Re: [Asterisk-Users] Music On Hold and ATA-186 w/Silence Supression

[Robert Lawrence]

Alejandro G wrote:
I have a problem with ATA-186 configured for silence supression (AudioMode
bit 0 = 1). When enabled and listening music on hold no sound is heared (if
I talk I began to hear the music and again mutes when I stop talking).
If I configure for silence supression off everything goes fine. Any hint?
 

Don't use silence supression.  Asterisk doesn't support it.
Robert
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[Asterisk-Users] how to configure groups using a sip phone

[deepak . dhiman]

hi friends ! 

i am facing a problem from one week and now required ur help urgently.
Actually, i want to configure asterisk for two groups javgroup and 
linuxgroup.
i also have constraint to use only sip phone (esatara ). now, please help me 
is it possible to configure astersik in that way or that kind of facility is 
given in zapata.conf.
tell me in detail abt the configurations of the sip.conf and 
extensions.conf. 

thanks 

Deepak Dhiman
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Re: [Asterisk-Users] how to configure groups using a sip phone

[Rod Bacon]

Can you be more specific?
What are you trying to achieve with the creation of such groups?
- Original Message - 
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 3:50 PM
Subject: [Asterisk-Users] how to configure groups using a sip phone


hi friends !
i am facing a problem from one week and now required ur help urgently.
Actually, i want to configure asterisk for two groups javgroup and 
linuxgroup.
i also have constraint to use only sip phone (esatara ). now, please help 
me is it possible to configure astersik in that way or that kind of 
facility is given in zapata.conf.
tell me in detail abt the configurations of the sip.conf and 
extensions.conf.
thanks
Deepak Dhiman
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Re: [Asterisk-Users] Asterisk Realtime Capabilities

[Rod Bacon]

The term RTCache has never been mentioned in the WIKI or these forums. I 
assume that it's some sort of function to speed up realtime db access by 
keeping transactions in RAM and writing periodically? If so, I can 
understand why this would need to be flushed.

- Original Message - 
From: Matthew Boehm [EMAIL PROTECTED]
To: Asterisk Users asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 3:01 PM
Subject: Re: [Asterisk-Users] Asterisk Realtime Capabilities


to a load-balanced (not sure which mechanism I'll empoy here yet)
   I was quoted a $21,000 layer-7 switch from F5 Networks to do SIP load
balancing.
outside)? In other words, can the registering server update a USRLOC
type database on the fly, so all other servers know where to route calls
   As long as all * servers share the same central database; this way when
SIP 1 registers via RealTime at server A, server B (using same db) should 
be
able to see the registration. You may not be able to use RTCache though...

Also, I will be using multiple * boxes as media gateways. Is there an
existing mechanism whereby a given server can record the number of
busy/available ZAP channels to a central database for the purpose of
call routing?
   Nothing built-in comes to mind, but Im sure you could AGI something.
-Matthew
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[Asterisk-Users] Previous sip reload not yet done

[administrator tootai]

Hi list,
We are running a CVS version of 03-30-2005 but also had this behaviour 
on previous versions.

From time to time, after a period of not making calls (eg a night or 
few hours), we have no dialtone when we want to call. SIP show peers 
show EP registered with status OK but nothing happend. Nothing special 
in logs. After a SIP reload, everything is again working fine.

So we add a SIP reload each morning in crontab. But this is not 
solving the problem: it's not always efficient and when we try to re-run 
this command from CLI, we get a Previous sip reload not yet done. Only 
solution is to restart asterisk.

Does anyone else have this problem? Is there a workaround?
Thanks for any hints.
--
Daniel
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[Asterisk-Users] RE: AS5300+SIP+ASTERISK or AS5300+MGCP

[jafar mohammed]

AS5300 setup

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2005.04.04 09:37:31
=~=~=~=~=~=~=~=~=~=~=~=
sh runn
Building configuration...

Current configuration : 11599 bytes
!
! Last configuration change at 03:26:25 GMT Mon Apr 4
2005 by charles
! NVRAM config last updated at 03:06:50 GMT Mon Apr 4
2005 by charles
!
version 12.3
service timestamps debug datetime localtime
show-timezone
service timestamps log datetime localtime
show-timezone
service password-encryption
!
hostname 66.178.100.66
!
boot-start-marker
boot-end-marker
!
!
!
resource-pool enable
 --More-- !
resource-pool throttle 20 default
clock timezone GMT 3
clock calendar-valid
!
aaa new-model
!
!
aaa group server radius cdt-1
 server 159.148.8.108 auth-port 2362 acct-port 2363
!
aaa group server radius cdt-2
 server 62.85.77.82 auth-port 2362 acct-port 2363
!
aaa group server radius tsl
 server 62.56.250.200 auth-port 1812 acct-port 1813
!
aaa authentication login h323 group radius
aaa accounting send stop-record authentication failure

aaa accounting connection h323 stop-only broadcast
group cdt-1 group cdt-2 group tsl
aaa nas port voip
aaa session-id common
 --More-- ip subnet-zero
ip telnet source-interface FastEthernet0
ip name-server 66.178.100.68
!
!
!
trunk group  mgcp
!
isdn switch-type primary-net5
!
voice rtp send-recv
!
voice service pots 
!
voice service voip 
 cause-code legacy
 h323
  h225 timeout setup 8
  session transport udp
 sip
  min-se  600 
!
voice class codec 2
 --More--  codec preference 1 gsmfr
!
!
voice class permanent 1
 signal timing idle suppress-voice 5
 signal timing oos suppress-all 30
 signal timing oos timeout 120
!
!
voice class h323 1
 h225 timeout tcp establish 30
 h225 timeout connect 60
 h225 timeout setup 30
  call start fast
!
voice class h323 2
  call start slow
!
voice class h323 1001
  call start fast
!
voice class h323 10
!
 --More-- !
voice class busyout 1
!
!
voice class dualtone-detect-params 1
!
!
!
!
!
fax interface-type modem
!
!
controller E1 0
 clock source line primary
 ds0-group 0 timeslots 1-15,17-31 type r2-digital
!
!
!
translation-rule 22
 Rule 0 22254 254
!
!
!
interface Tunnel1
 ip address 192.168.44.1 255.255.255.0
 tunnel source Ethernet0
 tunnel destination 217.21.95.9
!
interface Tunnel17
 --More--  ip address 10.1.17.2 255.255.255.0
 shutdown
 tunnel source 212.165.147.254
 tunnel destination 66.92.133.199
 tunnel mode nos
!
interface Tunnel18
 no ip address
!
interface Ethernet0
 ip address 195.202.73.106 255.255.255.248
 no ip mroute-cache
!
interface Serial0
 no ip address
 no ip mroute-cache
 clockrate 2015232
 no fair-queue
!
interface Serial1
 no ip address
 no ip mroute-cache
 clockrate 2015232
 --More--  no fair-queue
!
interface Serial2
 no ip address
 no ip mroute-cache
 clockrate 2015232
 no fair-queue
!
interface Serial3
 no ip address
 no ip mroute-cache
 shutdown
 clockrate 2015232
 fair-queue 100 256 0
 ip rtp priority 1 1 75
!
interface Serial2:15
 no ip address
 isdn switch-type primary-net5
 no cdp enable
!
interface FastEthernet0
 ip address 172.16.202.90 255.255.255.0 secondary
 --More--  ip address 66.178.100.66
255.255.255.248
 ip access-group 1 in
 ip access-group 1 out
 no ip mroute-cache
 duplex auto
 speed auto
 h323-gateway voip interface
 h323-gateway voip id gk0 ipaddr 216.52.153.203 1719
 h323-gateway voip h323-id ngins
 ip rtp priority 16384 16383 400
!
ip classless
ip route 0.0.0.0 0.0.0.0 66.178.100.65
no ip http server
!
!
no logging trap
access-list 101 permit ip any any
!
route-map VOIP permit 20
 match ip address 101
!
route-map VOIP permit 100
 --More-- !
!
radius-server attribute 44 include-in-access-req
radius-server host 159.148.8.108 auth-port 2362
acct-port 2363 key 7 065E582A585C51411F0317
radius-server host 62.85.77.82 auth-port 2362
acct-port 2363 key 7 014B510F4F195E573B584B
radius-server host 62.56.250.200 auth-port 1812
acct-port 1813 key 7 121500031F0E050A
radius-server retransmit 10
radius-server timeout 120
radius-server vsa send accounting
radius-server vsa send authentication
call threshold global total-calls low 60 high 90
busyout
!
call application voice kenya flash:kenya.tcl
!
call application voice kenya1 flash:kenya.tcl
!

!
voice-port 0:0
 compand-type a-law
 connection plar 9001
!
!
mgcp call-agent 62.56.250.198 2427 service-type mgcp
version 1.0
mgcp dtmf-relay voip codec all mode out-of-band
mgcp restart-delay 2
mgcp codec g711ulaw packetization-period 10
mgcp package-capability dtmf-package
mgcp package-capability line-package
mgcp package-capability rtp-package
mgcp package-capability nas-package
mgcp package-capability script-package
mgcp sdp simple
 --More-- no mgcp validate domain-name
mgcp endpoint offset
mgcp bind control source-interface FastEthernet0
mgcp bind media source-interface FastEthernet0
mgcp behavior signals v0.1
!
mgcp profile default
!
dial-peer cor custom
!
!
!
!
dial-peer voice 271 pots
 permission orig
 huntstop
 

Re: [Asterisk-Users] V92 modem with asterisk

[Rod Bacon]

No.
- Original Message - 
From: Alexandre Charles [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 3:48 PM
Subject: [Asterisk-Users] V92 modem with asterisk


Hi everyone,
I just install Linux and asterisk on one of my pc. I
want to run some basic functionality tests.
Is it possible to use a v92 modem as a FXO or FXS
card. If yes how do we configure and install the card?
What are the commands?
Thanks in advance for your help
AC
__
Lèche-vitrine ou lèche-écran ?
magasinage.yahoo.ca
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[Asterisk-Users] Wellgate 3701

[Asterisk user list]

Hi everyone

I'm trying to setup this Welltech Wellgate 3701 box.

If I got to the proxy setup it seems to work but the Pstn incoming call
always got a voice prompt from the Wellgate.

Going to peer to peer mode seems to be better but I couldn't find any
working configuration inside Asterisk.

I do not really suffer from the registration problem because I doing all
those trials with no password for the 3701 line configuration since I'm
in a closed environment.

Thanks for any help.


Ml 
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Re: [Asterisk-Users] Buying some Polycom IP300s

[Rod Bacon]

Over the last few weeks/months I have been testing 
phones and ATAs from Grandstream (BT101, GXP2000, 286, 488), SNOM (190), Zyxel 
(Piece of Crap), Sipura (SPA-2000, SPA-841) and I personally feel that the 
Sipura SPA-841 is the best value, good quality phone that I have used. I haven't 
used the polycoms yet, but I plan to in the next few weeks.




  - Original Message - 
  From: 
  Dan Morin 
  
  To: asterisk-users@lists.digium.com 
  
  Sent: Sunday, April 03, 2005 10:22 
  AM
  Subject: [Asterisk-Users] Buying some 
  Polycom IP300s
  
  Sorry =or the 
  double post, I tried to paste and accidently sent the 
email
  
  I've been playing with Asterisk for a few =eeks now, 
  and I've gotten everything to work well with softphones, so I'm ready to =ove 
  on to normal VoIP phones. I've been looking around and reading =omments 
  that people have had, and I was convinced that the Polycom IP300 was a great 
  =hone for a good price. But, then I ran into this page, which has been 
  =pdate in the last few days:
  
  http://w=w.voip-info.org/wiki-Polycom+SoundPoint+IP+500=DIV 

  
  The page in the wiki used to say that the =erson 
  would not recomed Polycom phones to anyone. So anyway, I just want to 
  =ake sure that the IP300 is a good choice. I don't want to get cheap 
  phones =hat aren't business quality, since I do play on using them for my 
  business =fter testing. Also, is the IP500 worth the extra money? 
  What can =t do that the IP300 can't. And finally, will the IP300 do ulaw 
  encoding?
  
  Thanks in advance.
  
  

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Re: [Asterisk-Users] AGI Dial Plan

[Jean-Michel Hiver]

Lee Lee wrote:
Hi everyone
 
Presently all our calls are channel to one provider and we would like 
to change that based on LCR.
 
the following is what we have presently;
 

# Dial the requested number, if we got something from the caller.
if ($dialto != )
{
$AGI-exec('SetAccount', $accountnum);
if ($debug) { $AGI-exec('NoOp', \Dialing $dialto... \); }
   if ($dialto =~ /^416/) {
$AGI-exec('Dial', Zap/g2/$dialto|30|C);
   } else { $AGI-exec('Dial', Zap/g1/$dialto|30|C) }
}
$AGI-hangup();

Cheers,
Jean-Michel.
--
Ykoz Un Max - La VoIP en pré-payé!
Essayez gratuitement - 5 crédits offerts.
--- http://ykoz.net/voip/max ---

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RE: [Asterisk-Users] Buying some Polycom IP300s

[Paul Hales]

My personal opinion is that the Polycom IP-300 is a slightly better phone than 
the Sipura, but I would be happy to be proved wrong on that.
 
later,
 
PaulH



From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Monday, 4 April 2005 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Buying some Polycom IP300s


Over the last few weeks/months I have been testing phones and ATAs from 
Grandstream (BT101, GXP2000, 286, 488), SNOM (190), Zyxel (Piece of Crap), 
Sipura (SPA-2000, SPA-841) and I personally feel that the Sipura SPA-841 is the 
best value, good quality phone that I have used. I haven't used the polycoms 
yet, but I plan to in the next few weeks.

