[Asterisk-Users] SIP/grandstream not registering

[Chandra]

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] Call recording/SIP not loggin IN

[CW_ASN]

- Original Message -
From: "Chandra" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, January 03, 2004 1:34 AM
Subject: Re: [Asterisk-Users] Call recording/SIP not loggin IN


> My sip.conf
> [general]
> port = 5060 ; Port to bind to
> bindaddr = 0.0.0.0
> disallow=all; Disallow all codecs
> allow=ulaw  ; Allow codecs in order of preference
>
> dtmfmode=rfc2833
>
> [xlite1]
> type=user
> host=dynamic
> secret=xlite1
> context=outgoing
> reinvite=no
> canreinvite=no
> qualify=60
>
> [xlite1]
> type=peer
> host=dynamic
> secret=xlite1
> reinvite=no
> canreinvite=no
> qualify=60
>
> In xlite i have User=xlite1, Pwd=xlite1 and SIP Proxy=IP of my * box, Out
> bound Proxy= IP of my * box
>
> netstat -na gives
>
> [EMAIL PROTECTED] root]# netstat -na
> Active Internet connections (servers and established)
> Proto Recv-Q Send-Q Local Address   Foreign Address State
> tcp0  0 0.0.0.0:32768   0.0.0.0:*   LISTEN
> tcp0  0 0.0.0.0:22305   0.0.0.0:*   LISTEN
> tcp0  0 0.0.0.0:22273   0.0.0.0:*   LISTEN
> tcp0  0 127.0.0.1:32769 0.0.0.0:*   LISTEN
> tcp0  0 0.0.0.0:33060.0.0.0:*   LISTEN
> tcp0  0 0.0.0.0:111 0.0.0.0:*   LISTEN
> tcp0  0 0.0.0.0:20000.0.0.0:*   LISTEN
> tcp0  0 0.0.0.0:56800.0.0.0:*   LISTEN
> tcp0  0 0.0.0.0:80  0.0.0.0:*   LISTEN
> tcp0  0 0.0.0.0:22321   0.0.0.0:*   LISTEN
> tcp0  0 0.0.0.0:22289   0.0.0.0:*   LISTEN
> tcp0  0 0.0.0.0:21  0.0.0.0:*   LISTEN
> tcp0  0 0.0.0.0:22  0.0.0.0:*   LISTEN
> tcp0  0 0.0.0.0:23  0.0.0.0:*   LISTEN
> tcp0  0 0.0.0.0:443 0.0.0.0:*   LISTEN
> tcp0128 202.51.xx.xx1:22202.51.xx.xx0:3148
> ESTABLISHED
> udp0  0 0.0.0.0:32769   0.0.0.0:*
> udp0  0 0.0.0.0:50360.0.0.0:*
> udp0  0 0.0.0.0:50600.0.0.0:*
> udp0  0 0.0.0.0:45690.0.0.0:*
> udp0  0 0.0.0.0:111 0.0.0.0:*
> udp0  0 0.0.0.0:11770   0.0.0.0:*
> udp0  0 0.0.0.0:11771   0.0.0.0:*
> udp0  0 0.0.0.0:24270.0.0.0:*
> Active UNIX domain sockets (servers and established)
> Proto RefCnt Flags   Type   State I-Node Path
> unix  2  [ ACC ] STREAM LISTENING 1504   /dev/gpmctl
> unix  2  [ ACC ] STREAM LISTENING 1775
> /tmp/.font-unix/fs7100
> unix  2  [ ACC ] STREAM LISTENING 1520
> /var/lib/mysql/mysql.sock
> unix  2  [ ACC ] STREAM LISTENING 1885
> /var/run/asterisk.ctl
> unix  2  [ ACC ] STREAM LISTENING 1621
> /tmp/.iroha_unix/IROHA
> unix  2  [ ACC ] STREAM LISTENING 1593   /tmp/cd_sockV4
> unix  2  [ ACC ] STREAM LISTENING 1671   /tmp/kd_sockV4
> unix  2  [ ACC ] STREAM LISTENING 1699   /tmp/td_sockV4
> unix  2  [ ACC ] STREAM LISTENING 1565   /tmp/jd_sockV4
> unix  7  [ ] DGRAM1094   /dev/log
> unix  3  [ ] STREAM CONNECTED 1889
> /var/lib/mysql/mysql.sock
> unix  3  [ ] STREAM CONNECTED 1888
> unix  2  [ ] DGRAM1778
> unix  2  [ ] DGRAM1645
> unix  2  [ ] DGRAM1406
> unix  2  [ ] DGRAM1160
> unix  2  [ ] DGRAM1110
> [EMAIL PROTECTED] root]#
>
>
> my grandstream is also not registering to *.
>

You have two entries for [xlite1].
In order to test, first remove 'qualify' and 'reinvite' from the sip.conf,
reload and try again.
If you don't use NAT, then you should delete OutBoundProxy from xlite
config., and set 'Use OutboundProxy' as 'Never'.
Make sure that xlite is setted as Send internal IP Always.

Assuming that you have only one IP address (and a loopback) in your box,
'netstat' looks good.

Next steps could be dump the traces in xlites, and * box, to see whats wrong
more deeply.


Hope this helps, please advice.

Gus




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RE: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)

[David Gomillion]

Just to second this post: I had the same symptoms and resolved them by
tweaking my firewall.

Hope this helps,
David Gomillion

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of SW
Sent: Friday, January 02, 2004 11:07 PM
To: "John Coll"; [EMAIL PROTECTED]
Subject: [Asterisk-Users] Newbie - getting two local phones to
communicate would be a good start :)

Hi John,

If your effort is to make calls between two GS phones via *, here is
what
you need.
You need all three devices in the same LAN, so set both phones and * to
10.0.1.98/24.

After that from your asterisk Linux box ping both phones. If that is
successful you know your layer 1,2 and 3 are ok. Disable all fire-walls,
iptables, ipchains in Linux box.

Now in * you need two files in /etc/asterisk

sip.conf and extensions.conf.

rename or delete both those existing files.

Here are the minimum you probably need in these two files.

sip.conf :

[general]
port=5060
allow=all
maxexpirey=180
defaultexpirey=160

[5702]
type=friend
username=5702
context=internal
dtmfmode=info

[5703]
type=friend
username=5703
context=internal
dtmfmode=info


And extensions.conf

[internal]

exten => _57XX,1,Dial(SIP/${EXTEN})


Save both files and issue command reload from * CLI.

Now you should be able to call from one phone to another.

while making calls enable sip debug and study the messages going in and
out.
Also if you have ethereal fire that up and capture SIP packets. and see
how
the SIP negotiation goes on. This will help you when you start moving to
fwd, IAXTEL etc. etc.

good luck.

SW




From: "John Coll" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Date: Fri, 2 Jan 2004 22:57:28 -
Subject: [Asterisk-Users] Newbie - getting two local  phones to
communicate
would be a good start :)
Reply-To: [EMAIL PROTECTED]

Hi

This is hard work :) I have read the Asterisk Handbook, BudgeTone User
Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource
Pages
and more.

I am not a linux newbie but am new to Asterisk. I have failed to find
any
docs that explain how to get a very very simple, minimal, system up and
I am
trying to get the following to work:

2 BudgePhone 102D connected on a LAN to a linux RH9 server running
Asterisk
IP 10.0.1.198 - I just want to be able to dial from one phone and talk
to
the other :) I have another phone connected to FWD sucesfully and the
LAN is
NATed at the PC that is acting as the Asteriski server and firewall. But
for
now its just two phones on a LAN - I'll conquer FWD and IAX later

The extensions are 5702 and 5703. I can "dial" direct from one phone to
the
other (not using Asterisk) and the other one rings and answers fine with
a
voice path.

When I dial "5703" from 5702 (thus via asterisk), 5703 rings. If I take
it
off hook it stops ringing but I can still hear ringing on 5702. After a
few
seconds I get the "rapid-beep" tone on both phones. No voice.

I get this from asterisk CLI

*CLI>
*CLI>
-- Executing Macro("SIP/5702-a5be", "exten-aa|5703") in new stack
-- Executing SetMusicOnHold("SIP/5702-a5be", "random") in new stack
-- Executing Macro("SIP/5702-a5be", "dial|20|tr|5703") in new stack
-- Executing AGI("SIP/5702-a5be", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  dialparties.agi: Caller ID name is 'John workroom #1' number is '5702'
--  dialparties.agi: Added extension 5703 to extension map
--  dialparties.agi: Extension 5703 cf is disabled
--  dialparties.agi: Extension 5703 do not disturb is disabled
--  dialparties.agi: DbSet CALLTRACE/5703 to 5702
  dialparties.agi: About to execute Dial(SIP/5703|20|tr)
-- AGI Script Executing Application: (Dial) Options:
(SIP/5703|20|tr)
-- Called 5703
-- SIP/5703-5fdc is ringing
-- SIP/5703-5fdc answered SIP/5702-a5be
-- Attempting native bridge of SIP/5702-a5be and SIP/5703-5fdc
WARNING[81926]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for
seqno
36119 (Response)
  == Spawn extension (macro-dial, s, 1) exited non-zero on
'SIP/5702-a5be'
in macro 'dial'
  == Spawn extension (macro-exten-aa, s, 2) exited non-zero on
'SIP/5702-a5be' in macro 'exten-aa'
  == Spawn extension (johnhome, s, 1) exited non-zero on 'SIP/5702-a5be'

*CLI>
*CLI>

I've turned on SIP debug but can not see any errors reported. This look
like
the moment of failure:

Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.202
From: "John Coll 5702"
;tag=bfbd6f17-1d79-ed6b-1710-239de572455
9
To: ;tag=as3835ce1f
Call-ID: [EMAIL PROTECTED]
CSeq: 28108 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Type: application/sdp
Content-Length: 176

v=0
o=root 27210 27211 IN IP4 10.0.1.198
s=session
c=IN IP4 10.0.1.198
t=0 0
m=audio 18922 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

 to 10.0.1.202:5060
WARNING[81926]: File chan_sip.c,

[Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)

[SW]

Hi John,

If your effort is to make calls between two GS phones via *, here is what
you need.
You need all three devices in the same LAN, so set both phones and * to
10.0.1.98/24.

After that from your asterisk Linux box ping both phones. If that is
successful you know your layer 1,2 and 3 are ok. Disable all fire-walls,
iptables, ipchains in Linux box.

Now in * you need two files in /etc/asterisk

sip.conf and extensions.conf.

rename or delete both those existing files.

Here are the minimum you probably need in these two files.

sip.conf :

[general]
port=5060
allow=all
maxexpirey=180
defaultexpirey=160

[5702]
type=friend
username=5702
context=internal
dtmfmode=info

[5703]
type=friend
username=5703
context=internal
dtmfmode=info


And extensions.conf

[internal]

exten => _57XX,1,Dial(SIP/${EXTEN})


Save both files and issue command reload from * CLI.

Now you should be able to call from one phone to another.

while making calls enable sip debug and study the messages going in and out.
Also if you have ethereal fire that up and capture SIP packets. and see how
the SIP negotiation goes on. This will help you when you start moving to
fwd, IAXTEL etc. etc.

good luck.

SW




From: "John Coll" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Date: Fri, 2 Jan 2004 22:57:28 -
Subject: [Asterisk-Users] Newbie - getting two local  phones to communicate
would be a good start :)
Reply-To: [EMAIL PROTECTED]

Hi

This is hard work :) I have read the Asterisk Handbook, BudgeTone User
Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages
and more.

I am not a linux newbie but am new to Asterisk. I have failed to find any
docs that explain how to get a very very simple, minimal, system up and I am
trying to get the following to work:

2 BudgePhone 102D connected on a LAN to a linux RH9 server running Asterisk
IP 10.0.1.198 - I just want to be able to dial from one phone and talk to
the other :) I have another phone connected to FWD sucesfully and the LAN is
NATed at the PC that is acting as the Asteriski server and firewall. But for
now its just two phones on a LAN - I'll conquer FWD and IAX later

The extensions are 5702 and 5703. I can "dial" direct from one phone to the
other (not using Asterisk) and the other one rings and answers fine with a
voice path.

When I dial "5703" from 5702 (thus via asterisk), 5703 rings. If I take it
off hook it stops ringing but I can still hear ringing on 5702. After a few
seconds I get the "rapid-beep" tone on both phones. No voice.

I get this from asterisk CLI

*CLI>
*CLI>
-- Executing Macro("SIP/5702-a5be", "exten-aa|5703") in new stack
-- Executing SetMusicOnHold("SIP/5702-a5be", "random") in new stack
-- Executing Macro("SIP/5702-a5be", "dial|20|tr|5703") in new stack
-- Executing AGI("SIP/5702-a5be", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  dialparties.agi: Caller ID name is 'John workroom #1' number is '5702'
--  dialparties.agi: Added extension 5703 to extension map
--  dialparties.agi: Extension 5703 cf is disabled
--  dialparties.agi: Extension 5703 do not disturb is disabled
--  dialparties.agi: DbSet CALLTRACE/5703 to 5702
  dialparties.agi: About to execute Dial(SIP/5703|20|tr)
-- AGI Script Executing Application: (Dial) Options: (SIP/5703|20|tr)
-- Called 5703
-- SIP/5703-5fdc is ringing
-- SIP/5703-5fdc answered SIP/5702-a5be
-- Attempting native bridge of SIP/5702-a5be and SIP/5703-5fdc
WARNING[81926]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno
36119 (Response)
  == Spawn extension (macro-dial, s, 1) exited non-zero on 'SIP/5702-a5be'
in macro 'dial'
  == Spawn extension (macro-exten-aa, s, 2) exited non-zero on
'SIP/5702-a5be' in macro 'exten-aa'
  == Spawn extension (johnhome, s, 1) exited non-zero on 'SIP/5702-a5be'

*CLI>
*CLI>

I've turned on SIP debug but can not see any errors reported. This look like
the moment of failure:

Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.202
From: "John Coll 5702"
;tag=bfbd6f17-1d79-ed6b-1710-239de5724559
To: ;tag=as3835ce1f
Call-ID: [EMAIL PROTECTED]
CSeq: 28108 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Type: application/sdp
Content-Length: 176

v=0
o=root 27210 27211 IN IP4 10.0.1.198
s=session
c=IN IP4 10.0.1.198
t=0 0
m=audio 18922 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

 to 10.0.1.202:5060
WARNING[81926]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno
28108 (Response)


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[Asterisk-Users] Re: Cisco SIP license?

[Adthrawn]

Josh,

This is what I've understood it to be so far...

The phone(s) are available in two flavours:

79xx with Call Manager Single User License

79xx without Call Manager Single User License

These optional licenses (which can also be purchase separately, and are 
approx £10/$15) are to upgrade the number of users on the Cisco Call 
Manager Platform.

If you think about it, a number of other brands of PBX utilise the 
Cisco 79xx's, and Cisco does sell them with the aim of using them in 
non-Call Manager situations (otherwise, they'd have just stuck to SCCP, 
and never release MGCP or SIP firmware), so a cheaper version without 
the legality of Call Manager Licensing was produced. Not much saving? 
Well, depends how many you buy! And of course, how much more you spent 
for the extra licenses with the Call Manager itself (you don't want to 
double-spend when you're buying a $15,000 PBX).

The firmware is a different issue. To legally get the firmware, you 
must have a support contract. Where I am hazey, is whether any support 
contract between you and Cisco that gives you access to the TAC counts, 
or whether you need a TAC contract for every device you plan to load 
the firmware onto. Either way, with Cisco, if you can't get something 
without paying for it, then you can count on it being illegal if you 
don't!

I'll know on Monday, when I speak to my Cisco Rep next.

