[asterisk-users] (no subject)

2011-10-31 Thread Karim Mardhani 
Karim Mardhani karim at vertexcommunication.ca
http://lists.digium.com/mailman/listinfo/asterisk-users wrote:
* Hi everyone,** ** I am trying to get Meetme to return back to the context 
from where it** joined the meetme.  For example a user uses the following 
context to join a** conference, once user hangs up I would like to continue 
executing the rest** of the dialplan.  But when caller hangs up from the 
conference I see on CLI** that meetme exited with non-zero status but none of 
the rest of the** dialplan is executed.  Please help.  I am using asterisk 
1.6.2.20** ** [default]** exten = _,1,MeetMe(1000,1pdMX)** exten = 
_,n,noop(returned from meetme) ;After user hangs up should** come here** 
exten = _,n,SoftHangup(${ORIG_CALLER})** exten = 
_,n,SoftHangup(${CONF_CALLER})** exten = _,n,Hangup** exten = 
h,1,noop(default-end)** exten = h,n,SoftHangup(${ORIG_CALLER})** exten = 
h,n,SoftHangup(${CONF_CALLER})** exten = h,n,Hangup*
That's not how Asterisk works. When the caller hangs up, execution of
the current dialplan extension stops, and control passes to the 'h'
extension, if one exists in the current context.

Any processing you want to do when the caller hangs up must be done
in the 'h' extension. Cheers

Thanks Tony for the quick response.  As you would see I have the h
extension defined but execution doesn't go to that either.

 Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk
http://lists.digium.com/mailman/listinfo/asterisk-users -
http://www.softins.co.uk
Play: tony at mountifield.org
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-- 
Karim Mardhani

Vertex Communication Ltd.
18667552554 ext. 103
www.vertexcommunication.ca
sip: ka...@sip2.vertexcommunication.ca
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[asterisk-users] Meetme does not return back to the dialplan

2011-10-30 Thread Karim Mardhani 
Hi everyone,

I am trying to get Meetme to return back to the context from where it
joined the meetme.  For example a user uses the following context to join a
conference, once user hangs up I would like to continue executing the rest
of the dialplan.  But when caller hangs up from the conference I see on CLI
that meetme exited with non-zero status but none of the rest of the
dialplan is executed.  Please help.  I am using asterisk 1.6.2.20

[default]
exten = _,1,MeetMe(1000,1pdMX)
exten = _,n,noop(returned from meetme) ;After user hangs up should
come here
exten = _,n,SoftHangup(${ORIG_CALLER})
exten = _,n,SoftHangup(${CONF_CALLER})
exten = _,n,Hangup
exten = h,1,noop(default-end)
exten = h,n,SoftHangup(${ORIG_CALLER})
exten = h,n,SoftHangup(${CONF_CALLER})
exten = h,n,Hangup


-- 
Karim Mardhani
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[asterisk-users] Unable to get Fax t38 working with IrisTel trunk

2011-01-11 Thread Karim Mardhani 
Hi everyone,
I have been trying to get T.38 Faxing to work with Iristel sip trunks for
last few days but havn't been sccussful.  I am using Asterisk 1.6.2.8 and
SpanDSP 0.6.  Here is what I see in the tcpdump capture:
1.  Call come in from the trunk as regular voice call with g.711 codec
2.  Asterisk answers the call and recognizes the CNG and sends the call to
fax extension
3.  Eventually Receive fax is called with a file name
4.  Asterisk sends update message to remote gateway with T38 codec
information
5. Remote server doesn't respond.  Asterisk resends update messages multiple
times.
6. Eventually remote gateway sends the invite with T38 codecs listed in the
SDP
7. Asterisk Responds back with 488 Not acceptable here
8. Another invite is send from the remote gateway with T38 codecs in the SDP
9. Asterisk sends OK but g.711 codecs listed in SDP.
10. remote gateway sends the BYE message and call is completed.

The questions I have are:
Why Asterisk sends update message in step 4 above instead of send an invite?
Why Asterisk responds back with 488 in step 7 above?
Why asterisk sends g.711 codecs in step 9 above?