 
 
 

- Original Message - 
From: Dan Morin mailto:[EMAIL PROTECTED]  
To: asterisk-users@lists.digium.com 
Sent: Sunday, April 03, 2005 10:22 AM
Subject: [Asterisk-Users] Buying some Polycom IP300s

Sorry =or the double post, I tried to paste and accidently sent the 
email
 
I've been playing with Asterisk for a few =eeks now, and I've gotten 
everything to work well with softphones, so I'm ready to =ove on to normal VoIP 
phones.  I've been looking around and reading =omments that people have had, 
and I was convinced that the Polycom IP300 was a great =hone for a good price.  
But, then I ran into this page, which has been =pdate in the last few days:
 
http://w=w.voip-info.org/wiki-Polycom+SoundPoint+IP+500 
http://www.voip-info.org/wiki-Polycom+SoundPoint+IP+500 =DIV 
 
The page in the wiki used to say that the =erson would not recomed 
Polycom phones to anyone.  So anyway, I just want to =ake sure that the IP300 
is a good choice.  I don't want to get cheap phones =hat aren't business 
quality, since I do play on using them for my business =fter testing.  Also, is 
the IP500 worth the extra money?  What can =t do that the IP300 can't.  And 
finally, will the IP300 do ulaw encoding?
 
Thanks in advance.



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RE: [Asterisk-Users] Asterisk@Home Question

[Kerry Garrison]

You would use the caller ID to route the call.
-Kerry
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Sunday, April 03, 2005 10:17 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] Question

 I was wondering if there is a way to select the outbound trunk based 
 on the extension that making the call.

Set the context in the sip.conf file for that user to a context in
extensions.conf that only has entries for dialing out through specific
providers.

Nabeel
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RE: [Asterisk-Users] Previous sip reload not yet done

[Nabeel Jafferali]

 Does anyone else have this problem? Is there a workaround?

Yeah, I had this problem when I added a lot of SIP register statements
and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the
problem. It seem * was getting stuck waiting for DNS lookups.

Nabeel
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Re: [Asterisk-Users] at-320 phone configuration difficulty

[clive]

Mishehu

try 19750407

Also to get palmtool to work you need to play with the debug 
settings on the phone first.

koltov
Clive






On 2 Apr 2005 at 0:30, I put the Who? in Mishehu wrote:

 Hi guys,
 
 I just got a Netweb 401 (AT-320) phone.  It came with firmware 1.38 on 
 it, and it has since been updated after failed attempts to configure, 
 and now has 1.42 (IAX2) from centrality (P1688S).  According to 
 voip-info, atcom's docs, etc, there are two passwords for it - one is 
 1234, and the superuser password is supposed to be 12345678.  Only 1234 
 works, and I get codec configuration, IP configuration, 
 firmware/ringtone/dialplan update options.  But nowhere do I find where 
 to set information about my asterisk box I want this phone to connect 
 to.  I've tried using Palmtool 1.42, and anytime I try to query the 
 phone's settings, I get Cannot connect to Palm1.  The person who sold 
 me sent no documentation or discs with it, and now on top of it, all the 
 buttons such as Local Num and Local IP are all switched around.  I am 
 very unhappy, and have wasted 4 hours already trying to work on this.  
 If anybody can assist, I'll be very grateful.
 
 -mishehu
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[Asterisk-Users] How to send email from the dial plan?

[Ronald Wiplinger]

I would like to get a notice by email, if we run out of gateways!
exten = _9011Z.,410,Busy
exten = _9011Z.,411,EMAIL    =  How to?
bye
Ronald
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[Asterisk-Users] error while compiling asterisk-1.0.7

[Kamran Ahmad]

gcc -shared -Xlinker -x -o cdr_odbc.so cdr_odbc.o
-lodbc  -L/usr/lib/pgsql
gcc -pipe  -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations -g 
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE  -O6
-march=i686 -DASTERISK_VERSION=\1.0.7\
-DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\
-DASTLIBDIR=\/usr/lib/asterisk\
-DASTVARLIBDIR=\/var/lib/asterisk\
-DASTVARRUNDIR=\/var/run\
-DASTSPOOLDIR=\/var/spool/asterisk\
-DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\
-DBUSYDETECT_MARTIN  -fPIC-c -o cdr_tds.o
cdr_tds.c
cdr_tds.c: In function `mssql_connect':
cdr_tds.c:415: error: `TDSCONNECTINFO' undeclared
(first use in this function)
cdr_tds.c:415: error: (Each undeclared identifier is
reported only once
cdr_tds.c:415: error: for each function it appears
in.)
cdr_tds.c:415: error: `connection' undeclared (first
use in this function)
cdr_tds.c:460: warning: implicit declaration of
function `tds_free_connect'
cdr_tds.c: At top level:
cdr_tds.c:71: warning: `connect_time' defined but not
used
make[1]: *** [cdr_tds.o] Error 1
make[1]: Leaving directory `/asterisk-1.0.7/cdr'
make: *** [subdirs] Error 1


helo

i am getting this error while compiling
asterisk-1.0.7. any expert tell me what is this.

regrads
Kamran



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Re: [Asterisk-Users] Time sync on PRI

[Tobias Jönsson]

On Thu, 31 Mar 2005, Peter Svensson wrote:
It would not be very hard to add both features to libpri. Libpri already 
has a function to decode and dump the time/date information. If I 
remember correctly the time/date IE should be added to the SETUP 
messages. I have been thinking about adding it, but have not had the 
time.
It's already there, in bristuff patches. Please encourage Digium to add 
Junghanns' patches to the asterisk code :)

--
Best Regards,
Tobias Jönsson, Lund SE___
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Re: [Asterisk-Users] How to send email from the dial plan?

[Olle E. Johansson]

Ronald Wiplinger wrote:
I would like to get a notice by email, if we run out of gateways!
exten = _9011Z.,410,Busy
exten = _9011Z.,411,EMAIL    =  How to?
  -= Info about application 'System' =-
[Synopsis]:
Execute a system command
[Description]:
  System(command): Executes a command  by  using  system(). Returns -1 on
failure to execute the specified command. If  the command itself executes
but is in error, and if there exists a priority n + 101, where 'n' is the
priority of the current instance, then  the  channel  will  be  setup  to
continue at that priority level.  Otherwise, System returns 0.
/O
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[Asterisk-Users] Zaptel group members - dial out on a availible port via trial and error?

[Etienne Pretorius]

Hi ya-all.
Little question that has been bothering me somewhot.
Say I have only 2 out going analog phone lines.
Some1 in the office decides to call their a client...
so the Dial command it using a group and it will start at the first Zap 
channel listed
in the group.

But now what if I disconnect the that line and he dials up again... then 
it tries the first
Zap channel again - but why does it not time out or try the other Zap 
channel?
Btw - I am using a TDM400P with the FXS ports on channel 3 and 4;

*Asterisk*
Urgent handler
   -- Executing Dial(Zap/1-1, Zap/g2/$EXTEN) in new stack
Urgent handler
Urgent handler
   -- Called g2/$EXTEN
Urgent handler
   -- Zap/3-1 answered Zap/1-1
Urgent handler
   -- Attempting native bridge of Zap/1-1 and Zap/3-1
Urgent handler
[And it hangs...]
*extensions.conf*
[outgoing]  ;Dial 
0 on the phone for external line
   ;SIP 
Phones need another was... they act like a cell phone

exten = _0,1,Dial(Zap/g2/$EXTEN)
;exten = _0,2,NoOp(DIALSTATUS=${DIALSTATUS})
exten = _0,3,Goto(_0-${DIALSTATUS},1)  ;Jump 
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten = _0-ANSWER,1,Goto(_0,102)
exten = _0-.,1,Goto(_0,1)  ;Try 
another line

exten = _0,102,Congestion
exten = _0,103,Hangup
Ps - I cant get a log to see the DIALSTATUS. Also I would have 
expected that Asterisk would then just try the other group member but
it does not Any help in this regard would be greatly apricaited.

--
Kind Regards
Etienne
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Re: [Asterisk-Users] Previous sip reload not yet done

[administrator tootai]

Nabeel Jafferali a écrit :
Does anyone else have this problem? Is there a workaround?
   

Yeah, I had this problem when I added a lot of SIP register statements
and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the
problem. It seem * was getting stuck waiting for DNS lookups.
 

Thanks. We put everywhere it was accepted the IP address and will see. 
FYI, sipgate.de doesn't accept to register with IP address. CLI SIP 
reload command is now applied much faster as with FQDNs in sip.conf
--
Daniel
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Re: [Asterisk-Users] Q.931 to SIGTRAN interface

[Dinesh Nair]

On 04/02/05 10:11 Mike Mueller said the following:
I don't think an Asterisk box can generate enough calls to cause sockets
related performance penalties.  Five packets per phone call.  What's the
max call rate an Asterisk box can support?
i think that would require an OS dependent answer.
but generally, i'm more than interested in contributing resources towards 
an open sourced implementation of SS7 with asterisk.

--
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo The opinions here in no way reflect the opinions of my $a $b.  |
| done; done  |
+=+
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Re: [Asterisk-Users] Sangoma VS. Digium

[Dinesh Nair]

On 04/01/05 00:00 Matthew Boehm said the following:
Steve Underwood wrote:

And your EU bias is clearly demonstrated by this. I've never seen a
BRI product outside he EU. :-)

Come to Houston, TX. We were running a BRI for quite some time before
upgrading to a T1.

ahem, ISDN BRIs are fairly common here in asia too. but i guess that asia 
don't count now, does it ? :)

--
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo The opinions here in no way reflect the opinions of my $a $b.  |
| done; done  |
+=+
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Re: [Asterisk-Users] problems with call-forward from ccme to * on sip trunk

[Andrea Riela]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Uhmm ...
maybe a connection plar from ccme to an * number (like 511 on my conf), 
then a simple forward from 511 to 601 on ccme?
Something like:

exten = _511,1,Dial(SIP/601,45)
I need help ... :D
Andrea
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (Darwin)
iD8DBQFCUPWwMakHrsrHP9wRAuSVAKCMyKYIVSP8B+Tc0losELtmJovsEQCcDoOi
gp1ZxZqe+G9hdAK6nEoqlaI=
=D68e
-END PGP SIGNATURE-
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RE: [Asterisk-Users] how to configure groups using a sip phone

[Deepak Dhiman]

Hi Bacon
Thanks for the quick response.
Actually I want to confirm that whether it is possible to divide logical
channels into group just like physiacl channels in zapata.

Deepak Dhiman

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Monday, April 04, 2005 12:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] how to configure groups using a sip phone


Can you be more specific?

What are you trying to achieve with the creation of such groups?


- Original Message - 
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 3:50 PM
Subject: [Asterisk-Users] how to configure groups using a sip phone


 hi friends !

 i am facing a problem from one week and now required ur help urgently.

 Actually, i want to configure asterisk for two groups javgroup and 
 linuxgroup. i also have constraint to use only sip phone (esatara ). 
 now, please help me is it possible to configure astersik in that way 
 or that kind of facility is given in zapata.conf.
 tell me in detail abt the configurations of the sip.conf and 
 extensions.conf.
 thanks
 Deepak Dhiman
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Re: [Asterisk-Users] Previous sip reload not yet done

[Olle E. Johansson]

administrator tootai wrote:
Nabeel Jafferali a écrit :
Does anyone else have this problem? Is there a workaround?
  

Yeah, I had this problem when I added a lot of SIP register statements
and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the
problem. It seem * was getting stuck waiting for DNS lookups.
 

Thanks. We put everywhere it was accepted the IP address and will see. 
FYI, sipgate.de doesn't accept to register with IP address. CLI SIP 
reload command is now applied much faster as with FQDNs in sip.conf

Changing register= statements to IP addresses is a bad idea. SIP is 
domain name based and (as proved by sipgate) an IP address points to 
*one* host, whereas a SIP domain by using SRV records can point to many 
IP addresses and servers. There's a huge difference between sending a 
REGISTER to [EMAIL PROTECTED] and [EMAIL PROTECTED]

See this as a short time fix. We need to make a better solution on the 
REGISTER parsing to prevent this from happening, it's clearly a bug.

