Re: [asterisk-users] transfer=mediaonly : can't hear nothing
Tim Panton ha scritto: I'd be tempted to simplify things even more by removing the codec negotiation and have all the boxes be _forced_ to use alaw. Tim The same, can't hear nothing (also upgraded to 1.4.2) I still have quite a bad feeling about opening a bug like mediaonly doesn't works in the simplest of cases (same codec, same net, no jitterbuffer). Really noone is using this new feature ? If you're using it and it works what's your config ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] transfer=mediaonly : can't hear nothing
Kevin P. Fleming ha scritto:
OK, then you'll need to get a verbose/debug console trace, and
preferably a packet capture of the IAX2 traffic on 'Server', and post a
bug on bugs.digium.com with those files attached.
___
While setting up the servers to gather the logs I've tryed a
configuration which is so hello world it seems unprobable to me it
can't work due to a bug.
I post once again here, sorry for the verbosity, if then in your opinion
there's still something wrong with * internals and not with my
understanding of the configs I'll open the bug.
I anticipate that only with mediaonly (when I can't hear) I get these
messages : Received iseqno 4 not within window 5-5 which seems to
remand to bug number 0006808, but I've tested also with jitterbuffer=no
on all machines and the problem remains.
Also I get some Subclass: (38?) packets, only in mediaonly mode.
3 machines, all on the same class C net (192.168.52.x), 2 are clients
(C001 and C002) and one is the server
C001 has two nics, the second being 192.168.0.1 connected to a switch
with a linksys pap in it, which generates the call:
C001 and C002 sip.conf, iax.conf and extensions.conf are the same
(except of course for IPs where to listen and credentials)
C00x sip.conf:
[general]
context=default ; Default context for incoming calls
realm=retireti.it
bindport=5060 ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=192.168.0.1; IP address to bind to (0.0.0.0
binds to all)
srvlookup=no
tos_sip=cs3; Sets TOS for SIP packets.
tos_audio=ef ; Sets TOS for RTP audio packets.
disallow=all
allow = alaw
language=it
dtmfmode = inband
progressinband=no
canreinvite=no
qualify=yes
jbenable = no
jbforce = no
jbmaxsize = 400
jbimpl = adaptive
[0100x01]
type=friend
secret=0100x00
context=outgoing
callerid=(whatever 0100x01)
host=dynamic
C00x iax.conf:
[general]
bindport=4569
bindaddr=192.168.52.9x (C001 .94 and C002 .95)
language=it
disallow=all
allow = alaw
allow = gsm
jitterbuffer = yes
forcejitterbuffer = no
maxjitterbuffer = 400
dropcount=2
maxjitterinterps=10
resyncthreshold=1000
maxexcessbuffer=80
minexcessbuffer=10
jittershrinkrate=1
autokill=yes
auth=md5
register = 0100x01:[EMAIL PROTECTED]
[server]
type=friend
context=incoming
secret=pwd
auth=md5
host=192.168.52.56
disallow=all
allow=alaw
allow=gsm
C00x extensions.conf :
[general]
static = yes
writeprotect = no
clearglobalvars = no
[globals]
CODACCOUNT = 0100x01
PWD = 0100x00
SERVER = 192.168.52.56
[outgoing]
exten = _X.,1,NoOp(esco)
;exten = _X.,n,Dial(IAX2/${CODACCOUNT}:[EMAIL PROTECTED]/${EXTEN})
exten = _X.,n,Dial(IAX2/${CODACCOUNT}:[EMAIL PROTECTED]/${EXTEN})
exten = _X.,n,Hangup
[incoming]
exten = _X.,1,NoOp(entro)
exten = _X.,n,Answer
exten = _X.,n,Playback(tt-weasels)
exten = _X.,n,Echo
exten = _X.,n,Hangup
now Server configs :
iax.conf :
[general]
bindport=4569
bindaddr=192.168.52.56
language=it
disallow=all
allow=alaw
allow=gsm
jitterbuffer = yes
forcejitterbuffer = no
maxjitterbuffer = 100
dropcount=2
maxjitterinterps=10
resyncthreshold=1000
maxexcessbuffer=80
minexcessbuffer=10
jittershrinkrate=1
context=default
autokill=yes
[0100101]
username=0100101
type=friend
secret=0100100
auth=md5
host=dynamic
context=default
callerid=0100101
transfer=no
qualify=yes
[0100201]
username=0100201
type=friend
secret=0100200
auth=md5
host=dynamic
context=default
callerid=0100201
transfer=no
qualify=yes
extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
[default]
exten = _X.,1,NoOp(here we are)
exten = _X.,n,Dial(IAX2/server:[EMAIL PROTECTED]/${EXTEN})
exten = _X.,n,Hangup
As you can see I've removed the realtime engine, and I've no input
client and termination clients difference, C001 calls the server,
which calls C002, which playback something and then Echoes, anyway both
C001 and C002 are the same type of registered, monitored friends for
the Server.