Best,
Ad.
On 3 Jan 2004, at 2:02 am, [EMAIL PROTECTED] 
wrote:

Message: 4
From: "Josh Edwards" <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Date: Sat, 03 Jan 2004 01:48:38 +
Subject: [Asterisk-Users] Cisco SIP license?
Reply-To: [EMAIL PROTECTED]
In order to use a cisco phone and the SIP image, do you need a 
license, or
just the firmware.

Is this like saying that you can get music, all you need is something 
like
kazza?  Or if you get the phone and the image are you legit?

What does it take to get the lic?

Josh

_
Get reliable dial-up Internet access now with our limited-time 
introductory
offer.  http://join.msn.com/?page=dept/dialup
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[Asterisk-Users] Asterisk Gotoif / last called

[vocalvoip]

Hi guys

Ive been trying to get this to work for ages now, basicaly im trying to do if 
${woteva} = "" (nothing), or its none existenant then do label 1, else label 2. for my 
last called function, so it will play a different message if theres no last call in 
the system or it was anonymous.
ive tried 

exten => 1000,1,GotoIf($[${last-call${CALLERIDNUM}} = ""]?4:5)
and heaps of other combinations, most times it ses syntax error when im in asterisk 
verbose more when it loads the gotoif. I know the variable ${last-call${CALLERIDNUM}} 
as i tested it before it loads the gotoif line. Does anyone know howto do this ??
Also say in my example above, its true so it goes to 4, does that mean it goes to 
extension 4.. or rule 4 in that dialplan ? 

thanks heaps for all your help :)

p.s i dunno if asterisk comes with a last caller function, and theres probs heaps 
better ways to do it, but if u wana do it all i did was 



;line 2 and 3, makes a variable last-call2001 = whoever called, and sets the time.

exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,SetGlobalVar(last-call${EXTEN}=${CALLERIDNUM})
exten => 2001,3,SetGlobalVar(2001date=${DATETIME})
exten => 2001,4,Voicemail(u2001)
exten => 2001,104,Voicemail(b2001)
exten => 2001,105,Hangup

; say your last call ; gets the callerid and then plays last-call'callerid'

exten => 128,1,Answer
exten => 128,2,Wait(1)
exten => 128,3,Playback(lastmisscall)
exten => 128,4,SayDigits(${last-call${CALLERIDNUM}})
exten => 128,5,Playback(lastcallat)
exten => 128,6,Datetime(${2001date})
exten => 128,7,Playback(thankyoucome)
exten => 128,8,Wait(1)
exten => 128,9,Hangup


oh while im asking.. is there a way to make global changes to a context.. cause ive 
got a dialplan for each user in a context obviously, but instead of having to put 
these 2 lines
exten => 2001,2,SetGlobalVar(last-call${EXTEN}=${CALLERIDNUM})
exten => 2001,3,SetGlobalVar(2001date=${DATETIME})
in each user dialplan could i just put it once and make it load everytime someone in 
that context is called ?

thanks again heaps :)
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Re: [Asterisk-Users] Call recording/SIP not loggin IN

[Chandra]

My sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0
disallow=all; Disallow all codecs
allow=ulaw  ; Allow codecs in order of preference

dtmfmode=rfc2833

[xlite1]
type=user
host=dynamic
secret=xlite1
context=outgoing
reinvite=no
canreinvite=no
qualify=60

[xlite1]
type=peer
host=dynamic
secret=xlite1
reinvite=no
canreinvite=no
qualify=60

In xlite i have User=xlite1, Pwd=xlite1 and SIP Proxy=IP of my * box, Out
bound Proxy= IP of my * box

netstat -na gives

[EMAIL PROTECTED] root]# netstat -na
Active Internet connections (servers and established)
Proto Recv-Q Send-Q Local Address   Foreign Address State
tcp0  0 0.0.0.0:32768   0.0.0.0:*   LISTEN
tcp0  0 0.0.0.0:22305   0.0.0.0:*   LISTEN
tcp0  0 0.0.0.0:22273   0.0.0.0:*   LISTEN
tcp0  0 127.0.0.1:32769 0.0.0.0:*   LISTEN
tcp0  0 0.0.0.0:33060.0.0.0:*   LISTEN
tcp0  0 0.0.0.0:111 0.0.0.0:*   LISTEN
tcp0  0 0.0.0.0:20000.0.0.0:*   LISTEN
tcp0  0 0.0.0.0:56800.0.0.0:*   LISTEN
tcp0  0 0.0.0.0:80  0.0.0.0:*   LISTEN
tcp0  0 0.0.0.0:22321   0.0.0.0:*   LISTEN
tcp0  0 0.0.0.0:22289   0.0.0.0:*   LISTEN
tcp0  0 0.0.0.0:21  0.0.0.0:*   LISTEN
tcp0  0 0.0.0.0:22  0.0.0.0:*   LISTEN
tcp0  0 0.0.0.0:23  0.0.0.0:*   LISTEN
tcp0  0 0.0.0.0:443 0.0.0.0:*   LISTEN
tcp0128 202.51.xx.xx1:22202.51.xx.xx0:3148
ESTABLISHED
udp0  0 0.0.0.0:32769   0.0.0.0:*
udp0  0 0.0.0.0:50360.0.0.0:*
udp0  0 0.0.0.0:50600.0.0.0:*
udp0  0 0.0.0.0:45690.0.0.0:*
udp0  0 0.0.0.0:111 0.0.0.0:*
udp0  0 0.0.0.0:11770   0.0.0.0:*
udp0  0 0.0.0.0:11771   0.0.0.0:*
udp0  0 0.0.0.0:24270.0.0.0:*
Active UNIX domain sockets (servers and established)
Proto RefCnt Flags   Type   State I-Node Path
unix  2  [ ACC ] STREAM LISTENING 1504   /dev/gpmctl
unix  2  [ ACC ] STREAM LISTENING 1775
/tmp/.font-unix/fs7100
unix  2  [ ACC ] STREAM LISTENING 1520
/var/lib/mysql/mysql.sock
unix  2  [ ACC ] STREAM LISTENING 1885
/var/run/asterisk.ctl
unix  2  [ ACC ] STREAM LISTENING 1621
/tmp/.iroha_unix/IROHA
unix  2  [ ACC ] STREAM LISTENING 1593   /tmp/cd_sockV4
unix  2  [ ACC ] STREAM LISTENING 1671   /tmp/kd_sockV4
unix  2  [ ACC ] STREAM LISTENING 1699   /tmp/td_sockV4
unix  2  [ ACC ] STREAM LISTENING 1565   /tmp/jd_sockV4
unix  7  [ ] DGRAM1094   /dev/log
unix  3  [ ] STREAM CONNECTED 1889
/var/lib/mysql/mysql.sock
unix  3  [ ] STREAM CONNECTED 1888
unix  2  [ ] DGRAM1778
unix  2  [ ] DGRAM1645
unix  2  [ ] DGRAM1406
unix  2  [ ] DGRAM1160
unix  2  [ ] DGRAM1110
[EMAIL PROTECTED] root]#


my grandstream is also not registering to *.

- Original Message -
From: "CW_ASN" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, January 02, 2004 9:14 PM
Subject: Re: [Asterisk-Users] Call recording


> - Original Message -
> From: "Chandra" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, January 02, 2004 9:30 AM
> Subject: Re: [Asterisk-Users] Call recording
>
>
> > xlite saying login timed out. contact network admin.
> >
> > how to get rid of this. * is not behind NAT.
> >
> > DIAX works fine
> >
>
> Could you especify a bit more?
> Send sip.conf, 'netstat -na' from you linux box, xlite config, etc...
>
> Regards,
>
> Gus
>
>
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>
>


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Re: [Asterisk-Users] mini-ITX suggestions

[Patrick Cantwell]

On Fri, 2 Jan 2004, Steven Critchfield wrote:

> On Fri, 2004-01-02 at 12:34, Gary Gapinski wrote:
> > Does anyone have recommendations for (or against) mini-ITX platforms to
> > be used with Wildcard X100P and TDM400P cards?
> >
> > I am considering the use of systems using VIA EPIA CL and Epia M as
> > small, quiet platforms on which to host Asterisk.
>
> It has been covered here before that those are i586 level chips, so be
> prepared for the actual horsepower you will get out of them. If you
> don't do much VoIP, it will probably be fine. I think there is even a
> person or 2 here already using them as thats why we know about the i586
> problem.


What about the newer chip (the Nehemiah)? I have an 800mhz version of the
older system/core that's clearly i586 architecture doing some PVR stuff (I
have a tuner card with mpeg encoding builtin, so the lack of horsepower is
a non-issue), however I *believe* the newer Nehemiah core is i686
compatable.  I do know they've added hardware RNG, SSE/SIMD extensions,
and a full speed floating point unit to the processor. The motherboard
also has an updated north/south bridge, utilizing PC2100/DDR266 RAM, comes
with USB2.0 and firewire integrated, and moves from 1 ATA100 to 2 ATA133
channels.  It'd sure be a waste to couple all of that nice hardware to an
antiquated i586 chip. (and the best is a board+chip+case can be
accomplished for around $200USD)
Can anyone confirm if they've added what was necessary to bring the chip
up to the i686 level?

A brief overview of the new Nehemiah features can be seen at
http://www.mini-itx.com/reviews/nehemiah/?page=3#s05

I'll have one of these machines in hand probably early next week, so maybe
I can do some testing before I move it into the role of my new PVR (those
damn trans-encodes from mpeg to streamable formats take too darn long on
the 800! :)

-Pat



> --
> Steven Critchfield  <[EMAIL PROTECTED]>
>
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>


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[Asterisk-Users] Newbridge Mainstreet 3624 Manual

[scheesman]

Hi all,

I have posted a copy of the 3624 manual on the web.  It's 11MB and over
650 pages, so not exactly light reading!  You can grab it at
http://www.caeveo.com/files/newbridge3624.pdf.  Please be kind and save
it to your local machine instead of reading it from the web!  Thanks!

Sean
<>

Re: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)

[Rich Adamson]

> This is hard work :) I have read the Asterisk Handbook, BudgeTone User
> Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages
> and more.
> 
> I am not a linux newbie but am new to Asterisk. I have failed to find any
> docs that explain how to get a very very simple, minimal, system up and I am
> trying to get the following to work:

> I've been at this off and on for two weeks    Linux admin and firewalls
> seem trivial compared to this so I must be missing something pretty basic :)

Careful, that's the wrong thing to say on this list; but, the exact same
thing has been reiterated at least several thousand times (minimum) in 
the last few months. The underlying problem truly is that even for those of 
us that have been professionally involved with telephony (for years), the 
initial learning curve for * is far steeper then the average implementor 
can begin to comprehend. 

Please folks, let's not start the _weekly_ read the code/docs war once
again; for the experienced ones that really want to click on reply, 
"please don't"

The bottom line is that unless you can read/comprehend code rather quickly, 
the technical documentation does not exist in any reasonable form. Lots 
of very good people are trying very very hard, but the fact is that far more 
technical doc exists only in the code then one would expect from such an 
excellent application. (The subject really has been covered in very negative 
terms many times, if one can find it. One of the better choices for newbie
research really is http://www.voip-info.org/tiki-index.php , but even this 
is very much a 'work-in-progress'. That's a Good Thing!!! There is also a
fair number of folks on the list that are trying to earn a living via * 
that won't take the time to respond to even the most basic questions for
obvious reasons. Their signatures will become very apparent.)

Not all of the documentation problem is really related to *; there really
is a lot of interpretation/advancement/research going on with SIP vendors
that frequently initiate postings related to problems/comments on the list.
Once you get a basic * system working, you'll find significant issues with
the SIP standards in terms of NAT and many many other items. That's not
putting * down, its just the nature of non-commercial internet standards.
I do believe that most implementors find the /usr/src/asterisk/README.*
to be helpful, and some other directories that contain sample configs
(of which the directory names are so unobvious I can't find them after
a couple of beers. ;)

You will find that not all SIP vendors interpret the exact same standards
in the same way. For those of us that have tried, software/hardware SIP
phones vary dramatically in terms of interoperability with * (and other
telephony apps). Some get it reasonably right, and other vendors try to
advance the standards with their own interpretations. And, a few are 
obviously basement operations with minimal informed staff.

There really are only a few _aggressive_ responders that will abrasively
tell you to read the docs, but what they really mean is read the code. If
that's not appropriate, then simply delete their replies; they really won't
mind even a little tiny bit. It's just the nature of this list.

But, keep the faith, asterisk is really very good and stable once past
that initial vary-steep learning curve.

Rich


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[Asterisk-Users] Cisco SIP license?

[Josh Edwards]

In order to use a cisco phone and the SIP image, do you need a license, or 
just the firmware.

Is this like saying that you can get music, all you need is something like 
kazza?  Or if you get the phone and the image are you legit?

What does it take to get the lic?

Josh

_
Get reliable dial-up Internet access now with our limited-time introductory 
offer.  http://join.msn.com/?page=dept/dialup

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[Asterisk-Users] echo

[Sean Adams]

Read the faq, checked the config files... can't find anything about an 
echo problem like this. Here's what I've got:

4 channel t1 card, span 2 going to channel bank with both fxs and fxo 
lines
Polycom IP600 phones on same LAN with asterisk
iax connection to voicepulse (T1 going out on another router)
Asterisk on a 2.4GHz machine

what I hear:

FXO <-> FXSno echo at all
FXO or FXS <-> voicepulse	no echo at all
IP600 <-> voicepulse		no echo at all
IP600 <-> FXO or FXS		echo heard by IP600 caller, no echo heard by 
remote party
IP600 <-> IP600			can't get it to work yet - SIP times out (separate 
issue I guess)

I also did a few more tests:

- I made an extension that just does Wait(). Called it from the ip600. 
No echo.

- Used the built in Echo function. I only hear one echo. I measured the 
latency by recording it with a microphone. It is 100ms, which seems a 
bit excessive for ethernet.

The fact that I hear only one echo when doing the Echo test, and no 
echo anywhere except in the IP600 <-> POTS path, would lead me to 
believe that the source of the echo is within asterisk (not in 
transmission or in the phones), and only when bridging SIP<->POTS. Any 
ideas? 

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RE: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)

[David J Carter]

John,

Try these files.

They work for me.


Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Coll
Sent: 02 January 2004 23:42
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Newbie - getting two local phones to
communicate would be a good start :)


Robert, OK :) I'll go there yet again tomorrow.

Do you know of a complete set of files and instructions that will work with
a most basic no-frills system - a basis that I can easily understand and
build on?  I just pray for the Nutshell Asterisk Handbook!

thanks again

john


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of rnc Info
Lists
Sent: 02 January 2004 23:26
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie - getting two local phones to
communicate would be a good start :)


John wrote:
> Hi
>
> This is hard work :) I have read the Asterisk Handbook, BudgeTone User
> Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource
> Pages
> and more.
>
> I am not a linux newbie but am new to Asterisk. I have failed to find any
> docs that explain how to get a very very simple, minimal, system up and I
> am trying to get the following to work:

< cut  john >


John,
What is the dialparties.agi? You didn't mention that in your description
(or I missed it)

 Used in Zac's system - I've got the perl code and installed - seems
to work without complaint. nice :)

 I have used * with 2 GS phones with no problem.

My suggestion is to go back to the simple extensions.conf file  and try it
again.  Take out all of the fancy stuff until you get the basic phone
working.

If it still doesn't work then post all relevant parts of your
extensions.conf and any changes you made in sip.conf along with the trace.

My GS SIP.conf for one of the phones is:
[2001]
type=friend
username=2001
secret=test2
host=dynamic
context=local-extensions   <--this will probably be different in your setup

Extension.conf for ringing that phone is:
exten => 2001,1,Dial(SIP/2001,20,Ttr)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup

They probably aren't perfect but they do work.

