Cisco 
7960
  
  I bought a 7960 it was running version 3.3 of the 
  SIP software. It worked fine. Me being the idiot I am upgraded to 5.1. 
  Now it downloads the configs and then reboots. if I unplug the ethernet it 
  doesn't rebootor if I remove all the lines in the SIP config it won't 
  reboot. Since this is used cisco won't give me any support. For now I am 
  running the MGCP version but eh asterisk seems to have some issues with 
  it.
   Thanks,
   
Will

Re: [Asterisk-Users] Asterisk - AS5300 SIP Interoperability

2003-07-17 Thread Marsico, Gustavo - (Arg)




Adam:

The * configuration should be 
something like this:

extensions.conf:

[globals]

GW5300=xxx.xxx.xxx.xxx 
[5300 IP]

[carriers]

exten = 
_100.,1,Dial,SIP/[EMAIL PROTECTED]In the 5300 (enable 
mode):

sip-ua
 retry invite 
4
 retry 
response 3
 retry bye 
2
 retry cancel 
2
 sip-server 
ipv4:xxx.xxx.xxx.xxx 
[Asterisk IP]
!

dial-peer voice 8 
voip
 application 
session
 
destination-pattern 555693..
 
translate-outgoing called 1001
 voice-class 
codec 1
 session 
protocol sipv2
 session 
target 
ipv4:xxx.xxx.xxx.xxx[Asterisk 
IP]
!


Hope this 
help,

Regards,

Gustavo


- Greetings,- - 
I am attempting to configure an AS5300 to provide a SIP based gateway to 
the PSTN from Asterisk. I have been unable to identify through the docs how 
specifically this - should be 
configured in Asterisk and have not been able to get things working through 
trial and error.- - I am sure I am missing something fairly 
obvious here but any guidance (or example cfgs) would be much 
appreciated.- - Rgds,- - 
Adam

Este mensaje es confidencial. El mismo contiene información reservada 

y que no puede ser difundida. Si usted ha recibido este e-mail 

por error, por favor avísenos inmediatamente vía e-mail y tenga la 

amabilidad de eliminarlo de su sistema; no deberá copiar el mensaje 

ni divulgar su contenido a ninguna persona. Muchas gracias.

 

This message is confidential. It contains information that is privileged and

legally exempt from disclosure. If you have received this e-mail by mistake, 

please let us know immediately by e-mail and delete it from your system; 

you should also not copy the message nor disclose its contents to anyone. 

Thank You.

[Asterisk-Users] E1 R2 on Asterisk

2003-07-17 Thread LQ (Asterisk)

Dear fellows,

I need to use Asterisk with an E1 card with CAS R2 signalling for Argentina.
I know that the E100P don't support it right now.
Does anybody developing R2 drivers?

Any other alternatives/prices to handle an E1 R2 on Asterisk?

Thanks in advance!.

Best regards,
Pablo.


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[Asterisk-Users] slightly OT /how to obtain 900 number

2003-07-17 Thread firedude

I'm interested in offering fee based support services by telephone.  Does 
anyone have any suggestions on how I could obtain a 900 number to do this?  
My initial thought is to have the 900 number terminate to my asterisk 
server which will connect callers to cus service reps / techs in various 
locations through IAX or SIP.  

I really would prefer to do this via 900 to avoid reps losing time having 
to enter and validate cc information but if anyone is familiar with an 
easy way to do this other than 900 please let me know.
AJ

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[Asterisk-Users] outgoing callerid string

2003-07-17 Thread firedude

Is there a way for me to set my outgoing callerid string so that all 
callers outside of my pbx see our callerid string as company name main 
company number but callers inside our telephone network see extension 
holders name extension number?  In looking at the references it looks like 
I can do one or the other but not both.  Does anyone know how I might 
accomplish this?
Thanks for any suggestions.
AJ

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RE: [Asterisk-Users] Asterisk - AS5300 SIP Interoperability

Thanks Daniel  Gustavo,

I had the AS5300 configured ok and could make calls PSTN  AS5300  ASTERISK  7940 no 
problem but outbound from Asterisk to the AS5300 wasn't working ... until now (wasn't 
sure about the sip.conf)  ... thanks again gents !

 -Original Message-
 From: Daniel Concepcion [mailto:[EMAIL PROTECTED] 
 Sent: 17 July 2003 13:40
 To: [EMAIL PROTECTED]; Low, Adam
 Subject: Re: [Asterisk-Users] Asterisk - AS5300 SIP Interoperability

* DISCLAIMER * 

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intended recipient, please telephone or email the sender and delete this message and 
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Re: [Asterisk-Users] E1 R2 on Asterisk

2003-07-17 Thread Steve Underwood

LQ (Asterisk) wrote:

Dear fellows,

I need to use Asterisk with an E1 card with CAS R2 signalling for Argentina.
I know that the E100P don't support it right now.
Correct

Does anybody developing R2 drivers?

Yes.

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[Asterisk-Users] outgoing callerid string

2003-07-17 Thread Derek Beaumont

You can set this in zapata.conf.
Basically, you just define the callerid before you define the channel.
Outgoing channels will have the callerid of the company, and internal
channels will have the callerid of the extension.

In zapata.conf
Example:
;I don't think you have to change anything in the fxo_ks channels
because that will be handled by the phone company.

signalling=fxs_ks
callerid=Receptionist 0
channel=4

callerid=Adam West 555
channel=5


Is there a way for me to set my outgoing callerid string so that all 
callers outside of my pbx see our callerid string as company name main 
company number but callers inside our telephone network see extension 
holders name extension number?  In looking at the references it looks
like 
I can do one or the other but not both.  Does anyone know how I might 
accomplish this?
Thanks for any suggestions.
AJ

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[Asterisk-Users] UK Gateway

2003-07-17 Thread Linus Surguy

We're in the process of testing some equipment and configurations and to do
this we have setup a UK PSTN Gateway to Free World Dialup.

Simply dial 0845 004 5566 (UK local rate call) and at the prompt enter the
FWD subscriber number - within a couple of seconds you should be connected.

We can also terminate UK 0800/0808 numbers for SIP/IAX - PSTN calls, at the
moment we don't have an FWD number setup for this, but simply use the
username/password of guest/guest and point your connection at
voip-gw1.magrathea-telecom.co.uk and send the number as 0800xxx

Of course, this is all a trial at the moment, so no commerical warranties
are available!

Finally, if you wanted to trial your own personal 0870 number pointing to
FWD, drop me an e-mail and I'll tell you how to do it.

Linus
Magrathea



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[Asterisk-Users] Echo on incoming calls (PRI-SIP) but not on outgoing (SIP-PRI)

2003-07-17 Thread fredrik . hedberg

Hello all!
 We have an E400P on a dual Xeon box with four EuroISDN PRI to the PSTN
and Cisco ATA-186 as SIP UA.  We experience very large amount of echo (on -some- 
calls) when we're
doing PRI-SIP calls but not when doing SIP-PRI calls.
 We don't think the problem is IP latency related (When calls are made
PRI-*-PRI we still experience the echo). What do you think?  Regards
Fredrik Hedberg
 









Hello all!



We have an E400P on a dual
Xeon box with four EuroISDN PRI to the PSTN and Cisco ATA-186 as SIP UA. 



We experience very large
amount of echo (on -some- calls) when were doing PRI-SIP calls but
not when doing SIP-PRI calls.



We dont think the
problem is IP latency related (When calls are made PRI-*-PRI we still
experience the echo). What do you think? 



Regards
Fredrik Hedberg

Re: [Asterisk-Users] Cisco 7960

2003-07-17 Thread Pavel Litvinenko

William Carlson wrote:

lol well I probaly should ask a question lol. Any idea what could be 
causing this? Also I cannot call from my pingtel phone to the 7960 but 
I can call the other way around. any ideas on that?
  Thanks,
 Will
 
- Original Message -

*From:* William Carlson mailto:[EMAIL PROTECTED]
*To:* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
*Sent:* Thursday, July 17, 2003 7:34 AM
*Subject:* [Asterisk-Users] Cisco 7960
I bought a 7960 it was running version 3.3 of the SIP software. 
It worked fine. Me being the idiot I am upgraded to 5.1. Now it
downloads the configs and then reboots. if I unplug the ethernet
it doesn't reboot or if I remove all the lines in the SIP config
it won't reboot. Since this is used cisco won't give me any
support. For now I am running the MGCP version but eh asterisk
seems to have some issues with it.
   Thanks,
 Will

I was trying to get work 7940 MGCP with * - bad idea  :) ... but it was 
3.x version

--

-
Best Regards,
Pavel Litvinenko.
ICQ: 16224754
Ph: (8632) 923962, 923640
sip:[EMAIL PROTECTED]


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Re: [Asterisk-Users] Cisco 7960

2003-07-17 Thread Dan

Hi Will,

Take care that starting with the version 5.x you cannot do downgrades
anymore.
You're stuck with this version till a new release will be available.

BR,
Dan

- Original Message - 
From: William Carlson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 17, 2003 2:34 PM
Subject: [Asterisk-Users] Cisco 7960


I bought a 7960 it was running version 3.3 of the SIP software.  It worked
fine. Me being the idiot I am upgraded to 5.1. Now it downloads the configs
and then reboots. if I unplug the ethernet it doesn't reboot or if I remove
all the lines in the SIP config it won't reboot. Since this is used cisco
won't give me any support. For now I am running the MGCP version but eh
asterisk seems to have some issues with it.
   Thanks,
 Will


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RE: [Asterisk-Users] Cisco 7960

Title: Message



William,

I am running 7960/7940's with 5.1 (Asterisk SIP) without problems 
although I did have some issues (too numerous to mention)with new phones 
that had never been operated on a CallManager network first. It seems the 
firmware must be upgraded to support SIP and this can only be done with 
CallManager (apparently).

The only way I managed to figure everything out was with a packet 
analyser, I don't suppose you have the possibility of doing that 
?