/O
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Re: [Asterisk-Users] problems with call-forward from ccme to * on sip trunk

[Andrea Riela]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Apr 4, 2005, at 10:07 AM, Andrea Riela wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Uhmm ...
maybe a connection plar from ccme to an * number (like 511 on my 
conf), then a simple forward from 511 to 601 on ccme?
Something like:

exten = _511,1,Dial(SIP/601,45)
Ok, it works with this workaround ...
But it's a workaround .. I hope some expert could help me to configure 
* correctly :)
Regards
Andrea
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (Darwin)

iD8DBQFCUP4IMakHrsrHP9wRAv5JAKCGor2S+v45KOs1g1mZ6iJiDWUgSQCgrupc
z3UGbpMfaUbZf2ROxdxuW4U=
=Rdk2
-END PGP SIGNATURE-
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Re: [Asterisk-Users] Previous sip reload not yet done

[administrator tootai]

Olle E. Johansson a écrit :
administrator tootai wrote:
Nabeel Jafferali a écrit :
Does anyone else have this problem? Is there a workaround?
  

Yeah, I had this problem when I added a lot of SIP register statements
and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved 
the
problem. It seem * was getting stuck waiting for DNS lookups.
 

Thanks. We put everywhere it was accepted the IP address and will 
see. FYI, sipgate.de doesn't accept to register with IP address. CLI 
SIP reload command is now applied much faster as with FQDNs in 
sip.conf

Changing register= statements to IP addresses is a bad idea. SIP is 
domain name based and (as proved by sipgate) an IP address points to 
*one* host, whereas a SIP domain by using SRV records can point to 
many IP addresses and servers. There's a huge difference between 
sending a REGISTER to [EMAIL PROTECTED] and [EMAIL PROTECTED]

See this as a short time fix. We need to make a better solution on the 
REGISTER parsing to prevent this from happening, it's clearly a bug.
Well noticed. Should I concider bugs #3850 and #3933 including this 
matter or should I open a new one?
--
Daniel
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Re: [Asterisk-Users] Sangoma VS. Digium

[Michael Bielicki]

BRI's are in use in roughly 2/3 of the world with the US and I think
China being the main exceptions.

On Apr 4, 2005 9:37 AM, Dinesh Nair [EMAIL PROTECTED] wrote:
 
 
 On 04/01/05 00:00 Matthew Boehm said the following:
  Steve Underwood wrote:
 
 
 And your EU bias is clearly demonstrated by this. I've never seen a
 BRI product outside he EU. :-)
 
 
  Come to Houston, TX. We were running a BRI for quite some time before
  upgrading to a T1.
 
 ahem, ISDN BRIs are fairly common here in asia too. but i guess that asia
 don't count now, does it ? :)
 
 --
 Regards,   /\_/\   All dogs go to heaven.
 [EMAIL PROTECTED](0 0)http://www.alphaque.com/
 +==oOO--(_)--OOo==+
 | for a in past present future; do|
 |   for b in clients employers associates relatives neighbours pets; do   |
 |   echo The opinions here in no way reflect the opinions of my $a $b.  |
 | done; done  |
 +=+
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-- 
Michal Bielicki
http://www.aefirion.org/
http://www.asterisk.com.pl/
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RE: [Asterisk-Users] Router with QoS recommendations

[Shaoul Jacobson - TELLINK]

Hi,

QoS is nice (and important) but only works within a FULLY controlled end to
end link.

Inside a BIG enterprise LAN, on leased lines its OK.
Using end to end MPLS should also be ok
Mind that some provider sell MPLS but it is not their own MPLS end to end.
Going from one provider on MPLS to another on MPLS, you lose all the
benefits. No control.

Using the World Wide Wait (Internet) it will not help.

A waste of money.
My 2 cents.



Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel :   +32 3 201 96 36
Fax :   +32 3 227 09 81
e-mail  [EMAIL PROTECTED]


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[Asterisk-Users] Asterisk+Sipgate - just one step away..

[Razvan Cosma]

 Hello all,
I have a working Asterisk setup, also a working sipgate.co.uk account 
(tested with a GrandStream ATA 486), but got stuck in forwarding calls 
from local users to sipgate. Very frustrating, since I feel there's just 
one silly error somewhere.. story follows:
REGISTER both of the local user to * and of the * to sipgate.co.uk is 
successful
but when dialing some random phone number in Linphone in the form 
sip:[EMAIL PROTECTED] (1.2.3.4 is the * box) I get

   -- Executing SetCallerID(SIP/user-733d, [EMAIL PROTECTED]) 
in new stack
   -- Executing Dial(SIP/user-733d, 
SIP/[EMAIL PROTECTED]|30|tr) in new stack
Outgoing Call for 
 is not a local user
   -- Called [EMAIL PROTECTED]
Failed to authenticate on INVITE to ''[EMAIL PROTECTED] 
sip:[EMAIL PROTECTED];tag=as319c47a2'
 this I think is the problem - while the call is redirected, the 
correct number is dialed, Asterisk says it changed the callerid, but 
yy is the local username and 1.2.3.4 is the * address, shouldnt' 
it be ''[EMAIL PROTECTED] ?

sip.conf:
[general]
register = :[EMAIL PROTECTED]/xx
[sipgate]
type=peer
username=
secret=pp
host=sipgate.co.uk
fromuser=
fromdomain=sipgate.co.uk
nat=no
authuser=
dtmfmode=info
context=incomingsipgate
context=default
insecure=very
canreinvite=yes
disallow=all
allow=ulaw
allow=alaw
extensions.conf:
[general]
static=yes
writeprotect=yes
[incomingsipgate]
exten = h,1,Hangup
exten = xxx,1,Dial(SIP/102,20,tr)
[sipgate]
exten = _9.,1,SetCallerID([EMAIL PROTECTED])
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
exten = _9.,3,Playback(invalid)
exten = _9.,4,Hangup
Any hints please?
Thank you very much
Razvan
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Re: [Asterisk-Users] Router with QoS recommendations

[tim panton]

On 4 Apr 2005, at 09:25, Shaoul Jacobson - TELLINK wrote:
Hi,
QoS is nice (and important) but only works within a FULLY controlled 
end to
end link.
Inside a BIG enterprise LAN, on leased lines its OK.
Using end to end MPLS should also be ok
Mind that some provider sell MPLS but it is not their own MPLS end to 
end.
Going from one provider on MPLS to another on MPLS, you lose all the
benefits. No control.
Using the World Wide Wait (Internet) it will not help.

A waste of money.
My 2 cents.
I'm not sure I totally agree. It is also useful if you control the 
narrowest pipe.
Take the example of several sub-offices joined to a head office PBX over
'public' ADSL lines. Let's say the company buys all the ADSL lines from 
the
same provider.
In such a set-up, the uplink side of the sub-office ADSL links are
likely to be the main bandwidth limit.
A well configured router there will slow outgoing email etc to preserve
the quality of current VOIP sessions.

Sure, the provider may have internal bandwidth constrictions, but
they are unlikely to kick in before the 256k up channel of
a typical ADSL.
Oh, and, the web and the internet are not the same thing.
Think like that and you'll forget mail. Which is a huge bandwidth
consumer, and can stand being delayed by a second or
two.
Tim.
http://www.westhawk.co.uk/
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[Asterisk-Users] Planet VIP 450

[Altus Snyman]

Good day all
Did someone get the planet VIP 450 working
Thanks
Altus

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Re: [Asterisk-Users] Detecting when a called mobile is not reachable?

[David John Walsh]

I guess I should have added that this is based on the European, and
specifically UK model, but I would have expected it to have been
deemed best practice by most operators.



On Apr 4, 2005 4:04 AM, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
 Rod Bacon wrote:
  This is quite interesting.
 
  I tested calls to 2 mobiles that I knew were off, and not diverted to
  voicemail. 1 with Telstra, the other with vodafone (I'm in Australia).
  Via ISDN, both calls were shown as unanswered by asterisk. When the
  calls went to voicemail, the call was deemed to be answered.
 
  Via analogue circuits, the call is shown as answered, no matter what.
 
 That's what I would expect.
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Re: [Asterisk-Users] Asterisk+Sipgate - just one step away..

[administrator tootai]

Razvan Cosma a écrit :
 Hello all,
I have a working Asterisk setup, also a working sipgate.co.uk account 
(tested with a GrandStream ATA 486), but got stuck in forwarding calls 
from local users to sipgate. Very frustrating, since I feel there's 
just one silly error somewhere.. story follows:
REGISTER both of the local user to * and of the * to sipgate.co.uk is 
successful
but when dialing some random phone number in Linphone in the form 
sip:[EMAIL PROTECTED] (1.2.3.4 is the * box) I get

   -- Executing SetCallerID(SIP/user-733d, [EMAIL PROTECTED]) 
in new stack
   -- Executing Dial(SIP/user-733d, 
SIP/[EMAIL PROTECTED]|30|tr) in new stack
Outgoing Call for 
 is not a local user
   -- Called [EMAIL PROTECTED]
Failed to authenticate on INVITE to ''[EMAIL PROTECTED] 
sip:[EMAIL PROTECTED];tag=as319c47a2'
 this I think is the problem - while the call is redirected, the 
correct number is dialed, Asterisk says it changed the callerid, but 
yy is the local username and 1.2.3.4 is the * address, 
shouldnt' it be ''[EMAIL PROTECTED] ?

sip.conf:
[general]
register = :[EMAIL PROTECTED]/xx
[sipgate]
type=peer
username=
secret=pp
host=sipgate.co.uk
fromuser=
fromdomain=sipgate.co.uk
nat=no
authuser=
dtmfmode=info
context=incomingsipgate
context=default
insecure=very
canreinvite=yes
disallow=all
allow=ulaw
allow=alaw
extensions.conf:
[general]
static=yes
writeprotect=yes
[incomingsipgate]
exten = h,1,Hangup
exten = xxx,1,Dial(SIP/102,20,tr)
[sipgate]
exten = _9.,1,SetCallerID([EMAIL PROTECTED])
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
exten = _9.,3,Playback(invalid)
exten = _9.,4,Hangup
Any hints please?
according to your sip.conf, should be
[...]
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
in extensions.conf
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[Asterisk-Users] Manipulation based on SIP extension

[Irakli Natsvlishvili]

Hello there,
How do I configure any type of action based caller's extension and dialed 
number? For example if someone on extension 1777 calls extension 1777 this 
should be treated as accessing his voicemail box, so he won't need to call 
voicemail and entering mailbox number and password.

I.N. 

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[Asterisk-Users] Difficulty in configuring Asterisk to ensure that Call Transfers (SIP phone) are properly recorded and billable

[Peter Dean]

I am hoping someone in the * community has come across this problem before.

Problem:

Person SIP Phone A (SIPA)
Person SIP Phone B (SIPB)
SIP Phone C (SIPC PSTN Line)

SIPA calls a billable phone number via SIPC
exten = _123456/_1XX,1,SetAccount(${ACCOUNTCODE_COMPANYZ})
exten = _123456/_1XX,2,SetAMAFlags(billing) 
exten = _123456/_1XX,3,Macro(spa3k_pstn_out,${EXTEN}) 

Then SIPA, whom has generated charges decides to transfer the call the
COMPANYY, and this is where the billing traceable problems begin;

SIPA, transfers the call to SIPB, and the transaction information
appears to be incorrect;

a) SIPA is recorded as connecting with SIPB, and not SIPC.
b) SIPC is then recorded as connecting with SIPB.

Is there a way of ensuring that;
a) SIPA transaction is recorded as SIPA called SIPC.
b) SIPB transaction is recorded as SIPB called SIPC.

A simular scenerio is also happening the other way round. i.e. outside
dialling in via SIPC, SIPA answers, then transfers to SIPB.

Also I need to ensure that the accountcode  billing flags are
correctly set when the call transfer has occurred, but I have not be
sucessful as yet as I am lacking a bit of experience with the inner
workings of configuration.

If there is someone whom has experienced this problem before or
something simular, I would be interested in knowing how you managed to
resolve it.

- Peter

Info:
--
Asterisk:   v1.0.7
SIPA  B: Polycom SoundPoint IP 300
SIPC:Sipura 3000
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Re: [Asterisk-Users] Authenticating username

[David John Walsh]

Nabeel,

Could you expand on your comments, or provide a link / paste in a
sample extensions.conf to show how this would be set up?

David

On Apr 4, 2005 12:57 AM, Nabeel Jafferali [EMAIL PROTECTED] wrote:
  Dial(SIP/904)calls whoever logged on as john.
 
 You could define a variable in extensions.conf.
 
 Nabeel
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[Asterisk-Users] Realtime voicemail

[Ronald Wiplinger]

I tried to use ONE entry of my voicemail.conf to put into the database:
[other]
;602=1357,Ronald Wiplinger 602,[EMAIL PROTECTED]
INSERT INTO `voicemail_users` ( `uniqueid` , `customer_id` , `context` , 
`mailbox` , `password` , `fullname` , `email` , `pager` , `stamp` , 
`attach` , `saycid` , `hidefromdir` )
VALUES ('1', '602', 'other', '602', '3525', 'Ronald Wiplinger', 
'[EMAIL PROTECTED]', '', NOW( ) , 'no', 'yes', 'no')

extconfig.conf includes:
voicemail = mysql,astconf,voicemail_users
*CLI reload
-- Executing VoiceMail(SIP/601-a9a3, b602) in new stack
Apr  4 17:48:34 WARNING[18977]: app_voicemail.c:2227 leave_voicemail: No 
entry in voicemail config file for '602'

What do I miss?
bye
Ronald
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Re: [Asterisk-Users] Music On Hold and ATA-186 w/Silence Supression

[Irakli Natsvlishvili]

Hello, Alejandro!
AG I have a problem with ATA-186 configured for silence supression
Don't!
I.N.
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[Asterisk-Users] Difficulty in configuring Asterisk to ensure that Call Transfers (SIP phone) are properly recorded and billable

[Peter Dean]

I am hoping someone in the * community has come across this problem before.