transfer=no, and all works ok, with debug,verbose and 'iax2 set debug' I
see in Server's CLI :
*CLI Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX
Subclass: NEW
Timestamp: 00010ms SCall: 6 DCall: 0 [192.168.52.94:4569]
VERSION : 2
CALLED NUMBER : 12
CODEC_PREFS : (alaw|gsm)
CALLING NUMBER : 0100101
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME: whatever
LANGUAGE: it
USERNAME: 0100101
FORMAT : 8
CAPABILITY : 57354
ADSICPE : 2
DATE TIME : 2007-03-20 12:16:30
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
Timestamp: 00013ms SCall: 3 DCall: 6 [192.168.52.94:4569]
AUTHMETHODS : 2
CHALLENGE : 347981677
USERNAME: 0100101
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
Timestamp: 00030ms SCall: 6
[asterisk-users] transfer=mediaonly : can't hear nothing
I've setup this simple configuration to test the new mediaonly iax
feature in 1.4 :
Input (client) - Server (routing) - Termination
transfer=notransfer=mediaonly transfer=no
all the machines are in the same 192.168.0.x net
the routing Server in the middle has iaxusers realtime backend on mysql
the call is originated with a sip phone registered on the Input client
Server's credentials are hardcoded in iax.conf on the Termination server
codecs allowed are alaw and gsm, I see all the traffic as alaw
( Ooh, voice format changed to 8 )
with Server's transfer=no all works ok
with Server's transfer=yes all works ok I can hear, some seconds (~12)
after the call is answered the Server spits a
-- Channel 'IAX2/[Termination IP]:4569-2' ready to transfer
-- Channel 'IAX2/[Client IP]:4569-1' ready to transfer
and inserts the cdr (in mysql), I can still hear
(of course the cdr is shorter than the actual call, or we would'nt be
testing mediaonly :)
with Server's transfer=mediaonly I have quite immediatly the 'ready to
transfer' message but cant' hear nothing (I see udp activity on I and T
anyway)
10% of the tryes the transfer is not so immediate and I can hear a
couple of seconds of monkey screams
(the cdr works ok, billing the entire call)
client I extensions.conf
[paps]
exten = _X.,1,Dial(IAX2/${LOGIN}:[EMAIL PROTECTED]/${EXTEN})
exten = _X.,n,Hangup
routing server S extensions.conf
[default]
exten = _X.,n,Dial(IAX2/server:[EMAIL PROTECTED]/${EXTEN})
exten = _X.,n,Hangup
termination server T extensions.conf
[incoming]
exten = _X.,1,Answer
exten = _X.,n,Playback(tt-monkeys) -- .gsm
exten = _X.,n,Echo
exten = _X.,n,Hangup
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Re: [asterisk-users] transfer=mediaonly : can't hear nothing
Kevin P. Fleming ha scritto: I've setup this simple configuration to test the new mediaonly iax feature in 1.4 : What version of Asterisk exactly? 1.4.1 Input (client) - Server (routing) - Termination transfer=notransfer=mediaonly transfer=no This doesn't make any sense; the 'Server' is going to have two 'friend' entries for the other systems. Are you saying that you have transfer=mediaonly set for both 'friends'? In our setup calls are always in the arrows direction, Input never receives a call, Termination never starts one. So the config is : realtime mysql users on the server to auth the customers (Input) and one user entry in iax.conf on the Termination to auth the Server transfer=mediaonly is set in [general] I tried to add an entry on the Server for the Termination (as a peer, not