Thanks

Karim
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[Asterisk-Users] Can't compile asterisk-oh323 on Mandrake 10

2005-08-07 Thread Karim Mardhani 




  
  Hi All:
  
  I am trying to compile asterisk with oh323 but I can't 
  compile it. I am using instruction provided at http://www.oinko.net/astrecipes/index.php?from=1q=astrecipes/compiling+asterisk+with+oh323. 
  The compile error I am getting is as follows. Quite a few other people 
  are getting exactly same error but no one has posted a fix for this error 
  yet. Any help would greatly be appreciated.
  
  gcc -Wall -pipe -Wall -Wstrict-prototypes 
  -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE 
  -I/usr/include/asteri\sk -I../wrapper -g -c -o chan_oh323.o 
  chan_oh323.cIn file included from 
  /usr/include/string.h:33, 
  from 
  chan_oh323.c:34:/usr/lib/gcc-lib/i586-mandrake-linux-gnu/3.3.2/include/stddef.h:213: 
  error: syntax error before "typedef"In file included from 
  chan_oh323.c:34:/usr/include/string.h:38: error: syntax error before 
  "extern"/usr/include/string.h:39: error: parse error before 
  "__THROW"/usr/include/string.h:43: error: parse error before 
  "__THROW"/usr/include/string.h:56: error: parse error before 
  "__BEGIN_NAMESPACE_STD"/usr/include/string.h:58: error: syntax error 
  before "extern"
  
  My gcc version is 3.3.2
  
  Thanks
  
  Karim

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[Asterisk-Users] Do sipura 200 and linksys pap2 ATAs send their mac address in REGISTER message?

2005-05-15 Thread Karim Mardhani 




  HI All:
  
  I was just wondering, Do sipura 200 and 
  linksys pap2 ATAs send their mac address in REGISTER message? Is their 
  any other way to get the MAC address of sip peer who is trying to 
  register?
  
  Thanks
  
  Karim

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[Asterisk-Users] Data calls with Asterisk

2005-01-23 Thread Karim Mardhani 




  Hi All: I am new to Asterisk so if my question sounds too 
  newbeeish then pleasebear with me. I have about 10 remote 
  locations which are collecting some data. Iwould like to upload that 
  data every night. All remote locations have56K modem. I was 
  wondering can Asterisk be used to receive this data? Basically I will have 
  an asterisk with 1 FXO card and have it receivedata calls. Can 
  asterisk receive data calls? Thanks in advance for your 
  responsesRegards,Karim MardhaniZeeCore 
Consulting
  

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[Asterisk-Users] Music on hold problem

2004-09-07 Thread Karim Mardhani 
  Hi All:

  I am having problems with getting music on hold to work.  I get following
error

  Executing SetMusicOnHold(SIP/2000-2ddf, default) in new stack
  -- Executing Answer(SIP/2000-2ddf, ) in new stack
  -- Executing WaitMusicOnHold(SIP/2000-2ddf, 30) in new stack
  Sep  7 15:50:47 WARNING[1190886320]: res_musiconhold.c:329 moh1_exec:
Unable to start music on hold (class '30') on channel SIP/2000-2ddf
== Spawn extension (from-sip, 4999, 3) exited non-zero on
'SIP/2000-2ddf'
  asterisk*CLI

  I have made sure that I have mpg123 in /usr/bin.  The version of mpg123 I
have is 0.59r.   Following is my extension.conf:

  exten = 4999,1,SetMusicOnHold(default)
  exten = 4999,2,Answer
  exten = 4999,3,WaitMusicOnHold(30)
  exten = 4999,4,Hangup

  When I change WaitMusicOnHold with
MP3Player(/var/lib/asterisk/mohmp3/sample-hold.mp3) I can hear the
sample-hold file when I dail extension 4999.

  Do both MP3Player and MOH use mpg123?

  Any help in this regard would greatly be appreciated.

  Thanks

  Karim




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[Asterisk-Users] Asterisk behind openBSD firewall/NAT

2004-07-30 Thread Karim Mardhani 

Hi All:

  Has anybody been able to get Asterisk work behind a openBSD
firewall/NAT?  If you have then would it be possible to share your
pf.config file?