Problem:

Person SIP Phone A (SIPA)
Person SIP Phone B (SIPB)
SIP Phone C (SIPC PSTN Line)

SIPA calls a billable phone number via SIPC
   exten = _123456/_1XX,1,SetAccount(${ACCOUNTCODE_COMPANYZ})
   exten = _123456/_1XX,2,SetAMAFlags(billing)
   exten = _123456/_1XX,3,Macro(spa3k_pstn_out,${EXTEN})

Then SIPA, whom has generated charges decides to transfer the call the
COMPANYY, and this is where the billing traceable problems begin;

SIPA, transfers the call to SIPB, and the transaction information
appears to be incorrect;

a) SIPA is recorded as connecting with SIPB, and not SIPC.
b) SIPC is then recorded as connecting with SIPB.

Is there a way of ensuring that;
a) SIPA transaction is recorded as SIPA called SIPC.
b) SIPB transaction is recorded as SIPB called SIPC.

A simular scenerio is also happening the other way round. i.e. outside
dialling in via SIPC, SIPA answers, then transfers to SIPB.

Also I need to ensure that the accountcode  billing flags are
correctly set when the call transfer has occurred, but I have not be
sucessful as yet as I am lacking a bit of experience with the inner
workings of configuration.

If there is someone whom has experienced this problem before or
something simular, I would be interested in knowing how you managed to
resolve it.

Info:
--
Asterisk:   v1.0.7
SIPA  B: Polycom SoundPoint IP 300
SIPC:Sipura 3000
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Re: [Asterisk-Users] Asterisk+Sipgate - just one step away..

[Razvan Cosma]

On 04/04/2005 12:46 PM, administrator tootai wrote:
according to your sip.conf, should be
[...]
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
in extensions.conf
Ye :) Thank you very much!
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RE: [Asterisk-Users] Router with QoS recommendations

[Shaoul Jacobson - TELLINK]

Hi,

 I'm not sure I totally agree.
Good, we do agree on some :)
I also agree with some of your remarks
(no flame war)

 It is also useful if you control the narrowest pipe.
I agree. But I disagree about the definition of the narrowest pipe.

 A well configured router there will slow outgoing email etc 
 to preserve the quality of current VOIP sessions.
agreed

 Let's say the company buys all the ADSL lines from 
 the same provider.
Buying all connection to the same provider is a wise decision.
It does not give any guaranty but this can be discussed :)
You are also a bigger customer.
So you could negociate some QoS, sla, ... (read my thought after my sig)

Most broadband (cable, xdsl) connection should provide enough bandwidth.
If you use 70% or more of your bandwidth then I agree QoS will definitively
help. (look during peaks  for each up  down link)
Otherwise, not much.

You share the bandwidth with other customers on your provider's backbone.
And your ISP decides how to shape traffic. Some VoIP providers in the US are
suing some ISP's because their VoIP traffic is degraded. 

The situation can be even worse with a cable connection as you share the
bandwidth AT your end-point not at the backbone.


Regards, 

Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel :   +32 3 201 96 36
Fax :   +32 3 227 09 81
e-mail  [EMAIL PROTECTED]

PS

I have worked in close relations with some 'big' providers.
They accept sla's, backup circuits even when they know they cannot provide.
The customer is billed for this extra 'service'
Extra billing is the only extra service the customer gets.
Beside the false safety he things he got.

If an accident happens, the isp pays for the lack of service.
This is far cheaper than implementing the needed technology.

I won't give names here, but this was the ways at some big international
isp's, not a small local isp.

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[Asterisk-Users] Supervised transfer problems

[Daniele Gallina - 3P System S.r.l.]

Hi all, when I try to transfer a call asterisk say me:

-- Executing SetCallerID(SIP/20012-cb87, Gallina Daniele
20012) in new stack
-- Executing Dial(SIP/20012-cb87, SIP/20013) in new stack
-- Called 20013
-- SIP/20013-034d is ringing
-- SIP/20013-034d answered SIP/20012-cb87
-- Attempting native bridge of SIP/20012-cb87 and SIP/20013-034d
-- Started music on hold, class '3psystem', on SIP/20013-034d
Apr  4 12:12:39 NOTICE[4710]: chan_sip.c:5161 get_refer_info: Supervised
transfer requested, but unable to find callid
'[EMAIL PROTECTED]'
Apr  4 12:12:40 NOTICE[4710]: chan_sip.c:5161 get_refer_info: Supervised
transfer requested, but unable to find callid
'[EMAIL PROTECTED]'
Apr  4 12:12:42 NOTICE[4710]: chan_sip.c:5161 get_refer_info: Supervised
transfer requested, but unable to find callid
'[EMAIL PROTECTED]'

Why? Any ideas?

Daniele


-- 
Daniele Gallina
3P System S.r.l. - Software Developer
Web: http://www.3psystem.net
E-Mail: [EMAIL PROTECTED]
Tel: 041.8626401 Scelta 2
Fax: 041.5161655


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[Asterisk-Users] SIP Jitter buffer

[1 2]

Hi 

I am using CVS latest

Is it correct there is no jitter buffer for SIP (RTP)

Are there any plans for this?

prob a stupid question:
Is it required / do the endpoints handle this - if the
src and destination are both SIP and there is no
transcoding but asterisk is still in the media path? 

Thanks

Jack

__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
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[Asterisk-Users] How do you do Line Hunting in Asterisk?

[Etienne Pretorius]

I have come accoross the fact that * can't handle if there is no 
dialtone
So out of interist, can you do Line hunting in * in a sequencial manner 
and can you
also do so in a random fasion?

--
Kind Regards
Etienne

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Re: [Asterisk-Users] Time sync on PRI

[Peter Svensson]

On Mon, 4 Apr 2005, Tobias Jönsson wrote:

 On Thu, 31 Mar 2005, Peter Svensson wrote:
  It would not be very hard to add both features to libpri. Libpri already 
  has a function to decode and dump the time/date information. If I 
  remember correctly the time/date IE should be added to the SETUP 
  messages. I have been thinking about adding it, but have not had the 
  time.
 
 It's already there, in bristuff patches. Please encourage Digium to add 
 Junghanns' patches to the asterisk code :)

Since Junhhanns will not assign a transferable / resellable license to 
Digium and Digium will not accept any code that is not under such a 
license there is a bit of a stalemate.

The Junghanns patch is not available for cvs head either.

Peter

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[Asterisk-Users] Zap - What is going on?

[Etienne Pretorius]

Ok - I was told that you set a group for Zap channels.
So I tried to make use of my Zap channels so the 2 I am interisted 
in is channel 3 and channel 4.
I make Channel 3 in use bu calling a line... then I try to call another 
line so expecting to have Zap channel 4
open and allowing me to make a call, but it just keeps on ringing... and 
then times out. Can anyone please
shed some light on this for me?

extensions.conf
[outgoing]  ;Dial 
0 on the phone for external line
   ;SIP 
Phones need another way... they act like a cell phone
exten = _0,1,Dial(Zap/g2/$EXTEN,20,tr) ;Try 
finding a line...
exten = _0,2,Goto(_0-${DIALSTATUS},1)  ;Jump 
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten = _0-ANSWER,1,Goto(_0,102)
exten = _0-.,1,Goto(_0,1)  ;Try 
another line

exten = _0,102,Congestion
exten = _0,103,Hangup
Asterisk Console:
   -- Starting simple switch on 'Zap/1-1'
   -- Executing Dial(Zap/1-1, Zap/g2/$EXTEN|20|tr) in new stack
   -- Called g2/$EXTEN
   -- Nobody picked up in 2 ms
   -- Hungup 'Zap/4-1'
   ===That channel is free and 
has a seperate phone line connected to it.
   -- Executing Goto(Zap/1-1, _0-NOANSWER|1) in new stack

--
Kind Regards
Etienne

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Re: [Asterisk-Users] Asterisk@Home Question

[Tony Davidson]

I'd think about using a prefix for each trunk as a form of password.  At 
home I have to dial 1 then the number to use one of my trunks, or 2 
then the number for a different trunk.  If you gave them a code of say 
666 they would have to dial that then the number.  If you had a code for 
your trunk they wouldn't be able to use your trunk unless they knew the 
code.

Probably more elegant solutions but just a quick suggestion.
tony
I am using [EMAIL PROTECTED] 0.8
I was wondering if there is a way to select the outbound trunk based on the
extension that making the call.
Here is why I ask. Since I am already running my Asterisk server for my own
use, I also wanted to let friends and family in on the action but I don't
want to pay for their calls. So if I ask them to buy talk time from a
termination provider and then setup a separate trunk for them, how do I make
sure that only their calls use that outbound trunk?
Any ideas?
 

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Re: [Asterisk-Users] Manipulation based on SIP extension

[Henry Devito]

- Original Message - 
From: Irakli Natsvlishvili [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 4:52 AM
Subject: [Asterisk-Users] Manipulation based on SIP extension


Hello there,
How do I configure any type of action based caller's extension and dialed 
number? For example if someone on extension 1777 calls extension 1777 this 
should be treated as accessing his voicemail box, so he won't need to call 
voicemail and entering mailbox number and password.

I.N.
Try something like this
 exten = 1777,1,GotoIf($[${CALLERIDNUM} = 1777]?5:2)
 exten = 1777,2,Dial(SIP/177),15,rt
 exten = 1777,3,Voicemail(u${EXTEN})
 exten = 1777,4,Hangup
 exten = 1777,5,VoicemailMain(s${EXTEN})
 exten = 1777,6,Hangup
 Henry 

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[Asterisk-Users] Re: error while compiling asterisk-1.0.7

[Tony Mountifield]

In article [EMAIL PROTECTED],
Kamran Ahmad [EMAIL PROTECTED] wrote:
 gcc -shared -Xlinker -x -o cdr_odbc.so cdr_odbc.o
 -lodbc  -L/usr/lib/pgsql
 gcc -pipe  -Wall -Wstrict-prototypes
 -Wmissing-prototypes -Wmissing-declarations -g 
 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE  -O6
 -march=i686 -DASTERISK_VERSION=\1.0.7\
 -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\
 -DASTLIBDIR=\/usr/lib/asterisk\
 -DASTVARLIBDIR=\/var/lib/asterisk\
 -DASTVARRUNDIR=\/var/run\
 -DASTSPOOLDIR=\/var/spool/asterisk\
 -DASTLOGDIR=\/var/log/asterisk\
 -DASTCONFPATH=\/etc/asterisk/asterisk.conf\
 -DASTMODDIR=\/usr/lib/asterisk/modules\
 -DASTAGIDIR=\/var/lib/asterisk/agi-bin\
 -DBUSYDETECT_MARTIN  -fPIC-c -o cdr_tds.o
 cdr_tds.c
 cdr_tds.c: In function `mssql_connect':
 cdr_tds.c:415: error: `TDSCONNECTINFO' undeclared
 (first use in this function)
 cdr_tds.c:415: error: (Each undeclared identifier is
 reported only once
 cdr_tds.c:415: error: for each function it appears
 in.)
 cdr_tds.c:415: error: `connection' undeclared (first
 use in this function)
 cdr_tds.c:460: warning: implicit declaration of
 function `tds_free_connect'
 cdr_tds.c: At top level:
 cdr_tds.c:71: warning: `connect_time' defined but not
 used
 make[1]: *** [cdr_tds.o] Error 1
 make[1]: Leaving directory `/asterisk-1.0.7/cdr'
 make: *** [subdirs] Error 1
 
 
 helo
 
 i am getting this error while compiling
 asterisk-1.0.7. any expert tell me what is this.

Looks like it is trying to compile cdr_tds without the correct version
of FreeTDS being installed. The makefile is finding /usr/include/tds.h
or /usr/local/include/tds.h, which tells it to compile cdr_tds.c

If you don't need to log CDRs to a MSSQL or Sybase server, the easiest
solution if to comment out the two MODS+= lines under the heading
FreeTDS stuff.

If you do need it, then you will probably have to get a newer FreeTDS.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [Asterisk-Users] Zap - What is going on?

[Rob Scott]

For a start it should be 

${EXTEN}

You have to realize that ALL variables look like that.
Dollar-open-curly-brackets-variablename-close-curly-brackets.

So it didn't see your text as a variable and it tried to call the number
$EXTEN on Zap/g2.
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Etienne
Pretorius
Sent: 04 April 2005 13:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Zap - What is going on?

Ok - I was told that you set a group for Zap channels.

