RE: [Asterisk-Users] password on radius authentication

2006-06-30 Thread Glenn Dalgliesh 

Well, I know to be compatible with porta-billing you need password to do ip
based auth. It's a bit goody but they basically seem to expect 

if trusted ip and no Digest support then radius auth has username=src_ip and
password=x. 

To put it another way it would be help full to porta-billing users to be
able set username and password fields on auth being sent via radius to
porta-billing. 

So in a round about way I would say yes

I can probably test the module against some things for you.

Thanks


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dennis Nacino
Sent: Wednesday, June 28, 2006 6:05 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] password on radius authentication

Hi,


It's kind of off-topic , but still within Asterisk. I developed an asterisk
module that send an
authentication to a radius server for call authorization and process its
reply (limited to
User-Name and Cisco or Quintum VSA h323 attribute). My question, is when it
make sense to use or
include the attribute Password/User-Password? Looking on PDF's of Quintum
and Cisco none of it
really make use of this attribute. Any comment?



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[Asterisk-Users] realtime and queues and persistantmembers in 1.2.5

2006-03-23 Thread Glenn Dalgliesh 








It appears that when realtime is enabled in queues.conf
persistantmembers no longer has effect on dynamically added members. I am
wondering if this is a intended or a bug. 








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[Asterisk-Users] Queue strategy

2005-02-15 Thread Glenn Dalgliesh 
Just woundering if the intentend functionality of leastrecent and
fewestcalls it to continually dial only the first chosen ext. in the queue.
In other words if a memeber is logged into the queue but doesn't answer the
call the call never moves on in my configuration from that ext.  This could
be really bad

Thanks

[support]
announce-frequency=45
strategy=leastrecent
music=default
monitor-join=1
retry=5

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[Asterisk-Users] pridialplan=unknown ?

2005-01-02 Thread Glenn Dalgliesh 
After setting the pridialplan=unknown I seeing the Called Number TON change
to Unknown Number Type but not the Calling Number TON. Should both be
following this parameter or not. If not is their another option to change
the Calling Number TON?


 Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0:
0   Location: User (0)
   Ext: 1  Progress Description: Calling
equipment is non-ISDN. (3) ]
 [28 10 b1 44 61 6c 67 6c 69 65 73 68 20 47 6c 65 6e 6e]
 Display (len=16) Charset: 31 [ Dalgliesh Glenn ]
 [6c 0c 21 83 34 31 30 37 33 35 38 35 35 30]
 Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
   Presentation: Presentation allowed of network
provided number (3) '4102228550' ]
 [70 0c 80 31 34 31 30 33 35 33 32 32 36 34]
 Called Number (len=14) [ Ext: 1  TON: Unknown Number Type (0)  NPI:
Unknown Number Plan (0) '1410264' ]
-- Called g1/1410264
 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 3171/0xC63) (Terminator)

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Re: [Asterisk-Users] Call Queue Question

2005-01-02 Thread Glenn Dalgliesh 
When I first started looking at a similar problem started out on the same
path with app_queue but even having access to a friend of mine who actually
help write some of the Queue code we decided it wasn't the right tool for
the job. We approached this a little differently using valetparking and agi.
This may not be exactly what you are looking for since my thoughts were
around trying only 1 number with a required response to bridge but most
likely it could be adapted.

app_valetparking: Alternative method for call parking (by bkw)

*-*-*-*-*-*-* /etc/asterisk/extensions.conf *-*-*-*-*-*-*-*-*-*
[forwardnum]
Exten = s,1,Read(ext|extension|4)
Exten = s,2,Authenticate(/etc/asterisk/passwd${ext}.conf)
Exten = s,3,Read(IAFORWARDNUM|telephone-number|10)
Exten = s,4,SayDigits(${IAFORWARDNUM})
Exten = s,5,Background(if-correct-press)
Exten = s,6,Background(digits/1)
Exten = s,7,Background(to-enter-a-diff-number)
Exten = s,8,Background(press-2)
Exten = 1,1,Playback(auth-thankyou)
Exten = 1,2,DBput(iaforward/${ext}num=${IAFORWARDNUM})
Exten = 1,3,Hangup
Exten = 2,1,Goto(forwardnum|s|3)

[macro-iaforward]
Exten = s,1,Wait(1)
Exten = s,2,Playback(pls-hold-while-try)
Exten = s,3,AGI(outconnect.agi|${ARG1})
Exten = s,4,ValetParkCall(${PARKID}|mylot|35|iaforward|5|default)
Exten = s,5,Voicemail(u${ARG1}); If
unavailable, send to voicemail w/ unavail announce
Exten = s,6,Hangup

[acceptout]
Exten = s,1,Wait(1)
Exten = s,2,Answer
Exten = s,3,DigitTimeout,5
Exten = s,4,ResponseTimeout,10
Exten = s,5,BackGround(doyouaccept)
Exten = 1,1,ValetUnParkCall(${PARKID}|mylot)
Exten = 2,1,Hangup
Exten = i,1,Hangup
Exten = t,1,Hangup

[default]
Exten = 4107358515,1,SetVar(exten=5001)
Exten = 4107358515,2,Macro(iaforward|${exten})


*-*-*-*-*-*-* /var/lib/asterisk/agi-bin/outconnect.agi *-*-*-*-*-*-*-*-*-*
#!/usr/bin/perl

use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();

my $exten = $ARGV[0];
my $calldir = /var/spool/asterisk/outgoing;
my $calldir1 = /tmp;
$extnum = $exten.num;
$iaforwardnum = $AGI-database_get('iaforward',$extnum);
$parkid = $AGI-database_get('VALETPARK','SEQ');
$parkid = ($parkid  9998) ? 1 : $parkid+1;
$AGI-database_put('VALETPARK','SEQ', $parkid);

$AGI-set_variable(PARKID, $parkid);
$callerid = $input{'callerid'};

open (CALL, $calldir/call.$$);

   print CALL qq{
Channel: SIP/[EMAIL PROTECTED]
MaxRetries: 0
Callerid: 999$callerid

Context: acceptout
Extension: s
Priority: 1
SetVar: PARKID=$parkid
};
close();

exit(0);

- Original Message - 
From: Michael Wareman [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, January 02, 2005 9:30 PM
Subject: [Asterisk-Users] Call Queue Question


 Hi,

 I've been fussing with this for a while now - and cannot seem to get
 it to work correctly - or rather as I desire it..

 I'm trying to implement a 'find-me' feature in my voicemail.
 Basically - pressing 1 at the voicemail puts the caller into a queue.
 The members of the queue are my cell phone and my work DID..  I need
 the queue to call both numbers at the same time and wait for a #
 confirmation before bridging the call. This will be repeated with 1..
 2 etc for different family members

 If I use AddQueueMember to add the members to the queue - or define
 the members directly in queues.conf (like member =
 IAX2/[EMAIL PROTECTED]/12125551212) - I do not get the ability to use #
 to confirm before the call is bridged. Every other feature works
 perfectly though. It seems the only way I can get the # is to use
 AgentCallbackLogin.

 Problem is - I don't want anyone to have to do anything to log in. The
 agents should always be on...  Can anyone offer any ideas on keeping
 an agent logged in thru AgentCallbackLogin without the user having to
 do anything - that is survivable thru reloads etc.. or any other
 workarounds..

 Many thanks,

 Michael.
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[Asterisk-Users] ValetParking

2004-11-04 Thread Glenn Dalgliesh 
Does anyone that the source for app_valetparking.c 

Thanks
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[Asterisk-Users] app_valetparking

2004-10-27 Thread Glenn Dalgliesh 
Does anyone have a copy of lastest source I seem to have delete my copy and
http://www.bkw.org/app_valetparking.c seem to not exits at the moment.

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[Asterisk-Users] ValetParking

2004-10-13 Thread Glenn Dalgliesh 
First Thanks to brian for work on valetpark it seems to work really well

I was working on some apps using ValetParking and having good success but
was wondering when you think valetparking will make it into the
CVS/releases? So, I can build around it with a little more confidence.
Thanks

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Re: [Asterisk-Users] Telco POTS - FXO ?

2004-10-13 Thread Glenn Dalgliesh 
Yes,
- Original Message - 
From: Neil Cherry [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, October 13, 2004 10:54 AM
Subject: [Asterisk-Users] Telco POTS - FXO ?


 Maybe I'm just doing this wrong. Is the FXO card (X100P) used to
 connect to the telco pots line?

 -- 
 Linux Home Automation Neil Cherry   [EMAIL PROTECTED]
 http://home.comcast.net/~ncherry/   (Text only)
 http://hcs.sourceforge.net/ (HCS II)
 http://linuxha.blogspot.com/My HA Blog
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[Asterisk-Users] Authenticate cmd with db

2004-10-13 Thread Glenn Dalgliesh 
I want to authenticate against the asterisk internal database but don't seem
to be able to figure out the syntax for the Authenticate cmd. I am assuming
I have something wrong in line s,4

-- Executing Authenticate(SIP/5006-a54e, /iaforward/5001pass|d) in new
stack
-- Playing 'agent-pass' (language 'en')
-- Playing 'auth-incorrect' (language 'en')

Thanks

database show
/iaforward/5001pass   : 
/iaforward/5002pass   : 

Exten = s,1,Answer
Exten = s,2,Read(EXTENA|extension|4)
Exten = s,3,Background(vm-password)
Exten = s,4,Authenticate(/iaforward/${EXTENA}pass|d)
Exten = s,5,Read(${iaforwardnum}|telephone-number|10)

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[Asterisk-Users] cvsup options file for v1-0

2004-10-12 Thread Glenn Dalgliesh 
I want to dowload cvs of v1-0 with cvsup and was wondering what the options
file will look like to make this happen.

I am assuming the some thing on the line *default release=cvs tag=.