  I am trying to get Asterisk which is behind an openBSD firewall/NAT to
register with FWD but can't get it to talk.  I have captured IP traffic
on udp port 5060 using tcpdump on both internal and external interfaces
of my openBSD gateway (the logs are at the end of this e-mail).

  From the tcpdump logs I can see that a message is sent to FWD out from
the external interface and response is received from FWD on udp port
5060 but the response is not forwarded to Asterisk.  Here are my NAT and
FILTER rules:  (tl0 is the external interface, xl0 is the internal
interface)

nat on tl0 inet from 192.168.0.0/24 to any - (tl0) round-robin
rdr on xl0 inet proto tcp from any to any port = ftp - 127.0.0.1 port 8021
rdr pass on tl0 inet proto tcp from any to (tl0) port = sip - 192.168.0.3
rdr pass on tl0 inet proto udp from any to 209.89.66.243 port = sip -
192.168.0.3 port 5060

Tcpdump output on the external interface of gateway (filtered for udp port
5060):

tcpdump: listening on tl0
05:51:58.622714 243.209-89-66-0.interbaun.com.57199  192.246.69.223.sip: 
udp 376 (DF) (ttl 63, id 23576)
05:51:58.716031 192.246.69.223.sip  243.209-89-66-0.interbaun.com.sip: 
udp 462 (DF) (ttl 47, id 0)
05:51:59.622771 243.209-89-66-0.interbaun.com.57199  192.246.69.223.sip: 
udp 376 (DF) (ttl 63, id 23577)
05:51:59.716004 192.246.69.223.sip  243.209-89-66-0.interbaun.com.sip: 
udp 462 (DF) (ttl 47, id 0)
05:52:00.623539 243.209-89-66-0.interbaun.com.57199  192.246.69.223.sip: 
udp 376 (DF) (ttl 63, id 23578)
05:52:00.719989 192.246.69.223.sip  243.209-89-66-0.interbaun.com.sip: 
udp 462 (DF) (ttl 47, id 0)
05:52:01.624328 243.209-89-66-0.interbaun.com.57199  192.246.69.223.sip: 
udp 376 (DF) (ttl 63, id 23579)
05:52:01.716980 192.246.69.223.sip  243.209-89-66-0.interbaun.com.sip: 
udp 462 (DF) (ttl 47, id 0)
05:52:02.624107 243.209-89-66-0.interbaun.com.57199  192.246.69.223.sip: 
udp 376 (DF) (ttl 63, id 23580)
05:52:02.715968 192.246.69.223.sip  243.209-89-66-0.interbaun.com.sip: 
udp 462 (DF) (ttl 47, id 0)
05:52:03.623884 243.209-89-66-0.interbaun.com.57199  192.246.69.223.sip: 
udp 376 (DF) (ttl 63, id 23581)
05:52:03.715954 192.246.69.223.sip  243.209-89-66-0.interbaun.com.sip: 
udp 462 (DF) (ttl 47, id 0)
05:52:18.645348 243.209-89-66-0.interbaun.com.57199  192.246.69.223.sip: 
udp 376 (DF) (ttl 63, id 23582)
05:52:18.737143 192.246.69.223.sip  243.209-89-66-0.interbaun.com.sip: 
udp 462 (DF) (ttl 47, id 0)

Tcpdump output at the internal interface

tcpdump: listening on xl0
06:05:00.451172 192.168.0.3.sip  fwd.pulver.com.sip:  udp 376 (DF) (ttl
64, id 23811)
06:05:01.450934 192.168.0.3.sip  fwd.pulver.com.sip:  udp 376 (DF) (ttl
64, id 23812)
06:05:02.450711 192.168.0.3.sip  fwd.pulver.com.sip:  udp 376 (DF) (ttl
64, id 23813)
06:05:03.451502 192.168.0.3.sip  fwd.pulver.com.sip:  udp 376 (DF) (ttl
64, id 23814)
06:05:04.451286 192.168.0.3.sip  fwd.pulver.com.sip:  udp 376 (DF) (ttl
64, id 23815)



Regards,

Karim Mardhani
ZeeCore Consulting


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