Robert


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;
; Static extension configuration files, used by
; the pbx_config module.
;
; The "General" category is for certain variables.  
;
[general]
;
static=yes
;
writeprotect=no
;
; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1  ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]

;
; Any category other than "General" and "Globals" represent 
; extension contexts, which are collections of extensions.  
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal.  In patterns, some characters have special meanings:
;
;   X - any digit from 0-9
;   N - any digit from 2-9
;   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
;   . - wildcard, matches anything remaining (e.g. _9011. matches anything starting 
with 9011 including 9011)
;
[iaxtel700]
exten => _91700NXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])

[macro-stdexten];
;
; Standard extension macro:
;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
;   ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20)   ; Ring the interface, 
20 seconds maximum
exten => s,2,voicemail2(u${ARG1})   ; If unavailable, send 
to voicemail2 w/ unavail announce
exten => s,3,Goto(default,s,1)  ; If they press #, return to 
start
exten => s,102,voicemail2(b${ARG1}) ; If busy, send to 
voicemail2 w/ busy announce
exten => s,103,Goto(default,s,1); If they press #, 
return to start
;
;
[demo]
;
; Create an extension, 5000, for dialing the
; Asterisk demo.
;
exten => 5000,1,Playback(demo-abouttotry)   ; Let 
them know what's going on
exten => 5000,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])  ; Call the Asterisk 
demo
exten => 5000,3,Playback(remote_unavail)   
 ; Couldn't connect to the demo site
exten => 5000,4,Goto(s,6)   

RE: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)

[John Coll]

Robert, OK :) I'll go there yet again tomorrow.

Do you know of a complete set of files and instructions that will work with
a most basic no-frills system - a basis that I can easily understand and
build on?  I just pray for the Nutshell Asterisk Handbook!

thanks again

john


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of rnc Info
Lists
Sent: 02 January 2004 23:26
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie - getting two local phones to
communicate would be a good start :)


John wrote:
> Hi
>
> This is hard work :) I have read the Asterisk Handbook, BudgeTone User
> Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource
> Pages
> and more.
>
> I am not a linux newbie but am new to Asterisk. I have failed to find any
> docs that explain how to get a very very simple, minimal, system up and I
> am trying to get the following to work:

< cut  john >


John,
What is the dialparties.agi? You didn't mention that in your description
(or I missed it)

 Used in Zac's system - I've got the perl code and installed - seems
to work without complaint. nice :)

 I have used * with 2 GS phones with no problem.

My suggestion is to go back to the simple extensions.conf file  and try it
again.  Take out all of the fancy stuff until you get the basic phone
working.

If it still doesn't work then post all relevant parts of your
extensions.conf and any changes you made in sip.conf along with the trace.

My GS SIP.conf for one of the phones is:
[2001]
type=friend
username=2001
secret=test2
host=dynamic
context=local-extensions   <--this will probably be different in your setup

Extension.conf for ringing that phone is:
exten => 2001,1,Dial(SIP/2001,20,Ttr)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup

They probably aren't perfect but they do work.

Robert


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Re: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)

[rnc Info Lists]

John wrote:
> Hi
>
> This is hard work :) I have read the Asterisk Handbook, BudgeTone User
> Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource
> Pages
> and more.
>
> I am not a linux newbie but am new to Asterisk. I have failed to find any
> docs that explain how to get a very very simple, minimal, system up and I
> am
> trying to get the following to work:
>
> 2 BudgePhone 102D connected on a LAN to a linux RH9 server running
> Asterisk
> IP 10.0.1.198 - I just want to be able to dial from one phone and talk to
> the other :) I have another phone connected to FWD sucesfully and the LAN
> is
> NATed at the PC that is acting as the Asteriski server and firewall. But
> for
> now its just two phones on a LAN - I'll conquer FWD and IAX later
>
> The extensions are 5702 and 5703. I can "dial" direct from one phone to
> the
> other (not using Asterisk) and the other one rings and answers fine with a
> voice path.
>
> When I dial "5703" from 5702 (thus via asterisk), 5703 rings. If I take it
> off hook it stops ringing but I can still hear ringing on 5702. After a
> few
> seconds I get the "rapid-beep" tone on both phones. No voice.
>
> I get this from asterisk CLI
>
> *CLI>
> *CLI>
> -- Executing Macro("SIP/5702-a5be", "exten-aa|5703") in new stack
> -- Executing SetMusicOnHold("SIP/5702-a5be", "random") in new stack
> -- Executing Macro("SIP/5702-a5be", "dial|20|tr|5703") in new stack
> -- Executing AGI("SIP/5702-a5be", "dialparties.agi") in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
>   dialparties.agi: Caller ID name is 'John workroom #1' number is '5702'
> --  dialparties.agi: Added extension 5703 to extension map
> --  dialparties.agi: Extension 5703 cf is disabled
> --  dialparties.agi: Extension 5703 do not disturb is disabled
> --  dialparties.agi: DbSet CALLTRACE/5703 to 5702
>   dialparties.agi: About to execute Dial(SIP/5703|20|tr)
> -- AGI Script Executing Application: (Dial) Options: (SIP/5703|20|tr)
> -- Called 5703
> -- SIP/5703-5fdc is ringing
> -- SIP/5703-5fdc answered SIP/5702-a5be
> -- Attempting native bridge of SIP/5702-a5be and SIP/5703-5fdc
> WARNING[81926]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
> exceeded on call [EMAIL PROTECTED] for seqno
> 36119 (Response)
>   == Spawn extension (macro-dial, s, 1) exited non-zero on 'SIP/5702-a5be'
> in macro 'dial'
>   == Spawn extension (macro-exten-aa, s, 2) exited non-zero on
> 'SIP/5702-a5be' in macro 'exten-aa'
>   == Spawn extension (johnhome, s, 1) exited non-zero on 'SIP/5702-a5be'
>
> *CLI>
> *CLI>



John,
What is the dialparties.agi? You didn't mention that in your description
(or I missed it)  I have used * with 2 GS phones with no problem.

My suggestion is to go back to the simple extensions.conf file  and try it
again.  Take out all of the fancy stuff until you get the basic phone
working.

If it still doesn't work then post all relevant parts of your
extensions.conf and any changes you made in sip.conf along with the trace.

My GS SIP.conf for one of the phones is:
[2001]
type=friend
username=2001
secret=test2
host=dynamic
context=local-extensions   <--this will probably be different in your setup

Extension.conf for ringing that phone is:
exten => 2001,1,Dial(SIP/2001,20,Ttr)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup

They probably aren't perfect but they do work.

Robert


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RE: [Asterisk-Users] Newbridge Mainstreet 3624 T1 channel bank no w Alcatel

[scheesman]

I have a manual for the 3624 in PDF format that I bought.  Anyone have
any opinions on the legitimacy of sharing it?!  I'll be happy to post it
somewhere if the consensus is positive!

Sean

-Original Message-
From: TC [mailto:[EMAIL PROTECTED] 
Sent: Friday, January 02, 2004 4:07 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbridge Mainstreet 3624 T1 channel bank
no w Alcatel


>  Certainly a bit of Googling can lead to partial free documentation 
> (I'd recomment the Newbridge section of
> http://www.at2.com/pages/DocumentLibrary.cfm) and I suspect the
information
> overlaps between models particularly with respect to the console 
> commands, but alas the PDFs that are floating around out there are 
> only the insert/remove updates to the core manuals. I do also have 
> limited pinout data for the 3624 if anyone needs it.
Just for the archives
http://www.at2.com/downloads/Documents/TechManuels/newbridge3600tech.pdf
http://www.at2.com/downloads/Documents/TechManuels/newbridge3624techmanu
el.p
df
but not exactly what we want

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[Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)

[John Coll]

Hi

This is hard work :) I have read the Asterisk Handbook, BudgeTone User
Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages
and more.

I am not a linux newbie but am new to Asterisk. I have failed to find any
docs that explain how to get a very very simple, minimal, system up and I am
trying to get the following to work:

2 BudgePhone 102D connected on a LAN to a linux RH9 server running Asterisk
IP 10.0.1.198 - I just want to be able to dial from one phone and talk to
the other :) I have another phone connected to FWD sucesfully and the LAN is
NATed at the PC that is acting as the Asteriski server and firewall. But for
now its just two phones on a LAN - I'll conquer FWD and IAX later

The extensions are 5702 and 5703. I can "dial" direct from one phone to the
other (not using Asterisk) and the other one rings and answers fine with a
voice path.

When I dial "5703" from 5702 (thus via asterisk), 5703 rings. If I take it
off hook it stops ringing but I can still hear ringing on 5702. After a few
seconds I get the "rapid-beep" tone on both phones. No voice.

I get this from asterisk CLI

*CLI>
*CLI>
-- Executing Macro("SIP/5702-a5be", "exten-aa|5703") in new stack
-- Executing SetMusicOnHold("SIP/5702-a5be", "random") in new stack
-- Executing Macro("SIP/5702-a5be", "dial|20|tr|5703") in new stack
-- Executing AGI("SIP/5702-a5be", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  dialparties.agi: Caller ID name is 'John workroom #1' number is '5702'
--  dialparties.agi: Added extension 5703 to extension map
--  dialparties.agi: Extension 5703 cf is disabled
--  dialparties.agi: Extension 5703 do not disturb is disabled
--  dialparties.agi: DbSet CALLTRACE/5703 to 5702
  dialparties.agi: About to execute Dial(SIP/5703|20|tr)
-- AGI Script Executing Application: (Dial) Options: (SIP/5703|20|tr)
-- Called 5703
-- SIP/5703-5fdc is ringing
-- SIP/5703-5fdc answered SIP/5702-a5be
-- Attempting native bridge of SIP/5702-a5be and SIP/5703-5fdc
WARNING[81926]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno
36119 (Response)
  == Spawn extension (macro-dial, s, 1) exited non-zero on 'SIP/5702-a5be'
in macro 'dial'
  == Spawn extension (macro-exten-aa, s, 2) exited non-zero on
'SIP/5702-a5be' in macro 'exten-aa'
  == Spawn extension (johnhome, s, 1) exited non-zero on 'SIP/5702-a5be'

*CLI>
*CLI>

I've turned on SIP debug but can not see any errors reported. This look like
the moment of failure:

Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.202
From: "John Coll 5702"
;tag=bfbd6f17-1d79-ed6b-1710-239de5724559
To: ;tag=as3835ce1f
Call-ID: [EMAIL PROTECTED]
CSeq: 28108 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Type: application/sdp
Content-Length: 176

v=0
o=root 27210 27211 IN IP4 10.0.1.198
s=session
c=IN IP4 10.0.1.198
t=0 0
m=audio 18922 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

 to 10.0.1.202:5060
WARNING[81926]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno
28108 (Response)




The Grandstream phones are configured like this:

Login password  xxx
MAC 00.0B.82.00.4B.57
IP  10.0.1.202
Subnet  255.255.255.0
Default router  10.0.1.198
DNS server #1   10.0.1.198
DNS Server #2   158.152.1.43

SIP Server: 10.0.1.198
Outbound Proxy:
SIP User ID:5702
Authenticate ID:5702
Authenticate Password:  xxx  (same if this is set to an empty string)
Name:   John Coll 5703
TimezoneGMT
SIP User ID is
phone number:   yes

And sip.conf contains this

[general]
port = 5060
bindaddr = 0.0.0.0
externip = 10.0.1.198

[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0  ; Address to bind to
externip = 10.0.1.198   ; Addres

[5702]
type=friend
host=dynamic
context=johnhome
reinvite=no
canreinvite=no
qualify=300
callerid="John workroom #1" <5702>
mailbox=5702
nat=no

[5703] is similar



extensions.conf is currently slightly modified verson of Zac Sprackett's
file http://sprackett.com/asterisk/ - its a bit long so I won't paste yet.
However I have had the same result with a much simpler extensions.conf -
some days ago.

 Any help would really be appreciated as I am stuck and finding the process
hard because I can't seem to find a basic introduction aimed at getting me
up and running with the most basic of systems. Perhaps you can point me to a
BASIC and minimal set of configuration files for example for a SIP phone or
two on a NAT LAN with an X100P plugged into PSTN. I guess that is where most
people start - or should I start somewhere else?

I've been at this off and on for two weeks    Linux admin and firewalls
seem trivial compared to this so I must be missing something pretty basic :)

thanks

john


___

Re: [Asterisk-Users] hangup detection

[Sean Adams]

Okay, I'm an idiot. The tones are picked up just fine by asterisk with 
no changes.

It helps if you understand the syntax of zapata.conf. I thought 
busydetect=yes just had to be under the context line. I didn't realize 
how the "channels=" is actually the delimiter that includes the stuff 
above it (I had busydetect below that line).

I should add that I find the asterisk config files to be very whacky in 
general.

On Jan 2, 2004, at 12:34 PM, Martin Pycko wrote:

If the on/off times are diffrent you need to edit Makefile and 
uncomment
BUSYDETECT_TONES_ONLY flag or something like that ... and then you can
change the MAX/MIN values in dsp.c too. That should help you with
busycount=10 and busydetect=yes

regards
Martin
On Fri, 2 Jan 2004, Sean Adams wrote:

Here's a recording:

http://www.seanadams.com/hangup_tones.aif

(sorry - recorded from speakerphone - skip to the end)

The following numbers are not real precise, I just got this from
visually looking at the spectrum on my computer:
The tones appear to consist of 2600, 2440, 2000, and 1400 Hz.

The timing is 120ms on, 80ms off.

I'll take a look at dsp.c and see if I can make it work. Thanks for 
the
pointers.



On Jan 2, 2004, at 10:46 AM, Martin Pycko wrote:

busydetect should help you. Set busycount=10 busydetect=yes in
zapata.conf
and measure the length of the tone .. should be equal the pause too.
Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example 
like
this: your result - 100, your result + 100 [ms]

regards
Martin
On Fri, 2 Jan 2004, Sean Adams wrote:

So I made the mistake of buying a Carrier Access channel bank 
without
noticing the page on the wiki about the fact that they don't support
disconnect supervision (bastards!). However, apart from that, I do
have
it working fine for incoming calls.

Is there some trick to get asterisk to detect the hangup tones from
SBC? I've tried busydetect and callprogress as suggested, but 
neither
seems to work.  The tone is not a busy tone, but that ear-piercing
high
pitched buzzer. It goes "if you'd like to make a call, please hang 
up
and try again. If you need help, hang up and then dial your 
operator.
BEEP BEEP BEEP etc."

I am set up here with recording gear and spectrum analyzer software,
so
I can identify the tones and timing if necessary. However I'm not 
sure
how to make asterisk detect the tones, or if this work has already
been
done. Anyone know?

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Re: [Asterisk-Users] one way choppy sound problem !

[Andres]

On Friday 02 January 2004 13:46, Nicolas Gudino wrote:
> Hi Steven,
>
> On Fri, 2004-01-02 at 14:55, Steven Critchfield wrote:
> > What is the ping times between your 2 asterisk servers? In the archive I
> > have documented before that IAX jitter buffer sometimes has problems on
> > short ping time links. At the time we where on a private T1 with 4ms
> > ping times. We re enabled our jitter buffer now that we are on a DSL
> > connection and our ping time is between 56 and 70 ms.
>
> The ping time is about 35 ms, one server is on ADSL and the other a T1.
> I tried with different jitter buffer settings, but I really don't know
> how to tune them.  I also tried disabling jitter buffers. I even tried
> using a sip call directly, without using IAX2 (so no jitter buffers
> apply, at least no iax jitter buffers), always with the same result:
> choppy sound from sip to pstn and perfect sound from pstn to sip. Using
> alaw or ulaw the choppiness is tolerable, with other codecs is prety
> bad. Are there any documents on how to tune jitter buffers? Thanks!