- options file for cvsup to download cvs head 
*default host=cvs.digium.com
*default base=/usr/src
*default release=cvs tag=.
*default delete use-rel-suffix
asterisk
libpri
zaptel

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Re: [Asterisk-Users] Beyond T1

2004-09-16 Thread Glenn Dalgliesh 
Well, you might be better off at that scale to use a cisco as5850 or equiv
with SER and Asterisk. I might not work so well with 672 calls going thru 1
asterisk box.

ds3 - Cisco as5850 - Asterisk (Possible multiple depending on actual
config and use)


- Original Message - 
From: Marcelo Pacheco [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, September 16, 2004 12:52 PM
Subject: Re: [Asterisk-Users] Beyond T1


 I'm no E1 expert, but as I understand one channel is wasted with framing,
so
 it is as 2048000 bps link, where one 64000 bps channel is wasted with
 signalling. So there's 31 channels left. If you use EM, FXS or FXO, you
 could get 31 voice channels, with PRI or MFC/R2D you get 30 voice
channels.

 I now that from the fact that a full E1 with EuroISDN gives you 30 voice
 channels.

 An a full E1 with Brazilian R2D also gives you 30 voice channels, as one
 channel is used for signalling as CAS (Channel Associated Channeling),
where
 each 4 bits is used for each channel.

 The only situation where you get closer to actual 2mbps out of an E1
channel
 is when you run SyncPPP, Frame Relay or another bit synchronous protocol
on
 the full trunk/link, where you throw away the channelling and use the
whole
 link as one big synchronous bit pipe.

 Marcelo Pacheco

 Em Qui 16 Set 2004 13:26, Andrew Kohlsmith escreveu:
  On Thursday 16 September 2004 12:17, Andrew Thompson wrote:
   Depending on where you using the circuits, you might try an E1. It
uses
   the same total bandwidth as a T1(I think), but splits the channels at
   56K instead of 64K, yielding more channels. (And now I can't remember
   the number.)
 
  uh, no.  This is definitely NOT correct.
 
  T1 is 24 8-bit channels + 1 framing bit sent 8000 times a second.
24*8+1 =
  193 bits per frame * 8000 = 1554000bps.
 
  E1 is 32 8-bit channels + 1 framing bit sent 8000 times a second.
32*8+1 =
  2056000bps.
 
  (my E1 knowlege is poor, I hope I am not furthering the misinformation
  here)
 
  In both cases you get 64kbit clean channels unless you're doing
robbed-bit
  (inband) signalling.
 
  -A.
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Re: [Asterisk-Users] Grandstream Budgetone BT-101 and VoipJet

2004-08-24 Thread Glenn Dalgliesh 
I would make the following changes to your sip.conf and restart asterisk.  I
have seen alot of issues with regard to codec with * and grandstream. both
have a little to do with it but this should keep it working.

[general]
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm


- Original Message - 
From: William Suffill [EMAIL PROTECTED]
To: John Week [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion [EMAIL PROTECTED]
Sent: Tuesday, August 24, 2004 7:24 PM
Subject: Re: [Asterisk-Users] Grandstream Budgetone BT-101 and VoipJet


 I had this issue with a grandstream as well a week or so ago and have
 yet to solve the issue. Until I get my Budgetone here physically again
 I won't be able to mess with it hands on. What did you use for
 codec/signaling and did your asterisk box see any warnings or errors?

 On Tue, 24 Aug 2004 15:56:38 -0700, John Week [EMAIL PROTECTED] wrote:
  Is anyone using this combination successfully?  I have a dell 500sc
  running rh9 and asterisk 1.0rc1.  It is configured with an x100p.  I
  have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone
  BT-101.  I have signed up with Voipjet (they use iax2).  I also have
  an FWD number via iax2.  I can sucessfully call back and forth to all
  devices via zap, sip, and fwd.  I can successfully place calls using
  voipjet with everything except the grandstream.  When I place a
  voipjet call with the grandstream, I can hear the party I'm calling,
  but they can't hear me.  I have tried all the different codecs the
  grandstream supports without luck.  I am running the 1.0.5.10
  firmware.  I've emailed voipjet support about it, but they don't have
  one.
 
  thanks,
 
  John
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Re: [Asterisk-Users] How to share Zap channels in 2 Asterisk servers

2004-05-22 Thread Glenn Dalgliesh 
Please reply with sip.conf  extension.conf for both servers. Hard to tell
what the problem is without see config info
- Original Message - 
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 22, 2004 11:39 AM
Subject: [Asterisk-Users] How to share Zap channels in 2 Asterisk servers



 Hello

 I am trying to setup Asterisk on 2 servers PBX300 and PBX200.
 PBX300 has X100P card with 1 telephone line. PBX200 don't have any Zap
device.
 Softphone from PBX200 can talk to softphone on PBX300 but no outgoing call
from
 PBX200.
 I can call from PBX300 outside but I am unable to configure soft Phone
defined
 in PBX200 to dial out side using PBX300 Zap devices.

 I am geting error message  Rejected connect attempt from PBX200.

 Please help if this is possible.

 Thanks

 Deepak



 
 This message was sent using IMP, the Internet Messaging Program.

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[Asterisk-Users] rtpmap issue w/Grandstreams

2004-05-20 Thread Glenn Dalgliesh 
Sent this to grandstream support but has anyone else seen this issue. All of
the previous sdp rtpmap are correct until the grandstream sends this. I have
been using disallow=gsm and canreinvite=no to get around the problem.

- Original Message - 
From: Glenn Dalgliesh
To: [EMAIL PROTECTED]
Sent: Thursday, May 20, 2004 4:34 PM
Subject: rtpmap issue


I am seeing ATA-286, running Program--1.0.4.55, send and INVITE message with
rtpmap:3 PCMU/8000. 3 is a well-known port and should be mapped to GSM not
PCMU. 162.33.165.203 = HandyTone ATA-286162.33.165.198 = Asterisk SIP
MESSAGE 16   162.33.165.203:5060(3) - 162.33.165.198:5060(2) UDP
Frame 16   20/May/04 15:26:16.0653 TimeFromPreviousSipFrame=0.0043
TimeFromStart=2.4311 SIP/2.0 200 OK Via: SIP/2.0/UDP
162.33.165.198:5060;branch=z9hG4bK2f81a326 From:
sip:[EMAIL PROTECTED];tag=as4fb7ebf5 To:
sip:[EMAIL PROTECTED];tag=9ce2cb909d0a97ff Call-ID:
[EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream
HT286 1.0.4.55 Contact: sip:[EMAIL PROTECTED] Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type:
application/sdp Content-Length: 147  v=0 o=5003 8000 8000 IN IP4
162.33.165.203 s=SIP Call c=IN IP4 162.33.165.203 t=0 0 m=audio 5004 RTP/AVP
3 a=rtpmap:3 PCMU/8000 a=ptime:20

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[Asterisk-Users] SDP messages relating to rtpmap Question

2004-05-11 Thread Glenn Dalgliesh 
SDP question if * recieves a=rtpmap:103 telephone-event/8000 it shouldn't
it send out the same  a=rtpmap:103 telephone-event/8000  to the other side
of the connection?  and not  something like a=rtpmap:101
telephone-event/8000?

Thanks

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[Asterisk-Users] SIP re-INVITES problem

2004-04-20 Thread Glenn Dalgliesh 
When a call is place to xxx9931211 from the pstn the call proceeds normally
until asterisk issues the Second INVITE, which is MESSAGE 14, and instead of
call being sent with INVITE sip:[EMAIL PROTECTED] SIP/2.0. It gets
sent with INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 and this seems
to cause SNOM proxy to return the packet without a Record-Route entry and
then asterisk starts sending the packets to the UA directly. Not sure if
this is a bug or not but it seems odd to me that the INVITE and re-INVITE
messages have different fields in them.

Also, if I test the same scenario with canreinvite=no since * doesn't issue
a re-INIVTE the call completes properly and all messages go thru the SNOM
proxy to reach the UA.

Any insight would be appreciated.

Thanks

Glenn

pstn- asterisk - snom  UA
(xxx.99.77.23) (xxx.93.91.74)(yyy.33.165.201)

 SIP MESSAGE 3xxx.99.77.23:5060(2) - xxx.93.91.74:5060(4)
 UDP Frame 319/Apr/04 18:17:47.9517
TimeFromPreviousSipFrame=0.1666 TimeFromStart=0.1676
INVITE sip:[EMAIL PROTECTED] SIP/2.0

- Re-Invite
 SIP MESSAGE 14   xxx.99.77.23:5060(2) - xxx.93.91.74:5060(4)
 UDP Frame 14   19/Apr/04 18:17:50.4408
TimeFromPreviousSipFrame=0.0003 TimeFromStart=2.6566
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0


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[Asterisk-Users] Re: SIP re-invite

2004-04-20 Thread Glenn Dalgliesh 
Trouble getting chan_sip2 to compile

below is what I have done

-download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software
cp /root/software/chan_sip2s.c /usr/src/asterisk/channels
cd /usr/src/asterisk/
patch -p0 acl.c /root/software/acl.c.patch
cd /usr/src/asterisk/include/asterisk
patch -p0 acl.h /root/software/acl.h.patch
- added the follow to /usr/src/asterisk/channels/Makefile
chan_sip2.so: chan_sip2.o
cd /usr/src/asterisk
make
make install

I assume that problem is with what did or didn't add to the Makefile

Thank for any help
- Original Message - 
From: Olle E. Johansson [EMAIL PROTECTED]
To: Glenn Dalgliesh [EMAIL PROTECTED]
Sent: Tuesday, April 20, 2004 1:29 PM
Subject: SIP re-invite


 Could you please test this with my chan_sip2. I have a hunch it will work
with
 that channel.