friend) and have T register, but the result is the same just to be clear (and verbose) : iax.conf on the Server : [general] bindport=4569 bindaddr=0.0.0.0 language=it disallow=all allow=alaw allow=gsm jitterbuffer = yes forcejitterbuffer = no maxjitterbuffer = 100 dropcount=2 maxjitterinterps=10 resyncthreshold=1000 maxexcessbuffer=80 minexcessbuffer=10 jittershrinkrate=1 ;transfer=no ;transfer=yes transfer=mediaonly context=default autokill=yes and iaxusers table : +-+--++-+--+-+-+-+--++--+ | name| username | type | secret | auth | host| context | ipaddr | port | regseconds | callerid | +-+--++-+--+-+-+-+--++--+ | 0100101 | 0100101 | friend | 0100100 | md5 | dynamic | default | 0.0.0.0 |0 | 0 | 0100101 | | 0100102 | 0100102 | friend | 0100100 | md5 | dynamic | default | |0 | 0 | 0100102 | | 0100103 | 0100103 | friend | 0100100 | md5 | dynamic | default | |0 | 0 | 0100103 | | 0100203 | 0100203 | friend | 0100200 | md5 | dynamic | default | |0 | 0 | 0100203 | | 0100202 | 0100202 | friend | 0100200 | md5 | dynamic | default | |0 | 0 | 0100202 | | 0100201 | 0100201 | friend | 0100200 | md5 | dynamic | default | |0 | 0 | 0100201 | | t01001 | t01001 | peer | t01001 | md5 | dynamic | default | 0.0.0.0 |0 | 0 | 701001 | +-+--++-+--+-+-+-+--++--+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk upgrade
at the moment (fortunately) i'm not experiencing any kind of particular problem, do you suggest me to upgrade asterisk? #1 sysadmin rule: If it's not broken, just don't fix it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER with multiple asterisk deployment
Adi Simon ha scritto: Hi, Did anyone actually manage setting up a single SER with multiple Asterisk boxes? I particulary have a problem of keeping the session alive and by that I mean directing all the following sip messages to the same asterisk box the first signal was sent (randomally). record_route() and loose_route() should help you, AFAIK. They don't? Cheers, Simone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] lost packets when bridging zap and iax
We have a machine with a TE410P in it acting as a client to route calls via iax2 to our central server, caller -- ( zap - iax ) --- ( iax - whatever ) -- called client server often the called can't hear the caller (both machines on public ip) 'iax2 show netstats on client machine shows more and more dropped packets on the local side if we use sip as the entering point for the calls all works well : caller -- ( sip - iax ) --- ( iax - whatever ) -- called client server seems something in the bridging between zap and iax screws up, but I don't know if it's a bug or a misconfiguration, my conf files follows, someone has similar experiences to share ? /etc/asterisk# cat iax.conf [general] bindport=4569 bindaddr=xxx.xx.xx.xxx disallow=all allow=alaw jitterbuffer=yes forcejitterbuffer=no tos=lowdelay autokill=yes language=it notransfer=yes /etc/asterisk# cat sip.conf [general] context=invalid bindport=5060 bindaddr=xxx.xx.xx.xxx srvlookup=no disallow=all allow=alaw progressinband=no canreinvite=no language=it [authentication] [some-ip] type=friend context=ip host=some-ip /etc/asterisk# cat zapata.conf [channels] language=it context=default switchtype=euroisdn pridialplan=national prilocaldialplan=national signalling=pri_cpe immediate=no callerid=asreceived