So I tried to make use of my Zap channels so the 2 I am interisted
in is channel 3 and channel 4.
I make Channel 3 in use bu calling a line... then I try to call another
line so expecting to have Zap channel 4 open and allowing me to make a
call, but it just keeps on ringing... and then times out. Can anyone
please shed some light on this for me?

extensions.conf

[outgoing]  ;Dial 
0 on the phone for external line
;SIP
Phones need another way... they act like a cell phone
exten = _0,1,Dial(Zap/g2/$EXTEN,20,tr) ;Try 
finding a line...
exten = _0,2,Goto(_0-${DIALSTATUS},1)  ;Jump 
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten = _0-ANSWER,1,Goto(_0,102)
exten = _0-.,1,Goto(_0,1)  ;Try 
another line

exten = _0,102,Congestion
exten = _0,103,Hangup

Asterisk Console:

-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, Zap/g2/$EXTEN|20|tr) in new stack
-- Called g2/$EXTEN
-- Nobody picked up in 2 ms
-- Hungup 'Zap/4-1'
===That channel is free and
has a seperate phone line connected to it.
-- Executing Goto(Zap/1-1, _0-NOANSWER|1) in new stack


-- 
Kind Regards
Etienne



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Re: [Asterisk-Users] Previous sip reload not yet done

[Cirelle Internet Products]

administrator tootai wrote:
Olle E. Johansson a écrit :
administrator tootai wrote:
Nabeel Jafferali a écrit :
Does anyone else have this problem? Is there a workaround?
  


Yeah, I had this problem when I added a lot of SIP register statements
and SIP peers. Changing the hostnames (FQDNs) to IP addresses 
solved the
problem. It seem * was getting stuck waiting for DNS lookups.
 

Thanks. We put everywhere it was accepted the IP address and will 
see. FYI, sipgate.de doesn't accept to register with IP address. CLI 
SIP reload command is now applied much faster as with FQDNs in 
sip.conf


Changing register= statements to IP addresses is a bad idea. SIP is 
domain name based and (as proved by sipgate) an IP address points to 
*one* host, whereas a SIP domain by using SRV records can point to 
many IP addresses and servers. There's a huge difference between 
sending a REGISTER to [EMAIL PROTECTED] and [EMAIL PROTECTED]

See this as a short time fix. We need to make a better solution on 
the REGISTER parsing to prevent this from happening, it's clearly a bug.

Well noticed. Should I concider bugs #3850 and #3933 including this 
matter or should I open a new one?
We had the same problem, on two different hardware platforms.
2 flavors of pentium 4/board combos
grandstream and sipura (handset/ata) devices
the only thing that has worked for us was to eliminate the
registration process all together. This has been going on since
last October that I am aware of which means it has been in every
cvs since then.
sip.conf
host=device ip (not dynamic)
qualify=yes
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[Asterisk-Users] Sending faxes and call accounting

[Chris Mason (Lists)]

In the Asterisk system I am testing for implementation at a small
luxury resort, there are four fax machines that the guests can use for sending
and receiving faxes. Because they require confidentiality, we cannot use
hylafax or other method than a stand alone fax.

I would just connect these faxes to the PSTN lines directly
but we would then have call accounting issues as the calls would not appear in
the CDRs and with long distance costing from $1/min upwards, it could get
costly quickly.

How reliably can I do an analogue in/out connection with
call accounting? I am using TDM400 cards or an Adtrans 600 channel bank.



Thanks



Chris






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Re: [Asterisk-Users] Zap - What is going on?

[Etienne Pretorius]

Dope --- *sheepish grin*. Sorry. Thanks for the help.
Kind Regards
Etienne
Technical Support
Kingsley Technologies
Rob Scott wrote:
For a start it should be 

${EXTEN}
You have to realize that ALL variables look like that.
Dollar-open-curly-brackets-variablename-close-curly-brackets.
So it didn't see your text as a variable and it tried to call the number
$EXTEN on Zap/g2.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Etienne
Pretorius
Sent: 04 April 2005 13:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Zap - What is going on?
Ok - I was told that you set a group for Zap channels.
So I tried to make use of my Zap channels so the 2 I am interisted
in is channel 3 and channel 4.
I make Channel 3 in use bu calling a line... then I try to call another
line so expecting to have Zap channel 4 open and allowing me to make a
call, but it just keeps on ringing... and then times out. Can anyone
please shed some light on this for me?
extensions.conf
[outgoing]  ;Dial 
0 on the phone for external line
   ;SIP
Phones need another way... they act like a cell phone
exten = _0,1,Dial(Zap/g2/$EXTEN,20,tr) ;Try 
finding a line...
exten = _0,2,Goto(_0-${DIALSTATUS},1)  ;Jump 
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten = _0-ANSWER,1,Goto(_0,102)
exten = _0-.,1,Goto(_0,1)  ;Try 
another line

exten = _0,102,Congestion
exten = _0,103,Hangup
Asterisk Console:
   -- Starting simple switch on 'Zap/1-1'
   -- Executing Dial(Zap/1-1, Zap/g2/$EXTEN|20|tr) in new stack
   -- Called g2/$EXTEN
   -- Nobody picked up in 2 ms
   -- Hungup 'Zap/4-1'
   ===That channel is free and
has a seperate phone line connected to it.
   -- Executing Goto(Zap/1-1, _0-NOANSWER|1) in new stack

 

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Re: [Asterisk-Users] SIP Jitter buffer

[Rich Adamson]

 I am using CVS latest
 
 Is it correct there is no jitter buffer for SIP (RTP)
 
 Are there any plans for this?
 
 prob a stupid question:
 Is it required / do the endpoints handle this - if the
 src and destination are both SIP and there is no
 transcoding but asterisk is still in the media path? 

My understanding is the new jitterbuffer code (in cvs-head) has 
been applied to iax connections, and the objective is to make it 
available for sip/rtp (and possibly other channel types) after 
things are cool in iax.

A jitterbuffer is only required when the delivery of rtp 
packets is inconsistent (eg, jerky).

Its my understanding that sip phones have at least some sort
of jitterbuffer built into firmware. Don't know how effective
they are for large variations in packet delivery though.


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[Asterisk-Users] Best way for nated sip peers thru a database

[Laurent FOULONNEAU]

Hello list,
Newbie questions
Seems that  nated sip peers/friends are not functional with RealTime 
because the database peers/users are not kept in memory.

On the other side the dynamic config  (MYSQL_FRIENDS)  system does not 
support the nat option.

Not sure  but may be ast_data is the good way for that , or may be thru 
radius with PortaOne's Radius client ?

Would like to know  what's the best way to have nated peers in a database 
instead of flat files.

Thanks in advance.
Laurent
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 01/04/2005
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Re: [Asterisk-Users] Previous sip reload not yet done

[administrator tootai]

[...]
See this as a short time fix. We need to make a better solution on 
the REGISTER parsing to prevent this from happening, it's clearly a 
bug.

Well noticed. Should I concider bugs #3850 and #3933 including this 
matter or should I open a new one?

We had the same problem, on two different hardware platforms.
2 flavors of pentium 4/board combos
grandstream and sipura (handset/ata) devices
the only thing that has worked for us was to eliminate the
registration process all together. This has been going on since
last October that I am aware of which means it has been in every
cvs since then.
Bug #3946 was open for this and Mantis.
--
Daniel
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Re: [Asterisk-Users] Supervised transfer problems

[Josiah Bryan]

On Monday 04 April 2005 6:23 am, Daniele Gallina - 3P System S.r.l. wrote:
 Hi all, when I try to transfer a call asterisk say me:

 -- Executing SetCallerID(SIP/20012-cb87, Gallina Daniele
 20012) in new stack
 -- Executing Dial(SIP/20012-cb87, SIP/20013) in new stack
 -- Called 20013
 -- SIP/20013-034d is ringing
 -- SIP/20013-034d answered SIP/20012-cb87
 -- Attempting native bridge of SIP/20012-cb87 and SIP/20013-034d
 -- Started music on hold, class '3psystem', on SIP/20013-034d
 Apr  4 12:12:39 NOTICE[4710]: chan_sip.c:5161 get_refer_info: Supervised
 transfer requested, but unable to find callid
 '[EMAIL PROTECTED]'

snip

I've been having the same 'problem' with my Polycom SoundPoint IP500 phone. 
When our receptionist hits transfer, ext, transfer - then I see a notice 
just as above. The odd thing is that the call ID is given as on the phone, 
not on the server. E.g. the IP in  ''[EMAIL PROTECTED] is the IP 
of the Polycom phone, not the * box. Is there any way to fix this? Rewrite 
sip headers? Any ideas?

-josiah

-- 
Josiah Bryan
IT Coordinator
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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[Asterisk-Users] Re: ASTCC question: Trunk LOCAL

[Ronald Wiplinger]

Darren Wiebe wrote:
That capability is not there yet.  I would personally recommend 
using the 'Local' channel and routing your calls via the 
extensions.conf file.  This is totally up to you but I find it gives 
me more flexibility.  That would also make it easier to do something 
like you are looking to do with the setgroup and checkgroup 
commands.

With the cards we have a field In Use. I would like to add this 
field to the TRUNK, so that I can jump to the next trunk instead, 
e.g., if I have several gateways available, but only a few ports at 
each gateway, than I need to jump to the next gateway. If I could add 
a field in Trunks, than I believe I could ask this field first, 
before I choose the trunk to dial, ... (not completely thought thru yet)

Can you tell me more about it, please. It sounds interesting!

In my extensions.conf I have the following lines:
[default-outgoing]
exten = _1NXXNXX,1,SetCIDNum(${CALLERIDNUM}|a)
exten = _1NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
...
Then I have a trunk setup for the default-outgoing context.  That 
way it uses the information above to route my calls.  I would 
recommend looking the the wiki for the group commands as I've only 
used them a little bit.


I tried this one, but it does not work!!!
In Trunks I put in:
1-800-xxx   Local  Line-optimize
I have a context [Line-optimize]
but the real world says:
* CLI
   -- AGI Script Executing Application: (DIAL) Options: 
(Local/011886228357765/Line-optimize|30|HL(59994:6:3))
   -- Limit Data:
   -- timelimit=59994
   -- play_warning=6
   -- play_to_caller=yes
   -- play_to_callee=no
   -- warning_freq=3
   -- start_sound=UNDEF
   -- warning_sound=timeleft
   -- end_sound=UNDEF
Apr  4 20:26:55 NOTICE[1487]: chan_local.c:436 local_alloc: No such 
extension/context [EMAIL PROTECTED] creating local channel
Apr  4 20:26:55 NOTICE[1487]: app_dial.c:936 dial_exec_full: Unable to 
create channel of type 'Local' (cause 0)
 == Everyone is busy/congested at this time (1:0/0/1)
   -- AGI Script astcc.agi completed, returning 0



Do you have any ideas???

bye
Ronald
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RE: [Asterisk-Users] Realtime mysql problem?

[Matt Schulte]

Sorry for the delay, do you have any clue when realtime will get added
to stable? I never did get this working but before I go too much further
I'd like to run production on a stable version..


I'll try out SIP today and let you know, the reason I'm using IAX is
because everything SIP we do is through SER. Not to mention since
realtime doesn't support qualify= and NAT mode must be manually set,
it's kind of pointless to use Asterisk for SIP. :-)

 Just for curiosity sake, have you tried any SIP RealTime stuff?
Perhaps this is an IAX problem? I remember  helping a guy a few weeks
ago get his SIP RealTime working. This is the first IAX I've dealt with.
And I have  no IAX stuff to test with.


-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, March 30, 2005 9:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime mysql problem?


Matt Schulte wrote:
 How do you toggle the realtime cache?

Check in the configs/iax.conf.sample file of a recent CVS download
and it should be in there.

 If there were too many fields
 in the table, could you foresee this being a problem?

No, because I have lots of extra company specific fields in my
sip_users table that asterisk doesn't use at all and I've had no
problems.

 ie iax users have peercontext and auth.

Just for curiosity sake, have you tried any SIP RealTime stuff?
Perhaps this is an IAX problem? I remember helping a guy a few weeks ago
get his SIP RealTime working. This is the first IAX I've dealt with. And
I have no IAX stuff to test with.

-Matthew

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Re: [Asterisk-Users] Router with QoS recommendations

[Bruno Hertz]

tim panton [EMAIL PROTECTED] writes:

 On 4 Apr 2005, at 09:25, Shaoul Jacobson - TELLINK wrote:

 Hi,

 QoS is nice (and important) but only works within a FULLY controlled
 end to
 end link.
 Inside a BIG enterprise LAN, on leased lines its OK.
 Using end to end MPLS should also be ok
 Mind that some provider sell MPLS but it is not their own MPLS end
 to end.
 Going from one provider on MPLS to another on MPLS, you lose all the
 benefits. No control.
 Using the World Wide Wait (Internet) it will not help.

 A waste of money.
 My 2 cents.


 I'm not sure I totally agree. It is also useful if you control the
 narrowest pipe.
 Take the example of several sub-offices joined to a head office PBX over
 'public' ADSL lines. Let's say the company buys all the ADSL lines
 from the
 same provider.
 In such a set-up, the uplink side of the sub-office ADSL links are
 likely to be the main bandwidth limit.
 A well configured router there will slow outgoing email etc to preserve
 the quality of current VOIP sessions.

 Sure, the provider may have internal bandwidth constrictions, but
 they are unlikely to kick in before the 256k up channel of
 a typical ADSL.