Rgds, Adam

  
  -Original Message-From: William Carlson 
  [mailto:[EMAIL PROTECTED] Sent: 17 July 2003 13:40To: 
  [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Cisco 
  7960
  lol well I probaly should ask a question lol. Any 
  idea what could be causing this? Also I cannot call from my pingtel phone to 
  the 7960 but I can call the other way around. any ideas on that?
   Thanks,
   Will
  
  - Original Message - 
  
From: 
William Carlson 

To: [EMAIL PROTECTED] 

Sent: Thursday, July 17, 2003 7:34 
AM
Subject: [Asterisk-Users] Cisco 
7960

I bought a 7960 it was running version 3.3 of 
the SIP software. It worked fine. Me being the idiot I am upgraded to 
5.1. Now it downloads the configs and then reboots. if I unplug the ethernet 
it doesn't rebootor if I remove all the lines in the SIP config it 
won't reboot. Since this is used cisco won't give me any support. For now I 
am running the MGCP version but eh asterisk seems to have some issues with 
it.
 Thanks,
 
  Will



* 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

RE: [Asterisk-Users] Help Needed

2003-07-17 Thread Arun Kumar Sharma, Noida

Thanks Adam,

This document provides me a high level architecture of Asterisk. Can you
please tell me if I want to evaluate Asterisk on an Intel PC which Quicknet
hardware will be required to just run a POTS to SIP call?

Thank you once again for very fast response.

Regards
Arun


-Original Message-
From: Low, Adam [mailto:[EMAIL PROTECTED]
Sent: 17 July 2003 19:11
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] [Asterisk-Users]Help Needed


http://www.digium.com/handbook-draft.pdf

 -Original Message-
 From: Arun Kumar Sharma, Noida [mailto:[EMAIL PROTECTED] 
 Sent: 17 July 2003 15:27
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] [Asterisk-Users]Help Needed


* 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
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Re: [Asterisk-Users] grandstream sip phone (NTP)

2003-07-17 Thread Stephen R. Besch

I have solved the time server problem with the Grandstream by having my 
* box's NTP service mirror a public NTP server.  I had to do this 
because my phones are all on the 192.168 subnet, which is non-routable. 
For example, assuming that the NTP service is configured and running on 
your * box, create an NTP mirror which allows access from machines on 
192.168.10.X by adding the following line to the ntp.conf file:

restrict 192.168.10.0 mask 255.255.255.0 notrust nomodify notrap

The IP range and netmask arguments are obvious.  The 3 option flags tell 
the ntp daemon that none of the machines that might communicate over 
this subnet are to be trusted as time servers, none of them are to be 
allowed to update the ntp daemon running on the asterisk server, and 
none of them will be able to use the trap service for logging purposes.

Finally, I also like to set up a different (from the one used by the 
phones for SIP and RTP) IP address for the NTP server (so the * box has 
2 addresses on the 192.168 net). It goes without saying that the 
asterisk box must also have a public IP address so that it can 
synchronize itself with a remote time server. In my setup, I have one 
net card for the public address, while the 2 192.168 addresses are on a 
second card.

--
Stephen R. Besch, Ph.D.
SachsLab
320 Cary Hall
SUNY at Buffalo
Buffalo, NY 14214
(716) 829-3289 x106
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[Asterisk-Users] H323/No one is available to answer at this time

2003-07-17 Thread isamar



Hi folks,

When dialing H323, I'm getting:
No one is available to answer at this time.

Anybody knows why this happen?

Thanks in advance,

Isamar

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RE: [Asterisk-Users] Help Needed

Not me I'm afraid, I'm running Asterisk -SIP- Cisco AS5300 -E1- PSTN .. no Quicknet 
hardware for me ...

 -Original Message-
 From: Arun Kumar Sharma, Noida [mailto:[EMAIL PROTECTED] 
 Sent: 17 July 2003 15:49
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Help Needed

 Thanks Adam,

 This document provides me a high level architecture of 
 Asterisk. Can you
 please tell me if I want to evaluate Asterisk on an Intel PC 
 which Quicknet
 hardware will be required to just run a POTS to SIP call?

 Thank you once again for very fast response.

 Regards
 Arun

* 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 
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Re: [Asterisk-Users] Call Pickup

2003-07-17 Thread Martin Pycko

You need to have a pending call in the system (some extensions that is
ringing to test that). If you have 3 FXS ports try to place a call from
the first one to the 2nd and then instead of taking the 2nd off hook dial
*8 on the 3rd phone

Martin

On Thu, 17 Jul 2003, Jay Tyndall wrote:

   Hi,

 I have been trying to workout how to use the call pickup.

 So Far, I have the following in zapata.conf
 [channels]
 signalling = fxo_ks
 context = local
 pickupgroup=1
 callgroup=1
 channel = 1-3


 When I dial *8# all I hear is busy tone.

 What have I missed?

 thanks
 Jay.
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[Asterisk-Users] AVM Fritz! to connect LAN with ISDN line?

2003-07-17 Thread Achim J. Latz

Hello,

Is it possible to use * as a gateway in the following setup:

   LAN (with Windows NT/Linux PCs) 
 |
Ethernet (IP)
 |
  Linux PC with * and AVM Fritz! ISDN Adapter
 |
   ISDN
 |
   Someone with a analog/digital phone (POTS)

Basically, people sitting on their PCs will wear a headset, and whenever 
they want to call someone, they start a phone application (e.g. 
Openphone) and dial the external/internal number. This software contacts 
*, and * establishes the connection (notifying the local/LAN user, or 
making a call through the ISDN interface to the external number).

Additionally, incoming calls to the * gw are routed to the LAN PC where 
the user with the corresponding extension is logged on. (optional)

Has someone done this? Does anyone have a (more or less) detailed
instruction routine (e.g., what client software, codecs, ...)

Thank you very much,

Achim


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[Asterisk-Users] Any dialing tricks...

2003-07-17 Thread Kim C. Callis









Alright, I am basically cheap, and I have a cellular plan
which allows for free incoming calls (Nextel). I was wondering if there was any
way to do sort of a dialback trick in the extensions.conf I call into the system from my cell
phone (maybe via DISA), I dial an internal extension, and dial a phone number
Then * sends to my cellphone the number dialed thus
giving me a in call on the cell. Or maybe have a call
back with a DISA and then just dial my phone number I am trying to reach



Just a thought!



Kim Callis

Re: [Asterisk-Users] AVM Fritz! to connect LAN with ISDN line?

2003-07-17 Thread Rainer Jochem


LAN (with Windows NT/Linux PCs) 
  |
 Ethernet (IP)
  |
   Linux PC with * and AVM Fritz! ISDN Adapter
  |
ISDN
  |
Someone with a analog/digital phone (POTS)

Sure. Works fine here with an Fritz!Card PCI v2.2
 

 Has someone done this? Does anyone have a (more or less) detailed
 instruction routine (e.g., what client software, codecs, ...)

We use 

Cisco 7960 / kphone (SIP) - Linux / X-Lite (SIP) - Windows
 |
LAN
 |
 Asterisk on a Debian box with an AVM Fritz!
 |
   ISDN

You should use chan_capi, CAPI for the Fritz!-Card 
(available from AVM) and don't forget to turn SMP off
in your kernel. (Otherwise the AVM CAPI-driver won't
work)
 


-- 
http://graphics.cs.uni-sb.de/VoIP/
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[Asterisk-Users] Can I interoperate with public PSTN gateways ?

2003-07-17 Thread David Boreham

Apologies if this is an FAQ, I wasn't able to find an answer googling:

Will any of the public PSTN/VoIP gateway services (Vonage, Packet8 etc)
interoperate with * ?

I'd like to deploy a box which provides PBX service for analog handsets,
and handles inbound/outbound calls via both analog PSTN lines, and,
say Packet8 VoIP service. I understand that I can do this by connecting
the analog side of an ATA to an analog card in a machine running *,
but I'd prefer to terminate the VoIP traffic directly---I figure that eventually
these services will allow multiple concurrent calls, and at that time I'd
need multiple ATA type boxen, which seems rather silly.

Hops that made sense, and thanks.


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Re: [Asterisk-Users] Any dialing tricks...

On Thu, 2003-07-17 at 10:38, Kim C. Callis wrote:
 Alright, I am basically cheap, and I have a cellular plan which allows
 for free incoming calls (Nextel). I was wondering if there was any way
 to do sort of a dialback trick in the extensions.conf I call into the
 system from  my cell phone (maybe via DISA), I dial an internal
 extension, and dial a phone number Then * sends to my cellphone the
 number dialed thus giving me a in call on the cell. Or maybe have a
 call back with a DISA and then just dial my phone number I am trying
 to reach

It is possible for your asterisk box to detect your callerid provided it
is available on your asterisk line, and use that as a pattern match on
incoming calls to direct you to a slightly different extension. From
there you can drop a sample call in the outgoing queue and then answer
the line long enough to hang it up. This should make your outgoing call
be about 1 second long, and allow you positive knowledge that the
asterisk system received and acknowledged your call. The sample.call
file can call you back then and drop you in a context where you can
access anything you are interested in.

See easy. 
-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] Cisco 7960g

2003-07-17 Thread Siggi Langauf

On Thu, 17 Jul 2003, Yifang Dai wrote:

 On Wed, Jul 16, 2003 at 12:32:41PM +0200, Siggi Langauf wrote:
   Has anybody tried Cisco 7960G? Or 7940?
 
  sure, using them all the time here (the Skinny version, which requires
  Cisco CallManager which in turn connects to asterisk via H.323).
 
 This is very interesting... Can you provide some more details on how you
 connect Cisco CallManager with asterisk via H.323? Thanks!

There's not much to it: just configure * as an H.323 gateway in
CallManager for the appropriate extensions.
If you need to route calls from * to CCM, just use something like

Dial(OH323/callto:[EMAIL PROTECTED])

in /etc/Asterisk/extensions.conf.
${EXTEN} is the CallManager extension you're going to dial, and
callmanager.your-domain.com is the CCM's host name (IP address is safer.)

The * part is a bit trickier: I had to use current CVS versions of both
asterisk and openh323/pwlib. Moreover, the H323 channel driver that comes
with asterisk will _not_ work with CCM. (It requires an older version of
openh323, and it will send voice Data to the call manager instead of the
telephone, which makes it 'one way'.)

The current 0.5.3 release of Michael's OH323 channel driver
(http://www.inaccessnetworks.com/projects/asterisk-oh323/) works fine.