Are your "rxgain"  and "txgain" values different than zero in zapata.conf?  If 
so then repeat your calls setting them to "0" and see if it helps. 
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[Asterisk-Users] Grandstream Flash Button

[Stephen R. Besch]

I don't know how I managed to mess up sending this last time, but 
somehow it got attached to the AgentCallbackLogin thread. Since the 
indended audience may not see it there, please indulge me by tolerating 
this second copy:

Here's a little tidbit about the non-functional flash key on the 
Budgetone 100's.  I have 20 of these phones. On some, the flash key 
works, and on some it does not. Since the problem is utterly independent 
of the firmware revision, I suspected that it was hardware based.  So, 
in the interest of scientific inquiry, I took one of the non-functional 
ones apart to see what I could see.

The results were very interesting. During the original assembly of the 
phone, several parts, mostly wires, were drizzled with some kind of 
semi-elastic adhesive, no doubt with the intention of improving the 
physical shock resistance of the phone. The problem is that the 
drizzler, whoever he/she may have been, was exceedingly sloppy.  The 
result is that the adhesive was drizzled right over the contact area on 
the keypad circuit board, preventing contact from being made when the 
key is pressed.  There were actually several such keys on the phone I 
examined, one of which was the flash key.

So, I took some acetone, which dissolves the goo nicely, and a cotton 
swab (several actually) and cleaned the stuff away.  While I was at it, 
I cleaned the contact area of all the keys.  The result is that the 
flash key now works on that phone.  Now if you decide to see if this is 
the issue with your own copy of the phone, be careful to not spill the 
acetone on the case since it will probably dissolve it as well - I 
completely removed the whole mess from the case. Also, take extra care 
to not leave any fingerprints on the contact areas since acids from the 
fingers are known to be associated with long term degradation of circuit 
board contacts.  And, finally, be careful when replacing the screws 
holding the circuitboard in the case. They are very easily stripped. I 
know this because I had to repair one of them with a dab of glue - much 
more carefully placed, I might add.

Simple? Yes.  Irritating? Yes. Poor quality control? Well that's real 
understatement. Are you listening Grandstream?

Stephen R. Besch

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[Asterisk-Users] Grandstream Flash Button

[Stephen R. Besch]

Here's a little tidbit about the non-functional flash key on the 
Budgetone 100's.  I have 20 of these phones. On some, the flash key 
works, and on some it does not. Since the problem is utterly independent 
of the firmware revision, I suspected that it was hardware based.  So, 
in the interest of scientific inquiry, I took one of the non-functional 
ones apart to see what I could see.

The results were very interesting. During the original assembly of the 
phone, several parts, mostly wires, were drizzled with some kind of 
semi-elastic adhesive, no doubt with the intention of improving the 
physical shock resistance of the phone. The problem is that the 
drizzler, whoever he/she may have been, was exceedingly sloppy.  The 
result is that the adhesive was drizzled right over the contact area on 
the keypad circuit board, preventing contact from being made when the 
key is pressed.  There were actually several such keys on the phone I 
examined, one of which was the flash key.

So, I took some acetone, which dissolves the goo nicely, and a cotton 
swab (several actually) and cleaned the stuff away.  While I was at it, 
I cleaned the contact area of all the keys.  The result is that the 
flash key now works on that phone.  Now if you decide to see if this is 
the issue with your own copy of the phone, be careful to not spill the 
acetone on the case since it will probably dissolve it as well - I 
completely removed the whole mess from the case. Also, take extra care 
to not leave any fingerprints on the contact areas since acids from the 
fingers are known to be associated with long term degradation of circuit 
board contacts.  And, finally, be careful when replacing the screws 
holding the circuitboard in the case. They are very easily stripped. I 
know this because I had to repair one of them with a dab of glue - much 
more carefully placed, I might add.

Simple? Yes.  Irritating? Yes. Poor quality control? Well that's real 
understatement. Are you listening Grandstream?

Stephen R. Besch

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Re: [Asterisk-Users] Newbridge Mainstreet 3624 T1 channel bank no w Alcatel

[TC]

>  Certainly a bit of Googling can lead to partial free documentation (I'd
> recomment the Newbridge section of
> http://www.at2.com/pages/DocumentLibrary.cfm) and I suspect the
information
> overlaps between models particularly with respect to the console commands,
> but alas the PDFs that are floating around out there are only the
> insert/remove updates to the core manuals. I do also have limited pinout
> data for the 3624 if anyone needs it.
Just for the archives
http://www.at2.com/downloads/Documents/TechManuels/newbridge3600tech.pdf
http://www.at2.com/downloads/Documents/TechManuels/newbridge3624techmanuel.p
df
but not exactly what we want

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Re: [Asterisk-Users] AgentCallbackLogin.

[Philipp von Klitzing]

Hi!

> I'm using the AgentCallbackLogin function to log my agents onto multiple 
> call queues.
> 
> exten => 3001,1, AgenCallbackLogin(1001,@sip). This works very well.
> 
> I can not work out how to log them back out? On of the forum members was 
> kind enough to point me into the directions of 'dial a null extension and 
> press * to logout'.
> 
> I don't seem to be able to translate this into Syntax.

Mr. Wiki can:
http://www.voip-info.org/wiki-Asterisk+cmd+AgentCallbackLogin

Philipp


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[Asterisk-Users] AgentCallbackLogin.

[Shad Mortazavi]

Dear Forum,

 

I'm using the AgentCallbackLogin function to log my
agents onto multiple call queues.

 

exten =>
3001,1, AgenCallbackLogin(1001,@sip). This works very well.

 

I can not work out how to log them back out? On of the forum
members was kind enough to point me into the directions of 'dial a null extension
and press * to logout'.

 

I don't seem to be able to translate this into Syntax.


Can some one help?


Warm Regards

 

---

Shad Mortazavi

US Technical Manager

Nexus Management

 

Re: [Asterisk-Users] hangup detection

[Martin Pycko]

If the on/off times are diffrent you need to edit Makefile and uncomment
BUSYDETECT_TONES_ONLY flag or something like that ... and then you can
change the MAX/MIN values in dsp.c too. That should help you with
busycount=10 and busydetect=yes

regards
Martin

On Fri, 2 Jan 2004, Sean Adams wrote:

>
> Here's a recording:
>
> http://www.seanadams.com/hangup_tones.aif
>
> (sorry - recorded from speakerphone - skip to the end)
>
> The following numbers are not real precise, I just got this from
> visually looking at the spectrum on my computer:
>
> The tones appear to consist of 2600, 2440, 2000, and 1400 Hz.
>
> The timing is 120ms on, 80ms off.
>
> I'll take a look at dsp.c and see if I can make it work. Thanks for the
> pointers.
>
>
>
> On Jan 2, 2004, at 10:46 AM, Martin Pycko wrote:
>
> > busydetect should help you. Set busycount=10 busydetect=yes in
> > zapata.conf
> > and measure the length of the tone .. should be equal the pause too.
> >
> > Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like
> > this: your result - 100, your result + 100 [ms]
> >
> > regards
> > Martin
> >
> > On Fri, 2 Jan 2004, Sean Adams wrote:
> >
> >>
> >> So I made the mistake of buying a Carrier Access channel bank without
> >> noticing the page on the wiki about the fact that they don't support
> >> disconnect supervision (bastards!). However, apart from that, I do
> >> have
> >> it working fine for incoming calls.
> >>
> >> Is there some trick to get asterisk to detect the hangup tones from
> >> SBC? I've tried busydetect and callprogress as suggested, but neither
> >> seems to work.  The tone is not a busy tone, but that ear-piercing
> >> high
> >> pitched buzzer. It goes "if you'd like to make a call, please hang up
> >> and try again. If you need help, hang up and then dial your operator.
> >> BEEP BEEP BEEP etc."
> >>
> >> I am set up here with recording gear and spectrum analyzer software,
> >> so
> >> I can identify the tones and timing if necessary. However I'm not sure
> >> how to make asterisk detect the tones, or if this work has already
> >> been
> >> done. Anyone know?
> >>
> >> ___
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> >>
> >
> > ___
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> >
>
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[Asterisk-Users] Re: Video

[Matt Lawson]

Linphone and H.263 over SIP and IAX2

On Thu, 1 Jan 2004, Olle E. Johansson wrote:


[EMAIL PROTECTED] wrote:
 

> What is the best software video client that is compatible with * ?
   

To add a follow-up?
Which channels support video and how?
I know that there's support in H.323 and SIP. Anything else?
 

I



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Re: [Asterisk-Users] hangup detection

[Steven Critchfield]

On Fri, 2004-01-02 at 13:02, Sean Adams wrote:
> >
> > Are the tones increasing in pitch?
> 
> No, the beeps are the same pitch - sounds like it was deliberately  
> designed to be a loud and awful sounding as possible through an  
> off-hook phone, to get your attention to go hang it up. My ears tell me  
> it's roughly 250ms on, 250ms off and so on.

Okay.

> Taking my first peek at the code now...

Always a good thing.

> > BTW, which CAC channel bank did you buy? The ADIT 600 should do
> > disconnect supervision, and I thought the AB1 did too.
> 
> It's the AB1 with 8 fxo, 16xfs. Here's the page I was talking about:
> 
> http://www.voip-info.org/wiki-Asterisk+hardware
> 
> Also, others have reported this problem but I can't find a resolution:
> 
> http://www.mail-archive.com/[EMAIL PROTECTED]/msg18626.html
> 
> > Are you also sure
> > you have that on your line so as to be detected? Your other option  
> > might
> > be to switch to groundstart lines which detect hangup much easier. May
> > be difficult to get unless you are a business though.
> 
> I just have regular business lines without any special provisioning. I  
> don't understand why a $20 answering machine can do this but an  
> expensive channel bank can't. :(

The difference is acceptable failure. If your $20 answering machine
fails by hanging up early, they only one really annoyed is the person
leaving a message and they will think they hit a record length limit
unless it was pretty short. If you are placing a call though the machine
and it thought the other side hung up and so it disconnected your
conversation, you would consider that unacceptable.

The other part is that disconnect supervision is something that
basically breaks the loop long enough, or reverse polarity for a moment
to let the other side disconnect. Think about how a relay would work,
reverse polarity or disconnect battery and it will disconnect the
points. Now days, that type of technology is rarely used, and therefore
not implemented unless asked for. 

It is highly probably that you don't have disconnect supervision on your
phone line. You should be able to hook up your test equipment and see
it. I think it has been discussed here before about using a phone that
takes power from the line to light up, if it blinks when the other side
hangs up, you have disconnect supervision. Otherwise, it will always be
a problem detecting hangup without waiting for those tones and matching
on them. 
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] hangup detection

[Sean Adams]

Not having any luck with just tweaking those values. I'm a bit confused 
still as to how the different busy detection choices are supposed to 
work - I've uncommented a few of the #if 0 to see if it's doing 
anything, and I can't see any indiciation that it is. Don't the 
specific off-hook tones need to be in dsp.c, or is it intended that 
asterisk should match the signal just by the timing?

Here's some information I found which confirms the tones I measured:

http://www.hackfaq.org/telephony-27.shtml

--
Receiver Off-Hook Tone
This tone is used to cause off-hook customers to replace the receiver 
on-hook on a permanent signal call and to signal a non-PBX off-hook 
line when ringing key is operated by a switchboard operator.

Receiver Off-Hook Tone is 1400 Hz, 2060 Hz, 2450 Hz and 2600 Hz at 0 
dBm0/frequency on and off every .1 second. On some older space division 
switching systems Receiver Off-Hook was 1400 Hz, 2060 Hz, 2450 Hz and 
2600 Hz at +5 VU on and off every .1 second. On a No. 5 ESS this 
continues for 30 seconds. On a No. 2/2B ESS this continues for 40 
seconds. On some other AT&T switches there are two iterations of 50 
seconds each.
-



On Jan 2, 2004, at 10:46 AM, Martin Pycko wrote:

busydetect should help you. Set busycount=10 busydetect=yes in 
zapata.conf
and measure the length of the tone .. should be equal the pause too.

Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like
this: your result - 100, your result + 100 [ms]
regards
Martin
On Fri, 2 Jan 2004, Sean Adams wrote:

So I made the mistake of buying a Carrier Access channel bank without
noticing the page on the wiki about the fact that they don't support
disconnect supervision (bastards!). However, apart from that, I do 
have
it working fine for incoming calls.

Is there some trick to get asterisk to detect the hangup tones from
SBC? I've tried busydetect and callprogress as suggested, but neither
seems to work.  The tone is not a busy tone, but that ear-piercing 
high
pitched buzzer. It goes "if you'd like to make a call, please hang up
and try again. If you need help, hang up and then dial your operator.
BEEP BEEP BEEP etc."

I am set up here with recording gear and spectrum analyzer software, 
so
I can identify the tones and timing if necessary. However I'm not sure
how to make asterisk detect the tones, or if this work has already 
been
done. Anyone know?

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RE: [Asterisk-Users] Newbridge Mainstreet 3624 T1 channel bank no w Alcatel

[Kris Boutilier]

 In the 'me too' vein, I also have an untested 3624 here on the shelf and am
waiting on a shipment of T100 cards to play with.

 Documentation is very hard to come by. Alcatel are certainly the owners of
the Mainstreet product line but, without a support contract, any
documentation they may have is essentially unavailable as their per-incident
fees for support cost more than most of the competitors entire channel
banks! The support dispatcher I spoke with was very sympathetic but unable
to help.

 There is an eBook reseller online who purports to have the manual
(http://www.telemanuals.com/catalog/default.php?cPath=40) however I am
having "difficulties" obtaining it from them at the moment. There are also
resellers who list paper documentation for these units, but in the us$250
range.

 Certainly a bit of Googling can lead to partial free documentation (I'd
recomment the Newbridge section of
http://www.at2.com/pages/DocumentLibrary.cfm) and I suspect the information
overlaps between models particularly with respect to the console commands,
but alas the PDFs that are floating around out there are only the
insert/remove updates to the core manuals. I do also have limited pinout
data for the 3624 if anyone needs it.

 If anyone out there familiar with these units would be willing to go
through some orientation it would be very appreciated - they certainly seem
to be a unit of choice for people trying to do larger proof of concept
trials with Asterisk...

k.


-Original Message-
From: Jeff Roberts [mailto:[EMAIL PROTECTED]
Sent: January 2, 2004 11:39 AM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbridge Mainstreet 3624 T1 channel bank
now Alcatel


On Thu, 01 Jan 2004 14:54:45 -0500
Lance Arbuckle <[EMAIL PROTECTED]> wrote:

> 
> 
> TC wrote:
> > 
> > Hi
> > I just came accross this
> > Newbridge Mainstreet 3624 but the Alctel site  appears to have zip for
> > reference/user manuals
> > Anyone by chance have 1 of these or a url for the docs ?
> 
> Anyone know how to reset the passwords on the 3624 ?
> 
> Also, is the 3624 suitable for use with the T100P and Asterisk.  I was
> considering one of these for my first asterisk channel bank since they
> show up on Ebay regularly and fiarly cheap.
> 
> -- 
>   .~.Triad Internet Systems, Inc.
>   /V\Lance C. Arbuckle
>  // \\   3315 Anderson Drive
> /(   )\  Winston-Salem, NC 27127
>  ^'~'^   336-771-2090
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> 
I have one of these as well, its been sitting on a shelf till I get around
to messing with it.  I wasnt able to come up with any docs that would do me
any good either.  If anyone has had success with it and * I'd be interested
to hear about it.
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Re: [Asterisk-Users] Newbridge Mainstreet 3624 T1 channel bank now Alcatel

[Jeff Roberts]

On Thu, 01 Jan 2004 14:54:45 -0500
Lance Arbuckle <[EMAIL PROTECTED]> wrote:

> 
> 
> TC wrote:
> > 
> > Hi
> > I just came accross this
> > Newbridge Mainstreet 3624 but the Alctel site  appears to have zip for
> > reference/user manuals
> > Anyone by chance have 1 of these or a url for the docs ?
> 
> Anyone know how to reset the passwords on the 3624 ?
> 
> Also, is the 3624 suitable for use with the T100P and Asterisk.  I was
> considering one of these for my first asterisk channel bank since they
> show up on Ebay regularly and fiarly cheap.
> 
> -- 
>   .~.Triad Internet Systems, Inc.
>   /V\Lance C. Arbuckle
>  // \\   3315 Anderson Drive
> /(   )\  Winston-Salem, NC 27127
>  ^'~'^   336-771-2090
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> 
I have one of these as well, its been sitting on a shelf till I get around to messing 
with it.  I wasnt able to come up with any docs that would do me any good either.  If 
anyone has had success with it and * I'd be interested to hear about it.
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RE: [Asterisk-Users] mini-ITX suggestions

[Steven Sokol]

I believe that the newer versions of the EPIA platform (those with the
"Nehemiah" version of the core) are actually i686 compliant.  There is
an article about EPIA & Linux at www.mini-itx.com that covers the
specifics of dealing with the older boards, including a review of
distros that are known to have issues.