 /Olle


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Re: [Asterisk-Users] Re: SIP re-invite

2004-04-20 Thread Glenn Dalgliesh 
okay add chan_sip2.so under CHANNEL_LIBS= and it compiles

Ran a test call with the same conditions and see the same results as with
sip_chan

FYI: I believe the bug report indication that these messages don't indicate
a problem is that so
  == Parsing '/etc/asterisk/sip.conf':   == Parsing
'/etc/asterisk/sip.conf': Found
Apr 20 16:16:48 WARNING[1184099120]: chan_sip2.c:7683 build_user: Ignoring
type in user definition of snom
Apr 20 16:16:48 WARNING[1184099120]: chan_sip2.c:7683 build_user: Ignoring
host in user definition of snom
Apr 20 16:16:48 WARNING[1184099120]: chan_sip2.c:8293 build_peer: Ignoring
unknown option type
Apr 20 16:16:48 WARNING[1184099120]: chan_sip2.c:7683 build_user: Ignoring
type in user definition of 555
Apr 20 16:16:48 WARNING[1184099120]: chan_sip2.c:7683 build_user: Ignoring
host in user definition of 555
Apr 20 16:16:48 WARNING[1184099120]: chan_sip2.c:7683 build_user: Ignoring
username in user definition of 555
Apr 20 16:16:48 WARNING[1184099120]: chan_sip2.c:8293 build_peer: Ignoring
unknown option type

- Original Message - 
From: Glenn Dalgliesh [EMAIL PROTECTED]
To: Olle E. Johansson [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Tuesday, April 20, 2004 3:30 PM
Subject: [Asterisk-Users] Re: SIP re-invite


 Trouble getting chan_sip2 to compile

 below is what I have done

 -download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software
 cp /root/software/chan_sip2s.c /usr/src/asterisk/channels
 cd /usr/src/asterisk/
 patch -p0 acl.c /root/software/acl.c.patch
 cd /usr/src/asterisk/include/asterisk
 patch -p0 acl.h /root/software/acl.h.patch
 - added the follow to /usr/src/asterisk/channels/Makefile
 chan_sip2.so: chan_sip2.o
 cd /usr/src/asterisk
 make
 make install

 I assume that problem is with what did or didn't add to the Makefile

 Thank for any help
 - Original Message - 
 From: Olle E. Johansson [EMAIL PROTECTED]
 To: Glenn Dalgliesh [EMAIL PROTECTED]
 Sent: Tuesday, April 20, 2004 1:29 PM
 Subject: SIP re-invite


  Could you please test this with my chan_sip2. I have a hunch it will
work
 with
  that channel.
 
  /Olle
 

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Re: [Asterisk-Users] Auto connect to voicemail

2004-04-05 Thread Glenn Dalgliesh 



I think this is what you are looking 
for

Exten = 1000,1,Answer,1Exten = 
1000,2,Wait,1Exten = 1000,3,Voicemailmain([EMAIL PROTECTED])

  - Original Message - 
  From: 
  Mitchell S. Sharp 
  To: [EMAIL PROTECTED] 
  
  Sent: Monday, April 05, 2004 5:27 
PM
  Subject: Re: [Asterisk-Users] Auto 
  connect to voicemail
  On Mon, 2004-04-05 at 15:57, Brian Rathman wrote: 
  I have the voicemail setup working in that I get the MWI and it emails the
message correctly. When I pressed the MWI button on my SNOM 200, it dials
into the voicemail system and prompts me for a mailbox and password. I know
there is a way to automatically connect directly into the mailbox via the
extension.conf file, but I can not find the documentation I am looking for
in reference to variables and macros for the extensions file. Can someone
please help me with this issue?

Thanks,
BrianBrian,At the CLI, type 'show 
  application VoiceMailMain'. You can use the CLI 'show applications' 
  command to list all available apps. If you hit tab, it acts just like 
  BASH's auto complete. Wonderful feature!Mitch 
  SharpInnovative Solutions

[Asterisk-Users] canreinvite and transcoding

2004-03-27 Thread Glenn Dalgliesh 
Does anyone know if it is possible to force a extension to not allow
transcoding? If you spec canreinvite=yes the cal may still transcoded if the
parties do not choose a the same code on each end. In my situation it is
better that the call fail than have it transcoded.

Also, I see some limited reference to canreinvite=update. Does this command
exist and if so what does it do.

Thanks

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[Asterisk-Users] Watchguard Firebox 1000 and Asterisk

2004-03-26 Thread Glenn Dalgliesh 
Has any had any experiences with Watchguard Firebox 1000 and Asterisk. I
have asterisk on public side and phones on the private side. I am able to
get the phones to register and make outbound calls but the inbound calls are
intermittent. I have NAT enable in asterisk and on the Cisco 7960.

Any insight would be appreciated.

Thanks

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[Asterisk-Users] Redhat lastest kernel problems with Zaptel drivers kernel 2.4.20-30.9

2004-03-18 Thread Glenn Dalgliesh 
I ran the up2date and installed newest kernel 2.4.20-30.9 and rebooted the
modules for the zaptel drivers wcfxo and wcfxs didn't load I reboot with
kernel 2.4.20-28.9 and all is working fine. I didn't have time to work in
the issue but have seen it on two systems.

Anyone have any idea what the issue is?

Thanks

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Re: [Asterisk-Users] Redhat lastest kernel problems with Zaptel drivers kernel 2.4.20-30.9

2004-03-18 Thread Glenn Dalgliesh 
kernel that work is 2.4.20-20.9 not kernel 2.4.20-28.9(haven't tested)
- Original Message - 
From: Glenn Dalgliesh [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 18, 2004 9:45 PM
Subject: [Asterisk-Users] Redhat lastest kernel problems with Zaptel drivers
kernel 2.4.20-30.9


 I ran the up2date and installed newest kernel 2.4.20-30.9 and rebooted the
 modules for the zaptel drivers wcfxo and wcfxs didn't load I reboot with
 kernel 2.4.20-28.9 and all is working fine. I didn't have time to work in
 the issue but have seen it on two systems.

 Anyone have any idea what the issue is?

 Thanks

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Re: [Asterisk-Users] help me: warnings on Read error on sound device, Ignoring rxwink

2004-03-18 Thread Glenn Dalgliesh 
I updgraded to kernel 2.4.20-28.9 to kernel 2.4.20-30.9 and my digium card
drivers refused to load I then rebooted with the previous kernel and all
work fine. Not sure if it is related or not  thought it might help
- Original Message - 
From: Michael Zheng [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 18, 2004 4:19 PM
Subject: [Asterisk-Users] help me: warnings on Read error on sound device,
Ignoring rxwink


 Hi all!

 I am frustrated. I am new to asterisk. My system is
 REDHAT Linux V9 (linux-2.4.20-30.9) and I just
 installed a sound-card (AudioExcel AV512,CMedia
 8738-6ch MX) and X100P card and compiled Asterisk.

 When I started (asterisk -c), I got problems
 related to the sound device and rewink:

 WARNING[73738]: chan_oss.c:238 sound_thread: Read
 error on sound device: Resource temporarily
 unavailable

 WARNING[8192]: chan_zap.c:7794 setup_zap: Ignoring
 rxwink

 I can't figure out why these happy. Could you anyone
 so kind help me to solve these problems?

 Thank you all.

 Michael

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Re: [Asterisk-Users] Redhat lastest kernel problems with Zaptel drivers kernel 2.4.20-30.9

2004-03-18 Thread Glenn Dalgliesh 
Okay, I need to sleep. I just need to recompile the drivers!!!

Sorry for the false alarm
- Original Message - 
From: Glenn Dalgliesh [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 18, 2004 10:05 PM
Subject: Re: [Asterisk-Users] Redhat lastest kernel problems with Zaptel
drivers kernel 2.4.20-30.9


 kernel that work is 2.4.20-20.9 not kernel 2.4.20-28.9(haven't tested)
 - Original Message - 
 From: Glenn Dalgliesh [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Thursday, March 18, 2004 9:45 PM
 Subject: [Asterisk-Users] Redhat lastest kernel problems with Zaptel
drivers
 kernel 2.4.20-30.9


  I ran the up2date and installed newest kernel 2.4.20-30.9 and rebooted
the
  modules for the zaptel drivers wcfxo and wcfxs didn't load I reboot with
  kernel 2.4.20-28.9 and all is working fine. I didn't have time to work
in
  the issue but have seen it on two systems.
 
  Anyone have any idea what the issue is?
 
  Thanks
 
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[Asterisk-Users] Directory App (Possible bug or undocumented feature)

2004-03-17 Thread Glenn Dalgliesh 
Can anyone verify this?

I have 2 voicemail context and when using the Directory app I seeing odd
results.
If I spesify the context as (default) I can only access default context
users as expected and it uses default extension.conf context to dial
If I specify the context as (group1) I can access that voicemail.conf users
in that context but if a duplicate mailbox number exists in default context
of voicemail.conf it will use that greeting and not group1 use greeting.

So, below if you dial 411 then select user1(501) you will hear greeting from
usera(501)

Also, in the show application Directory and other docs is seems that to be
very vague about context meaning for this app. according to my results it
appear for the most part to be both extensions and voicemail context. This
might need to be clarified a little more.

Thanks

Asterisk CVS-03/17/04-11:25:09 built by [EMAIL PROTECTED] on a i686 running Linux
(stable tree downloaded today)

extensions.conf
[group1]
include = default

[default]
Exten = 598,1,Directory(default)
Exten = 411,1,Directory(group1)
Exten = _5XX,1,Macro(stdexten,${EXTEN})

voicemail.conf
[default]
501 = 501,usera,[EMAIL PROTECTED],attach=yes

[group1]
500 = 500,Reception,,attach=no
501 = 501,user1,[EMAIL PROTECTED],attach=yes
502 = 502,user2,[EMAIL PROTECTED],attach=no

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[Asterisk-Users] CVS login

2004-03-01 Thread Glenn Dalgliesh 



I seem to be having trouble with cvs login. anyone 
having similar problems

It just hangs after entering the 
password

Re: [Asterisk-Users] Asterisk stable how to compile ?

2004-03-01 Thread Glenn Dalgliesh 
I have been doing the following and it seems to work fine# cd /usr/src#
export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot# cvs login   -
the password is anoncvs.# cvs checkout zaptel libpri# cvs checkout -r
v1-0_stable asterisk This will create just the asterisk directory.
Compiling them is generally quite straightforward. Just change to each
directory and type make install, in this order. Compile zaptel, then libpri,
and then asterisk.
# cd zaptel# make clean ; make install# cd ../libpri# make clean ; make
install# cd ../asterisk# make clean ; make install- Original
Message - 
From: SamW [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March 01, 2004 11:50 AM
Subject: [Asterisk-Users] Asterisk stable how to compile ?