Cheers,
Siggi

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[Asterisk-Users] error WARNING[28697]: File app_dial.c, Line 304 (wait_for_answer):Unable to forward voice

2003-07-17 Thread Cristi

I am trying to put a call on a E1 ISDN :
The configuration are simple:
zapata.conf :
[channels]
context=inbound
switchtype=euroisdn
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
;echocancel=no
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
;immediate=yes
immediate=no
callerid = asreceived
amaflags = billing
usecallerid=yes
overlapdial=yes
; Span 1
group=1
context=inbound
signalling=pri_cpe
channel = 1-15
channel = 17-31
; Span 2
group=2
context=inbound
signalling=pri_cpe
channel = 32-46
channel = 48-62
; Span 3
group=3
;context=h323
context=outbound
signalling=pri_cpe
channel = 63-77
;channel = 79-93
;group=5
;context=h323
;signalling=pri_cpe
channel = 79-93
; Span 4
group=4
context=outbound
signalling=pri_cpe
channel = 94-108
channel = 110-124
extension.conf
exten = ,1,Wait(1)
exten = ,2,Answer
exten = ,3,Playback(beep)
exten = ,4,Dial(Zap/g3/0007352638)
and I get this error

You can see the output from pri debug span3:
*CLI pri debug span 3
Enabled debugging on span 3
 Protocol Discriminator: Q.931 (8)  len=41
 Call Ref: len= 2 (reference 10309/0x2845) (Originator)
 Message type: SETUP (5)
 Sending Complete (len= 4)
 Bearer Capability (len= 3) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: 3.1kHz audio (16)
  Ext: 1  Trans mode/rate: 64kbps, 
circuit-mode (16)
  Ext: 1  User information layer 1: A-Law (35)
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Preferred 
Dchan: 0
ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3
   Ext: 1  Channel: 13 ]
 Progress Indicator (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0) 
0: 0   Location: Public network serving the remote user (4)
   Ext: 1  Progress Description: Calling 
equipment is non-ISDN. (3) ]
 Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI: 
Unknown Number Plan (0)
   Presentation: Unknown (3) '0212318657' ]
 Called Number (len= 7) [ Ext: 1  TON: Unknown Number Type (0)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '' ]
-- Making new call for cr 10309
-- Processing Q.931 Call Setup
-- Processing IE 33 (Sending Complete)
-- Processing IE 4 (Bearer Capability)
-- Processing IE 24 (Channel Identification)
-- Processing IE 30 (Progress Indicator)
-- Processing IE 108 (Calling Party Number)
-- Processing IE 112 (Called Party Number)
   -- Accepting call from '0212318657' to '' on channel 13, span 3
 Protocol Discriminator: Q.931 (8)  len=14
 Call Ref: len= 2 (reference 43077/0xA845) (Terminator)
 Message type: SETUP ACKNOWLEDGE (13)
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
Exclusive Dchan: 0
ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3
   Ext: 1  Channel: 13 ]
 Progress Indicator (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard 
(0) 0: 0   Location: Private network serving the local user (1)
   Ext: 1  Progress Description: Called 
equipment is non-ISDN. (2) ]
   -- Executing Wait(Zap/75-1, 1) in new stack
   -- Executing Answer(Zap/75-1, ) in new stack
 Protocol Discriminator: Q.931 (8)  len=14
 Call Ref: len= 2 (reference 43077/0xA845) (Terminator)
 Message type: CONNECT (7)
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
Exclusive Dchan: 0
ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3
   Ext: 1  Channel: 13 ]
 Progress Indicator (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard 
(0) 0: 0   Location: Private network serving the local user (1)
   Ext: 1  Progress Description: Called 
equipment is non-ISDN. (2) ]
   -- Executing Playback(Zap/75-1, beep) in new stack
   -- Playing 'beep'
 Protocol Discriminator: Q.931 (8)  len=5
 Call Ref: len= 2 (reference 10309/0x2845) (Originator)
 Message type: CONNECT ACKNOWLEDGE (15)
   -- Executing Dial(Zap/75-1, Zap/g3/0007894638) in new stack
Making new call for cr 32771
 Protocol Discriminator: Q.931 (8)  len=45
 Call Ref: len= 2 (reference 3/0x3) (Originator)
 Message type: SETUP (5)
 Bearer Capability (len= 3) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: Speech (0)
  Ext: 1  Trans mode/rate: 64kbps, 
circuit-mode (16)
  Ext: 1  User information layer 1: A-Law (35)
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
Exclusive Dchan: 0
ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3
   Ext: 1  Channel: 1 ]
 Display (len= 1) [ 1 ]
 Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI: 
ISDN/Telephony

Re: [Asterisk-Users] Cisco 7960

2003-07-17 Thread Anton L. Kapela

William Carlson said:
 lol well I probaly should ask a question lol. Any idea what could be
 causing this? Also I cannot call from my pingtel phone to the 7960 but
 I can call the other way around. any ideas on that?
   Thanks,
  Will

Will,

My experiences with the 7940's and 7960's using Asterisk tell me that
the common syntax examples (the line allow=all) won't work in many
cases. That is to say, calling a 7940 to 7940 and 7940 to the PBX
(trying to check voicemail, for example) would never work.. I'd see
errors like:

-- Executing Ringing(SIP/2000-ab10, ) in new stack
-- Executing Wait(SIP/2000-ab10, 2) in new stack
-- Executing VoiceMailMain(SIP/2000-ab10, ) in new stack
  == Parsing '/etc/asterisk/voicemail.conf': Found
-- Playing 'vm-login'
WARNING[950286]: File app_voicemail.c, Line 1907 (vm_execmain):
Couldn't read username
  == Spawn extension (internal, 2999, 3) exited non-zero on
'SIP/2000-ab10'

or like this from extension to extension calls:

-- Executing Dial(SIP/2000-317a, SIP/2001|30|tr) in new stack
-- Called 2001
-- Got SIP response 488 Not Acceptable Here back from 172.16.0.253
  == No one is available to answer at this time
-- Executing VoiceMail(SIP/2000-317a, u2001) in new stack
  == Parsing '/etc/asterisk/voicemail.conf': Found
-- Playing 'vm/2001/unavail'
  == Spawn extension (internal, 2001, 2) exited non-zero on
'SIP/2000-317a'

The _only_ usefull shred of log data here (withough enabling sip
debug) is the Got SIP response 488 Not Acceptable Here string. What
I've found this seems to mean is the codec the phone is
offering/attempting to use is invalid, it otherwise not usable.

I find everything SIP-related works perfectly when I force ulaw, alaw,
or gsm.

My advice would be to try this in your [general] section of the sip.conf:

allow=gsm
allow=ulaw
allow=alaw

Let us know if it works then :)

--Tk
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RE: [Asterisk-Users] Cisco 7960

2003-07-17 Thread Matthew Hardeman










Ive run into this before, and its
a pain to debug



Be sure that your eth0 interface (primary,
first interface) is set to your internal address space (of the same subnet that
you assign to the phone). You can add an
IP alias on eth0:1 if you need an external IP on that box as well, but you must
have them in that order: internal = eth0, externals, others eth0:1+



Try that



Matt

PaperSoft





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William Carlson
Sent: Thursday, July 17, 2003 6:35
AM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco
7960





I bought a 7960 it was running
version 3.3 of the SIP software. It worked fine. Me being the idiot I am
upgraded to 5.1. Now it downloads the configs and then reboots. if I unplug the
ethernet it doesn't rebootor if I remove all the lines in the SIP config
it won't reboot. Since this is used cisco won't give me any support. For now I
am running the MGCP version but eh asterisk seems to have some issues with it.





 Thanks,





 Will

[Asterisk-Users] Sip call question

2003-07-17 Thread Skuse, Phil


There's something that I want to set up in our lab for testing purposes, but
I'm not sure how to do it.

I would like to be able to call an asterisk extension, and then enter a SIP
address using DTMF, and then have asterisk make a SIP transfer to that
address.

For example:

If I dial extn followed by *192*168*0*10*5060 I would like to be
transferred to sip://[EMAIL PROTECTED]:5060. But I don't want to have to
register the IP address beforehand in any config files.

Any idea how I would do this? I'm guessing that I either need to collect the
DTMF, format it into a sip address and then somehow get asterisk to dial
that address, or perhaps I can take it from $EXTEN somehow.

Is this possible with the existing apps/scripts/macros, or do I need to
write some new ones?

Phil Skuse [EMAIL PROTECTED]

 UNIX System Administrator, Vicorp Group Limited.   
 Tel  +44 (0)1753 660523  Fax +44 (0)1753 660501
 The Telephony Engine Company http://www.vicorp.com

 
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RE: [Asterisk-Users] Any dialing tricks...

2003-07-17 Thread Matthew Hardeman









Hey Kim!



I used to run that scam myself! You go! Back in the day, actually, I had our
legacy lucent merlin phone system wired up to a modem on a webserver which
could config it And with some voicemail tricks and the like, it was
possible for me to visit a little WAP site on my phone, and have it dial the
number and bridge the call to me



In asterisk it would be a lot more
graceful You could build a
little script to look for a two-way message from you and use the outbound call
spool to set up a call


Matt Hardeman

PaperSoft







-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kim C. Callis
Sent: Thursday, July 17, 2003 10:38 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Any
dialing tricks...



Alright, I am basically cheap, and I
have a cellular plan which allows for free incoming calls (Nextel). I was
wondering if there was any way to do sort of a dialback trick in the
extensions.conf I call into the system from my cell phone (maybe via DISA), I dial
an internal extension, and dial a phone number Then * sends to my
cellphone the number dialed thus giving me a in call on the cell. Or maybe have
a call back with a DISA and then just dial my phone number I am trying to
reach



Just a thought!



Kim Callis

[Asterisk-Users] random hangups

Hi ,

I''m getting random hangups on zap channels with long calls. It seems that the
hungup happens after 10 minutes or so. AbsoluteTimeout is set to 0. Any other
thing I should be configuring?

Thanks!

PHM

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Re: [Asterisk-Users] conference problem without zapata interface

2003-07-17 Thread Robert Hajime Lanning

You need to load the ztdummy kernel driver.  It will provide the pseudo
timing needed to sync the conference channel.

It is a driver that creates a dummy Zaptel hardware interface.

quote who=Andrzej Radke
 Hello !