I played with an older EPIA (the 900 MHz i586 version) and found it to
be weak in the VoIP department (given to hiccups, etc.) and effectively
useless with the X100P since it only has one attached PCI slot (although
it can be expanded to two using a riser card and some jumpers).

If you do go with a Mini ITX, I would highly recommend using the newer 1
GHz+ Nehemiah boards.

Just my .02

Steve

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steven Critchfield
> Sent: Friday, January 02, 2004 12:47 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] mini-ITX suggestions
> 
> On Fri, 2004-01-02 at 12:34, Gary Gapinski wrote:
> > Does anyone have recommendations for (or against) mini-ITX platforms
to
> > be used with Wildcard X100P and TDM400P cards?
> >
> > I am considering the use of systems using VIA EPIA CL and Epia M as
> > small, quiet platforms on which to host Asterisk.
> 
> It has been covered here before that those are i586 level chips, so be
> prepared for the actual horsepower you will get out of them. If you
> don't do much VoIP, it will probably be fine. I think there is even a
> person or 2 here already using them as thats why we know about the
i586
> problem.
> --
> Steven Critchfield  <[EMAIL PROTECTED]>
> 
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Re: [Asterisk-Users] hangup detection

[Sean Adams]

Here's a recording:

http://www.seanadams.com/hangup_tones.aif

(sorry - recorded from speakerphone - skip to the end)

The following numbers are not real precise, I just got this from 
visually looking at the spectrum on my computer:

The tones appear to consist of 2600, 2440, 2000, and 1400 Hz.

The timing is 120ms on, 80ms off.

I'll take a look at dsp.c and see if I can make it work. Thanks for the 
pointers.



On Jan 2, 2004, at 10:46 AM, Martin Pycko wrote:

busydetect should help you. Set busycount=10 busydetect=yes in 
zapata.conf
and measure the length of the tone .. should be equal the pause too.

Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like
this: your result - 100, your result + 100 [ms]
regards
Martin
On Fri, 2 Jan 2004, Sean Adams wrote:

So I made the mistake of buying a Carrier Access channel bank without
noticing the page on the wiki about the fact that they don't support
disconnect supervision (bastards!). However, apart from that, I do 
have
it working fine for incoming calls.

Is there some trick to get asterisk to detect the hangup tones from
SBC? I've tried busydetect and callprogress as suggested, but neither
seems to work.  The tone is not a busy tone, but that ear-piercing 
high
pitched buzzer. It goes "if you'd like to make a call, please hang up
and try again. If you need help, hang up and then dial your operator.
BEEP BEEP BEEP etc."

I am set up here with recording gear and spectrum analyzer software, 
so
I can identify the tones and timing if necessary. However I'm not sure
how to make asterisk detect the tones, or if this work has already 
been
done. Anyone know?

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Re: [Asterisk-Users] hangup detection

[Sean Adams]

Are the tones increasing in pitch?
No, the beeps are the same pitch - sounds like it was deliberately  
designed to be a loud and awful sounding as possible through an  
off-hook phone, to get your attention to go hang it up. My ears tell me  
it's roughly 250ms on, 250ms off and so on.

Are they the Special Information
Tones (SIT) that are also on the message when you dial a number that  
has
been disconnected?
No, not like that at all. I'll make a recording.

If so, then they are defined somewhere in the code, at least as part of
app_zapateller since that is how it tries to get rid of telemarketers.
You could then see about adding that to the dsp routines to detect the
SIT tones and determine what to do at that time.
Taking my first peek at the code now...

BTW, which CAC channel bank did you buy? The ADIT 600 should do
disconnect supervision, and I thought the AB1 did too.
It's the AB1 with 8 fxo, 16xfs. Here's the page I was talking about:

http://www.voip-info.org/wiki-Asterisk+hardware

Also, others have reported this problem but I can't find a resolution:

http://www.mail-archive.com/[EMAIL PROTECTED]/ 
msg18626.html

Are you also sure
you have that on your line so as to be detected? Your other option  
might
be to switch to groundstart lines which detect hangup much easier. May
be difficult to get unless you are a business though.
I just have regular business lines without any special provisioning. I  
don't understand why a $20 answering machine can do this but an  
expensive channel bank can't. :(

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Re: [Asterisk-Users] one way choppy sound problem !

[Steven Critchfield]

On Fri, 2004-01-02 at 12:46, Nicolas Gudino wrote:
> Hi Steven,
> 
> On Fri, 2004-01-02 at 14:55, Steven Critchfield wrote:
> > What is the ping times between your 2 asterisk servers? In the archive I
> > have documented before that IAX jitter buffer sometimes has problems on
> > short ping time links. At the time we where on a private T1 with 4ms
> > ping times. We re enabled our jitter buffer now that we are on a DSL
> > connection and our ping time is between 56 and 70 ms. 
> 
> The ping time is about 35 ms, one server is on ADSL and the other a T1.
> I tried with different jitter buffer settings, but I really don't know
> how to tune them.  I also tried disabling jitter buffers. I even tried
> using a sip call directly, without using IAX2 (so no jitter buffers
> apply, at least no iax jitter buffers), always with the same result:
> choppy sound from sip to pstn and perfect sound from pstn to sip. Using
> alaw or ulaw the choppiness is tolerable, with other codecs is prety
> bad. Are there any documents on how to tune jitter buffers? Thanks!

Not that I know of. Sorry.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] T400P & E400P second source

[Linus Surguy]

From: "Scott Stingel" <[EMAIL PROTECTED]>


> I understand that there also is a new board from Digium, 
> the TE405P, which is like the 3.3v TE410P, but uses a 5-volt supply.

Any ideas if this is actually shipping yet though? If not when?

Linus


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Re: [Asterisk-Users] mini-ITX suggestions

[Steven Critchfield]

On Fri, 2004-01-02 at 12:34, Gary Gapinski wrote:
> Does anyone have recommendations for (or against) mini-ITX platforms to 
> be used with Wildcard X100P and TDM400P cards?
> 
> I am considering the use of systems using VIA EPIA CL and Epia M as 
> small, quiet platforms on which to host Asterisk.

It has been covered here before that those are i586 level chips, so be
prepared for the actual horsepower you will get out of them. If you
don't do much VoIP, it will probably be fine. I think there is even a
person or 2 here already using them as thats why we know about the i586
problem.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] hangup detection

[Martin Pycko]

busydetect should help you. Set busycount=10 busydetect=yes in zapata.conf
and measure the length of the tone .. should be equal the pause too.

Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like
this: your result - 100, your result + 100 [ms]

regards
Martin

On Fri, 2 Jan 2004, Sean Adams wrote:

>
> So I made the mistake of buying a Carrier Access channel bank without
> noticing the page on the wiki about the fact that they don't support
> disconnect supervision (bastards!). However, apart from that, I do have
> it working fine for incoming calls.
>
> Is there some trick to get asterisk to detect the hangup tones from
> SBC? I've tried busydetect and callprogress as suggested, but neither
> seems to work.  The tone is not a busy tone, but that ear-piercing high
> pitched buzzer. It goes "if you'd like to make a call, please hang up
> and try again. If you need help, hang up and then dial your operator.
> BEEP BEEP BEEP etc."
>
> I am set up here with recording gear and spectrum analyzer software, so
> I can identify the tones and timing if necessary. However I'm not sure
> how to make asterisk detect the tones, or if this work has already been
> done. Anyone know?
>
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Re: [Asterisk-Users] one way choppy sound problem !

[Nicolas Gudino]

Hi Steven,

On Fri, 2004-01-02 at 14:55, Steven Critchfield wrote:
> What is the ping times between your 2 asterisk servers? In the archive I
> have documented before that IAX jitter buffer sometimes has problems on
> short ping time links. At the time we where on a private T1 with 4ms
> ping times. We re enabled our jitter buffer now that we are on a DSL
> connection and our ping time is between 56 and 70 ms. 

The ping time is about 35 ms, one server is on ADSL and the other a T1.
I tried with different jitter buffer settings, but I really don't know
how to tune them.  I also tried disabling jitter buffers. I even tried
using a sip call directly, without using IAX2 (so no jitter buffers
apply, at least no iax jitter buffers), always with the same result:
choppy sound from sip to pstn and perfect sound from pstn to sip. Using
alaw or ulaw the choppiness is tolerable, with other codecs is prety
bad. Are there any documents on how to tune jitter buffers? Thanks!



-- 
Nicolas Gudino <[EMAIL PROTECTED]>
House Internet S.R.L.

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Re: [Asterisk-Users] hangup detection

[Steven Critchfield]

On Fri, 2004-01-02 at 12:25, Sean Adams wrote:
> So I made the mistake of buying a Carrier Access channel bank without 
> noticing the page on the wiki about the fact that they don't support 
> disconnect supervision (bastards!). However, apart from that, I do have 
> it working fine for incoming calls.
> 
> Is there some trick to get asterisk to detect the hangup tones from 
> SBC? I've tried busydetect and callprogress as suggested, but neither 
> seems to work.  The tone is not a busy tone, but that ear-piercing high 
> pitched buzzer. It goes "if you'd like to make a call, please hang up 
> and try again. If you need help, hang up and then dial your operator. 
> BEEP BEEP BEEP etc."
> 
> I am set up here with recording gear and spectrum analyzer software, so 
> I can identify the tones and timing if necessary. However I'm not sure 
> how to make asterisk detect the tones, or if this work has already been 
> done. Anyone know?

Are the tones increasing in pitch? Are they the Special Information
Tones (SIT) that are also on the message when you dial a number that has
been disconnected?

If so, then they are defined somewhere in the code, at least as part of
app_zapateller since that is how it tries to get rid of telemarketers.
You could then see about adding that to the dsp routines to detect the
SIT tones and determine what to do at that time.

BTW, which CAC channel bank did you buy? The ADIT 600 should do
disconnect supervision, and I thought the AB1 did too. Are you also sure
you have that on your line so as to be detected? Your other option might
be to switch to groundstart lines which detect hangup much easier. May
be difficult to get unless you are a business though.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] mini-ITX suggestions

[Gary Gapinski]

Does anyone have recommendations for (or against) mini-ITX platforms to 
be used with Wildcard X100P and TDM400P cards?

I am considering the use of systems using VIA EPIA CL and Epia M as 
small, quiet platforms on which to host Asterisk.

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[Asterisk-Users] hangup detection

[Sean Adams]

So I made the mistake of buying a Carrier Access channel bank without 
noticing the page on the wiki about the fact that they don't support 
disconnect supervision (bastards!). However, apart from that, I do have 
it working fine for incoming calls.

Is there some trick to get asterisk to detect the hangup tones from 
SBC? I've tried busydetect and callprogress as suggested, but neither 
seems to work.  The tone is not a busy tone, but that ear-piercing high 
pitched buzzer. It goes "if you'd like to make a call, please hang up 
and try again. If you need help, hang up and then dial your operator. 
BEEP BEEP BEEP etc."

I am set up here with recording gear and spectrum analyzer software, so 
I can identify the tones and timing if necessary. However I'm not sure 
how to make asterisk detect the tones, or if this work has already been 
done. Anyone know?

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Re: [Asterisk-Users] one way choppy sound problem !

[Steven Critchfield]

On Fri, 2004-01-02 at 11:35, Nicolas Gudino wrote:
> I have a similar problem, with GS phones, X-Lite or Kphone. I tried all
> the codecs with the same result. Choppy sound in the direction SIP-Phone
> -> pstn, but crystal clear sound the other way around. The only
> difference in my case is that I have two asterisks servers connected
> together via IAX2, the PSTN call is received in one asterisk, while the
> sip phones are in the other asterisk. Ex: 
> 
> pstn -> * --iax2--> * ->sip phone (GS, Xlite or Kphone)
> 
> If I use an Xlite in the same asterisk as the pstn line, the sound is
> perfect in both ways. But when I answer the call in the second asterisk,
> the sound from the sip phone to pstn is choppy, with or without silence
> detection, and the sound from pstn to sip phone is perfect.
> 
> The asterisk server with the pstn line is an old pentium 133, maybe
> thats the problem, I will try with a better machine and see how it goes.

What is the ping times between your 2 asterisk servers? In the archive I
have documented before that IAX jitter buffer sometimes has problems on
short ping time links. At the time we where on a private T1 with 4ms
ping times. We re enabled our jitter buffer now that we are on a DSL
connection and our ping time is between 56 and 70 ms. 

> On Fri, 2004-01-02 at 06:23, Dawid Mielnik wrote:
> > Hi all,
> > 
> > I have my asterisk setup as following:
> > 
> > IP   2 x E1
> > x-lite <---> Asterisk ---> PSTN
> > 
> > 
> > When I place a call from x-lite to PSTN, the quality of the sound in the
> > direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user,
> > heard by the PSTN user is choppy and makes communication not very pleasant.
> > The sound is choppy as if bits of data were lost. The strange thing is that
> > the x-lite user hears the PSTN user fine !
> > 
> > In x-lite, I have swithed off sience detection (transmit silence - yes),
> > this has improved the sound quality but did not eliminated the problem. I
> > have fed a countinious sound into the microphone and still got chops in the
> > sound. I have also tried changing the codecs gsm, alaw, ulaw - but I get the
> > same problem with all of them. Maybe the problem lies somewhere in audio
> > buffering settings on x-lite ?
> > 
> > Has anyone ever had this sort of problem and managed to deal with it ? I
> > would greatly appreciate your help !
> > 
> > Best regards,
> > 
> > Dave
> > 
> > 
> > ___
> > Asterisk-Users mailing list
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> > 
> 
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-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] T400P & E400P second source

[Scott Stingel]

Title: Message



I 
understand that there also is a new board from Digium, the TE405P, which is like 
the 3.3v TE410P, but uses a 5-volt supply.
 
Just 
so you have another choice!  
 
regards,
Scott
 
 
Scott M. 
Stingel Emerging Voice Technology Inc.Palo Alto, California and London, England
Email:  [EMAIL PROTECTED]  URL:    www.evtmedia.com  

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of John 
  TernovasSent: Friday, January 02, 2004 5:29 PMTo: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] T400P 
  & E400P second source
  Since i'm sure there are others out there in the same position as me, 
  being disappointed that the original T400P and E400P cards are no longer 
  available from Digium, I thought I would pass on a place I found to get 
  them.  I needed the older card, since I didn't want to have to get a 
  motherboard with 3.3v pci slots on it.
   
  http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3068850018&category=3309&rd=1
   
  This guy has been selling replicas of the X100P for quite some 
  time.  I bought a few of those from him, and was extremely pleased with 
  the results.
   