 Oh, and, the web and the internet are not the same thing.
 Think like that and you'll forget mail. Which is a huge bandwidth
 consumer, and can stand being delayed by a second or
 two.

 Tim.


I agree, especially qos on upstream might be beneficial, and surely
is in a cable modem setup. E.g. my modem has a 10 Mbit LAN interface,
but uplink is limited to 256Kbit. So when I have many things
going out, uplink will be much sooner saturated than the LAN link,
and cable modem buffers run full leading to looong latencies and
maybe even package loss. Putting a router before the modem shaping
the upstream traffic solves that problem.

Regards, Bruno.

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Re: [Asterisk-Users] Realtime mysql problem?

[Wojciech Tryc]

What is your problem with IAX in realtime? I have it working (finally).
Wojtek
- Original Message - 
From: Matt Schulte [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 9:01 AM
Subject: RE: [Asterisk-Users] Realtime mysql problem?

Sorry for the delay, do you have any clue when realtime will get added
to stable? I never did get this working but before I go too much further
I'd like to run production on a stable version..
I'll try out SIP today and let you know, the reason I'm using IAX is
because everything SIP we do is through SER. Not to mention since
realtime doesn't support qualify= and NAT mode must be manually set,
it's kind of pointless to use Asterisk for SIP. :-)
Just for curiosity sake, have you tried any SIP RealTime stuff?
Perhaps this is an IAX problem? I remember  helping a guy a few weeks
ago get his SIP RealTime working. This is the first IAX I've dealt with.
And I have  no IAX stuff to test with.
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 30, 2005 9:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime mysql problem?
Matt Schulte wrote:
How do you toggle the realtime cache?
   Check in the configs/iax.conf.sample file of a recent CVS download
and it should be in there.
If there were too many fields
in the table, could you foresee this being a problem?
   No, because I have lots of extra company specific fields in my
sip_users table that asterisk doesn't use at all and I've had no
problems.
ie iax users have peercontext and auth.
   Just for curiosity sake, have you tried any SIP RealTime stuff?
Perhaps this is an IAX problem? I remember helping a guy a few weeks ago
get his SIP RealTime working. This is the first IAX I've dealt with. And
I have no IAX stuff to test with.
-Matthew
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[Asterisk-Users] Asterisk@Home Question

[Jeff R Glassman]

Message: 9
Date: Sun, 3 Apr 2005 23:52:39 -0500
From: * KAPIL * [EMAIL PROTECTED]
Subject: [Asterisk-Users] [EMAIL PROTECTED] Question
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1

Greetings!

This is my first post to the list...and I'm kinda' new to Asterisk, so
please be kindI did a fair amount of Googling but was not able to
find an answer.

I am using [EMAIL PROTECTED] 0.8

I was wondering if there is a way to select the outbound trunk based
on the extension that making the call.

Here is why I ask. Since I am already running my Asterisk server for
my own use, I also wanted to let friends and family in on the action
but I don't want to pay for their calls. So if I ask them to buy talk
time from a termination provider and then setup a separate trunk for
them, how do I make sure that only their calls use that outbound
trunk?

Any ideas?
Set up a trunk for them

Set up a route for them.

Setup
Outbound Routing

Add Route

Name your Route

Dial Pattern



79|NXX
79|NXXNXX
79|1NXXNXX
79|011.

Trunk Sequence
Select there Trunk

MAKE SURE they dial 79 for all there calls

Jeff


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RE: [Asterisk-Users] Asterisk@Home Question

[Giles Coochey]

 Greetings!
 
 This is my first post to the list...and I'm kinda' new to Asterisk, so
 please be kindI did a fair amount of Googling but was not able to
 find an answer.
 
 I am using [EMAIL PROTECTED] 0.8
 
 I was wondering if there is a way to select the outbound trunk based
 on the extension that making the call.
 
 Here is why I ask. Since I am already running my Asterisk server for
 my own use, I also wanted to let friends and family in on the action
 but I don't want to pay for their calls. So if I ask them to buy talk
 time from a termination provider and then setup a separate trunk for
 them, how do I make sure that only their calls use that outbound
 trunk?
 

Being rather new myself, but the first thing I thought about your
problem was putting those extensions in a different context to the one
where you define your trunk calls.

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RE: [Asterisk-Users] Realtime mysql problem?

[Matt Schulte]

Well, I made several posts. Basically realtime works fine on the system
you register to, if you try to contact that peer from another Ast server
(running realtime), it does a SELECT query and all finds the peer and
continues to say Unable to contact peer as if the user doesn't exist.
I even went as far as packet sniffing and noticed it doesn't ever go out
on port 4569 or anything. Again, I've made several posts about this
before for full details. :-)

Thanks, Matt

-Original Message-
From: Wojciech Tryc [mailto:[EMAIL PROTECTED] 
Sent: Monday, April 04, 2005 8:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime mysql problem?


What is your problem with IAX in realtime? I have it working (finally).
Wojtek
- Original Message - 
From: Matt Schulte [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 9:01 AM
Subject: RE: [Asterisk-Users] Realtime mysql problem?


Sorry for the delay, do you have any clue when realtime will get added
to stable? I never did get this working but before I go too much further
I'd like to run production on a stable version..


I'll try out SIP today and let you know, the reason I'm using IAX is
because everything SIP we do is through SER. Not to mention since
realtime doesn't support qualify= and NAT mode must be manually set,
it's kind of pointless to use Asterisk for SIP. :-)

 Just for curiosity sake, have you tried any SIP RealTime stuff?
Perhaps this is an IAX problem? I remember  helping a guy a few weeks
ago get his SIP RealTime working. This is the first IAX I've dealt with.
And I have  no IAX stuff to test with.


-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 30, 2005 9:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime mysql problem?


Matt Schulte wrote:
 How do you toggle the realtime cache?

Check in the configs/iax.conf.sample file of a recent CVS download
and it should be in there.

 If there were too many fields
 in the table, could you foresee this being a problem?

No, because I have lots of extra company specific fields in my
sip_users table that asterisk doesn't use at all and I've had no
problems.

 ie iax users have peercontext and auth.

Just for curiosity sake, have you tried any SIP RealTime stuff?
Perhaps this is an IAX problem? I remember helping a guy a few weeks ago
get his SIP RealTime working. This is the first IAX I've dealt with. And
I have no IAX stuff to test with.

-Matthew

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Re: [Asterisk-Users] Router with QoS recommendations

[David McNett]

On 03-Apr-2005, Tim Pushor wrote:
 I prefer PF's approach to security first, convenience second, and I 
 *really* like the fact that PF has a real parser. As the requements get 
 more complex, having everything in one file, and very readable and 
 structured is a huge plus. Also, the integration with ALTQ is nice, 
 especially for these types of applications.

I agree with everything Tim wrote above, and I'll add that the biggest
factor that influenced me in my move to OpenBSD for my firewall was that
it was the only free unix I found that could do bidirectional filtering
in bridged mode.  As in, when you're in a bridged configuration you can 
filter in and out on an interface.  Neither Linux nor FreeBSD could do
this.  It's certainly an edge case, but if you need that feature it's
invaluable.

I posted my asterisk altq experiments here:
  http://slacker.com/~nugget/asterisk4.php

-- 
David McNett [EMAIL PROTECTED]
http://slacker.com/~nugget/
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[Asterisk-Users] PRI: received SETUP message for call that is not a new call, wicked!

[Mark Elkins]

Hi list, I'm getting the message...
Apr  4 15:13:09 WARNING[1069]: chan_zap.c:7512 zt_pri_error: PRI:
received SETUP message for call that is not a new call, wicked!!!

This is running Asterisk 1.0.6-BRIstuffed-0.2.0-RC7k.

These messages happen when someone calls from the Telco on a BRI line...
but rather than asterisk simply immediately answering, they just hear
ringing

So really the new call IS a new call - but Asterisk things differently.

Anyone met and/or solved this problem?
This seems to be happening to 1 in 4 of all my calls??? - other calls
are fine.

-- 
  .  . ___. .__  Posix Systems - Sth Africa.  e.164 VOIP ready
 /| /|   / /__   [EMAIL PROTECTED]  -  Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496

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Re: [Asterisk-Users] V92 modem with asterisk

[Glenn]

Alexandre Charles said:
 Hi everyone,
 I just install Linux and asterisk on one of my pc. I
 want to run some basic functionality tests. Is it possible to use a v92
 modem as a FXO or FXS card. If yes how do we configure and install the
 card? What are the commands?
 Thanks in advance for your help

Most modems cannot be used as FXO's, since there are no drivers. The
WC_FXO drivers works with SOME Intel chipsets. ebay: digium fxo, you'll
find compatible modems for about $7 + SH.

cheers,
glenn

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Re: [Asterisk-Users] Router with QoS recommendations

[Tim Pushor]

I'm not sure about QoS, but I do run ATLQ on FreeBSD/PF. In a SOHO 
environment where there is likely to be DSL or cable, I find it very 
useful (on the upload side at least, which is usually a problem on 
asyncrhonous connections).

I can max out my pipe and hear no effect of it on the phone.
Shaoul Jacobson - TELLINK wrote:
Hi,
QoS is nice (and important) but only works within a FULLY controlled end to
end link.
Inside a BIG enterprise LAN, on leased lines its OK.
Using end to end MPLS should also be ok
Mind that some provider sell MPLS but it is not their own MPLS end to end.
Going from one provider on MPLS to another on MPLS, you lose all the
benefits. No control.
Using the World Wide Wait (Internet) it will not help.
A waste of money.
My 2 cents.

Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel :   +32 3 201 96 36
Fax :   +32 3 227 09 81
e-mail  [EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk Realtime Capabilities

[Matthew Boehm]

Rod Bacon wrote:
 The term RTCache has never been mentioned in the WIKI or these
 forums. I assume that it's some sort of function to speed up realtime
 db access by keeping transactions in RAM and writing periodically? If
 so, I can understand why this would need to be flushed.

RealTime Cache is a mechanism written to allow RealTime SIP/IAX
peers/users to work with NAT and recieve MWI. When RTC is enabled, peer/user
info is retreived from the database and stored in same list as sip.conf
peers/users.

-Matthew

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[Asterisk-Users] X-Lite to Zap, no Voice on other phone!

[Etienne Pretorius]

Hello,
The problem is - and i was wandering if anyone knows the solution - is 
that When I dial from my windows machine,
to an external phone line through Zap, then the receiving party does not 
hear my voice - but when the receiving party
calls me back, then we have voice on both sides. What makes the more 
currious is that internal numbers work fine, both
sides have voice.

--
Kind Regards
Etienne

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Re: [Asterisk-Users] Realtime mysql problem?

[Matthew Boehm]

 do you have any clue when realtime will get added to stable?

It won't.

 Not to mention since
 realtime doesn't support qualify= and NAT mode must be manually set,

Have you been using RTC? (RealTime Cache) It fixes the NAT/MWI problem.

-Matthew
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[Asterisk-Users] IAXy audio troubles (only on INCOMING calls)

[niels]

Hello All!

I just got my IAXy.. Configured it.. Got it Up and Running 

Calls OUT have no problems (that means from IAXy - Asterisk -
ZAP/SIPclient/IAXclient) 

Calls IN do have problems (that means from ZAP/SIPclient/IAXclient -
Asterisk - IAXy) 

On those incoming calls on my IAXy I hear the other party on my IAXy,
But this other party can't hear me (the audio that's beeing sent from
the IAXy to asterisk can't be heard)

Does anyone have any idea what I can do about this?

Is this a common problem?

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Re: [Asterisk-Users] Best way for nated sip peers thru a database

[Matthew Boehm]

Laurent FOULONNEAU wrote:
 Hello list,

 Newbie questions

 Seems that  nated sip peers/friends are not functional with RealTime
 because the database peers/users are not kept in memory.

*sigh* I'm quoting this wiki page:
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Sip

quoteNOTE: As of CVS-HEAD 3/16/05, if you enable RealTime caching in
your sip.conf, Voicemail MWI works and so does 'sip show peers'./quote

This also enables NAT because using the cache, peers/users are kept in
memory.

 Would like to know  what's the best way to have nated peers in a
 database instead of flat files.

You could always do this for your sip.conf:

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


-Matthew

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[Asterisk-Users] Browser based configuration of Asterisk

[Chuck Bunn]

Hi,
I have a Linux Fedora 3 Asterisk only box (2 FXO  2 FXS ports) with no 
GUI or WEB server running. I can get to it remotely using Putty but I 
want to add the capability to at least do Dial Plan configuration via a 
browser. Do any of the GUI based configurations support such a setup. I 
really do not want to install a GUI on the Asterisk box but installing 
Apache would be okay... I googled (  :list.digium.com) around the 
mailing lists and found several boxes that do this (Mediatrix and Epygi) 
but I do want to have to pay for a box when I already have an old server 
converted.