 In file app_meetme.c we can read
 A ZAPTEL INTERFACE MUST BE\n
 INSTALLED FOR CONFERENCING FUNCTIONALITY.\n

 I receive message, when I try conference
 WARNING[28686]: File app_meetme.c, Line 151 (build_conf): Unable to open
 pseudo channel
 -- Playing 'conf-invalid'


 Does it means that I cannot establish conference without
 any hardware zaptel interface ???

 What can I do if I want make conference only between my sip phones
 using asterisk ??  Buy it ???

 Greeting
 Andrzej Radke




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Re: [Asterisk-Users] Chan_H323, G729 (minor problem)

2003-07-17 Thread Dave Alan Caruana

ok ...
I removed the dtmfmode=inband
from the h323.conf file which resulted in the error messages vanishing ..
ya I thought ...

alas DTMF tones sent to an IVR at the other end of the connection
do not work either!!!

My incoming calls are coming from PSTN lines through an E1
so DTMF must be inline .. THe (thousands of) error messages
aren't really a problem, just annoying.

Dave

- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 15, 2003 4:28 PM
Subject: Re: [Asterisk-Users] Chan_H323, G729 (minor problem)


 You're trying to detect inband dtmfs from the codec stream.

 Martin

 On Tue, 15 Jul 2003, Dave Alan Caruana wrote:

  hi ..
 
  I have finally managed to get Chan_H323  G729 working
  flawlessly, thanks to some help from Jerry McNamara.
  For those out there who are stuck with the same problem
  the procedure is :
  1. install on RedHat 8.0 and nothing else (RH9 doesn't work!)
  2. Install asterisk, zaptel etc. the normal way
  3. Compile Pwlib  oH323 with versions taken from nufone's
  site (http://www.nufone.net/downloads) since the latest versions
  do not have support for G729. Remember to set the environment
  versions as described in the Readme files.
  4. Modify the makefile of chan_h323 (which is in
  /usr/src/asterisk/channels/h323)
  to re-enable the G729 code.
  5. in h323.conf put in allow=g729
  and you should have a working configuration ..
 
  now for my question ..
  during G729 calls I am getting repeatedly the message
 
  WARNING[311314]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
detect
  process 256 frames
 
  this scrolls up the screen at a very high rate of knots.. the call is
  unaffected and goes through normally.
  Is this something wrong? normal? can it be fixed/suppressed?
 
  cheers
  Dave
 
 
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Re: [Asterisk-Users] AVM Fritz! to connect LAN with ISDN line?

2003-07-17 Thread WipeOut .

Yup that will work.. I have the same setup on a Redhat9 system..


 Hello,
 
 Is it possible to use * as a gateway in the following setup:
 
LAN (with Windows NT/Linux PCs) 
  |
 Ethernet (IP)
  |
   Linux PC with * and AVM Fritz! ISDN Adapter
  |
ISDN
  |
Someone with a analog/digital phone (POTS)
 
 Basically, people sitting on their PCs will wear a headset, and whenever 
 they want to call someone, they start a phone application (e.g. 
 Openphone) and dial the external/internal number. This software contacts 
 *, and * establishes the connection (notifying the local/LAN user, or 
 making a call through the ISDN interface to the external number).
 
 Additionally, incoming calls to the * gw are routed to the LAN PC where 
 the user with the corresponding extension is logged on. (optional)
 
 Has someone done this? Does anyone have a (more or less) detailed
 instruction routine (e.g., what client software, codecs, ...)
 
 Thank you very much,
 
   Achim
 
 
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[Asterisk-Users] Video Phones?

2003-07-17 Thread Dave Packham

anyone using a SIP based video phone with * yet?   I would like to buy some but would 
like it to work with * first

Thanks

Dave Packham

ie p0lar

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Re: [Asterisk-Users] random hangups

2003-07-17 Thread Martin Pycko

do you have in zapata.conf

busydetect=yes
or
callprogress=yes ?

Martin

On Thu, 17 Jul 2003, Paulo H. Mannheimer wrote:

 Hi ,

 I''m getting random hangups on zap channels with long calls. It seems that the
 hungup happens after 10 minutes or so. AbsoluteTimeout is set to 0. Any other
 thing I should be configuring?

 Thanks!

 PHM

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Re: [Asterisk-Users] Chan_H323, G729 (minor problem)

2003-07-17 Thread Martin Pycko

dtmfmode=rfc2833

or

dtmfmode=info

try that instead

Martin

On Thu, 17 Jul 2003, Dave Alan Caruana wrote:

 ok ...
 I removed the dtmfmode=inband
 from the h323.conf file which resulted in the error messages vanishing ..
 ya I thought ...

 alas DTMF tones sent to an IVR at the other end of the connection
 do not work either!!!

 My incoming calls are coming from PSTN lines through an E1
 so DTMF must be inline .. THe (thousands of) error messages
 aren't really a problem, just annoying.

 Dave

 - Original Message -
 From: Martin Pycko [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Tuesday, July 15, 2003 4:28 PM
 Subject: Re: [Asterisk-Users] Chan_H323, G729 (minor problem)


  You're trying to detect inband dtmfs from the codec stream.
 
  Martin
 
  On Tue, 15 Jul 2003, Dave Alan Caruana wrote:
 
   hi ..
  
   I have finally managed to get Chan_H323  G729 working
   flawlessly, thanks to some help from Jerry McNamara.
   For those out there who are stuck with the same problem
   the procedure is :
   1. install on RedHat 8.0 and nothing else (RH9 doesn't work!)
   2. Install asterisk, zaptel etc. the normal way
   3. Compile Pwlib  oH323 with versions taken from nufone's
   site (http://www.nufone.net/downloads) since the latest versions
   do not have support for G729. Remember to set the environment
   versions as described in the Readme files.
   4. Modify the makefile of chan_h323 (which is in
   /usr/src/asterisk/channels/h323)
   to re-enable the G729 code.
   5. in h323.conf put in allow=g729
   and you should have a working configuration ..
  
   now for my question ..
   during G729 calls I am getting repeatedly the message
  
   WARNING[311314]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
 detect
   process 256 frames
  
   this scrolls up the screen at a very high rate of knots.. the call is
   unaffected and goes through normally.
   Is this something wrong? normal? can it be fixed/suppressed?
  
   cheers
   Dave
  
  
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[Asterisk-Users] Re: Strange behavior in latest CVS

2003-07-17 Thread Paul Cheng

Hi All,

A week ago, we had Asterisk working very stably with SIP, I4L (ISDN 
BRI) with a passive Eicon card, chan_h323, g729, etc. However, if 
Asterisk ran for long periods of time without a restart, there would be 
a build-up of unterminated SIP channels without the ability to do a 
soft hangup. Restarting periodically would solve the problem.

These artifact channels didn't really have an impact on Asterisk as far 
as we could tell.

After we downloaded the latest CVS over the last two days, the sip 
artifacts don't seem to be there anymore, but if * runs for a day or 
two, the g729 codec support starts exhibiting strange behavior and even 
if there is a free licensed channel, it will refuse the call. 
Restarting fixes this problem. In addition, our chan_h323 no longer 
works. Inbound voice works, but outbound no longer works.

Has anyone seen this behavior with the g729 codec or h323 with the 
latest CVS? We're using RH9 k2.4.18.

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[Asterisk-Users] Silly questions due to ingrained knowledge of analog phone use.

2003-07-17 Thread Benjamin Long

Greetings all!

I've got some really silly questions. I'm a technical guy, and I understand 
how the astrisk server works and how VOIP works, etc... The problem I have is 
that at my small company we have a phone system with analog lines and 
everyone here is comfortable with the concept of using them. I've never seen 
IP phones in action so I don't know how they work from a users point of view. 

For instance, we have 10 lines here. I'd someone calls for me, the person who 
answered the phone puts them on hold, intecoms me at my desk, if I'm not 
there he all pages me (announcement via speaker on all unused phones). If 
he gets ahold of me he tells me that so-and-so is on line X, and I can either 
pick up line X or tell him to put him into my voice mail. How would that 
sequance go if I had an astrisk PBX and IP phones everywhere? I need to wrap 
my brain around the concepts of *using* IP phones before I can go and think 
about setting up a system to use them.

I thank in advance anyone who can spare the time to help me understand this 
better. I really love what Astrisk could do for us here, and I'm hopeing that 
I can get it set up and useable without too much culture shock for the users. 
Thanks again!

Benjamin Long
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Re: [Asterisk-Users] random hangups

These lines are commented, I'm not setting either busydetect or callprogress.
I''m not sure what is their defaults.

Busydetect was set to yes before, but it gave much more random hangups.


 do you have in zapata.conf
 
 busydetect=yes
 or
 callprogress=yes ?
 
 Martin
 
 On Thu, 17 Jul 2003, Paulo H. Mannheimer wrote:
 
  Hi ,
 
  I''m getting random hangups on zap channels with long calls. It seems that
 the
  hungup happens after 10 minutes or so. AbsoluteTimeout is set to 0. Any
 other
  thing I should be configuring?
 
  Thanks!
 
  PHM
 
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[Asterisk-Users] Gnophone

2003-07-17 Thread Dave Cotton

I'm trying to compile Gnophone, but the file raw2h.c is corrupt, has anyone got a 
clean version?

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Re: [Asterisk-Users] Segmentation fault with chan_oh323

2003-07-17 Thread Michael Ulitskiy

That is another problem I hope the developers would pay attention to.
ulaw codec segfaulting when it is used by h323 side of connection for
both incoming and outgoing calls. At least with chan_oh323. 
If I set alaw codec for h323 it works fine regardless of codec on SIP 
side. 