 I want to build a stable asterisk to run, if some one can guide through
 how to compile will be useful. Currently available documentation do not
 show any good information about a correct how to. According to the
 Asterisk Web site, it indicate to download the Stable 1.0 use the
 following, cvs checkout -r v1-0_stable asterisk.  But Asterisk won't
 build on its own, it needs libpri and zaptel.

 There are 2 places to download libpri and zaptel,

 1. CVS
 cvs checkout libpri zaptel
 2. Use Download site, following 2 locations,
 ftp://ftp.asterisk.org/pub/telephony/libpri/libpri-0.5.2.tar.gz
 ftp://ftp.asterisk.org/pub/telephony/zaptel/zaptel-0.8.1.tar.gz


 Which one of the 2 above should be used for a stable Asterisk build. (I
 do not use digium hardware)

 I am currently seeing lot of segmentation faults (core-dump) when I
 running asterisk. Help is appreciated.

 - SamW

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Re: [Asterisk-Users] 4 port FXO

2004-02-19 Thread Glenn Dalgliesh 
I could be wrong but I think I remember seeing mention of recommendation
about the number per server although I don't remember the number.

- Original Message - 
From: Christian Hecimovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, February 19, 2004 12:26 PM
Subject: Re: [Asterisk-Users] 4 port FXO


 I used an external gateway - a Mediatrix 1204. It's nice because the voice
 streams are offloaded to it, reducing any load on the server. But it has a
 nightmare setup and interface, and it's kind of expensive.

 On Thursday 19 February 2004 00:27, Chad Brown wrote:
  What is my best bet If I want to get 4 port FXO on asterisk. Should I
  use 4 Digiums X100Ps or would I be better off in this situation going
  with a Dialogic card?
 
 
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[Asterisk-Users] ADSI ports

2004-02-18 Thread Glenn Dalgliesh 



Are their any options for ADSI capitableports 
indenisties of 20 to 50 ports that will work with asterisk and ADSI 
phones.

Re: [Asterisk-Users] ADSI ports

2004-02-18 Thread Glenn Dalgliesh 
Are there any special feature that the channel bank has to support? or as
long and it is connected a Wildcard T1 port it should fine with any channel
bank?

Thanks
- Original Message - 
From: Jon Pounder [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, February 18, 2004 5:23 PM
Subject: Re: [Asterisk-Users] ADSI ports


  Are their any options for ADSI capitable ports in denisties of 20 to 50
  ports that will work with asterisk and ADSI phones.



 2 t1 ports and a 48port channel bank
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Re: [Asterisk-Users] TDM card loses Dial tone

2004-02-11 Thread Glenn Dalgliesh 



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

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

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

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


  - Original Message - 
  From: 
  Bob Bevins 
  To: [EMAIL PROTECTED] 
  
  Sent: Wednesday, February 11, 2004 12:24 
  PM
  Subject: [Asterisk-Users] TDM card loses 
  Dial tone
  
  
  Hi,
  
  I have a redhat 9 asterisk server with 
  tdm400p three ports, and a x100p installed at home. I am not running X or 
  framebuffers. Every so often like 
  once a month, I lose dial tone on my channels. Asterisk still responds 
  at the cli. I don’t see any log entries pertaining 
  to this. If I restart asterisk it does not change. I have to reboot the 
  computer, which I would think would be a hardware problem, or an OS issue. I 
  can’t seem to make it happen when I want so troubleshooting is an issue. The 
  irq’s are ok as seen below. I am not doing smp, or multithreading as some posts would reveal that as 
  a problem.
  
  These are brand new cards from 
  Digium. The tdm is a new 
  card with the power connected. I tested the power supply and it is supplied 
  the correct voltages.
  
   
  CPU0
   0: 9298672 
  XT-PIC 
  timer
   1: 
  4 
  XT-PIC 
  keyboard
   2: 
  0 
  XT-PIC 
  cascade
   8: 
  1 
  XT-PIC 
  rtc
   9: 251135 
  XT-PIC 
  usb-uhci, 
  eth0
  10: 92378687 
  XT-PIC 
  wcfxs
  11: 92395359 
  XT-PIC 
  wcfxo
  12: 
  20 
  XT-PIC 
  PS/2 Mouse
  14: 75560 
  XT-PIC 
  ide0
  15: 
  0 
  XT-PIC 
  ide1
  NMI: 
  0
  ERR: 
  0
  
  
  Please respond if someone is aware 
  of these types of problems.
  
  Thanks in 
  advance,
  
  Bob

Re: [Asterisk-Users] asterisk-grandstream call

2004-02-09 Thread Glenn Dalgliesh 
Please include your sip.conf and extension.conf files. Hard to say what is
wrong without seeing the configuration

- Original Message - 
From: Bill Michaelson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 09, 2004 3:15 PM
Subject: [Asterisk-Users] asterisk-grandstream call


 I am trying to muddle my way tthrough getting something - actually
 anything to work - with Asterisk.  I've acquired a Grandstream phone and
 I've got * on a Red Hat 9 box.   I've gotten to a point where I can see
 (via ethereal) that the phone REGISTER's successfully with asterisk, and
 then I try to dial into voicemail.  This is what I observe in the packet
 trace...

 GS: INVITE - *
 *: Status 100 (Trying) - GS
 *: Status 200 (OK with session description) - GS

 So far, seems reasonable - but I'm a complete novice with this protocol.

 Then I see a huge stream of UDP packets sent by * to the GS on port
 5004, but the GS only replies with an ICMP destination unreachable to
 each packet.  I'm guessing that this is an RTP audio stream, but I don't
 know why the GS is not ready or otherwise unwilling to receive the
 packets.  Examining the GS config, I've confirmed that the local RTP
 port is set to 5004.

 I have many questions about how this should work, but I'll save some
 bandwidth and leave it to someone here to suggest what should be checked
 next.

 Thanks.

 -- 
 Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED]
 Thanks for putting up with my spam filter!


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Re: [Asterisk-Users] Release phone call

2004-02-05 Thread Glenn Dalgliesh 
Title: Message



I don't really have a answer for you on you issue 
but have a question about what "find-me" is. I see it on the feature list but am 
unable to find any real information about it. Is this simply call forward or is 
their more to it. 

thanks

  - Original Message - 
  From: 
  B. J. Bomar 
  
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, February 05, 2004 1:01 
  PM
  Subject: [Asterisk-Users] Release phone 
  call
  
  Hello all, I am 
  trying to figure out how to have * release a phone call. We are noticing 
  some call quality issues on people who have a "find-me" feature, and answer 
  the call through a cell phone. Here is the call path we are seeing, and 
  all VoIP connections are using SIP.
  
  PSTN --- Cisco 
  7206 --- * Server
  ^---| 
  ^-|
  
  Hopefully the 
  diagram makes sense, but in case it doesn't, let me try to explain. A 
  call comes in from PSTN into our Cisco7206 with PRI card. It then goes 
  to our * server, which then forwards the call back through the Cisco to a cell 
  phone on PSTN. I am wanting to have * release the call to the Cisco once 
  the call is connected. Any thoughts or ideas?
  
  Thanks.
  
  B. 
  J.

[Asterisk-Users] Vegastream 50 FXO with Asterisk

2004-02-05 Thread Glenn Dalgliesh 



Anyone have any 
experienceconfiguringVegaStream's with Asterisk.

Ihave run 
into afew of questions. 1. It appear that after turning on 
registrations I am seeing two request for registration per 
linesip:[EMAIL PROTECTED]sip:[EMAIL PROTECTED]What is 
purpose and how do I handle this?2. DTMF btw Asterisk and the Unit I was 
unable to get rfc2833 to work successfully with inbound or outbound DTMF. Is 
this a known issue?
3. How is the 
best way to deal with dialout and selecting a free channel on the 
VegaStream
Any general 
suggestions/experiences with regard to configuring a VegaStream 
withasteriskwould be 
appricated.Thanks

Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

2004-01-25 Thread Glenn Dalgliesh 
I have had several installations where I was unable in any configuration to
make the FVS318 work with VOIP traffic. I don't belive it is related to any
paticular Phones or VOIP GW have see same problems with even Cisco 7960's

Has anyone opened a ticket with Netgear on this issue?
- Original Message - 
From: Philipp von Klitzing [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 24, 2004 5:35 PM
Subject: Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)


 Hi!

   I've concluded that the Netgear router (FVS318) performing the NAT is
   corrupting the outgoing RTP packets.  Traces confirmed that the
BudgeTone
   is sending them out with a UDP checksum of 0 but the next hop after
the
   Netgear router they are set to a non-zero value (an incorrect one).
   Asterisk is never even seeing the packets because the kernel is
   recognizing them as corrupt and dropping them, hence the recvfrom()
   Resource temporarily unavailable errors in rtp.c.
 
  Here is Netgear's response:
   Original
Message 
  SIP VOIP phones do not work with netgear routers. The router will always
  set a value in the checksum.

 For the record: With a BT 101 behind NAT provided by a Netgear WGR614 I
 don't experience that error message.

 Philipp


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Re: [Asterisk-Users] Example of TDM20B

2004-01-25 Thread Glenn Dalgliesh 
This is my working config for x100p  tdm400 so if you change the channel
entries from 2-5 to 2-3 you should be good to go.

/etc/rc.d/rc.local
modprobe wcfxo
modprobe wcfxs
/usr/sbin/asterisk

/etc/zaptel.conf
fxsks=1
fxols=2-5
loadzone = us
defaultzone=us

/etc/asterisk/zapata.conf
[channels]
echocancel=yes
echocancelwhenbridged=yes
context=default
signalling=fxo_ls
;relaxdtmf=yes
group=2
channel= 2-5

signalling=fxs_ks
busydetect=no
context=incoming
channel= 1

- Original Message - 
From: Steven E. Frazier [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, January 25, 2004 6:12 PM
Subject: [Asterisk-Users] Example of TDM20B


 I am trying to find an example of how to set up my FXS Station Card in my
 Asterisk.

 I have (1) XP100P
 I have (1) tdm20B (2 Port FXS)

 Could someone tell me if this is correct?