Thanks
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Re: [Asterisk-Users] Router with QoS recommendations

[Arnaud PIGNARD]

At 15:36 04/04/2005, you wrote:
On 03-Apr-2005, Tim Pushor wrote:
 I prefer PF's approach to security first, convenience second, and I
 *really* like the fact that PF has a real parser. As the requements get
 more complex, having everything in one file, and very readable and
 structured is a huge plus. Also, the integration with ALTQ is nice,
 especially for these types of applications.
I agree with everything Tim wrote above, and I'll add that the biggest
factor that influenced me in my move to OpenBSD for my firewall was that
it was the only free unix I found that could do bidirectional filtering
in bridged mode.  As in, when you're in a bridged configuration you can
filter in and out on an interface.  Neither Linux nor FreeBSD could do
this.  It's certainly an edge case, but if you need that feature it's
invaluable.
I'm using ALTQ since FreeBSD 4.6 and it's also exist ALTQ+PF that's near 
the same as OpenBSD version.

And i confirm that's shapping with ALTQ work great ! Even with 32 Kbps.
You can easely shape around 1000 rules and have a full Fast Ethernet port 
on a dual PIII (FreeBSD ALTQ port without PF)
ALTQ have many shape algo, maybe the only one with such diversity.

You have some CD distribution with ALTQ enable.

I posted my asterisk altq experiments here:
  http://slacker.com/~nugget/asterisk4.php


--
David McNett [EMAIL PROTECTED]
http://slacker.com/~nugget/
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--
Arnaud Pignard ([EMAIL PROTECTED])
Frontier Online - Opérateur Internet
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[Asterisk-Users] SIP Absolute Timeout

[kaiser]

Hi,

I dial a number with following setting:

exten = _X.,1,Absolutetimeout(20)exten 
= _X.,2,dial(SIP/[EMAIL PROTECTED]|L(30))exten 
= T,1,BackGround(tt-weasels)exten = 
T,2,Hangup()

I find Absolute time out is not working , is it 
normal?


kaiser
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Re: [Asterisk-Users] Detecting when a called mobile is not reachable?

[Ian Hailey]

David John Walsh wrote:
I guess I should have added that this is based on the European, and
specifically UK model, but I would have expected it to have been
deemed best practice by most operators.

On Apr 4, 2005 4:04 AM, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
 

Rod Bacon wrote:
   

This is quite interesting.
I tested calls to 2 mobiles that I knew were off, and not diverted to
voicemail. 1 with Telstra, the other with vodafone (I'm in Australia).
Via ISDN, both calls were shown as unanswered by asterisk. When the
calls went to voicemail, the call was deemed to be answered.
Via analogue circuits, the call is shown as answered, no matter what.
 

That's what I would expect.
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Thaks for all your replies, adding the *r* seems to help.
Ian.
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[Asterisk-Users] SIP phones to Asterisk using MAC address instead of IP address

[Chuck Bunn]

Hi,
I know this can be done but I guess I am not understanding the few notes 
I have seen on this - can SIP phones be tied to Asterisk using a PC mac 
address instead of their IP address (obviously I am using DHCP). If 
someone could please point to the right Wiki or How to I would greatly 
appreciate it.

Thanks
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Re: [Asterisk-Users] problems with call-forward from ccme to * on sip trunk

[Andrea Riela]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
To help to find my mistake, I've two debugs:
1) isdn -- connection plar to 5600 on * -- 601 on cme -- vm 
call-forward to 5601 on *

ext.num 123456789 calls my ISDN number, on ccme there's a connection 
plar to internal 5600 (on asterisk), that dials automatically to 601 on 
ccme. The call-forward noan (no answer) to 5601 on * works great

debug: http://www.nesys.it/sipwork.txt
2) isdn -- connection plar to 601 on cme -- vm call-forward to 5601 *
ext.num 123456789 calls my ISDN number, on ccme there's a connection 
plar directly to internal 601. The call-forward noan to 5601 doesn't 
work correctly (the call goes to *, but the connection tears down)

debug: http://www.nesys.it/sipdnwork.txt
I think that's a debug quite interesting for all sip people.
Any advice will be appreciated
Thanks for your support
Regards
Andrea
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (Darwin)
iD8DBQFCUUl+MakHrsrHP9wRAjaWAJ9TX38RK7w8UxYSC52w8mKAU3vTjACgzSNl
lcsr7AsP5qC4MZrvEdcAldc=
=XAdb
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Re: [Asterisk-Users] Realtime mysql problem?

[Wojciech Tryc]

can you send me a dump from SQL for this account?
I have it working both ways,
W
- Original Message - 
From: Matt Schulte [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 9:34 AM
Subject: RE: [Asterisk-Users] Realtime mysql problem?

Well, I made several posts. Basically realtime works fine on the system
you register to, if you try to contact that peer from another Ast server
(running realtime), it does a SELECT query and all finds the peer and
continues to say Unable to contact peer as if the user doesn't exist.
I even went as far as packet sniffing and noticed it doesn't ever go out
on port 4569 or anything. Again, I've made several posts about this
before for full details. :-)
Thanks, Matt
-Original Message-
From: Wojciech Tryc [mailto:[EMAIL PROTECTED]
Sent: Monday, April 04, 2005 8:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime mysql problem?
What is your problem with IAX in realtime? I have it working (finally).
Wojtek
- Original Message - 
From: Matt Schulte [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 9:01 AM
Subject: RE: [Asterisk-Users] Realtime mysql problem?

Sorry for the delay, do you have any clue when realtime will get added
to stable? I never did get this working but before I go too much further
I'd like to run production on a stable version..
I'll try out SIP today and let you know, the reason I'm using IAX is
because everything SIP we do is through SER. Not to mention since
realtime doesn't support qualify= and NAT mode must be manually set,
it's kind of pointless to use Asterisk for SIP. :-)
Just for curiosity sake, have you tried any SIP RealTime stuff?
Perhaps this is an IAX problem? I remember  helping a guy a few weeks
ago get his SIP RealTime working. This is the first IAX I've dealt with.
And I have  no IAX stuff to test with.
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 30, 2005 9:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime mysql problem?
Matt Schulte wrote:
How do you toggle the realtime cache?
   Check in the configs/iax.conf.sample file of a recent CVS download
and it should be in there.
If there were too many fields
in the table, could you foresee this being a problem?
   No, because I have lots of extra company specific fields in my
sip_users table that asterisk doesn't use at all and I've had no
problems.
ie iax users have peercontext and auth.
   Just for curiosity sake, have you tried any SIP RealTime stuff?
Perhaps this is an IAX problem? I remember helping a guy a few weeks ago
get his SIP RealTime working. This is the first IAX I've dealt with. And
I have no IAX stuff to test with.
-Matthew
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Re: [Asterisk-Users] Sending faxes and call accounting

[Glenn]

Chris Mason (Lists) said:
 In the Asterisk system I am testing for implementation at a small luxury
 resort, there are four fax machines that the guests can use for sending
 and receiving faxes. Because they require confidentiality, we cannot use
 hylafax or other method than a stand alone fax.

I don't understand you're confidentiality arguement. If asterisk is
switching the call, it /can/ save a copy of the transmission.

None the less, you should be able to switch a fax call just like a voice
call.

cheers,
glenn


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[Asterisk-Users] Distributed services such as voicemail using Asterisk

[Chuck Bunn]

Hi,
Is it possible to distribute services used by Asterisk onto several 
boxes - similar to Pingtel (Pingtel is not an option for me since I need 
to tie analog phones into the system). The main service I want to 
distribute is the voice mail. I know that Mysql (have not tried 
PostGreSQL yet) can be used for voice mail and configuration files but 
can the MySQL server be on another server???

Thanks
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[Asterisk-Users] Asterisk Realtime - extensions configuration help

[Shaoul Jacobson - TELLINK]

Hi,

The wiki http://www.voip-info.org/wiki-Asterisk+RealTime+Extensions shows a
very trivial sample:

INSERT INTO `extensions_table` VALUES 
(1, 'mycontext', '_574555', 1, 'Wait', '2');


but how would you 'translate' an old definition as :
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)

I already found out that the commas need to be replaced by '|'.
(exten = ... Dial(SIP/1007,20,tr) becomes ..., 'Dial', '1007|20|tr' )
It is mentioned only for the 'goto' in the wiki.
Maybe is it worth to broaden up the sample.


Regards,


Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel :   +32 3 201 96 36
Fax :   +32 3 227 09 81
e-mail  [EMAIL PROTECTED]

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RE: [Asterisk-Users] SIP phones to Asterisk using MAC address insteadof IP address

[Giles Coochey]

 Hi,
 
 I know this can be done but I guess I am not understanding 
 the few notes 
 I have seen on this - can SIP phones be tied to Asterisk 
 using a PC mac 
 address instead of their IP address (obviously I am using DHCP). If 
 someone could please point to the right Wiki or How to I 
 would greatly 
 appreciate it.
 

I would do this by using IP reservations on the DHCP server. Most DHCP
servers will allow you to set a reservation of a paricular IP address to
a particular MAC address.

You may not be able to use this if you have more phones than available
IP addresses of course.

I couldn't see anything in
http://www.voip-info.org/wiki-Asterisk+config+sip.conf that would help
your cause directly.

Giles
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Re: [Asterisk-Users] Re: X100P interrupt load

[Jesse D. Guardiani]

On Mon, 2005-04-04 at 12:29 +0800, Dinesh Nair wrote:
 
 On 03/23/05 04:15 Jesse Guardiani said the following:
  This should be has some issues. I do not consider
  the FreeBSD zaptel support to be production quality
  in any way. I experienced reproducible system hangs
  (mostly after an asterisk restart), interrupt issues
  (audio skips and SSH pauses during typing), and
  general instability. This was with an up-to-date
  FreeBSD 5.3-SECURITY and the latest zaptel at
  asterisk from ports (1.0.6 for asterisk, and a
  significantly lower version for zaptel, I think).
  
  I do not recommend anyone run FreeBSD + Asterisk at
  this time.
 
 perhaps a post detailing how these hangs happenned and any CLI output 
 before these hangs would help in /eliminating/ this.

:) I doubt it. The zaptel driver for FreeBSD isn't up-to-date with the
Linux version, so I doubt Zaptel support on FreeBSD will ever be quite
as reliable as Linux.

But if you're curious: the hangs could be forced by restarting the
asterisk server. Sometimes it would survive one restart then crash
on the second. Sometimes it would crash for no reason at all.


  i'm running asterisk 
 on freebsd 4.x /with/ digium TDM cards without any problems. any problems i 
 faced were usually tied down the the digium hardware itself, instead of 
 asterisk or freebsd. note that noload = pbx_wilcalu needs to exist in 
 modules.conf, as detailed in the asterisk on freebsd wiki.
 
  not a hardware guy, so I don't know much about interrupts.
  Just that 1000 interrupts/sec is fairly high. :)
 
 those are the interrupts which the digium cards generate, and are used for 
 timing. it's not specifically a freebsd issue.
 
-- 
Jesse Guardiani, Systems Administrator
WingNET Internet Services,
P.O. Box 2605 // Cleveland, TN 37320-2605
423-559-LINK (v)  423-559-5145 (f)
http://www.wingnet.net

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[Asterisk-Users] configuring md5 authentication

[Dov Bigio]

Hello,

How does md5 authentication works?

I have created a user on my sip.conf like this:

[dov]type=friendhost=dynamicusername=dovauth=md5; echo -n "dov:myhost.com.br:dov" | md5summd5secret=a72d3b44ea28fc6515d922b21970b83c
;secret=dov
Where myhost is the real that I normally use on my SIP phone when I don't use md5 authentication. The echo line is the command I used to convert my user:realm:pwd into md5.

In my X-Ten phone I just enter my username "dov" and password "dov" as plain text.

It doesn't log in as I thought it should...
Is there any extra setting that I have to define?

Thank you
Dov


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RE: [Asterisk-Users] Realtime mysql problem?

[Matt Schulte]

 do you have any clue when realtime will get added to stable?

It won't.
why not?

 Not to mention since
 realtime doesn't support qualify= and NAT mode must be manually set,

Have you been using RTC? (RealTime Cache) It fixes the NAT/MWI
problem.

I haven't tried this yet because of my other issue, is RTC on by
default?
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Re: [Asterisk-Users] configuring md5 authentication

[Maik Schmitt]

 Where myhost is the real that I normally use on my SIP phone when I don't use 
 md5 authentication. The echo line is the command I used to convert my 
 user:realm:pwd into md5.
 
 In my X-Ten phone I just enter my username dov and password dov as plain 
 text.
 
 It doesn't log in as I thought it should...
 Is there any extra setting that I have to define?

Did you use the same realm, that was specified in the sip.conf? The
default is asterisk.

-- 
Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP


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[Asterisk-Users] newbie - want to use asterisk as an internal PBX

[mak kwak]

Hallo.
At the begining I would like to use asterisk as a VoIP server for some internal 
extensions inside one building without connection to external world. I planning 
to use kphone as soft phones. I tried to use configureation description that is 
described in 
http://asterisk.net.au/tutorial/1/

I'm running RH7.3, compiled and installed asterisk successfuly, compiled kphone.

I set up all scripts according to the link above.

/usr/sbin/asterisk -vvvgc : seems to be starging OK.