Michael

On Thursday 17 July 2003 03:36 am, Mark Thompson wrote:
 This also happened to me when I was using the same codec with both oh323
 and SIP, if I forced it to alaw on oh323 and ulaw on SIP the connection
 worked. I also tried h323 instead of oh323 which works okay but you have
 to use earlier versions of pwlib and openh323.
 Mark
 
 -Original Message-
 From: Michael Ulitskiy [mailto:[EMAIL PROTECTED] 
 Sent: 16 July 2003 23:44
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Segmentation fault with chan_oh323
 
 
 Hi,
 
 I'm trying to interconnect sip and h323 endpoints using asterisk and
 asterisk crashes with segmentation fault whenever h323 
 connection needs to be established. It registers with gatekeeper ok
 though. Here are the symptoms. If the call initiated by SIP device,
 asterisk replies to it Trying and then silently crashes (it launched
 as asterisk -cd). In debug log I can see the following: Jul 16
 18:11:52 DEBUG[196621]: File pbx.c, Line 1123 (pbx_extension_helper):
 Launching 'Dial' Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line
 1393 (oh323_request): In oh323_request. Jul 16 18:11:52 DEBUG[196621]:
 File chan_oh323.c, Line 1394 (oh323_request): type=oh323, format=4,
 data=phone number. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c,
 Line 1440 (oh323_request): Created new call structure 0 (2428 bytes).
 That's it. If the call initiated by H323 device, then I see
 *CLI   
 WrapH323Connection::WrapH323Connection: WrapH323Connection created.
 Segmentation fault and debug log shows: Jul 16 18:33:12 DEBUG[196621]:
 File chan_oh323.c, Line 2141 (init_h323_connection): In
 init_h323_connection... Jul 16 18:33:12 DEBUG[196621]: File
 chan_oh323.c, Line 2180 (init_h323_connection): Created new call
 structure 0 (2428 bytes). Jul 16 18:33:12 DEBUG[196621]: File
 chan_oh323.c, Line 1527 (copy_call_details): --- CALL DETAILS --- Jul 16
 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1528
 (copy_call_details): call_token = ip$192.168.0.227:5018/92 Jul 16
 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1529
 (copy_call_details): call_source_alias = tnt [192.168.0.227]
 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1530
 (copy_call_details): call_dest_alias = 12125551234  12125551234
 ip$192.168.0.70:1720
 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1531
 (copy_call_details): call_source_e164 = phone number Jul 16 18:33:12
 DEBUG[196621]: File chan_oh323.c, Line 1532 (copy_call_details):
 call_dest_e164 = 12125551234 That's it. And gatekeeper log shows that
 after normal ARQ-ACF exchange originating device immediately sent DRQ.
 If anybody knows a reason for this (and the way to fix it of course ;)),
 I'd appreciate if you let me know. If you need any additional info to
 troubleshoot it, let me know too. Thank a lot.
 
 Michael
 
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Re: [Asterisk-Users] E1 R2 on Asterisk

LQ (Asterisk) wrote:

Dear fellows,

I need to use Asterisk with an E1 card with CAS R2 signalling for Argentina.
I know that the E100P don't support it right now.
Correct

Does anybody developing R2 drivers?

Yes.
Interestingly terse reply; perhaps you can be more specific?

I have an interest in the same drivers, and there was some discussion 
a week ago (two weeks?) on the topic, specifically about how a driver 
might be written, but I heard no confirmation that there was progress 
or any timeframes.

Anyone have any encouraging updates for those of us waiting for R2?

JT
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Re: [Asterisk-Users] outgoing callerid string

You can use the SetCallerID application to do this, as well. 
Whenever a call is destined for the outside world, just 
SetCallerID(13334445)

Am I missing something more complex here?

Of course, you'll need to have a PRI to do this, and 1333444 
needs to be a permitted number for your outbound calls.

JT



You can set this in zapata.conf.
Basically, you just define the callerid before you define the channel.
Outgoing channels will have the callerid of the company, and internal
channels will have the callerid of the extension.
In zapata.conf
Example:
;I don't think you have to change anything in the fxo_ks channels
because that will be handled by the phone company.
signalling=fxs_ks
callerid=Receptionist 0
channel=4
callerid=Adam West 555
channel=5
Is there a way for me to set my outgoing callerid string so that all
callers outside of my pbx see our callerid string as company name main
company number but callers inside our telephone network see extension
holders name extension number?  In looking at the references it looks
like
I can do one or the other but not both.  Does anyone know how I might
accomplish this?
Thanks for any suggestions.
AJ
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Re: [Asterisk-Users] E1 R2 on Asterisk

2003-07-17 Thread Eduardo Goncalves

On Thu, 17 Jul 2003 13:11:52 -0700
John Todd [EMAIL PROTECTED] wrote:
 Interestingly terse reply; perhaps you can be more specific?
 
 I have an interest in the same drivers, and there was some discussion 
 a week ago (two weeks?) on the topic, specifically about how a driver 
 might be written, but I heard no confirmation that there was progress 
 or any timeframes.
 
 Anyone have any encouraging updates for those of us waiting for R2?
 
 JT


I've got a libr2 from cvs. It's in alpha stage. I could't test it yet.


Eduardo
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Re: [Asterisk-Users] Can I interoperate with public PSTN gateways?

Apologies if this is an FAQ, I wasn't able to find an answer googling:

Will any of the public PSTN/VoIP gateway services (Vonage, Packet8 etc)
interoperate with * ?
I'd like to deploy a box which provides PBX service for analog handsets,
and handles inbound/outbound calls via both analog PSTN lines, and,
say Packet8 VoIP service. I understand that I can do this by connecting
the analog side of an ATA to an analog card in a machine running *,
but I'd prefer to terminate the VoIP traffic directly---I figure 
that eventually
these services will allow multiple concurrent calls, and at that time I'd
need multiple ATA type boxen, which seems rather silly.

Hops that made sense, and thanks.
Vonage doesn't work, and has explicitly said that they will not work 
with Asterisk and SIP.  You must buy their ATA-186 and use it like an 
analog phone line.  This is frighteningly short-sighted, or an 
inability/unwillingness to develop alternate product and price 
strategies for different call patterns.

Packet8, in my conversations with their sales/support line, has said 
they don't give out the SIP data to customers.  However, I seem to 
recall from prior list postings that several people are using their 
service so there must be a backdoor or method to squeeze that info 
out of them.

iconnecthere.com (DeltaThree) works fine, and will give you the 
username/password for SIP use.  They will not allow multiple calls at 
once on the same account, though.

There are a growing number of IAX and SIP service providers.  Dig 
around on the archives a bit to find some.

JT

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Re: [Asterisk-Users] Silly questions due to ingrained knowledgeof analog phone use.

This is not an explicit answer to all of your questions, but...

1) There is currently no intercom functionality supported by Asterisk 
as an in-band method of communicating with phones.  There is the 
ability to make audio on a phone call appear out of the sound-out 
port on a soundcard, which may be what you're after if you have a PA 
system of some sort.

2) You can do everything you're looking for with Asterisk.  Spend a 
bit of money on some hardphones (Cisco ATA-186 is my personal bias, 
since they have 2 lines and they're cheap) and get an X100P analog 
adapter.  Everything you've mentioned can be demo'ed with that 
configuraion.

JT


Greetings all!

	I've got some really silly questions. I'm a technical guy, 
and I understand
how the astrisk server works and how VOIP works, etc... The problem I have is
that at my small company we have a phone system with analog lines and
everyone here is comfortable with the concept of using them. I've never seen
IP phones in action so I don't know how they work from a users point of view.

For instance, we have 10 lines here. I'd someone calls for me, the person who
answered the phone puts them on hold, intecoms me at my desk, if I'm not
there he all pages me (announcement via speaker on all unused phones). If
he gets ahold of me he tells me that so-and-so is on line X, and I can either
pick up line X or tell him to put him into my voice mail. How would that
sequance go if I had an astrisk PBX and IP phones everywhere? I need to wrap
my brain around the concepts of *using* IP phones before I can go and think
about setting up a system to use them.
I thank in advance anyone who can spare the time to help me understand this
better. I really love what Astrisk could do for us here, and I'm hopeing that
I can get it set up and useable without too much culture shock for the users.
Thanks again!
Benjamin Long
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[Asterisk-Users] Example: Writing a click-to-call application using pbx_spool

2003-07-17 Thread John Laur

I have written a small perl CGI script that demonstrates how one might
use the asterisk spooler 'pbx_spool' to make a 'click-to-dial' type
application. The script is intended to be a demonstration example only
and since it has little security, should not be deployed. I was just
experimenting with the spooler and wrote this to try some things, and I
though it'd be a good example to share..

The file 'placecall-example.cgi.gz' can be downloaded here:

  http://www.blurbco.com/~gork/asterisk/ 

Hope it helps someone,
John


 Short readme embedded in the script:
This application is written to demonstrate how you can do PC-to-phone
integration with the asterisk outgoing queue mechanism, pbx_spool.

WARNING! Take note that this applicaiton provides no security by
default. Anyone using it can set up a call between any two extensions
in the list you specify, and thus it is not suitable for any kind of
real deployment! It is intended to serve only as an example.

some omitted instructions

Usage:

Load placecall.cgi in your web browser. Select the extension from which
you want the call to originate. Then, select the extension you wish to
call. Your phone will ring and the caller id will show as the value you
configured below. When you pick up, you'll hear Please hold while I try
that extension and then your call will be connected to the party you
wish to call.

You should be able to apply the same basic technique seen here to write
your own click-to-call style application.

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Re: [Asterisk-Users] Silly questions due to ingrained knowledge ofanalog phone use.

VoIP works very similar to the analog component of asterisk. So with
that, lets talk about the analog side of this. First try and forget
which line a call is on. You will only use extensions from now on. For
hold, you place a call in parking, and a extension is read to you. This
extension is what you use to retrieve the call from anywhere else.

So your user experience to mimic what you are already doing is as
follows;
Call rings all phones in office.
First person to pick up gets call.
Call is determined to be for different user.
Call is parked via flash hook and dialing the parking extension.
Then user determines best method to contact you, either through calling
your extension, or paging you.
If contact is made, then you can dial the parked extension and pick up
the call.

That works for when you are available with the exception of a direct to
voicemail option, but that can be built into an extension too.

Other option is that you switch to using the IVR to help direct the user
to the right extension number, and your phone takes it from there. With
enough phone lines available, you can make use of the *72/*73 call
forwarding option so that when you want to go sit in another office for
a while, or you need the calls to go to your cell phone, the call will
go where you tell it to.


On Thu, 2003-07-17 at 14:16, Benjamin Long wrote:
 Greetings all!
 
   I've got some really silly questions. I'm a technical guy, and I understand 
 how the astrisk server works and how VOIP works, etc... The problem I have is 
 that at my small company we have a phone system with analog lines and 
 everyone here is comfortable with the concept of using them. I've never seen 
 IP phones in action so I don't know how they work from a users point of view. 
 