 /etc/zaptel.conf

 fxsks=1
 fxoks=2
 fxoks=3
 loadzone=us
 defaultzone=us

 /etc/asterisk/zapata.conf


 [channels]
 ;
 language=en
 ;
 ;X100P Port 1
 context=inbound-analog
 signalling=fxs_ks
 usecallerid=yes
 echocancel=yes
 echocancelwhenbridged=yes
 channel = 1
 ;
 ; FXS Port 1
 context=local
 signalling=fxs_ls
 usecallerid=yes
 echocancel=yes
 echocancelwhenbridged=yes
 ;
 ;FXS Port 2
 context=local
 signalling=fxs_ls
 usecallerid=yes
 echocancel=yes
 echocancelwhenbridged=yes


 /etc/asterisk/extensions.conf


 [local]
 exten = 2203,1,SetMusicOnHold,loud
 exten = 2203,2,Dial(Zap/2,15,Ttr)
 exten = 2203,102,Voicemail(2203)
 exten = 2203,Hangup


 exten = 2204,1,SetMusicOnHold,loud
 exten = 2204,2,Dial(Zap/2,15,Ttr)
 exten = 2204,102,Voicemail(2203)
 exten = 2204,Hangup


 Should this be enough for me to get dial tone on my FXS Cards?


 Thanks.
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Re: [Asterisk-Users] Mediatrix 1204 sip experience?

2004-01-23 Thread Glenn Dalgliesh 
Works okay but user interface is a little like using RegEdit to program your
router.
In the version of software the one I have it lack some security features and
I am unable to find any DMTF controls


- Original Message - 
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk-a-users-list [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 2:40 PM
Subject: [Asterisk-Users] Mediatrix 1204 sip experience?



 Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip
FXO
 4-port gateway?

 The archives tend to suggest the box is not very straight forward, and
possibly
 lacks some basic pstn interaction features.

 Thinking about trying one in place of a pair of x100p's (functioning fine
now).
 CallerId, etc, supported on this gateway?

 Rich


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Re: [Asterisk-Users] Channel Banks

2004-01-19 Thread Glenn Dalgliesh 
Title: Channel Banks



Well, you have several options. A T100P and a 
device such as a Adtran Altlas or simpler Channel bank. But since at this time 
as you point out Digium only has 1 FXOport per PCI slot(FYI I hear they 
are working on a 4 port per PCI slot). The other options are MediaTrix, 
VegaStream, or other devices like them. They are essentially FXO to SIP over 
Ethernet devices and although I really like the Digium T100P(It is 
solid)for whatyou are trying toaccomplish I think card and 
channel bank may not be the most elegant solution.

http://www.adtran.com/adtranpx/Rooms/DisplayPages/LayoutInitial?Container=com.webridge.entity.Entity%5BOID%5B4299336D011EC042A05F1D0C7B3E2AD9%5D%5D
www.vegastream.com
www.mediatrix.com

Just my opinion!
- Original Message - 

  From: 
  Kostur, 
  Andre 
  To: '[EMAIL PROTECTED]' 
  
  Sent: Monday, January 19, 2004 11:30 
  AM
  Subject: [Asterisk-Users] Channel 
  Banks
  
  OK, I'm having some trouble finding which equipment I need 
  
  What I'd like to do is take about a dozen incoming analog 
  lines and bring them into an * server. Of course one is going to have a 
  hard time fitting a dozen X100P cards in a case, so an alternative would be a 
  channel bank and a T100P in the * server. Now here's where my confusion 
  comes in. I _think_ I need a channel bank that has a T1 interface on one 
  side (to go to the * server), and FXO interfaces on the other (to accept the 
  incoming analog lines from the telco)? And are there suggestions out 
  there as to which channel banks one should select for this sort of 
  deployment?

[Asterisk-Users] Fw: Forward call with response required to accept

2004-01-12 Thread Glenn Dalgliesh 



Sorry, If this is a dual post, was having trouble 
with email.



I am looking for a way to Forward to a external or 
internal number and require a digit(s) in order to complete 
forward.

Example:

PSTN1Calls * dialsPSTN2
 
ifPSTN2pressesproper digitsbridge the PSTN1 and 
PSTN2
if 
no response return to a context

Reasons: 2 actually
 1. call is forwarded to cell 
phone but If cell is out of range, turned off, or they don't answer I don't want 
the calling party to get connected to the Cell phones VM
 2. Call is forwarded to outside 
number and I want a level of security that ensures that the person that the call 
is intended for is present.

Any help would be appreciated
Thanks

[Asterisk-Users] Forward call with response required to accept

2004-01-12 Thread Glenn Dalgliesh 



I am looking for a way to Forward to a external or 
internal number and require a digit(s) in order to complete 
forward.

Example:

PSTN1Calls * dialsPSTN2
 
ifPSTN2pressesproper digitsbridge the PSTN1 and 
PSTN2
if 
no response return to a context

Reasons: 2 actually
 1. call is forwarded to cell 
phone but If cell is out of range, turned off, or they don't answer I don't want 
the calling party to get connected to the Cell phones VM
 2. Call is forwarded to outside 
number and I want a level of security that ensures that the person that the call 
is intended for is present.

Any help would be appreciated
Thanks

[Asterisk-Users] Forward call with response required to accept

2004-01-11 Thread Glenn Dalgliesh 



I am looking for a way to Forward to a external or 
internal number and require a digit(s) in order to complete 
forward.

Example:

PSTN1Calls * dialsPSTN2
 
ifPSTN2pressesproper digitsbridge the PSTN1 and 
PSTN2
if 
no response return to a context

Reasons: 2 actually
 1. call is forwarded to cell 
phone but If cell is out of range, turned off, or they don't answer I don't want 
the calling party to get connected to the Cell phones VM
 2. Call is forwarded to outside 
number and I want a level of security that ensures that the person that the call 
is intended for is present.

Any help would be appreciated
Thanks

Re: [Asterisk-Users] SIP/grandstream not registering

2004-01-03 Thread Glenn Dalgliesh 
It looks like you have you * on public IP and your phones on private, most
likely behind NAT if so in your sip.conf under each [grandstreamX] you most
likely need:  nat=yes


- Original Message - 
From: Chandra [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 03, 2004 1:44 AM
Subject: [Asterisk-Users] SIP/grandstream not registering


 hi,

 i can't seem to register my grandstream SIP to * server...

 i have my grandstream IP as 192.168.0.11 want to register to * at
 202.51.xx.xxx.

 sip show peers says that my grand stream has unspecified IP but when i try
 to dial a number it gets this error...
 WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
 exceeded on call [EMAIL PROTECTED] for
seqno
 40939 (Response)


 my sip.conf is...
 [grandstream2]
 type=peer
 host=dynamic
 secret=grandstream2
 reinvite=no
 canreinvite=no
 qualify=60


 [grandstream2]
 type=user
 host=dynamic
 secret=grandstream2
 context=outgoing
 reinvite=no
 canreinvite=no
 qualify=60

 help


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Re: [Asterisk-Users] Grandstream budgetTone registration time out

2003-12-24 Thread Glenn Dalgliesh 



What version of the BudgeTone software are you 
running?


  - Original Message - 
  From: 
  Chandra 
  To: [EMAIL PROTECTED] 
  
  Sent: Wednesday, December 24, 2003 12:09 
  PM
  Subject: [Asterisk-Users] Grandstream 
  budgetTone registration time out
  
  hi,
  
  i have been using grandstream budgettone IP 
  phones and they work fine except that these phones times out after some 
  hours.. i ahve seen that the phones working ok are next day unregistered and 
  sip show peers do not show their IP and although these phones can make calls , 
  they cannot be called. They Sip show peers only shows their IP when i restart 
  the IP phones. This is really annoying me now. Is there any better solutions 
  than just restarting the phones every day?
  
  Any help is appreciated.
  
  cm

[Asterisk-Users] sip.conf and Codecs

2003-12-10 Thread Glenn Dalgliesh 



I have been doing some testing and have found issue 
with certain devices and negotiating codecs in doing this Ihave noticed 
something that seems peculiar to me. It seems that including allow=all 
yields different results than having no disallow or allows in the sip.conf. 
Could someone please explain why that is true?

Thanks

[Asterisk-Users] GrandStream Budgetone Phone DHCP General Observations

2003-12-05 Thread Glenn Dalgliesh 




Symptom: Phone after about 15mins 
will stop functioning 
Problem: DHCP lease renewed but default route 
dropped
Fix: Assign a static ip and problem 
is resolved. Upgrade to new firmware once it is 
released

It turn's out thatthese phones have a few 
issue in 1.0.3.81 firmware. Thephone may stop 
transmitting packets if configuredwith DHCP, if DHCP is being provided by 
certain devices.Netopia routers have been confirmed in this category. It 
turns out that there is some differences btw the implementation of DHCP btw 
different vendor and this is causing the phone to loose it default route and 
stop transmitting packets approx 15mins after the phone receives it's lease 
after reboot. GrandStream says this will be fixed in the next 
release.

Other Grandstream 
observations:

In many situations I have found added 
the following to * sip.confwill correct many problem btw Budgetone's and 
other phones registered with *.

[general]
disallow=adpcmdisallow=gsmdtmfmode=inband

Stun and Budgetone: I have observed 
odd behavior with the phone and various STUN server's.Leaving the phone 
behinda given router and pointing the phone's stun server at 
variousversions of STUNd reporting different results to the phone. I 
am not sure if this isa budgetone or stun server problem. but I do know 
that it causes problems it certain environments.

Re: [Asterisk-Users] GrandStream Budgetone Phone DHCP General Observations

2003-12-05 Thread Glenn Dalgliesh 
FYI you can't get back to the old firmware in some cases apparently.

- Original Message - 
From: Bartosz Jozwiak [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 11:53 AM
Subject: Re: [Asterisk-Users] GrandStream Budgetone Phone  DHCP  General
Observations


 I would love to try it out too!