When I run kphone I am able to login: trace from asterisk console:

  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/enum.conf': Found
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 1 - 2
Asterisk Ready.
*CLI -- Registered SIP 'kphone' at 10.1.3.154 port 5062 expires 180


Problems start now. According to extensions.conf:
; echo test, to make sure your phone works.
exten = 600,1,Playback(demo-echotest) ; Let them know what's going on
exten = 600,2,Echo ; Do the echo test
exten = 600,3,Playback(demo-echodone) ; Let them know it's over
exten = 600,4,Goto(s,6) ; Start over 


I GUESS that when I dial 600, I should be able to hear echo when I'm talking, 
but unfortunatelly I cannot even dial the number. I tried many ways (10.1.3.154 
- is my asterisk pbx):
600
[EMAIL PROTECTED]
sip:[EMAIL PROTECTED]

all the dials above produce asterisk log message:
Apr  4 16:54:25 NOTICE[27916]: pbx.c:1329 pbx_extension_helper: Cannot find 
extension context 'voipmenu'

and kphone says: call failed: not found

My sound card works fine. I do not know what can I do more.
Have You got any ideas.

Greetings


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Re: [Asterisk-Users] Asterisk Realtime - extensions configuration help

[Matthew Boehm]

Shaoul Jacobson - TELLINK wrote:
 Hi,

 The wiki http://www.voip-info.org/wiki-Asterisk+RealTime+Extensions
 shows a very trivial sample:

 INSERT INTO `extensions_table` VALUES
 (1, 'mycontext', '_574555', 1, 'Wait', '2');


 but how would you 'translate' an old definition as :
 exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)

 I already found out that the commas need to be replaced by '|'.
 (exten = ... Dial(SIP/1007,20,tr) becomes ..., 'Dial', '1007|20|tr' )
 It is mentioned only for the 'goto' in the wiki.
 Maybe is it worth to broaden up the sample.

You just answered your own question in the same post. so..why did you
even post this question if you answered it 2 lines later?

-Matthew

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[Asterisk-Users] Just a test

[Rick Baranowski]

Title: Just a test






Just testing our new subscription.

Ping J

Rick


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[Asterisk-Users] Does the agent queue app support Aftercall and AUX agent status?

[Steve Mann]

In most call centers 
I have worked in, the agents had the ability to change their status from "auto 
ready" or "available" into an AUX of After call state, Aftercall basically works 
like wrap time, in that the agent would not receive another call in the queue 
until their status was manually changed back to auto ready by a specific key 
combination on the dial pad. Aux worked in that the agent could change their 
state from auto ready into an AUX state where the press a code that indicates 
what type of AUX state they are in, an example would be Aux-Break, or 
Aux-Supervisor feedback (for tracking of time, etc.)

Example, I am an 
agent, I receive a call, while on the call, I dial a special key code, and then 
when the call disconnects, instead of going right back to the queue and 
receiving another call, I go into an After Call state, where I can write notes, 
and log the call, etc. Then I would dial another sequence to put me back into a 
ready state.

An example of Aux, I 
am an agent, I am in a queue, but not on a call, my supervisor calls my 
extension and says they need to discuss one of my previous calls with me. I dial 
a code, and my state is changed from auto ready into an Aux state, where I then 
dial an additional digit to indicate why I am in Aux, example: Feedback, Break, 
etc. I then get back to my desk, and dial a new code to place me back into a 
ready state.

I know that with 
Asterisk, you can program a wrap time option to allow the CSR X number of 
seconds or minutes of wrap time before receiving another call, but I am looking 
for the above functionality over and above the simple implementation of wrap 
time.

I do not want to 
just have the agents log in, and out when they don't want a call, but instead 
use the functionality I described above as a time keeping system for payroll, 
reporting, and agent tracking purposes.

I sent an email, 
with a more ambiguous subject line about this subject, and received no response, 
so I am hoping with a better subject line, someone may open the 
email.

In that previous 
email I mentioned that on the digium homepage's FAQ it listed some call center 
terminology that detailed the above mentioned functionality, but I can not find 
and documentation on it, so I am hoping it exists, but has not been documented 
yet, and that someone out there has used it, or knows if it truly does exist, or 
if I am out of luck.

The link to the FAQ 
section:
http://www.digium.com/index.php?menu=faq#General_7

If I am asking the 
wrong list, if someone knows a better place to ask this question, please let me 
know.

Thanks,

Steve MannNetwork 
AdministratorFineLine Solutions
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Re: [Asterisk-Users] Re: Snom and Multiple calls

[Josh Dady]

Okay, after talking with Sven today, it turns out my problem 
description is wrong (I was combining to cases, one of which does work 
in the current firmware):

  - Multiple incoming calls (works already)
  - Incoming call while dialing (or waiting for answer of) outgoing 
call (doesn't)

--
Joshua P. Dady


smime.p7s
Description: S/MIME cryptographic signature
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Re: [Asterisk-Users] Asterisk@Home Question

[Dalon Westergreen]

yip,  i think that is the best approach.

--Dalon

On Apr 4, 2005 6:33 AM, Giles Coochey
[EMAIL PROTECTED] wrote:
  Greetings!
 
  This is my first post to the list...and I'm kinda' new to Asterisk, so
  please be kindI did a fair amount of Googling but was not able to
  find an answer.
 
  I am using [EMAIL PROTECTED] 0.8
 
  I was wondering if there is a way to select the outbound trunk based
  on the extension that making the call.
 
  Here is why I ask. Since I am already running my Asterisk server for
  my own use, I also wanted to let friends and family in on the action
  but I don't want to pay for their calls. So if I ask them to buy talk
  time from a termination provider and then setup a separate trunk for
  them, how do I make sure that only their calls use that outbound
  trunk?
 
 
 Being rather new myself, but the first thing I thought about your
 problem was putting those extensions in a different context to the one
 where you define your trunk calls.
 
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[Asterisk-Users] Problem registering 'SJPhone'?

[Chuck Bunn]

Hi,
Has anyone had problems registering an SJPone software phone. I get  
lots of junk mail so I have some filters running in Thunderbird and I 
have not seen my registration acknowledge come through. Do they 
(www.sjlabs.com) use some other domain for registration?? Any one else 
had this problem??

Thanks
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Re: [Asterisk-Users] Asterisk-1.0.7 Build - Serious issues

[Dana Olson]

On Mar 31, 2005 1:24 PM, Dana Olson [EMAIL PROTECTED] wrote:
 On Thu, 31 Mar 2005 10:04:34 -0500, Kanuri, Seshu (Company IT)
 [EMAIL PROTECTED] wrote:
  Folks!
 
  I want to let everyone know that I have been trying to migrate from
  1.0.6 to 1.0.7 last few days and I have come across serious issues in
  the build 1.0.7. What I found are listed below. I would recommend
  everyone to hold off any upgrade till the next build.
 
  1)Voicemail - No Audio. Asterisk is not able to stream the voice to the
  Uas. 0-9 Digit files seem to be missing and Asterisk does not try to say
  extension numbers for the called user. My guess is all these .gsm files
  are corrupt and hence you don't hear anything.
 
  2)Music on hold - .MP3 files in the ../mohmp3 and other folders are
  corrupt. When we tried to play these files using a media player, all we
  hear is gibberish.
 
  3)DTMF is screwed up. Whatever worked in 1.06 does not work now when we
  configure this for RFC2833.
 
  Has anyone upgraded to 1.0.7 from 1.0.6 and had these issues and been
  able to find a fix?
 
  Seshu
 
 Just to add to this, I've been having some issues with audio with
 1.0.7 as well. I haven't yet downgraded back to 1.0.6 to see if it
 solves it, but basically, I'm hearing some artifacts, and these didn't
 occur in the last 3 or 4 builds. I'm using the same phone and same
 config as I had been on 1.0.6. Other people that I call say that my
 phone sounds like crap now too. Also, when dialing over a Zap channel,
 the audio seems to sorta stutter now, at the very first second or two
 of a call. This didn't happen prior to 1.0.7.
 
 When I get some time that the server is not in use, I will downgrade
 and try to confirm that it is definitely an issue with the new build,
 but so far it seems that way.
 --
 Dana


I rolled back to Asterisk 1.0.6 this morning and things seem back to
normal as far as voice quality goes.
--
Dana
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Re: [Asterisk-Users] Router with QoS recommendations

[James H. Thompson]

Any FreeBSD/OpenBSD solutions we should add to the list at the 
bottom of this page?

 http://www.voip-info.org/tiki-index.php?page=VOIP+Routers


Jim

James H. Thompson[EMAIL PROTECTED]

  - Original Message - 
  From: 
  Arnaud 
  PIGNARD 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Monday, April 04, 2005 3:57 
AM
  Subject: Re: [Asterisk-Users] Router with 
  QoS recommendations
  At 15:36 04/04/2005, you wrote:On 03-Apr-2005, Tim 
  Pushor wrote:  I prefer PF's approach to security first, 
  convenience second, and I  *really* like the fact that PF has a 
  real parser. As the requements get  more complex, having 
  everything in one file, and very readable and  structured is a 
  huge plus. Also, the integration with ALTQ is nice,  especially 
  for these types of applications.I agree with everything Tim 
  wrote above, and I'll add that the biggestfactor that influenced me in 
  my move to OpenBSD for my firewall was thatit was the only free unix I 
  found that could do bidirectional filteringin bridged mode. As 
  in, when you're in a bridged configuration you canfilter in and out on 
  an interface. Neither Linux nor FreeBSD could dothis. It's 
  certainly an edge case, but if you need that feature 
  it'sinvaluable.I'm using ALTQ since FreeBSD 4.6 and it's also 
  exist ALTQ+PF that's near the same as OpenBSD version.And i 
  confirm that's shapping with ALTQ work great ! Even with 32 Kbps.You can 
  easely shape around 1000 rules and have a full Fast Ethernet port on a 
  dual PIII (FreeBSD ALTQ port without PF)ALTQ have many shape algo, maybe 
  the only one with such diversity.You have some CD distribution with 
  ALTQ enable.I posted my asterisk altq experiments 
  here: http://slacker.com/~nugget/asterisk4.php--David 
  McNett [EMAIL PROTECTED]http://slacker.com/~nugget/___Asterisk-Users 
  mailing 
  listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 
  UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- 
  Arnaud Pignard ([EMAIL PROTECTED])Frontier 
  Online - Opérateur 
  Internet___Asterisk-Users 
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Re: [Asterisk-Users] Sangoma VS. Digium

[Dana Olson]

On Mar 31, 2005 1:44 PM, Dana Olson [EMAIL PROTECTED] wrote:
 On Thu, 31 Mar 2005 11:37:19 -0600, Rich Adamson [EMAIL PROTECTED] wrote:
 My understanding is that to an extent when we buy Sangoma
 we're putting the dagger to Digium.
   
If anything puts the dagger to Digium it'll be their own inability to
engineer reliable hardware.
   
I appreciate what Digium has done for Asterisk, but reliability 
expectations
for phone equipment are extremely high.  I sympathize with people who 
need
hardware that doesn't need to be restarted once a week just to do its 
job
properly.  If Digium can't deliver on those reliability expectations, 
and do
it soon, people are going to switch to companies that can.  And you know
what?  I don't blame them.
  
  
   The Digium boards need to be restarted once a week?
  
   Please clarify this. I was dead set on getting in a Sangoma A104 for a
   production Asterisk box, but then I read this thread and felt that it
   didn't matter so much what I would order... And so I was deciding to
   stick with Digium. And then I read your scary comment.
  
   I've currently got a Digium board filled with 3 T1s, but it hasn't
   been under heavy use right yet, due to my attention being pulled from
   * and put onto SER+AudioCodes devices for other applications, and I
   haven't had to restart yet. Is this going to change? What's the deal?
  
   Please clarify your statement for me, as I need reliability as well.
 
  I'll jump in here (but I'm not the original poster). The once a week
  thing relates to the digium TDM card (fxo and/or fxs modules). I don't
  believe the T1 cards are an issue that requires driver reloads.
 
 Alright, that helps clarify it a bit, but then again, I have been
 running Asterisk at home with a TDM card for a couple months and
 haven't had to restart it for a long time. Is it a requirement or just
 simply a recomendation?


I shouldn't have said anything. My incoming pots line stopped
responding this weekend. I found out when I got an email from someone
telling me that it just keeps ringing and ringing. This never happened
before...

Strange thing is though, today, my IAX number wasn't responding. It's
like I'm silently losing my registration or something.

Totally unrelated, I know. I'm gonna go back to 1.0.6 because I ran it
for a good while with no problems. I hope this solves it. It fixed the
issues I had with 1.0.7 at work.
--
Dana
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Re: [Asterisk-Users] Realtime mysql problem?

[Matthew Boehm]

Matt Schulte wrote:
 do you have any clue when realtime will get added to stable?

It won't.
 why not?

Now, this has been answered many, many, many times...in fact..I believe
Olle answered this in his Welcome to Asterisk post he sent out over the
weekend.

To summarize: The stable branch is for bug fixes only. New features will
never be added to stable. If you want new AND stable wait for 1.2.

 I haven't tried this yet because of my other issue, is RTC on by
 default?

No. You might want to check your configs/sip.conf.sample for the correct
settings.

-Matthew

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