 For instance, we have 10 lines here. I'd someone calls for me, the person who 
 answered the phone puts them on hold, intecoms me at my desk, if I'm not 
 there he all pages me (announcement via speaker on all unused phones). If 
 he gets ahold of me he tells me that so-and-so is on line X, and I can either 
 pick up line X or tell him to put him into my voice mail. How would that 
 sequance go if I had an astrisk PBX and IP phones everywhere? I need to wrap 
 my brain around the concepts of *using* IP phones before I can go and think 
 about setting up a system to use them.
 
 I thank in advance anyone who can spare the time to help me understand this 
 better. I really love what Astrisk could do for us here, and I'm hopeing that 
 I can get it set up and useable without too much culture shock for the users. 
 Thanks again!
 
 Benjamin Long
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Re: [Asterisk-Users] Cisco 7960

2003-07-17 Thread William Carlson

Cisco's website has some stuff on there website
which seems to indicate if the 7960 cannot contact the call manager server it
reboots. However to my knowledge this has never had call manager software before
and cisco doesn't mention this "feature" with the SIP firmware. I downgraded to
5.0 unfortunately due to only being able to run Secure images now thats as far
back as I can go. Thanks again cisco for this "feature".

From what I can tell the phone never talked to the
Asterisk box. If I turn on SIP debugging I do not see any traffic coming from
the cisco box. Although I did have them on seperate subnets. Let me try putting
them on the same subnet and see if that helps.
Thanks,
Will

- Original Message -
From:
Matthew
Hardeman
To: [EMAIL PROTECTED]

Sent: Thursday, July 17, 2003 12:58
PM
Subject: RE: [Asterisk-Users] Cisco
7960

Ive run into this
before, and its a pain to debug

Be sure that your
eth0 interface (primary, first interface) is set to your internal address
space (of the same subnet that you assign to the phone). You can add an IP alias on eth0:1 if
you need an external IP on that box as well, but you must have them in that
order: internal = eth0, externals, others
eth0:1+

Try
that

Matt
PaperSoft

-Original
Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William CarlsonSent: Thursday, July 17, 2003 6:35
AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Cisco
7960

I bought a 7960 it was running
version 3.3 of the SIP software. It worked fine. Me being the idiot I am
upgraded to 5.1. Now it downloads the configs and then reboots. if I unplug
the ethernet it doesn't rebootor if I remove all the lines in the SIP
config it won't reboot. Since this is used cisco won't give me any support.
For now I am running the MGCP version but eh asterisk seems to have some
issues with it.

Thanks,

Will

[Asterisk-Users] AGI Silence detection

2003-07-17 Thread Stuart Hirst

Does anyone how you might detect a period of x milliseconds of silence
using AGI ?

Rgds,

Stuart 


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RE: [Asterisk-Users] Cisco 7960

2003-07-17 Thread Matthew Hardeman










Heres a hint


Upgrade back to 5.1. It still has
bugs, but theyre no worse than 5.0



Put it in the same subnet as the asterisk
server, and make sure that the eth0 interface (the FIRST interface in the box)
is on that subnet. Generally, you
will want this to be your internal address space. 192.168.0.x If you asterisk server needs an external
IP on the same Ethernet, just do an alias as eth0:1 or something along those
lines. Make the primary hostname of
the server (as reflected by the hostname command) match up in /etc/hosts as the
internal IP address.



Asterisk apparently picks up the first IP
address on the system to use as its source IP address for all things SIP If you have a Cisco phone communicate
with Asterisk on another subnet known to the system (via an IP alias on the
same Ethernet card), Ive found that the Cisco 7960 will crash and burn. I suspect that if Asterisk were modified
to source the communications back over the interface it received them, the
crash would no longer happen.



Check your configuration files Ive
had these phones crash on me before if your networking isnt very
friendly to them, but never before just during the booting sequence Trust me; you can get this phone to work
Its just a matter of patience and experimenting, and lots of free time
wasted on Ethereal J



As an aside, Ive actually been
actively working with a Cisco developer (even today) to generate more debug
information for them on the network caused crash and reboot issue, and they
think theyve about got it licked I believe they will be sending me a
firmware image to test soon that will have at least that bug, and probably
more, fixed.



Matt Hardeman

PaperSoft



-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William Carlson
Sent: Thursday, July 17, 2003 5:10 PM
To:
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
Cisco 7960





Cisco's website has some stuff on
there website which seems to indicate if the 7960 cannot contact the call
manager server it reboots. However to my knowledge this has never had call
manager software before and cisco doesn't mention this feature with
the SIP firmware. I downgraded to 5.0 unfortunately due to only being able to
run Secure images now thats as far back as I can go. Thanks again cisco for
this feature.











From what I can tell the phone never
talked to the Asterisk box. If I turn on SIP debugging I do not see any traffic
coming from the cisco box. Although I did have them on seperate subnets. Let me
try putting them on the same subnet and see if that helps.





 Thanks,





 Will







- Original Message - 





From: Matthew Hardeman






To: [EMAIL PROTECTED] 





Sent: Thursday, July 17, 2003 12:58 PM





Subject: RE:
[Asterisk-Users] Cisco 7960









Ive run into this
before, and its a pain to debug



Be sure that your eth0
interface (primary, first interface) is set to your internal address space (of
the same subnet that you assign to the phone). You can add an IP alias on eth0:1 if you
need an external IP on that box as well, but you must have them in that order:
internal = eth0, externals, others eth0:1+



Try that



Matt

PaperSoft





-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William Carlson
Sent: Thursday, July 17, 2003 6:35 AM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco
7960





I bought a 7960 it was running
version 3.3 of the SIP software. It worked fine. Me being the idiot I am
upgraded to 5.1. Now it downloads the configs and then reboots. if I unplug the
ethernet it doesn't rebootor if I remove all the lines in the SIP config
it won't reboot. Since this is used cisco won't give me any support. For now I
am running the MGCP version but eh asterisk seems to have some issues with it.





 Thanks,





 Will

Re: [Asterisk-Users] AGI Silence detection

In AGI you can't as there is no access to the audio stream. In EAGI you
get access to the audio stream on FD 4, You can then use that for
detection on your own. But the best option might be to write a function
into agi that uses the silence detection that I believe is available via
dsp.c. I'm not sure what type of event you would want to through back at
the script when the threshold was exceeded. 
 
On Thu, 2003-07-17 at 17:14, Stuart Hirst wrote:
 Does anyone how you might detect a period of x milliseconds of silence
 using AGI ?
 
 Rgds,
 
 Stuart 
 
 
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[Asterisk-Users] TE410P startup (2 boards)

2003-07-17 Thread Alex Zarubin

Title: TE410P startup (2 boards)





Hello,


This message
 TE410P: Double/missed interrupt detected
is looping on the system console.


Do we need to keep Board ID = 0 on board 1 and set it to 1 on board 2?


Please, help.


Jul 17 17:12:09 mspgate03 kernel: Zapata Telephony Interface Registered on major 196
Jul 17 17:12:09 mspgate03 kernel: Found TE410P at base address fcf0, remapped to f89a6000
Jul 17 17:12:09 mspgate03 kernel: TE410P version c01a003a
Jul 17 17:12:09 mspgate03 kernel: FALC version: 0005, Board ID: 00
Jul 17 17:12:09 mspgate03 kernel: Reg 0: 0x36e81800
Jul 17 17:12:09 mspgate03 kernel: Reg 1: 0x36e81000
Jul 17 17:12:09 mspgate03 kernel: Reg 2: 0x07fc07fc
Jul 17 17:12:09 mspgate03 kernel: Reg 3: 0x
Jul 17 17:12:09 mspgate03 kernel: Reg 4: 0x
Jul 17 17:12:09 mspgate03 kernel: Reg 5: 0x
Jul 17 17:12:09 mspgate03 kernel: Reg 6: 0xc01a003a
Jul 17 17:12:09 mspgate03 kernel: Reg 7: 0x1000
Jul 17 17:12:09 mspgate03 kernel: Reg 8: 0x
Jul 17 17:12:09 mspgate03 kernel: Reg 9: 0x00ff
Jul 17 17:12:09 mspgate03 kernel: Reg 10: 0x
Jul 17 17:12:09 mspgate03 kernel: TE410P: Launching card: 0
Jul 17 17:12:09 mspgate03 kernel: TE410P: Setting up global serial parameters
Jul 17 17:12:09 mspgate03 kernel: TE410P: Timing from source 0
Jul 17 17:12:09 mspgate03 kernel: Found a Wildcard: Wildcard TE410P-Xilinx
Jul 17 17:12:09 mspgate03 kernel: Found TE410P at base address fc90, remapped to f89a8000
Jul 17 17:12:09 mspgate03 kernel: TE410P version c01a003a
Jul 17 17:12:09 mspgate03 kernel: FALC version: 0005, Board ID: 00
Jul 17 17:12:09 mspgate03 kernel: Reg 0: 0x36e92800
Jul 17 17:12:09 mspgate03 kernel: Reg 1: 0x36e92000
Jul 17 17:12:09 mspgate03 kernel: Reg 2: 0x07fc07fc
Jul 17 17:12:09 mspgate03 kernel: Reg 3: 0x
Jul 17 17:12:09 mspgate03 kernel: Reg 4: 0x
Jul 17 17:12:09 mspgate03 kernel: Reg 5: 0x
Jul 17 17:12:09 mspgate03 kernel: Reg 6: 0xc01a003a
Jul 17 17:12:09 mspgate03 kernel: Reg 7: 0x1000
Jul 17 17:12:09 mspgate03 kernel: Reg 8: 0x
Jul 17 17:12:09 mspgate03 kernel: Reg 9: 0x00ff
Jul 17 17:12:09 mspgate03 kernel: Reg 10: 0x
Jul 17 17:12:09 mspgate03 kernel: TE410P: Launching card: 0
Jul 17 17:12:09 mspgate03 kernel: TE410P: Setting up global serial parameters
Jul 17 17:12:09 mspgate03 kernel: TE410P: Timing from source 0
Jul 17 17:12:09 mspgate03 kernel: Found a Wildcard: Wildcard TE410P-Xilinx
Jul 17 17:12:09 mspgate03 kernel: Registered tone zone 0 (United States / North America)
Jul 17 17:12:09 mspgate03 kernel: TE410P: Span 1 configured for ESF/B8ZS
Jul 17 17:12:09 mspgate03 kernel: SPAN 1: Primary Sync Source
Jul 17 17:12:09 mspgate03 kernel: TE410P: Span 2 configured for ESF/B8ZS
Jul 17 17:12:09 mspgate03 kernel: TE410P: Span 3 configured for ESF/B8ZS
Jul 17 17:12:09 mspgate03 kernel: TE410P: Span 4 configured for ESF/B8ZS
Jul 17 17:12:09 mspgate03 kernel: TE410P: Span 1 configured for ESF/B8ZS
Jul 17 17:12:09 mspgate03 kernel: TE410P: Span 2 configured for ESF/B8ZS
Jul 17 17:12:09 mspgate03 kernel: TE410P: Span 3 configured for ESF/B8ZS
Jul 17 17:12:09 mspgate03 kernel: TE410P: Span 4 configured for ESF/B8ZS
Jul 17 17:12:09 mspgate03 kernel: Pre-interrupt
Jul 17 17:12:09 mspgate03 kernel: Got interrupt, status = 0001
Jul 17 17:12:09 mspgate03 kernel: Pre-interrupt
Jul 17 17:12:09 mspgate03 kernel: Got interrupt, status = 0001
Jul 17 17:12:09 mspgate03 kernel: TE410P: Double/missed interrupt detected
Jul 17 17:12:09 mspgate03 kernel: Pre-interrupt
Jul 17 17:12:09 mspgate03 kernel: Got interrupt, status = 0002
Jul 17 17:12:09 mspgate03 kernel: Pre-interrupt
Jul 17 17:12:09 mspgate03 kernel: Got interrupt, status = 0002
Jul 17 17:12:09 mspgate03 kernel: TE410P: Double/missed interrupt detected
Jul 17 17:12:09 mspgate03 kernel: Pre-interrupt
Jul 17 17:12:09 mspgate03 kernel: Got interrupt, status = 0001
Jul 17 17:12:09 mspgate03 kernel: Pre-interrupt
Jul 17 17:12:09 mspgate03 kernel: Got interrupt, status = 0001
Jul 17 17:12:09 mspgate03 kernel: TE410P: Double/missed interrupt detected
Jul 17 17:12:09 mspgate03 kernel: Pre-interrupt
Jul 17 17:12:09 mspgate03 kernel: Got interrupt, status = 0002
Jul 17 17:12:09 mspgate03 kernel: Pre-interrupt
Jul 17 17:12:09 mspgate03 kernel: Got interrupt, status = 0002
Jul 17 17:12:09 mspgate03 kernel: TE410P: Double/missed interrupt detected