  After speaking with him on the phone, he says he just wants 10 confirmed 
  orders before making a set of the cards.  I guess it makes sense, since 
  he can't be sure there is demand for the cards.
   
  I've placed my order for 2 of them, and am looking forward to him getting 
  the 10 necessary orders so that I can get my cards.
   
  Please, no flames, I'm just passing on information that might be helpful 
  to some,
  J.T.
  
  
  Do you Yahoo!?Find out what made 
  the Top Yahoo! Searches of 2003 

Re: [Asterisk-Users] T400P & E400P second source

[Steven Critchfield]

On Fri, 2004-01-02 at 11:28, John Ternovas wrote:
> Since i'm sure there are others out there in the same position as me,
> being disappointed that the original T400P and E400P cards are no
> longer available from Digium, I thought I would pass on a place I
> found to get them.  I needed the older card, since I didn't want to
> have to get a motherboard with 3.3v pci slots on it.
>  
> http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3068850018&category=3309&rd=1
>  
> This guy has been selling replicas of the X100P for quite some time. 
> I bought a few of those from him, and was extremely pleased with the
> results.
>  
> After speaking with him on the phone, he says he just wants 10
> confirmed orders before making a set of the cards.  I guess it makes
> sense, since he can't be sure there is demand for the cards.
>  
> I've placed my order for 2 of them, and am looking forward to him
> getting the 10 necessary orders so that I can get my cards.
>  
> Please, no flames, I'm just passing on information that might be
> helpful to some,

Be aware that the cards will probably NOT have FCC certification. You
can be fined for using the cards, and connecting them to the PSTN if
they do not have the required FCC approvals. As can be attested to in
the archive, those in Australia have had to be very cautious until they
had their approvals.  

Also be aware that this does not support Digium, nor our community.

BTW, this user seems like a troll as he only has posted to this list
twice to sell items that Digium has worked on at lower costs. Please do
not feed the trolls. 
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] * crash when forward voicemail --Nicolas Gudino

[Nicolas Gudino]

Well, Eric and James have answered already. Personally, I use redhat
(will upgrade to fedora soon), but using an unmodified kernel.org kernel
compiled from source. Best regards,

On Thu, 2004-01-01 at 15:25, JR Richardson wrote:
> Hey Nicolas,
> 
> That did it.  I ran that export command you suggested, then launched *,
> everything worked fine.  I'm still looking for info on what that command
> actually does.  Can you shed some light please?
> 
> Thanks.
> 
> JR
> 
> 
> Did you try with this line before launching asterisk (with stock redhat
> 9 kernels):
> 
> export LD_ASSUME_KERNEL=2.4.1
> 
> Best regards,

-- 
Nicolas Gudino <[EMAIL PROTECTED]>
House Internet S.R.L.

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Re: [Asterisk-Users] one way choppy sound problem !

[Nicolas Gudino]

I have a similar problem, with GS phones, X-Lite or Kphone. I tried all
the codecs with the same result. Choppy sound in the direction SIP-Phone
-> pstn, but crystal clear sound the other way around. The only
difference in my case is that I have two asterisks servers connected
together via IAX2, the PSTN call is received in one asterisk, while the
sip phones are in the other asterisk. Ex: 

pstn -> * --iax2--> * ->sip phone (GS, Xlite or Kphone)

If I use an Xlite in the same asterisk as the pstn line, the sound is
perfect in both ways. But when I answer the call in the second asterisk,
the sound from the sip phone to pstn is choppy, with or without silence
detection, and the sound from pstn to sip phone is perfect.

The asterisk server with the pstn line is an old pentium 133, maybe
thats the problem, I will try with a better machine and see how it goes.


On Fri, 2004-01-02 at 06:23, Dawid Mielnik wrote:
> Hi all,
> 
> I have my asterisk setup as following:
> 
>   IP   2 x E1
> x-lite <---> Asterisk ---> PSTN
> 
> 
> When I place a call from x-lite to PSTN, the quality of the sound in the
> direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user,
> heard by the PSTN user is choppy and makes communication not very pleasant.
> The sound is choppy as if bits of data were lost. The strange thing is that
> the x-lite user hears the PSTN user fine !
> 
> In x-lite, I have swithed off sience detection (transmit silence - yes),
> this has improved the sound quality but did not eliminated the problem. I
> have fed a countinious sound into the microphone and still got chops in the
> sound. I have also tried changing the codecs gsm, alaw, ulaw - but I get the
> same problem with all of them. Maybe the problem lies somewhere in audio
> buffering settings on x-lite ?
> 
> Has anyone ever had this sort of problem and managed to deal with it ? I
> would greatly appreciate your help !
> 
> Best regards,
> 
> Dave
> 
> 
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Re: [Asterisk-Users] Residential router w/ QoS support?

[Thilo Salmon]

Michael,

I just got mine. Do you recall how you managed to priortize RTP? Or do
you rely on the 'priortized switching port' feature? I tried that, but
perhaps my TOS value does not match the one this router expects. Even
sending a single, large email can kill the voice stream. Leave alone
BitTorrent. Unfortunately, the documentation is far from extensive...

Thilo

On Thu, 2003-12-18 at 17:15, Michael Graves wrote:
> I use a Linksys BEFSR81 which ia an 8 port model with QoS. I paid about
> $90 USD. I had to buy a QoS router when I first installed a Vonage line
> about a year ago. Without it using FTP to d/l loarge files would simply
> kill my calling.
> 
> Michael
> 
> On Wed, 17 Dec 2003 17:01:52 +0100, Thilo Salmon wrote:
> 
> >Did anybody ever come across an affordable, residential cable/dsl router
> >with support for QoS? 
> >
> >The ones I've seen so far (Netgear, D-Link and W-Linx) do not seem to
> >support it. I noticed that even email can damage a G.711 stream on an
> >128kbit uplink, leave alone file-sharing applications. I understand this
> >is strictly related to *, but nevertheless of interest to many of us.
> >
> >Thilo
> >
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> 
> --
> Michael Graves   [EMAIL PROTECTED]
> Sr. Product Specialist  www.pixelpower.com
> Pixel Power Inc.  [EMAIL PROTECTED]
>  FWD 54245
> 
> "Philosophers and plowmen, each must know their part to sow a new
> mentality, closer to the heart." - Geddy Lee, Rush
>  
> ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704
> 
> 
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[Asterisk-Users] T400P & E400P second source

[John Ternovas]

Since i'm sure there are others out there in the same position as me, being disappointed that the original T400P and E400P cards are no longer available from Digium, I thought I would pass on a place I found to get them.  I needed the older card, since I didn't want to have to get a motherboard with 3.3v pci slots on it.
 
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3068850018&category=3309&rd=1
 
This guy has been selling replicas of the X100P for quite some time.  I bought a few of those from him, and was extremely pleased with the results.
 
After speaking with him on the phone, he says he just wants 10 confirmed orders before making a set of the cards.  I guess it makes sense, since he can't be sure there is demand for the cards.
 
I've placed my order for 2 of them, and am looking forward to him getting the 10 necessary orders so that I can get my cards.
 
Please, no flames, I'm just passing on information that might be helpful to some,
J.T.
Do you Yahoo!?
Find out what made the Top Yahoo! Searches of 2003

Re: [Asterisk-Users] Slow wiki?

[Olle E. Johansson]

Philipp von Klitzing wrote:
Hi there,

is this a problem with the Wiki software or the DB? The delay is still 
tolerable, but not exactly nice to work with.

http://www.voip-info.org/tiki-index.php?page=Asterisk+billing
Page generated in: 2.35 seconds
The same page, 1.75 seconds for me.

The TikiWiki software is very slow. Jim has installed a PHP accelerator that helps
quite a lot, but not enough. I've tested the software locally and confirmed that
the coding is a bit awkward for most systems...
Also guess that there's some load on the system nowadays.
/Olle
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Re: [Asterisk-Users] Call recording

[Olle E. Johansson]

Sergio Serrano Revuelto wrote:
You must use Monitor Application

Happy New Year,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Edoardo
Borghesi [fabbricadigitale]
Enviado el: viernes, 02 de enero de 2004 12:33
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Call recording
Hello,

I need a way to record every call made to asterisk on a file. The
app_record application works but it is blocking, so I can't connect a
phone-operator and an user while recording. I thought to use the MeetMe
application and using a fake user to record the call but in this way I
can't know if the phone-operator is ready to answer or is answering
another user (i.e., the operator is always in conference and I obviously
don't want to have more than one user connected to the conference).
Does anyone know a way to achieve this goal? I can also modify some code
if this is needed.
See
http://www.bkw.org/~brian/asterisk-conf/bkw-exten.conf
for a good configuration example.

/O

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[Asterisk-Users] Malloc debug kills asterisk?

[Scott Stingel]

Hi-

In trying to track down a possible memory leak in asterisk, I've discovered
that the "show memory allocations" command crashes asterisk (causes it to
stop handling calls, although it doesn't seg fault).  The related "show
memory summary" works however.

Before I post this to the bugs list, can someone else please confirm this
problem?  You need to enable malloc debugging (symbol MALLOC_DEBUG) in the
Makefile, and then rebuild asterisk and try these commands.  I'm running
Fedora 1, and I want to make sure that it's not some anomaly with the kernel
rather than asterisk.

I tried this with the latest CVS, and also from about a month ago - same
result.

Thanks!
Scott

Scott M. Stingel 
Emerging Voice Technology Inc.
Palo Alto, California and London, England
Email:  scott "at" evtmedia.com  
URL:www.evtmedia.com  

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Re: [Asterisk-Users] asterisk dies while making calls

[Steven Critchfield]

On Fri, 2004-01-02 at 09:27, Asterisk List wrote:
> Hello:
> 
> It has happened while I was making 1000 outgoing calls, at a sustained rate 
> of 2 calls per second.
> Asterisk makes a SIP call to a CISCO router and this router is connected to 
> the PSTN line.
> 
> While putting files in the outgoig folder, I noticed that the files remained 
> there and the calls have stopped. Looking for the asterisk process, it was 
> gone.

> In /var/log/asterisk/messages file I find these lines:
> Jan  2 14:10:16 NOTICE[99500051]: File pbx_spool.c, Line 199 
> (attempt_thread): Call failed to go through, reason 5
> Jan  2 14:10:17 NOTICE[99483665]: File pbx_spool.c, Line 199 
> (attempt_thread): Call failed to go through, reason 5
> Jan  2 14:10:17 NOTICE[99532821]: File pbx_spool.c, Line 199 
> (attempt_thread): Call failed to go through, reason 5
> Jan  2 14:10:18 NOTICE[99516436]: File pbx_spool.c, Line 199 
> (attempt_thread): Call failed to go through, reason 5
> Jan  2 14:10:18 NOTICE[99565587]: File pbx_spool.c, Line 199 
> (attempt_thread): Call failed to go through, reason 5
> Jan  2 14:10:19 NOTICE[99549201]: File pbx_spool.c, Line 199 
> (attempt_thread): Call failed to go through, reason 5
> Jan  2 14:10:19 NOTICE[99581972]: File pbx_spool.c, Line 199 
> (attempt_thread): Call failed to go through, reason 5
> Jan  2 14:10:20 NOTICE[99598357]: File pbx_spool.c, Line 199 
> (attempt_thread): Call failed to go through, reason 5
> Jan  2 14:10:20 WARNING[81926]: File chan_sip.c, Line 450 (retrans_pkt): 
> Maximum retries exceeded on call 
> [EMAIL PROTECTED] for seqno 102 (Request)
> Jan  2 14:10:20 NOTICE[81926]: File sched.c, Line 209 (sched_settime): 
> Request to schedule in the past?!?!
> 
> This was the last line, asterisk died and the process dissapeared.
> What happened? Where can I find more asterisk log files to discover it?

The messages above appear to be failed threads. It seems you hit a limit
in your kernel or thread library for running processes. If you had been
on the end of one of the calls their towards the end, you might have
noticed a severe call quality drop as each of these processes can only
use a small fraction of the CPU time, and I bet you started to starve
each one for enough time to do any useful work. 

If you so wish to try that again, maybe you should have a window open
watching top as asterisk nears that point to see exactly how many
running processes there are at the time. 

Also as a general rule of thumb, if you experience a crash and are able
to reproduce it, make sure you have the ability to drop a core file. You
usually do this by running "ulimit -c unlimited", and then moving to a
directory that can handle a couple meg file when the failure occurs.
Then you run the application like normal and proceed to cause the
failure. You will then be left with the memory from the application in a
file at the point it failed. This is useful to then use gdb on the file
to determine what it was doing at the time of failure. Usually a simple
back trace is enough to get started, and a back trace is printed using
the command bt from within gdb. 
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] asterisk dies while making calls

[Asterisk List]

Hello:

It has happened while I was making 1000 outgoing calls, at a sustained rate 
of 2 calls per second.
Asterisk makes a SIP call to a CISCO router and this router is connected to 
the PSTN line.

While putting files in the outgoig folder, I noticed that the files remained 
there and the calls have stopped. Looking for the asterisk process, it was 
gone.

I found these lines in /var/log/asterisk/event_log:
Jan  2 14:10:15 asterisk[3271]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 0 attemp$
Jan  2 14:10:16 asterisk[3270]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 0 attemp$
Jan  2 14:10:16 asterisk[624]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 1 attempt$
Jan  2 14:10:16 asterisk[624]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 1 attempt$
Jan  2 14:10:16 asterisk[3277]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 0 attemp$
Jan  2 14:10:17 asterisk[3276]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 0 attemp$
Jan  2 14:10:17 asterisk[3283]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 0 attemp$
Jan  2 14:10:17 asterisk[624]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 1 attempt$
Jan  2 14:10:18 asterisk[3282]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 0 attemp$
Jan  2 14:10:18 asterisk[624]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 1 attempt$
Jan  2 14:10:18 asterisk[624]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 1 attempt$
Jan  2 14:10:18 asterisk[3289]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 0 attemp$
Jan  2 14:10:19 asterisk[3288]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 0 attemp$
Jan  2 14:10:19 asterisk[624]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 1 attempt$
Jan  2 14:10:19 asterisk[624]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 1 attempt$
Jan  2 14:10:19 asterisk[3294]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 0 attemp$
Jan  2 14:10:20 asterisk[3295]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 0 attemp$
Jan  2 14:10:20 asterisk[624]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 1 attempt$
Jan  2 14:10:20 asterisk[624]: Queued call to SIP/[EMAIL PROTECTED] expired 
without completion after 1 attempt$

Where 9 are valid phone numbers.

In /var/log/asterisk/messages file I find these lines:
Jan  2 14:10:16 NOTICE[99500051]: File pbx_spool.c, Line 199 
(attempt_thread): Call failed to go through, reason 5
Jan  2 14:10:17 NOTICE[99483665]: File pbx_spool.c, Line 199 
(attempt_thread): Call failed to go through, reason 5
Jan  2 14:10:17 NOTICE[99532821]: File pbx_spool.c, Line 199 
(attempt_thread): Call failed to go through, reason 5
Jan  2 14:10:18 NOTICE[99516436]: File pbx_spool.c, Line 199 
(attempt_thread): Call failed to go through, reason 5
Jan  2 14:10:18 NOTICE[99565587]: File pbx_spool.c, Line 199 
(attempt_thread): Call failed to go through, reason 5
Jan  2 14:10:19 NOTICE[99549201]: File pbx_spool.c, Line 199 
(attempt_thread): Call failed to go through, reason 5
Jan  2 14:10:19 NOTICE[99581972]: File pbx_spool.c, Line 199 
(attempt_thread): Call failed to go through, reason 5
Jan  2 14:10:20 NOTICE[99598357]: File pbx_spool.c, Line 199 
(attempt_thread): Call failed to go through, reason 5
Jan  2 14:10:20 WARNING[81926]: File chan_sip.c, Line 450 (retrans_pkt): 
Maximum retries exceeded on call 
[EMAIL PROTECTED] for seqno 102 (Request)
Jan  2 14:10:20 NOTICE[81926]: File sched.c, Line 209 (sched_settime): 
Request to schedule in the past?!?!