 - Original Message - 
 From: Senad Jordanovic [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Friday, December 05, 2003 1:38 PM
 Subject: RE: [Asterisk-Users] GrandStream Budgetone Phone  DHCP  General
 Observations


  Nicolas Bougues wrote:
   On Fri, Dec 05, 2003 at 10:42:02AM -0500, Glenn Dalgliesh wrote:
   Symptom: Phone after about 15mins will stop functioning
   Problem: DHCP lease renewed but default route dropped
   Fix: Assign a static ip and problem is resolved. Upgrade to new
   firmware once it is released
  
   It turn's out that these phones have a few issue in 1.0.3.81
   firmware.
   The phone may stop transmitting packets if configured with DHCP, if
   DHCP is being provided by certain devices. Netopia routers have been
   confirmed in this category. It turns out that there is some
   differences btw the implementation of DHCP btw different vendor and
   this is causing the phone to loose it default route and stop
   transmitting packets approx 15mins after the phone receives it's
   lease
   after reboot. GrandStream says this will be fixed in the next
   release.
  
  
   Interesting. We have 6 GS phones, one is 1.0.3.81 and has this
   behaviour, the others, 1.0.4.17 are ok. The DHCP server is Linux
   dhcpd.
  
   In a remote office, they have an Allied Telesyn router providing
   DHCP, and all the phones, no matter the version, work well.
  
   On a slightly different topic : does somebody know of a NAT-friendly
   (as Grandstream means it) tftpd server ? It seems theirs replies from
   port 69, which is the only thing their phones will accept.
  
   [ If anybody wants it, I can send the 1.0.4.17 firmware by email ].
 
  Hi,
  I would be interested in having 1.0.4.17 firmware. :)
  Thanks
 
  SJ
 
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Re: [Asterisk-Users] Newbie hardware question

2003-10-30 Thread Glenn Dalgliesh 



You could also look at products like
http://sales.netxusa.com/vegastream/vega50.php

  - Original Message - 
  From: 
  Andy 
  Hester 
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, October 30, 2003 3:46 
  PM
  Subject: RE: [Asterisk-Users] Newbie 
  hardware question
  
  
-Original Message-From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]On Behalf Of Just 
MESent: Thursday, October 30, 2003 11:00 AMTo: [EMAIL PROTECTED]Subject: 
[Asterisk-Users] Newbie hardware question
Hi,
I have scanned 
through the archives of this list and found a number of question about 
hardware, but I just can not find the answer to my question. I am new 
to phone systems, I got "drafted" to come up with a new phone system for our 
company (I guess they figure since I know computers I know phone systems as 
well :O).
We have 5 analog 
(I guess they are called PSTN lines) lines coming in and 16 clients 
(telephones) in our office. I am not worried about the minimum 
computer requirements because I have a coupleof spare P4 based servers 
with 512 megs of memory, but I need to know what cards should I be looking 
at using becauseI will run out of PCI slots if Iuse 4 TDM400P 
cards (for the clients) and 5 of the X100p (for the 
lines).
Any help or 
advice would be greatly appreciated.
Thanks

Jon 
Hoffman

Jon,
Steven 
just answered this question quite well, so I'll just refer 
tohim:

Andy
snip

You will want either a T100P, or a T400P. Then you will want a 
channel
bank that is modular enough to add a FXO card to it. With 5 lines of
FXO, the Adtran units will be a good choice as they are in units of 6
lines. The Adit cards are 8 lines at a time. The Adtran unit would 
let
you get 18 extensions and 6 incoming lines on a single T1 interface. 
Both of these units can be bought on Ebay for relatively inexpensive
compared to new prices. Then you will either have to scour the net 
for
the FXO card, or go pay full price for it. 
Either way, this gets you down to 1 PCI card. If you go the route of 
a
T400P card, adding more service later will be less of a hassle. You
could also use it to do your network routing if you decide to go frac 
T1
for data and some phone service tacked onto the same T1 interface. 
This
could potentially even be a better route as you wouldn't need to find
FXO interfaces anymore. You would also get the benefit of using the 
new
software fax setup to get yourself on the way to unified messaging.
-- 
Steven Critchfield 
[EMAIL PROTECTED]

Re: [Asterisk-Users] Am I missing somthing?

2003-10-29 Thread Glenn Dalgliesh 
No this most likely willn't work unless you have open the correct ports on
each NAT device. The problem is that NAT in general only allows packet in if
a packet has gone out first. I am assuming you have left have the fact that
* is used to setup the SIP call setup and then should drop out. If so when
you try to * tell the phone you are trying to contact what Ip and port to
contact you on the far end phone starts a RTP stream to that IP and port but
since your phone has not sent out an packet to the End phones ip on that
port the packets are just dropped. So, in most cases having NAT=yes and
CANREINVITE=no is the simplest although may not be the most efficient use of
bandwidth.

Below are some technical documents that help explain what the problems are
and how some people are dealing with it.

Document you should Read about NAT and SIP
http://corp.deltathree.com/technology/nattraversalinsip.pdf
http://www.ietf.org/proceedings/01aug/slides/avt-6/sld001.htm

--Info from FreeWorldDialup configuration with NAT
How do I go through a NAT?
Normally, if your SIP Telephone supports STUN, When using NAT (Network
Address Translation), both the SIP telephone and NAT sometimes need to be
configured. We have deployed a solution from Jasomi Networks
http://www.jasomi.com that helps FWD users traverse NATs and Firewalls but
if you SIP Telephone supports STUN like the Cisco 7960 and Cisco ATA-186
does, it is much preferred if you take the time to enable port forwarding
thru your NAT to your SIP Telephone.

Many NAT products have a web based configuration tool.

For example, on the Linksys NAT, Packet forwarding shows up under the
advanced tab on setup.

The key with getting the NAT problem solved on the ATA-186 is to forward the
SIP port: 5060 to the IP address assigned to the IP phone and then to
forward the media port range 16384 to 16391 to the IP address assigned to
the IP Phone. The Cisco 7960 requires the media port range of 16384 to 32768
to be fowarded to the IP Phone.

We recommend that members of the FWD Community consider using SIP Friendly
firewalls such as the products of InterTex http://www.internex.se and
Ingate.


Cisco ATA-186 NAT Notes: For the ATA 186, if you are using ATA firmware
version below 2.14, set the NATIP field to the NAT device's public IP
address, disable DHCP and set the StaticIP, StaticRoute, and StaticNetMask
fields for your private network's values. The NAT device must be configured
to forward the SIPPort and a range of 8 ports starting at MediaPort.

If you are using ATA firmware version 2.14 or above, you are not required to
set NATIP. Check out this link:
http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/atarn/186rn214.ht
mfor more information regarding release 2.14. In release 2.14, you may
leave the NAT IP address at the default value of 0 or 0.0.0.0 and let
the ATA automatically scan the Via header for a received= parameter when a
message is received. The parameter, if present, would indicate to the Cisco
ATA 186 that it is operating behind a firewall.

I'm still having NAT problems, now what? Note: If you are using ZoneAlarm
you will need to disable it, or at least be able to open up port: 5082.

You can visit the FWD QuickStart Guide
http://pulver.com/fwd/quick/nat.html and view the configuration for four
of the most popular FWD endpoints for somebody who is behind a NAT/Firewall
including: Windows Messenger, SJphone, ATA-186, Cisco 7960. Other models
most likely will work, but it will be up to the community to report their
success with us so we will know for certain.

If you find yourself still having trouble, please email [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED], with the subject: Need Alternative Proxy Help.
Jeff's .NET ID is [EMAIL PROTECTED] and he is available to provide limited
real-time support.

In your email, please let Jeff know your FWD Number and the kind of client
that you are trying to register on the FWD Network.


- Original Message - 
From: WipeOut [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 29, 2003 4:14 AM
Subject: [Asterisk-Users] Am I missing somthing?


 Should the following setup work?

 SIP UA---NAT---Internet---NAT---SIP UA

 If both UA's support STUN and report the external IP address in the SIP
 packet..

 I am trying to get away from using canreinvite=no so that traffic can go
 directly between the UA's and not via the central server but I can't
 seem to get it to work..

 Has anyone set this up and can give me some pointers??

 Later..

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[Asterisk-Users] Dial Multiple extension but require input not just off hook to bridge calls

2003-10-17 Thread Glenn Dalgliesh 



I am try to come up with a way to dial multiple ext 
and require one or more of the extension to require input before actually 
bridging the calls.

Example:
Acall from PSTN into * a match is made in 
extensions.conf and *then dials a 
local(fxs) ext 1 and a Cell Phone
If ext 1 picks up * bridges 
thecall
if Cell picks up require digit to in order 
tobridge thecall
if neither occurs and timeout is reach 
hang-updialed ext's and send call from the PSTN 
tovoicemail

The reason I am looking to do thisis that if 
you dial a cell phone and it is off or out of area the Voicemail of the cell 
phone will pickup immediatelyand then the voicemail will end up on cell 
phone and not *. If youcan require the dialed number torequire a 
digit in order to accept the bridgethen if no inputyou would 
reachtimeout value and get transferred to * voicemail.

Thanks for any 
help!!!

Re: [Asterisk-Users] X100P Config

2003-10-10 Thread Glenn Dalgliesh 
Title: Leterhead



What do you have configured in your 
/etc/zaptel.conf * /etc/asterisk/zapata.conf file. Also, submit a cat /proc/pci 
this should show adevice Tiger Jet Network Inc. if the pci bus recognized 
the card.


  - Original Message - 
  From: 
  David J Carter 
  To: [EMAIL PROTECTED] 
  
  Sent: Friday, October 10, 2003 2:05 
  PM
  Subject: [Asterisk-Users] X100P 
  Config
  
  
  
  Hiya 
  all,
  
  I 
  have just received my X100P telco card and I don’t seem to be able to talk to 
  it.
  
  I 
  am a bit of a numpty on Linux being from the Windows (wash my mouth with soap 
  and water) background, so any help would be 
  appreciated.
  
  I 
  have checked under YaST2 and I think it can see the card, but not 
  sure.
  
  My 
  * box is talking between 2 Grandstream phones no probs but now I would like to 
  talk to the outside world.
  
  Thanks 
  in anticipation.
  