Thank you.


Alex Zarubin
Webley Systems, Inc.

RE: [Asterisk-Users] AGI Silence detection

2003-07-17 Thread Stuart Hirst

What I am looking to do is to dial a number where I expect to get
someone's voicemail, wait for their greeting to finish and then play an
MP3 and hangup. So in answer to your question I am looking to detect a
silence for x milliseconds within a given time variable and if true
then I would assume that their greeting had finished and play my message
or if false, I might assume that something failed and redial the call.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: 17 July 2003 23:28
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] AGI  Silence detection


In AGI you can't as there is no access to the audio stream. In EAGI you
get access to the audio stream on FD 4, You can then use that for
detection on your own. But the best option might be to write a function
into agi that uses the silence detection that I believe is available via
dsp.c. I'm not sure what type of event you would want to through back at
the script when the threshold was exceeded. 
 
On Thu, 2003-07-17 at 17:14, Stuart Hirst wrote:
 Does anyone how you might detect a period of x milliseconds of silence

 using AGI ?
 
 Rgds,
 
 Stuart
 
 
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[Asterisk-Users] queue bug?

I may be tired after a long day, but this seems to be a bug.

I try twice to get someone to dial a valid extension, after failing I put it in
a queue. When the call get answered the operator cannot hear the other party. 

The script for someone who dials a valid extension is exactly the same, it gets
into the same queue but the operator can hear the other party.

Console shows calls being bridged in either situation.

any hint?
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Re: [Asterisk-Users] queue bug?

Found what is happening by zapbarging into a channel:

Some users hang up after trying the 1st time, but the script goes on and
transfers a dead call after the 2nd timeout ;-)

Nothing like a large cup of Brazilian Coffee to wake me up and clear my mind! ;-)

 I may be tired after a long day, but this seems to be a bug.
 
 I try twice to get someone to dial a valid extension, after failing I put it
 in
 a queue. When the call get answered the operator cannot hear the other party.
 
 
 The script for someone who dials a valid extension is exactly the same, it
 gets
 into the same queue but the operator can hear the other party.
 
 Console shows calls being bridged in either situation.
 
 any hint?
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Re: [Asterisk-Users] H323/No one is available to answer at this time

2003-07-17 Thread Jeremy McNamara

That is no where near enough information to help you..

Provide more information.



Jeremy McNamara

[EMAIL PROTECTED] wrote:

Hi folks,

When dialing H323, I'm getting:
No one is available to answer at this time.
Anybody knows why this happen?

Thanks in advance,

Isamar

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Re: [Asterisk-Users] H323/No one is available to answer at this time

2003-07-17 Thread isamar


Ok. I did a h.323 trace 5
Looks like the other side is kicking me off during the negotiation...
but I didn't figure out yet why.

Here is some info:

   }
  }
  0:30.215  H225 Caller:8105028 h323.cxx(1620)  H225
Handling
 PDU: CallProceeding callRef=25692
  0:30.215  H225 Caller:8105028  h323neg.cxx(334)   H245
Stopping
 MasterSlaveDetermination: state=Outgoing
  0:30.215  H225 Caller:8105028  h323neg.cxx(561)   H245
Stopping
 TerminalCapabilitySet: state=InProgress
  0:30.215  H225 Caller:8105028 h323.cxx(1880)  H225
Set prot
ocol version to 3 and implying H.245 version 5
  0:30.215  H225 Caller:8105028 h323.cxx(2071)  H225
Set remo
te party name: 200.221.40.50
  0:30.215  H225 Caller:8105028 h323.cxx(2079)  H225
Set remo
te application name: Internet Telephony Gateway -- Version 3.00
R8.0 Gat
eway (Build 4)  181/18247
  0:30.215  H225 Caller:8105028 h323.cxx(3876)  H323
Internal
EstablishedConnectionCheck: connectionState=AwaitingSignalConnect
fastStartState
=FastStartDisabled
  0:30.546  H225 Caller:8105028  h323pdu.cxx(474)   H225
Receivin
g PDU:
  {
 0:30.546  H225 Caller:8105028  h323pdu.cxx(474)   H225
Receivin
g PDU:
  {
q931pdu = {
  protocolDiscriminator = 8
  callReference = 25692
  from = destination
  messageType = ReleaseComplete
  IE: Cause - Normal call clearing = {
80 90  ..
  }
  IE: User-User = {
25 c0 06 00 08 91 4a 00  03 58 58 00 11 00 d2 25
%.J..XX%
de 20 6e b7 d7 11 87 e0  dc e4 32 ff e0 e7 08 80   .
n...2.
01 00  ..
  }
}
h225pdu = {
  h323_uu_pdu = {
h323_message_body = releaseComplete {
  protocolIdentifier = 0.0.8.2250.0.3
  reason = undefinedReason null
 callIdentifier = {
guid =  16 octets {
  d2 25 de 20 6e b7 d7 11  87 e0 dc e4 32 ff e0 e7   .%.
n...2..
.
}
  }
}
h245Tunneling = FALSE
  }
}
  }


  }
  0:30.549  H225 Caller:8105028 h323.cxx(1620)  H225
Handling
 PDU: ReleaseComplete callRef=25692
  0:30.549  H225 Caller:8105028 h323.cxx(1880)  H225
Set prot
ocol version to 3 and implying H.245 version 5
  0:30.549  H225 Caller:8105028   h323ep.cxx(1537)  H323
Clearing
 connection ip$localhost/25692 reason=EndedByRemoteUser
  0:30.549  H225 Caller:8105028 h323.cxx(1403)  H323
Call end
 reason for ip$localhost/25692 set to EndedByRemoteUser
  0:30.549  H225 Caller:8105028 h323.cxx(1421)  H225
Sending
release complete PDU: callRef=25692
  0:30.551  H225 Caller:8105028  h323pdu.cxx(474)   H245
Sending
PDU:
  command endSessionCommand disconnect null
  0:30.551  H225 Caller:8105028 h323.cxx(3085)  H245
Write PD
U fail: no control channel.
  0:30.552  H225 Caller:8105028  h323pdu.cxx(474)   H225



 0:30.215  H225 Caller:8105028 h323.cxx(2079)  H225Set
remo
te application name: Internet Telephony Gateway -- Version 3.00
R8.0 Gat
eway (Build 4)  181/18247
  0:30.215  H225 Caller:8105028 h323.cxx(3876)  H323Internal
EstablishedConnectionCheck: connectionState=AwaitingSignalConnect
fastStartState=FastStartDisabled
  0:30.546  H225 Caller:8105028  h323pdu.cxx(474)   H225
Receiving PDU:
q931pdu = {
  protocolDiscriminator = 8
  callReference = 25692
  from = destination
  messageType = ReleaseComplete
  IE: Cause - Normal call clearing = {
80 90  ..
  }
  IE: User-User = {
25 c0 06 00 08 91 4a 00  03 58 58 00 11 00 d2 25
%.J..XX%
de 20 6e b7 d7 11 87 e0  dc e4 32 ff e0 e7 08 80   .
n...2.
01 00  ..
  }
}
h225pdu = {
  h323_uu_pdu = {
h323_message_body = releaseComplete {
  protocolIdentifier = 0.0.8.2250.0.3
  reason = undefinedReason null
  callIdentifier = {
guid =  16 octets {
  d2 25 de 20 6e b7 d7 11  87 e0 dc e4 32 ff e0 e7   .%.
n...2..
.
}
h245Tunneling = FALSE
  }
}
  }








There other side is a Planet(http://www.planet.com.tw) VOIP-400FXO.
The ping time is aroung 400ms.
Its configuration is:
 h323 display_name   = hermann
 h323 h245_term_type = 60
 h323 rtp_port_base  = 3
 h323 out_fast_start = off
 h323 in_fast_start  = on
 h323 h245_tunneling   = off
 h323 cisco_t38   = off
 h323 callSignalPort  = 1720
 h323 nat_call   = off
 h323 call_name   =
 h323 local_alert = off
 h323 default_dtmf = H323 V2 Signal
 No Alternate IP Defined!
 h323 dns_ip = 200.221.11.98 Domain:
 h323 gk_mode= off
 h323

Re: [Asterisk-Users] E1 R2 on Asterisk

2003-07-17 Thread Steve Underwood

John Todd wrote:

LQ (Asterisk) wrote:

Dear fellows,

I need to use Asterisk with an E1 card with CAS R2 signalling for 
Argentina.
I know that the E100P don't support it right now.