This was the last line, asterisk died and the process dissapeared.
What happened? Where can I find more asterisk log files to discover it?
Thanks in advance,
Robert T.
_
Reserva y planifica tu viaje online. http://www.msn.es/Viajes/
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Re: [Asterisk-Users] Call recording

[CW_ASN]

- Original Message - 
From: "Chandra" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, January 02, 2004 9:30 AM
Subject: Re: [Asterisk-Users] Call recording


> xlite saying login timed out. contact network admin.
> 
> how to get rid of this. * is not behind NAT.
> 
> DIAX works fine
> 

Could you especify a bit more?
Send sip.conf, 'netstat -na' from you linux box, xlite config, etc...

Regards,

Gus


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Re: [Asterisk-Users] Prediction for 2004

[TC]

This is bizzare the following was removed from this post

--
And the answers are "standards", "professionalism", "comunications" and
"documentation

Note the section on Professionalism
http://users.tpg.com.au/adsl87w7/blog/


---

- Original Message -
From: "TC" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, January 02, 2004 6:21 AM
Subject: Re: [Asterisk-Users] Prediction for 2004


> Some more thoughts from Graig Sutherland of open h323 fame on how Open Src
> stays relevent
> in VoIP as the Big Boys start to move in on VoiP niche
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Re: [Asterisk-Users] unsubscribe

[Steven Critchfield]

On Fri, 2004-01-02 at 08:51, Ranga wrote:
> Sorry...I missed it. I wanted to change my email id. So unsubscribed and
> subscribed again.

Well then I guess it was a good thing I kept some composure instead of
flaming away as is the usual for those kinds of messages. 

Welcome back.

> - Original Message -
> From: "Steven Critchfield" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, January 02, 2004 7:52 PM
> Subject: Re: [Asterisk-Users] unsubscribe
> 
> 
> > On Fri, 2004-01-02 at 07:59, Ranga wrote:
> > > unsubscribe
> >
> > This is a function you do your self. You should see a URL in the footer
> > of this message that will show you how to do it. Had you been reading
> > copies of messages that where not in HTML, you would have seen this
> > message before.
> > --
> > Steven Critchfield <[EMAIL PROTECTED]>
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] * Stresstool Help required

[Alastair Maw]

On 02/01/04 14:24, Girish Gopinath wrote:

I gave the sip debug command, and one of the lines showed:"Ignoring this 
request"
Can you log the SIP debug messages to a file and put it up on the web 
somewhere? Or do an ethereal capture or similar. It's very hard to say 
what the problem might be without a full SIP trace.

It's likely that you're generating the same transaction ID for each SIP 
INVITE or something silly.

Regards,

Alastair
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Re: [Asterisk-Users] unsubscribe

[Ranga]

Sorry...I missed it. I wanted to change my email id. So unsubscribed and
subscribed again.
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, January 02, 2004 7:52 PM
Subject: Re: [Asterisk-Users] unsubscribe


> On Fri, 2004-01-02 at 07:59, Ranga wrote:
> > unsubscribe
>
> This is a function you do your self. You should see a URL in the footer
> of this message that will show you how to do it. Had you been reading
> copies of messages that where not in HTML, you would have seen this
> message before.
> --
> Steven Critchfield <[EMAIL PROTECTED]>
>
> ___
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>
>


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Re: [Asterisk-Users] License questioni supose ??

[Steven Critchfield]

On Fri, 2004-01-02 at 08:24, Andrew Thompson wrote:
> - Original Message -
> From: "Steven Critchfield" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, January 02, 2004 8:59 AM
> Subject: Re: [Asterisk-Users] License questioni supose ??
> 
> 
> > On Fri, 2004-01-02 at 06:54, Nicolas Bougues wrote:
> > > On Fri, Jan 02, 2004 at 12:23:52PM +0100, Michael Devenijn wrote:
> > > > I have some strange question bout the asterisk (gpl license ...) but
> i'm not an experienced linux user ...
> > > >
> > > > What happens if for example a big company buys digium , do we have a
> garantuee that asterisk stays opensource ???
> > > >
> > >
> > > Until the last released version, yes. Digium owns the Copyright : they
> > > can decide whenever they want that their next release will have any
> > > other kind of licence (open or not).
> > >
> 
> 
> Can someone define released version for me? Is that the 0.5 that's available
> as stable, or the most recent copy that's checked in to cvs, or something
> else?

AFAIK, since the CVS versions are distributed to users, and they are
released under the GPL, then basically all versions accessible via CVS
or tar ball should meet the definition as released. 
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] License questioni supose ??

[Andrew Thompson]

- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, January 02, 2004 8:59 AM
Subject: Re: [Asterisk-Users] License questioni supose ??


> On Fri, 2004-01-02 at 06:54, Nicolas Bougues wrote:
> > On Fri, Jan 02, 2004 at 12:23:52PM +0100, Michael Devenijn wrote:
> > > I have some strange question bout the asterisk (gpl license ...) but
i'm not an experienced linux user ...
> > >
> > > What happens if for example a big company buys digium , do we have a
garantuee that asterisk stays opensource ???
> > >
> >
> > Until the last released version, yes. Digium owns the Copyright : they
> > can decide whenever they want that their next release will have any
> > other kind of licence (open or not).
> >


Can someone define released version for me? Is that the 0.5 that's available
as stable, or the most recent copy that's checked in to cvs, or something
else?

-
Andrew Thompson http://aktzero.com/
Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
restful it is to watch the cursor blink. Close your eyes. The opinions
stated above are yours. You cannot imagine why you ever felt otherwise.



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[Asterisk-Users] * Stresstool Help required

[Girish Gopinath]

Hi all,

I am trying to write a program that sends SIP requests to asterisk. My aim 
is to make asterisk record as many voicemails it can at a time. The design 
of the program is like this:
There are two processes: One main process and a child process (No flames 
pls. I have very little idea about pthreads and dl modules)

The main program asks the user to input the number of test instances. When 
the user inputs that (valid instances are: 1 - 50), it will spawn that many 
number of child processes that communicate with *. All of them get their own 
sip ports, rtp ports, and user names (for REGISTERing with *). There is a 
delay of 4 seconds before spawning each process. When i input 1, everything 
works fine (i guess). * records the voicemail (i am sending the contents of 
a .wav file to asterisk) .

Here is the screen capture:
*CLI> -- Registered SIP 'gopi' at 192.168.68.15 port 5061 expires 120
 == Setting SIPDOMAIN to : 192.168.68.6
   -- Executing Dial("SIP/gopi-bddf", "SIP/stest|10|tr") in new stack
 == Everyone is busy at this time
   -- Executing Ringing("SIP/gopi-bddf", "") in new stack
   -- Executing Answer("SIP/gopi-bddf", "") in new stack
   -- Executing VoiceMail2("SIP/gopi-bddf", "u") in new stack
   -- Playing 'voicemail/default//unavail' (language 'en')
   -- Playing 'vm-intro' (language 'en')
   -- Playing 'beep' (language 'en')
WARNING[15376]: File app_voicemail.c, Line 1236 (leave_voicemail): No more 
messages possible
   -- Executing Hangup("SIP/gopi-bddf", "") in new stack
 == Spawn extension (stresstest, , 5) exited non-zero on 
'SIP/gopi-bddf'

The problem starts when i try to spawn more than one instance of the 
process. I tried with 2, both the instances got registered. The initial part 
of dialing is also ok. After that one of the child processes gets BYE 
request from *. The other child continues and * records voicemail for it.
Here is the screen capture of that:

   -- Registered SIP 'gopi' at 192.168.68.15 port 5061 expires 120
 == Setting SIPDOMAIN to : 192.168.68.6
   -- Executing Dial("SIP/gopi-7263", "SIP/stest|10|tr") in new stack
 == Everyone is busy at this time
   -- Executing Ringing("SIP/gopi-7263", "") in new stack
   -- Executing Answer("SIP/gopi-7263", "") in new stack
   -- Executing VoiceMail2("SIP/gopi-7263", "u") in new stack
   -- Playing 'voicemail/default//unavail' (language 'en')
   -- Registered SIP 'nath' at 192.168.68.15 port 5062 expires 120
 == Setting SIPDOMAIN to : 192.168.68.6
   -- Executing Dial("SIP/nath-bedf", "SIP/stest|10|tr") in new stack
 == Everyone is busy at this time
   -- Executing Ringing("SIP/nath-bedf", "") in new stack
   -- Executing Answer("SIP/nath-bedf", "") in new stack
   -- Executing VoiceMail2("SIP/nath-bedf", "u") in new stack
   -- Playing 'voicemail/default//unavail' (language 'en')
   -- Playing 'vm-intro' (language 'en')
   -- Playing 'vm-intro' (language 'en')
WARNING[5126]: File chan_sip.c, Line 469 (retrans_pkt): Maximum retries 
exceeded on call [EMAIL PROTECTED] for 
seqno 2 (Response)
WARNING[16400]: File file.c, Line 512 (ast_readaudio_callback): Failed to 
write frame
 == Spawn extension (stresstest, , 4) exited non-zero on 
'SIP/gopi-7263'
   -- Playing 'beep' (language 'en')
WARNING[17425]: File app_voicemail.c, Line 1236 (leave_voicemail): No more 
messages possible
   -- Executing Hangup("SIP/nath-bedf", "") in new stack
 == Spawn extension (stresstest, , 5) exited non-zero on 
'SIP/nath-bedf'

I gave the sip debug command, and one of the lines showed:"Ignoring this 
request"
Does that mean asterisk doesn't process 2 requests simultaneously, when it 
is sent from one machine? I know * is not a SIP proxy, it is a PBX. Is this 
problem related to that? If so, how * registered the two instances of the 
process? I tried with one instance of the test program from one machine, and 
SJPhone from another machine. Both worked fine.

Can anybody help me in figuring out the problem? I admit that there are many 
bugs in my program and i beleive that the problems are because of these bugs 
only. Still wanted to hear from you...

Warm Regards...

Girish

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Re: [Asterisk-Users] unsubscribe

[Steven Critchfield]

On Fri, 2004-01-02 at 07:59, Ranga wrote:
> unsubscribe

This is a function you do your self. You should see a URL in the footer
of this message that will show you how to do it. Had you been reading
copies of messages that where not in HTML, you would have seen this
message before.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Prediction for 2004

[TC]

Some more thoughts from Graig Sutherland of open h323 fame on how Open Src
stays relevent
in VoIP as the Big Boys start to move in on VoiP niche
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Re: [Asterisk-Users] Slow wiki?

[Steven Critchfield]

On Fri, 2004-01-02 at 07:59, Philipp von Klitzing wrote:
> Hi there,
> 
> is this a problem with the Wiki software or the DB? The delay is still 
> tolerable, but not exactly nice to work with.
> 
> http://www.voip-info.org/tiki-index.php?page=Asterisk+billing
> Page generated in: 2.35 seconds

Wiki software has a bad reputation of too many queries and there for
always slow operations. 

Of course when I hit that URL this morning it listed as 1.1 seconds. It
may well be that the recent mention is slashdot may have had increased
the visitors to the site at that time. 
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Ranga]

unsubscribe

Re: [Asterisk-Users] License questioni supose ??

[Steven Critchfield]

On Fri, 2004-01-02 at 06:54, Nicolas Bougues wrote:
> On Fri, Jan 02, 2004 at 12:23:52PM +0100, Michael Devenijn wrote:
> > I have some strange question bout the asterisk (gpl license ...) but i'm not an 
> > experienced linux user ...
> >  
> > What happens if for example a big company buys digium , do we have a garantuee 
> > that asterisk stays opensource ???
> >  
> 
> Until the last released version, yes. Digium owns the Copyright : they
> can decide whenever they want that their next release will have any
> other kind of licence (open or not).
> 
> But what's released under the GPL stays so. Anyone can continue using
> it, or even re-release it, provided that they still comply to the
> GPL terms. Nobody, even Digium, can revoke the GPL licence for GPL-ed
> software.

I'll answer this before Tilghman has to. Digium does not own all the
copyrights to asterisk. They own the largest portion, and they have been
granted permission by other contributers to use their code without
compensation.  Please read 1a of this document  
http://www.digium.com/disclaimer.txt.

1a allows Digium to sell licenses to the software even though they don't
own all the copyrights, but they have a license to use the parts they
don't own. You will see that unless you sign the above document, or the
other one they have, they don't include your changes into the main
repository.

So while there is the possibility that someone could purchase Digium and
take the software private, the last release before the act of taking it
private will always be available via a GPL license. It is possible for
us the community to at that point to take the last GPL version and
maintain it ourselves. Downside would be that those who would require a
commercial license to sidestep GPL requirements will be stuck trying to
deal with the company who bought Digium.

GPL guarantees you will have access to at least the version you are
running for as long as it is available, then you need to have made a
copy for yourself. 

Just think, the GPL puts you in a better position than Microsoft does
considering Windows ME is no longer available and is under 4 years old.
It is still possible to get older GPL software, even some that the
projects groups or companies that created them have been disbanded for
some time. 

Also remember the GPL is based on copyright law and builds extra support
by having many people with copyright interest in the software.

Here is an interesting article that should help explain the situation
for you. It is written by a lawyer for RedHat.
 http://www.nswscl.org.au/journal/51/Mark_H_Webbink.html
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] Slow wiki?

[Philipp von Klitzing]

Hi there,

is this a problem with the Wiki software or the DB? The delay is still 
tolerable, but not exactly nice to work with.

http://www.voip-info.org/tiki-index.php?page=Asterisk+billing
Page generated in: 2.35 seconds

Philipp


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RE: [Asterisk-Users] New to asterisk? RUN... don't walk.

[Paulo Mannheimer]

What about you drop your beer, stand up from your couch (if your fat
belly allows you to), turn off the damn TV and try to learn some basic
"C" programming. Then maybe you can help us in solving those "frequent
segmentation faults"  (if any).


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Me
Sent: quarta-feira, 31 de dezembro de 2003 17:37
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] New to asterisk? RUN... don't walk.


As a newcomer to Asterisk, you will not be welcomed
with open arms.  First, you will find almost no
documentation on it's features.  Second, if you try to
ask questions, you will be flamed and pointed to
worthless how-tos and 'the wiki'.  These worthless
documents can only be useful for explaining how things
work to those already in-the-know.  Lastly, Asterisk
is so bug ridden, expect frequent segmentation faults.
 With a community so 'anti-n00b', don't expect your
problems to be fixed anytime soon. 

RUN!!! Don't walk... away from Aterisk.

__
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Find out what made the Top Yahoo! Searches of 2003
http://search.yahoo.com/top2003
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[Asterisk-Users] SIP client not registering to *

[Chandra]

xlite saying login timed out. contact network admin.

how to get rid of this. * is not behind NAT.

also, the grandstream SIP phone also seems to fail to register. IAX phones
are all ok.

DIAX works fine


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Re: [Asterisk-Users] License questioni supose ??

[Nicolas Bougues]

On Fri, Jan 02, 2004 at 12:23:52PM +0100, Michael Devenijn wrote:
> I have some strange question bout the asterisk (gpl license ...) but i'm not an 
> experienced linux user ...
>  
> What happens if for example a big company buys digium , do we have a garantuee that 
> asterisk stays opensource ???
>  

Until the last released version, yes. Digium owns the Copyright : they
can decide whenever they want that their next release will have any
other kind of licence (open or not).

But what's released under the GPL stays so. Anyone can continue using
it, or even re-release it, provided that they still comply to the
GPL terms. Nobody, even Digium, can revoke the GPL licence for GPL-ed
software.