  
  Dave

[Asterisk-Users] Iconnect Incomming calls

2003-10-07 Thread Glenn Dalgliesh 




I have an IconnectHere account 
with a Inbound number and have setup the sip.conf to register and am recieving 
the call but When I answer the call it disconnect. I have tried sending the call 
to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon 
as I accept the call it disconnects. I believe it may be some type of codec 
issue but I am not very familiar with that layer.

Below is the .conf's  SIP 
debug

Thank for any 
help

; SIP Configuration for 
Asterisk;[general]port = 
5060 
; Port to bind tobindaddr = 
0.0.0.0 
; Address to bind tocontext = 
sipinbound ; 
Default for incoming callsregister = 
1410344:[EMAIL PROTECTED]/1410344

--=-=-=-= extentions.conf-=-=-=-=-=- have also 
tried sip phone same results
[sipinbound]Exten = 
_.,1,Dial,Zap/5-1


-=-=-=-=-=-=-=-=-=- upgraded to lastest cvs with 
same results 
pbx1*CLI show versionAsterisk 
CVS-10/03/03-13:40:08 built by [EMAIL PROTECTED] on a i686 
running Linux
-=-=-=-=-=-=-=-=-=-=

pbx1*CLI Sip read: INVITE 
sip:[EMAIL PROTECTED] SIP/2.0Record-Route: 
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176Via: 
SIP/2.0/UDP 
213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1Via: 
SIP/2.0/UDP 213.137.65.234:5060From: 
sip:[EMAIL PROTECTED];tag=17B49340-1BD5To: 
sip:[EMAIL PROTECTED]Date: Fri, 03 Oct 2003 20:37:52 
GMTCall-ID: [EMAIL PROTECTED]Supported: 
timer,100relMin-SE: 1800Cisco-Guid: 
1267048311-4111995351-2493635217-4243844325User-Agent: 
Cisco-SIPGateway/IOS-12.xAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, 
COMET, REFER, SUBSCRIBE, NOTIFY, INFOCSeq: 101 INVITEMax-Forwards: 
9Remote-Party-ID: 
sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=offTimestamp: 
1065213472Contact: sip:[EMAIL PROTECTED]:5060Diversion: 
sip:[EMAIL PROTECTED];reason=unconditionalExpires: 
180Allow-Events: telephone-eventContent-Type: 
application/sdpContent-Length: 332

v=0o=CiscoSystemsSIP-GW-UserAgent 7043 3136 IN 
IP4 213.137.65.234s=SIP Callc=IN IP4 213.137.65.234t=0 0m=audio 
16826 RTP/AVP 4 18 101 19c=IN IP4 213.137.65.234a=rtpmap:4 
G723/8000a=fmtp:4 annexa=noa=rtpmap:18 G729/8000a=fmtp:18 
annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=rtpmap:19 
CN/8000

23 headers, 14 linesUsing latest request as 
basis requestSending to 213.137.73.176 : 5060 (non-NAT)Found audio 
format ULAWFound audio format UNKNFound audio format UNKNFound audio 
format UNKNFound description format G723Found description format 
G729Found description format telephone-eventFound description format 
CNCapabilities: us - 524302, them - 257/0, combined - 0Non-codec 
capabilities: us - 1, them - 3, combined - 1Sip read: INVITE 
sip:[EMAIL PROTECTED] SIP/2.0Record-Route: 
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176Via: 
SIP/2.0/UDP 
213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1Via: 
SIP/2.0/UDP 213.137.65.234:5060From: 
sip:[EMAIL PROTECTED];tag=17B49340-1BD5To: 
sip:[EMAIL PROTECTED]Date: Fri, 03 Oct 2003 20:37:52 
GMTCall-ID: [EMAIL PROTECTED]Supported: 
timer,100relMin-SE: 1800Cisco-Guid: 
1267048311-4111995351-2493635217-4243844325User-Agent: 
Cisco-SIPGateway/IOS-12.xAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, 
COMET, REFER, SUBSCRIBE, NOTIFY, INFOCSeq: 101 INVITEMax-Forwards: 
9Remote-Party-ID: 
sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=offTimestamp: 
1065213472Contact: sip:[EMAIL PROTECTED]:5060Diversion: 
sip:[EMAIL PROTECTED];reason=unconditionalExpires: 
180Allow-Events: telephone-eventContent-Type: 
application/sdpContent-Length: 332

v=0o=CiscoSystemsSIP-GW-UserAgent 7043 3136 IN 
IP4 213.137.65.234s=SIP Callc=IN IP4 213.137.65.234t=0 0m=audio 
16826 RTP/AVP 4 18 101 19c=IN IP4 213.137.65.234a=rtpmap:4 
G723/8000a=fmtp:4 annexa=noa=rtpmap:18 G729/8000a=fmtp:18 
annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=rtpmap:19 
CN/8000

23 headers, 14 linesIgnoring this 
requestLooking for 14103445557 in sipinboundRDNIS is 
4103445557list_route: hop: 
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176list_route: 
hop: sip:[EMAIL PROTECTED]:5060Transmitting (no 
NAT):SIP/2.0 100 TryingVia: SIP/2.0/UDP 
213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1Via: 
SIP/2.0/UDP 213.137.65.234:5060From: 
sip:[EMAIL PROTECTED];tag=17B49340-1BD5To: 
sip:[EMAIL PROTECTED];tag=as72f8d457Call-ID: [EMAIL PROTECTED]CSeq: 
101 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, 
BYE, REFERContact: sip:[EMAIL PROTECTED]Content-Length: 
0

to 213.137.73.176:5060 -- Executing 
Dial("SIP/-080e9768", "Zap/5-1") in new stack -- Called 
5-1 -- Zap/5-1 is ringingTransmitting (no 
NAT):SIP/2.0 180 RingingVia: SIP/2.0/UDP 
213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1Via: 
SIP/2.0/UDP 213.137.65.234:5060From: 
sip:[EMAIL PROTECTED];tag=17B49340-1BD5To: 
sip:[EMAIL PROTECTED];tag=as72f8d457Call-ID: [EMAIL PROTECTED]CSeq: 
101 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, 
BYE, REFERContact: sip:[EMAIL PROTECTED]Content-Length: 
0

to

Re: [Asterisk-Users] New here

2003-10-03 Thread Glenn Dalgliesh 
Below are some links that should point you in the right direction. Assuming
you don't have any IP Phone on hand I would recomment starting with to
computers with softphone(one example
http://www.eutecticsinc.com/download6/DISK1/IPP200_SJSoftPhone.htm)  and
have them talk to each other.

http://www.digium.com/index.php?menu=documentation
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=3#config
http://www.megaglobal.net:8080/docs/asterisk/html/asterisk.txt


- Original Message - 
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, October 03, 2003 12:06 PM
Subject: [Asterisk-Users] New here


 Hi!

 I have downloaded asterisk... and I have installed it how I continue?
I
 don't know anything about the software and I don't really understand how
it
 works please any help will be ok.

 Does it exist some manual?

 Thanks a lot!


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[Asterisk-Users] Iconnect Incomming calls

2003-10-03 Thread Glenn Dalgliesh 



I have an IconnectHere account 
with a Inbound number and have setup the sip.conf to register and am recieving 
the call but When I answer the call it disconnect. I have tried sending the call 
to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon 
as I accept the call it disconnects. I believe it may be some type of codec 
issue but I am not very familiar with that layer.

Below is the SIP 
debug

Thank for any 
help

to 
162.33.165.195:5060Sip read: INVITE sip:[EMAIL PROTECTED] 
SIP/2.0Record-Route: 
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176Via: 
SIP/2.0/UDP 
213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1Via: 
SIP/2.0/UDP 213.137.65.234:5060From: 
sip:[EMAIL PROTECTED];tag=16A2EDA0-5B2To: 
sip:[EMAIL PROTECTED]Date: Fri, 03 Oct 2003 15:38:58 
GMTCall-ID: [EMAIL PROTECTED]Supported: 
timer,100relMin-SE: 1800Cisco-Guid: 
2316671854-4109242839-3208043153-4243844325User-Agent: 
Cisco-SIPGateway/IOS-12.xAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, 
COMET, REFER, SUBSCRIBE, NOTIFY, INFOCSeq: 101 INVITEMax-Forwards: 
9Remote-Party-ID: 
sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=offTimestamp: 
1065195538Contact: sip:[EMAIL PROTECTED]:5060Diversion: 
sip:[EMAIL PROTECTED];reason=unconditionalExpires: 
180Allow-Events: telephone-eventContent-Type: 
application/sdpContent-Length: 332

v=0o=CiscoSystemsSIP-GW-UserAgent 8647 4690 IN IP4 
213.137.65.234s=SIP Callc=IN IP4 213.137.65.234t=0 0m=audio 
16836 RTP/AVP 4 18 101 19c=IN IP4 213.137.65.234a=rtpmap:4 
G723/8000a=fmtp:4 annexa=noa=rtpmap:18 G729/8000a=fmtp:18 
annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=rtpmap:19 
CN/8000

23 headers, 14 linesUsing latest request as basis requestSending to 
213.137.73.176 : 5060 (non-NAT)Found audio format 4Found audio format 
18Found audio format 101Found audio format 19Found description 
format G723Found description format G729Found description format 
telephone-eventFound description format CNCapabilities: us - 524302, 
them - 257/0, combined - 0Non-codec capabilities: us - 1, them - 3, combined 
- 1Sip read: INVITE sip:[EMAIL PROTECTED] 
SIP/2.0Record-Route: 
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176Via: 
SIP/2.0/UDP 
213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1Via: 
SIP/2.0/UDP 213.137.65.234:5060From: 
sip:[EMAIL PROTECTED];tag=16A2EDA0-5B2To: 
sip:[EMAIL PROTECTED]Date: Fri, 03 Oct 2003 15:38:58 
GMTCall-ID: [EMAIL PROTECTED]Supported: 
timer,100relMin-SE: 1800Cisco-Guid: 
2316671854-4109242839-3208043153-4243844325User-Agent: 
Cisco-SIPGateway/IOS-12.xAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, 
COMET, REFER, SUBSCRIBE, NOTIFY, INFOCSeq: 101 INVITEMax-Forwards: 
9Remote-Party-ID: 
sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=offTimestamp: 
1065195538Contact: sip:[EMAIL PROTECTED]:5060Diversion: 
sip:[EMAIL PROTECTED];reason=unconditionalExpires: 
180Allow-Events: telephone-eventContent-Type: 
application/sdpContent-Length: 332

v=0o=CiscoSystemsSIP-GW-UserAgent 8647 4690 IN IP4 
213.137.65.234s=SIP Callc=IN IP4 213.137.65.234t=0 0m=audio 
16836 RTP/AVP 4 18 101 19c=IN IP4 213.137.65.234a=rtpmap:4 
G723/8000a=fmtp:4 annexa=noa=rtpmap:18 G729/8000a=fmtp:18 
annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=rtpmap:19 
CN/8000