Correct

Does anybody developing R2 drivers?

Yes.


Interestingly terse reply; perhaps you can be more specific?

I have an interest in the same drivers, and there was some discussion 
a week ago (two weeks?) on the topic, specifically about how a driver 
might be written, but I heard no confirmation that there was progress 
or any timeframes.

Anyone have any encouraging updates for those of us waiting for R2?
I've been more specific in the past. This was just a brief recap, since 
its the same question each time (OK, the country varies, but not much else).

The work was held up by SARS, as the testing has been done in China. Now 
the SARS threat has abated, I hope we will see a polished China R2 soon. 
Every other country requires modifications, as no two countries 
implement R2 in quite the same way. However, the software has been 
written with all the variants in mind, and completing support for other 
countries should be pretty straightforward, once it is a proven platform 
in China.

You will find some elements of R2ness in CVS. That is work I did over a 
year ago, then left unfinished. The DSP part of that is OK, although I 
have improved it recently. The rest was a lash-up, which has now been 
totally replaced by a solid implementation. I have a new channel driver 
for Asterisk, which supports R2 and PRI with the Zaptel drivers. My 
intention is to add other protocols, so this becomes a replacement for 
chan_zap. The way chan_zap works had some flexibility issues for me, so 
I have tried to steer everything through a generalised signalling API. 
On one side are plug in protocols - currently PRI and R2. On the other 
side is Asterisk. Hopefully, this will allow new protocols to plug in 
with little or no change to anything in Asterisk. This is somewhat like 
Dialogic's GlobalCall, but hopefully without so much of their 
clunkiness. :-)

FAQ: Can I help with testing, and be an early adopter?

No. I have all the testers I need right now. Until I have it well 
proven, with E1's full of calls pumping through it, I am not interesting 
in having other testers involved. I expect the assistance of other 
testers will be needed later, to deal with issues arising from the local 
variants of R2. Help at that time will be most appreciated.

Regards,
Steve
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Re: [Asterisk-Users] Can I interoperate with public PSTN gateways ?

2003-07-17 Thread David Boreham


 Packet8, in my conversations with their sales/support line, has said 
 they don't give out the SIP data to customers.  However, I seem to 
 recall from prior list postings that several people are using their 
 service so there must be a backdoor or method to squeeze that info 
 out of them.

Ah, that's good. I just signed up with them.

 iconnecthere.com (DeltaThree) works fine, and will give you the 
 username/password for SIP use.  They will not allow multiple calls at 
 once on the same account, though.
 
 There are a growing number of IAX and SIP service providers.  Dig 
 around on the archives a bit to find some.

Unfortunately I need a rate center in an out of the way place
(Montana). That limits my choices.

Thanks for the info.



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Re: [Asterisk-Users] Cisco 7960g

2003-07-17 Thread Yifang Dai

On Thu, Jul 17, 2003 at 06:50:41PM +0200, Siggi Langauf wrote:
 
 There's not much to it: just configure * as an H.323 gateway in
 CallManager for the appropriate extensions.

Thanks! I'll need to read more about how to accomplish this in CCM :) 

 If you need to route calls from * to CCM, just use something like
 
 Dial(OH323/callto:[EMAIL PROTECTED])
 
 in /etc/Asterisk/extensions.conf.
 ${EXTEN} is the CallManager extension you're going to dial, and
 callmanager.your-domain.com is the CCM's host name (IP address is safer.)
 
 The * part is a bit trickier: I had to use current CVS versions of both
 asterisk and openh323/pwlib. Moreover, the H323 channel driver that comes
 with asterisk will _not_ work with CCM. (It requires an older version of
 openh323, and it will send voice Data to the call manager instead of the
 telephone, which makes it 'one way'.)
 
 The current 0.5.3 release of Michael's OH323 channel driver
 (http://www.inaccessnetworks.com/projects/asterisk-oh323/) works fine.
 

I've downloaded openh323/pwlib/asterisk-oh323 from the above site, and
cvs co the asterisk modules, everything is looking good so far... I've
been putting off the asterisk project, since the only way to get to PTSN 
previously is the analog lines :)

-- 
Yifang Dai   |
eFax: (847)628-0255  |Debian GNU/Linux
[EMAIL PROTECTED] |



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[Asterisk-Users] ATA-186 software upgrade 2.16.1 - notes?

I see that there's now a 2.16.1 upgrade path for Cisco ATA-186
devices, dated (variously) July 11 or July 14 2003.

Here are some interesting bugs that claim to be fixed. Most notable
is CSCeb17953, at least from my perspective, as I've hit this bug
before.

CSCea42480 The Cisco ATA ignores the Require:100rel header and processes call.
CSCea69889 The Cisco ATA builds a 302 Moved Temporarily message
incorrectly after receiving a NOTIFY message.
CSCea93969 The Cisco ATA loses G.723 audio when call waiting occurs.
CSCeb01064 The Cisco ATA From header domain value changes SRV record name.
CSCeb17953 The Cisco ATA stops the registration process if it
receives an unexpected response to a REGISTER request.
CSCeb19228 The callback-on-busy feature does not work for calls to a PSTN.
CSCeb23060 Upon receiving a 4xx response to a REGISTER request from a
backup proxy, the Cisco ATA needs to continue retrying the request
with the primary proxy .
CSCeb24556 The Cisco ATA may fail to send a ring tone when acting as
a transfer target in a blind transfer.
CSCeb28218 The Cisco ATA, while in a call, detects audio from an incoming call.
CSCeb32210 When the SDP attribute a=fmtp appears before the attribute
a=rtpmap , the Cisco ATA will not send out-of-band DTMF digits.
CSCeb35955 Attended call transfers occur even when this feature is
disabled via the PaidFeature configuration parameter.
CSCeb36752 Call forwarding does not work when the Cisco ATA detects a
busy signal.
CSCeb37037 The Cisco ATA stops registering after a 2.16 upgrade is performed.
CSCeb37043 The call-waiting default user setting cannot be controlled
by the CallFeatures configuration parameter when the Cisco ATA
obtains its configuration file from the TFTP server.
CSCeb40099 The Cisco ATA plays an incorrect tone after unconditional
call forwarding is enabled or disabled.
CSCeb44406 Change the behavior of the Cisco ATA to not remove all
registrations.

Full information can be found here:

http://www.cisco.com/en/US/products/hw/gatecont/ps514/prod_release_note09186a00801a2519.html

JT
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[Asterisk-Users] serious dtmf recognition problem.

2003-07-17 Thread Joe Antkowiak









Hi,



I am using a channel bank and zaptel hardware. I have
a credit card machine on one of the channels that appears to be dialing too
soon for asterisk, every complete number recognized by asterisk is
missing the first 1-4 numbers. This is a serious problem for me, anyone
have any ideas on whats going on? The pstn picks up on the dtmf tones
just fine



I was able to get it to work 50% of the time by adding:

exten = _8XXNXX,1,Dial(Zap/g2/1${EXTEN}) 



but thats really ugly.



Zapata.conf:



;CC Machine 

context=cc-out 

signalling=fxo_ks

usecallerid=yes

hidecallerid=no

callwaiting=no

callwaitingcallerid=no

threewaycalling=no

transfer=no

cancallforward=no

callreturn=no 

echocancel=no 

echocancelwhenbridged=no

relaxdtmf=no

rxgain=6.0 

txgain=0.0

group=10

callgroup=10 

pickupgroup=10

immediate=no

amaflags=documentation

accountcode=cc-outbound

adsi=no

busydetect=no 

callprogress=no 

callerid=CC Machine

channel = 22

[Asterisk-Users] Speex support

2003-07-17 Thread Jan Rychter

What is the state of speex support in asterisk? I saw the codec seems to
be there.

Can speex be used on IAX2 links? Is there much work still to be done?

many thanks,
--J.
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[Asterisk-Users] Call Parking

2003-07-17 Thread Aaron Martin




How do I use parking? I thought all I had to 
do was hook flash, but this immediately cuts the other end of the call 
off..

Re: [Asterisk-Users] Call Parking

If you are on a zap device, make sure you add threewaycalling.

On Thu, 2003-07-17 at 22:25, Aaron Martin wrote:
 How do I use parking?  I thought all I had to do was hook flash, but
 this immediately cuts the other end of the call off..
  
-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] serious dtmf recognition problem.

2003-07-17 Thread Steve Underwood

Hi Joe,

Most auto-dialers will accept commas in the dial string, and insert 
delays where they occur. Will that work for you? Its normally used to 
insert a delay after a 9 on a PBX, to get a stable outside line before 
further dialing.

Regards,
Steve
Joe Antkowiak wrote:

Hi,

I am using a channel bank and zaptel hardware. I have a credit card 
machine on one of the channels that appears to be dialing too soon 
for asterisk, every complete number recognized by asterisk is missing 
the first 1-4 numbers. This is a serious problem for me, anyone have 
any ideas on whats going on? The pstn picks up on the dtmf tones just 
fine

I was able to get it to work 50% of the time by adding:

exten = _8XXNXX,1,Dial(Zap/g2/1${EXTEN})

but thats really ugly.



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Re: [Asterisk-Users] TE410P startup (2 boards)

2003-07-17 Thread Mark Spencer

   TE410P: Double/missed interrupt detected
 is looping on the system console.

 Do we need to keep Board ID = 0 on board 1 and set it to 1 on board 2?

No, that detection code was built around the assumption of a single card.
If you moved last0 into the t4 struct you could make it work.

Mark

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Re: [Asterisk-Users] Speex support