-- 
Nicolas Bougues
Axialys Interactive
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RE: [Asterisk-Users] SQL Updater Down!!!

[mattf]

Is the updater running? does the client machine have rights to access the
database machine? Please let me know what scripts are running on what
machines.

MATT---

-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Friday, January 02, 2004 1:43 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SQL Updater Down!!!


hi,

I am trying to install ASTGUICLIENT and when i run the
AST_WINphoneAPP_0.8.pl it opens a window  VICI Phone App -0.8 but i am
getting SQL Updater Down Mesasge. How can i solve this?


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Re: [Asterisk-Users] Call recording

[Chandra]

xlite saying login timed out. contact network admin.

how to get rid of this. * is not behind NAT.

DIAX works fine

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[Asterisk-Users] License questioni supose ??

[Michael Devenijn]

I have some strange question bout the asterisk (gpl license ...) but i'm not an 
experienced linux user ...
 
What happens if for example a big company buys digium , do we have a garantuee that 
asterisk stays opensource ???
 
Kind regards 
 
Michael Devenijn
<>

RE: [Asterisk-Users] Call recording

[Sergio Serrano Revuelto]

You must use Monitor Application


Happy New Year,
srsergio


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Edoardo
Borghesi [fabbricadigitale]
Enviado el: viernes, 02 de enero de 2004 12:33
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Call recording


Hello,

I need a way to record every call made to asterisk on a file. The
app_record application works but it is blocking, so I can't connect a
phone-operator and an user while recording. I thought to use the MeetMe
application and using a fake user to record the call but in this way I
can't know if the phone-operator is ready to answer or is answering
another user (i.e., the operator is always in conference and I obviously
don't want to have more than one user connected to the conference).

Does anyone know a way to achieve this goal? I can also modify some code
if this is needed.

Thanks

Edoardo

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Re: [Asterisk-Users] FW: [Asterisk-Dev] Strange bells in voicemail and voicemailmain ?

[Mark Spencer]

Sounds like something nasty being printed.  If you run asterisk in the
background (without -vvvgc) and don't attach to it do you hear it still?

Mark

On Fri, 2 Jan 2004, Patrick wrote:

> On Fri, 2004-01-02 at 08:31, Florian Overkamp wrote:
> > Hi,
> >
> > I've sent this to asterisk-dev recently, but seen no comments. Has anyone
> > else experienced this behaviour ? I've made a complete clean checkout of CVS
> > code, and it still happens
> >
> > > -Original Message-
> > > I've just made a new update from cvs on my devel box to play
> > > with, and I noticed that I get console bells when I start the
> > > voicemail app and asterisk seems to hang the channel.
> > >
> > > (I'm running asterisk non-detached in a screen, so my screen
> > > starts blinking madly and becoming unusable, have to kill the
> > > window and start again)
> > >
> > > Further debugging:
> > >
> > > I'm moved CVS back as far as 20/12 and it still doesn't work,
> > > so I'm curious as to what could be causing this...
> >
> > Thanks,
> >
> > Florian
> >
>
> Hi Florian,
>
> Yes I have seen this with a fresh checkout of 12/31 cvs. It happened
> with the second call mostly. First call would usually succeed without
> the console bell ringing. I started * in an xterm with asterisk -cg
> Also tried to reboot the box to see if that made it go away (maybe
> ztdummy had gone nuts) but that didn't make a difference. The previous
> release that (for me) did not have that issue was from 24/12. No idea
> what's causing it but it is annoying enough to unplug the speaker.
>
> Regards,
> Patrick
>
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>

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[Asterisk-Users] Call recording

[[fabbricadigitale]]

Hello,

I need a way to record every call made to asterisk on a file.
The app_record application works but it is blocking, so I can't connect
a phone-operator and an user while recording.
I thought to use the MeetMe application and using a fake user to record
the call but in this way I can't know if the phone-operator is ready to
answer or is answering another user (i.e., the operator is always in
conference and I obviously don't want to have more than one user
connected to the conference).

Does anyone know a way to achieve this goal? I can also modify some code
if this is needed.

Thanks

Edoardo

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Re: [Asterisk-Users] FW: [Asterisk-Dev] Strange bells in voicemail and voicemailmain ?

[Patrick]

On Fri, 2004-01-02 at 08:31, Florian Overkamp wrote:
> Hi,
> 
> I've sent this to asterisk-dev recently, but seen no comments. Has anyone
> else experienced this behaviour ? I've made a complete clean checkout of CVS
> code, and it still happens
> 
> > -Original Message-
> > I've just made a new update from cvs on my devel box to play 
> > with, and I noticed that I get console bells when I start the 
> > voicemail app and asterisk seems to hang the channel.
> > 
> > (I'm running asterisk non-detached in a screen, so my screen 
> > starts blinking madly and becoming unusable, have to kill the 
> > window and start again)
> > 
> > Further debugging:
> > 
> > I'm moved CVS back as far as 20/12 and it still doesn't work, 
> > so I'm curious as to what could be causing this...
> 
> Thanks,
> 
> Florian
> 

Hi Florian,

Yes I have seen this with a fresh checkout of 12/31 cvs. It happened
with the second call mostly. First call would usually succeed without
the console bell ringing. I started * in an xterm with asterisk -cg
Also tried to reboot the box to see if that made it go away (maybe
ztdummy had gone nuts) but that didn't make a difference. The previous
release that (for me) did not have that issue was from 24/12. No idea
what's causing it but it is annoying enough to unplug the speaker.

Regards,
Patrick

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Re: [Asterisk-Users] Anyone, ideas for incoming call management for CRM system

[Alastair Maw]

On 01/01/04 10:19, Olle E. Johansson wrote:

What I am looking for is a solution like this:
* Call comes in
* XXX on Line YYY answers
* A URL to a web page is transmitten on some channel, preferably the 
  VoIP channel
* The web page opens in a web window´
You're best off writing a separate application to do that - you probably 
don't want it tied into the VoIP communications layer because you're 
then tying yourself into using software VoIP clients. Most people prefer 
hardphones, so keep yourself flexible.

This would be really quite easy (couple of hours' coding, tops) to 
construct using a simple Perl script on the server listening to the 
Manager stream (telnet) for incoming calls, and a small C++/VB/whatever 
systray application on each Windows PC. The Perl script would connected 
to the appropriate PC whenever an incoming call appeared to tell it what 
URL to pop up.

There are security issues with people being able to pop up arbitrary web 
pages on each other's desktops and the like, but it basically wouldn't 
be very hard.

You can find information on the Manager interface here:

http://www.voip-info.org/tiki-index.php?page=Asterisk%20manager%20API

Regards,

Alastair
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Re: [Asterisk-Users] Programming an unlocked ADSI phone?

[Jayson Vantuyl]

On Thu, Jan 01, 2004 at 12:51:09PM -0700, Ken Godee wrote:
> Darren Nickerson wrote:
> >That worked a treat - thanks! Comedian Mail is now able to download to the
> >handset and there's a lot more functionality now.
> >
> >-d
> 
> I'd be interested in knowing if once you try to use Comedian mail
> softkeys if the 480 keypad goes dead?
I have this problem with the PT390s.

Do you have a problem during the login process or does your problem come
later when trying certain menus?

If it occurs during login, it is probably the same problem I used to
have.  It stems from Comedian not putting the ADSI phone back into Voice
Mode.  It is not clear to me if this should be necessary for you to hit
the buttons (although the Aastras seem to require it).

I've fixed it in my source and submitted a patch, but no one can seem to
confirm what the spec says should be correct (and the spec costs about
$2,000, so I can't afford to buy it).

> Mine and several others reported same, which makes it useless, a shame 
> to, I like the 480's ADSI function and haven't had a whole lot of time 
> to look into it.

It can be made to work.  Let me round off the rough edges of my patch
and I'll send it to you (and put it in the bugtracker again).  If I
recall, I tried to get it merged in but no one could test it with other
phones to see if it broke them.  Can anyone out there test a modified
patch to make sure it works for, say, the Telcordia phones and the like?

At this point, we seem to have only mediocre support for ADSI (you'll
have problems uploading large scripts as ADSIprog seems to have problems
splitting up large functions during upload).  Since the industry really
never intended for us to use it in this way, makes it really expensive
to implement it, and is very paranoid about supporting us in any way, I
suspect it may stay this way for a while.

To be sure, all of my contacts with Aastra and the like consist of "We
make a compliant phone.  It's been certified.  Buy the spec and embrace
it."  Mostly assertion that the phone has been certified so it must be
flawless and an insistence that I buy the spec and figure it out for
myself.  It took me three months to get a part number so I could order a
phone that they would give me the code for.  Even then, it was
interesting that it was an oddball pre-programmed phone for some PBX
system similar to us called SpheriCall.  I could only get that after
Aastra mistakenly routed me to Sayson (sayson.com) who they appear to
have offloaded the job of preprogramming their phones for end-users.

At any rate, I assure you that with a lot of work, I've made the Aastras
work with Comedian (and I could probably do so for you if you were
really, really nice or offered heaps of cash).  The problem I have not
been able to solve has been with setting the phone up to operate with a
default program.  It seems like it should be simple to set up a program
for when it's idle, when it's ringing, buttons for simple call functions
and the like.  It's not.  The self-load slot requires an idle time of
like 5 seconds before it kicks in.  Simply hanging up the phone and
immediately picking it up causes it to fall back into default mode where
your program isn't doing anything.

Frustratingly, the SpheriCall program that my phones come with have
SOLVED this problem.  I cannot, however, figure out how to do it with
ours.  I've come very close to buying some third-party programming
software (Black Dolphin something) just to program them.  The brochure
actually mentions support for the "special features" of Aastra
PowerTouches.  That doesn't really instill faith in me, though.

Good luck,

Jayson

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Re: [Asterisk-Users] one way choppy sound problem !

[WipeOut]

Dawid Mielnik wrote:

Hi all,

I have my asterisk setup as following:

IP   2 x E1
x-lite <---> Asterisk ---> PSTN
When I place a call from x-lite to PSTN, the quality of the sound in the
direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user,
heard by the PSTN user is choppy and makes communication not very pleasant.
The sound is choppy as if bits of data were lost. The strange thing is that
the x-lite user hears the PSTN user fine !
In x-lite, I have swithed off sience detection (transmit silence - yes),
this has improved the sound quality but did not eliminated the problem. I
have fed a countinious sound into the microphone and still got chops in the
sound. I have also tried changing the codecs gsm, alaw, ulaw - but I get the
same problem with all of them. Maybe the problem lies somewhere in audio
buffering settings on x-lite ?
Has anyone ever had this sort of problem and managed to deal with it ? I
would greatly appreciate your help !
Best regards,

Dave

 

I have the exact same problem with my Grandstream phones (Snom's are 
fine) and others have reported it as well with GS phones but this is the 
first time I have heard of this problem in X-Lite..

All I can suggest is that you use the latest version of x-lite and see 
if it helps, other than that I have not been able to find the answer..

Good luck, and if you find a solution let us know..

Later..

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[Asterisk-Users] one way choppy sound problem !

[Dawid Mielnik]

Hi all,

I have my asterisk setup as following:

IP   2 x E1
x-lite <---> Asterisk ---> PSTN


When I place a call from x-lite to PSTN, the quality of the sound in the
direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user,
heard by the PSTN user is choppy and makes communication not very pleasant.
The sound is choppy as if bits of data were lost. The strange thing is that
the x-lite user hears the PSTN user fine !

In x-lite, I have swithed off sience detection (transmit silence - yes),
this has improved the sound quality but did not eliminated the problem. I
have fed a countinious sound into the microphone and still got chops in the
sound. I have also tried changing the codecs gsm, alaw, ulaw - but I get the
same problem with all of them. Maybe the problem lies somewhere in audio
buffering settings on x-lite ?

Has anyone ever had this sort of problem and managed to deal with it ? I
would greatly appreciate your help !

Best regards,

Dave


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[Asterisk-Users] IAXy Release ?

[Kannaiyan Natesan]

Is there is any idea for IAXy Device to be released?
Anywhere I can find the specs for it?

Kannaiyan



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Re: [Asterisk-Users] sound driver advise needed

[Steven Critchfield]

On Fri, 2004-01-02 at 01:17, [EMAIL PROTECTED] wrote:
> On Thu, Jan 01, 2004 at 09:48:36PM -0700, Steve Murphy wrote:
> > Hello--
> > 
> [snip]
> > 
> > Trouble is, asterisk only sees the brain-dead interface. How do I
> > exorcise it from the kernel, or at least make the SB the first-priority
> > one?  rmmod didn't seem to do anything. Playing with the Redhat sound 
> > card detection stuff was useless. I've googled around the internet,
> > looking for tidbits, but nothing seems applicable. RedHat 8 Bible wasn't
> > very helpful.  Not much in the kernel source dirs, either, nor in the
> > source for the sound drivers.
> > 
> > Anybody have some experience with this sort of thing? It'd be neat to
> > put together an announcement functionality.
> > 
> > Any advise? Many thanks,
> 
> If you know the name of the module for the other soundcard, you could look 
> under /lib/modules/your_kernel/kernel/driver/sound and rename/move the file
> elsewhere so it doesn't get loaded. At least,  I think that'd work ;)
> 

While that would work, it is not a very good idea. It is better to
understand where and how the module was loaded and configure it to stop
loading. This gives the benefit of not causing headaches later if the
user chooses to use the card again. 

If you want to just not load the driver, remove its entry from
/etc/modules.conf, or possibly some RedHat specific file. 
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] FW: [Asterisk-Dev] Strange bells in voicemail and voicemailmain ?

[Florian Overkamp]

Hi,

I've sent this to asterisk-dev recently, but seen no comments. Has anyone
else experienced this behaviour ? I've made a complete clean checkout of CVS
code, and it still happens

> -Original Message-
> I've just made a new update from cvs on my devel box to play 
> with, and I noticed that I get console bells when I start the 
> voicemail app and asterisk seems to hang the channel.
> 
> (I'm running asterisk non-detached in a screen, so my screen 
> starts blinking madly and becoming unusable, have to kill the 
> window and start again)
> 
> Further debugging:
> 
> I'm moved CVS back as far as 20/12 and it still doesn't work, 
> so I'm curious as to what could be causing this...

Thanks,

Florian


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Re: [Asterisk-Users] Programming an unlocked ADSI phone?

[Tilghman Lesher]

On Thursday 01 January 2004 12:57, Darren Nickerson wrote:
> That worked a treat - thanks! Comedian Mail is now able to download
> to the handset and there's a lot more functionality now.

There's a patch on the bugtracker that should allow you to specify these
codes per user, as requested.

http://bugs.digium.com/bug_view_page.php?bug_id=733

Let me know how it works for you.

-Tilghman

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Re: [Asterisk-Users] sound driver advise needed

[andrewg]

On Thu, Jan 01, 2004 at 09:48:36PM -0700, Steve Murphy wrote:
> Hello--
> 
[snip]
> 
> Trouble is, asterisk only sees the brain-dead interface. How do I
> exorcise it from the kernel, or at least make the SB the first-priority
> one?  rmmod didn't seem to do anything. Playing with the Redhat sound 
> card detection stuff was useless. I've googled around the internet,
> looking for tidbits, but nothing seems applicable. RedHat 8 Bible wasn't
> very helpful.  Not much in the kernel source dirs, either, nor in the
> source for the sound drivers.
> 
> Anybody have some experience with this sort of thing? It'd be neat to
> put together an announcement functionality.
> 
> Any advise? Many thanks,

If you know the name of the module for the other soundcard, you could look 
under /lib/modules/your_kernel/kernel/driver/sound and rename/move the file
elsewhere so it doesn't get loaded. At least,  I think that'd work ;)

> 
> murf
> 
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