23 headers, 14 linesIgnoring this requestLooking for 14103445557 in 
sipinboundRDNIS is 4103445557list_route: hop: 
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176list_route: 
hop: sip:[EMAIL PROTECTED]:5060Transmitting (no 
NAT):SIP/2.0 100 TryingVia: SIP/2.0/UDP 
213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1Via: 
SIP/2.0/UDP 213.137.65.234:5060From: 
sip:[EMAIL PROTECTED];tag=16A2EDA0-5B2To: 
sip:[EMAIL PROTECTED];tag=as075e701dCall-ID: [EMAIL PROTECTED]CSeq: 
101 INVITEUser-Agent: Asterisk PBXContact: 
sip:[EMAIL PROTECTED]Content-Length: 0

to 213.137.73.176:5060 -- Executing 
Dial("SIP/-0810da50", "Zap/5-1") in new stack -- Called 
5-1 -- Zap/5-1 is ringingTransmitting (no 
NAT):SIP/2.0 180 RingingVia: SIP/2.0/UDP 
213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1Via: 
SIP/2.0/UDP 213.137.65.234:5060From: 
sip:[EMAIL PROTECTED];tag=16A2EDA0-5B2To: 
sip:[EMAIL PROTECTED];tag=as075e701dCall-ID: [EMAIL PROTECTED]CSeq: 
101 INVITEUser-Agent: Asterisk PBXContact: 
sip:[EMAIL PROTECTED]Content-Length: 0

to 213.137.73.176:5060 -- Zap/5-1 is 
ringing -- Zap/5-1 answered SIP/-0810da50We're at 
162.33.165.198 port 13196Answering with non-codec capability 1Reliably 
Transmitting (no NAT):SIP/2.0 200 OKVia: SIP/2.0/UDP 
213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1Via: 
SIP/2.0/UDP 213.137.65.234:5060Record-Route: 
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176From: 
sip:[EMAIL PROTECTED];tag=16A2EDA0-5B2To: 
sip:[EMAIL PROTECTED];tag=as075e701dCall-ID: [EMAIL PROTECTED]CSeq: 
101 INVITEUser-Agent: Asterisk PBXContact: 
sip:[EMAIL PROTECTED]Content-Type: 
application/sdpContent-Length: 167

v=0o=root 1387 1387 IN IP4

Re: [Asterisk-Users] Iconnect Incomming calls

2003-10-03 Thread Glenn Dalgliesh 



; SIP Configuration for 
Asterisk;[general]port = 
5060 
; Port to bind tobindaddr = 
0.0.0.0 
; Address to bind tocontext = 
sipinbound ; 
Default for incoming callsregister = 
1410344:[EMAIL PROTECTED]/1410344

--=-=-=-= extentions.conf-=-=-=-=-=- have also 
tried sip phone same results
[sipinbound]Exten = 
_.,1,Dial,Zap/5-1


-=-=-=-=-=-=-=-=-=- upgraded to lastest cvs with 
same results 
pbx1*CLI show versionAsterisk 
CVS-10/03/03-13:40:08 built by [EMAIL PROTECTED] on a i686 
running Linux
-=-=-=-=-=-=-=-=-=-=

pbx1*CLI Sip read: INVITE 
sip:[EMAIL PROTECTED] SIP/2.0Record-Route: 
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176Via: 
SIP/2.0/UDP 
213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1Via: 
SIP/2.0/UDP 213.137.65.234:5060From: 
sip:[EMAIL PROTECTED];tag=17B49340-1BD5To: 
sip:[EMAIL PROTECTED]Date: Fri, 03 Oct 2003 20:37:52 
GMTCall-ID: [EMAIL PROTECTED]Supported: 
timer,100relMin-SE: 1800Cisco-Guid: 
1267048311-4111995351-2493635217-4243844325User-Agent: 
Cisco-SIPGateway/IOS-12.xAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, 
COMET, REFER, SUBSCRIBE, NOTIFY, INFOCSeq: 101 INVITEMax-Forwards: 
9Remote-Party-ID: 
sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=offTimestamp: 
1065213472Contact: sip:[EMAIL PROTECTED]:5060Diversion: 
sip:[EMAIL PROTECTED];reason=unconditionalExpires: 
180Allow-Events: telephone-eventContent-Type: 
application/sdpContent-Length: 332

v=0o=CiscoSystemsSIP-GW-UserAgent 7043 3136 IN 
IP4 213.137.65.234s=SIP Callc=IN IP4 213.137.65.234t=0 0m=audio 
16826 RTP/AVP 4 18 101 19c=IN IP4 213.137.65.234a=rtpmap:4 
G723/8000a=fmtp:4 annexa=noa=rtpmap:18 G729/8000a=fmtp:18 
annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=rtpmap:19 
CN/8000

23 headers, 14 linesUsing latest request as 
basis requestSending to 213.137.73.176 : 5060 (non-NAT)Found audio 
format ULAWFound audio format UNKNFound audio format UNKNFound audio 
format UNKNFound description format G723Found description format 
G729Found description format telephone-eventFound description format 
CNCapabilities: us - 524302, them - 257/0, combined - 0Non-codec 
capabilities: us - 1, them - 3, combined - 1Sip read: INVITE 
sip:[EMAIL PROTECTED] SIP/2.0Record-Route: 
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176Via: 
SIP/2.0/UDP 
213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1Via: 
SIP/2.0/UDP 213.137.65.234:5060From: 
sip:[EMAIL PROTECTED];tag=17B49340-1BD5To: 
sip:[EMAIL PROTECTED]Date: Fri, 03 Oct 2003 20:37:52 
GMTCall-ID: [EMAIL PROTECTED]Supported: 
timer,100relMin-SE: 1800Cisco-Guid: 
1267048311-4111995351-2493635217-4243844325User-Agent: 
Cisco-SIPGateway/IOS-12.xAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, 
COMET, REFER, SUBSCRIBE, NOTIFY, INFOCSeq: 101 INVITEMax-Forwards: 
9Remote-Party-ID: 
sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=offTimestamp: 
1065213472Contact: sip:[EMAIL PROTECTED]:5060Diversion: 
sip:[EMAIL PROTECTED];reason=unconditionalExpires: 
180Allow-Events: telephone-eventContent-Type: 
application/sdpContent-Length: 332

v=0o=CiscoSystemsSIP-GW-UserAgent 7043 3136 IN 
IP4 213.137.65.234s=SIP Callc=IN IP4 213.137.65.234t=0 0m=audio 
16826 RTP/AVP 4 18 101 19c=IN IP4 213.137.65.234a=rtpmap:4 
G723/8000a=fmtp:4 annexa=noa=rtpmap:18 G729/8000a=fmtp:18 
annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=rtpmap:19 
CN/8000

23 headers, 14 linesIgnoring this 
requestLooking for 14103445557 in sipinboundRDNIS is 
4103445557list_route: hop: 
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176list_route: 
hop: sip:[EMAIL PROTECTED]:5060Transmitting (no 
NAT):SIP/2.0 100 TryingVia: SIP/2.0/UDP 
213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1Via: 
SIP/2.0/UDP 213.137.65.234:5060From: 
sip:[EMAIL PROTECTED];tag=17B49340-1BD5To: 
sip:[EMAIL PROTECTED];tag=as72f8d457Call-ID: [EMAIL PROTECTED]CSeq: 
101 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, 
BYE, REFERContact: sip:[EMAIL PROTECTED]Content-Length: 
0

to 213.137.73.176:5060 -- Executing 
Dial("SIP/-080e9768", "Zap/5-1") in new stack -- Called 
5-1 -- Zap/5-1 is ringingTransmitting (no 
NAT):SIP/2.0 180 RingingVia: SIP/2.0/UDP 
213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1Via: 
SIP/2.0/UDP 213.137.65.234:5060From: 
sip:[EMAIL PROTECTED];tag=17B49340-1BD5To: 
sip:[EMAIL PROTECTED];tag=as72f8d457Call-ID: [EMAIL PROTECTED]CSeq: 
101 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, 
BYE, REFERContact: sip:[EMAIL PROTECTED]Content-Length: 
0

to 213.137.73.176:5060 -- Zap/5-1 is 
ringing -- Zap/5-1 answered SIP/-080e9768We're at 
162.33.165.198 port 17288Answering with non-codec capability 1Reliably 
Transmitting (no NAT):SIP/2.0 200 OKVia: SIP/2.0/UDP 
213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1Via: 
SIP/2.0/UDP 213.137.65.234:5060Record-Route: 
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176From: 
sip:[EMAIL PROTECTED];tag=17B49340-1BD5To: 
sip:[EMAIL

Re: [Asterisk-Users] Hardware Question

2003-10-03 Thread Glenn Dalgliesh 
www.diguim.com

Wildcard T100P - Single t1
http://www.digium.com/downloads/product_sheets/T100P.pdf
Wildcard TE410P - 4 port - I believe support independant config of each T1
http://www.digium.com/downloads/product_sheets/TE410P.pdf
Wildcard T400P - 4 port -
http://www.digium.com/downloads/product_sheets/T400P.pdf
- Original Message - 
From: Jorge Daniel Cisneros Flores [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, October 03, 2003 5:38 PM
Subject: [Asterisk-Users] Hardware Question



 Hi

Somebody can tell me wich card i need to use that suport em signal,
and
 is posible to connect this card to any PBX

 Thanks


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