usecallingpres=yes echocancel=yes echocancelwhenbridged=no ;echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 group = 1 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 /etc/asterisk# cat /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 span=3,1,0,ccs,hdb3 bchan=63-77 dchan=78 bchan=79-93 span=4,1,0,ccs,hdb3 bchan=94-108 dchan=109 bchan=110-124 loadzone=it defaultzone=it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] -- Going to extension s|1 because of immediate=yes, but immediate is 'no'
We have an asterisk with a TE410P in it, when a call comes in it says :
-- Going to extension s|1 because of immediate=yes
-- Extension 's' in context 'default' from '[calling num]' does not
exist. Rejecting call on channel 0/27, span 2
but in zapata.conf immediate=no :
[channels]
language=it
context=default
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
signalling=pri_cpe
immediate=no
callerid=asreceived
usecallerid=yes
hidecallerid=yes
usecallingpres=yes
so I'm stuck, beacuse if in extension i put s,1,Dial(foobar/${EXTEN}) I
really dial 's' and if I put _X.,1,Dial(foobar/${EXTEN}) I don't even
get there because immediate=yes looks for 's'.
The strange thing is that this configuration works perfectly in other
places, can it be that the connected nortel forces in some way
immediate=yes ?
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[asterisk-users] asterisk sending connects when it shouldn't
When asterisk receives those messages you hear when calling an
unreacheable cellular phone it sends a 'connect' over the terminating
PRI line (digium TE410P), making the call seen as billed from customer's
perspective.
I don't know if this behaviour is a bug or something I can resolve with
some fine tuning, so I'm sending to both lists.
Since the calls comes from a SIP connected GSM gateway, is there some
SIP code which corresponds to the 'pass audio but don't connect' we want
here ?
that's roughly the extension :
exten = _X.,1,AGI(agi://127.0.0.1:54321/SomeAgiHere?someArgumentsHere)
exten = _X.,n,GotoIf($[${CALLABLE}=TRUE]?chkmax:hangup)
exten = _X.,n(chkmax),Set(GROUP()=${TECH_PRE})
exten = _X.,n,GotoIf($[${GROUP_COUNT(${TECH_PRE})} =
${MAX_CALLS}]?hangup:dial)
exten = _X.,n(dial),Dial(${STR_DIAL})
exten = _X.,n(hangup),Hangup
exten = h,1,Set(CDR(userfield)=${USERFIELD}-${HANGUPCAUSE})
Here the provider's trace of a call answered by asterisk :
/HDLU 4/Port
=== LAPD ===
--- ADDRESS ---
SAPI : 0 = call control procedures
CR : ..1.
EA0: ...0
TEI: 0 = non-automatic TEI assignment user equipment
EA1: ...1
--- CONTROL ---
--- I FRAME ---
I FORMAT : ...0
N(S) : 86
P : ...0
N(R) : 31
=== ETSI ISDN ===
PROT DISC : 08h = Q.931 user-network call control message
LEN CALL R : 2
SPARE : 0
FLAG : 1... = the message is sent to the side that
originates the call reference
CALL REF : 226
MESS TYPE : 07h = Connect
Here the complete trace :
/HDLU 4/Port
0 TEI: 0 CALL REF: 226 Setup '500' '[called number]'
0 TEI: 0 CALL REF: 226 Setup acknowledge
0 TEI: 0 CALL REF: 226 Call proceeding
0 TEI: 0 CALL REF: 226 Connect == should not
0 TEI: 0 CALL REF: 226 Connect acknowledge
0 TEI: 0 CALL REF: 226 Disconnect 16 normal call clearing
0 TEI: 0 CALL REF: 226 Release
0 TEI: 0 CALL REF: 226 Release complete
-
Here a trace from a correctly functioning non-voip system :
/HDLU 4/Port
0 TEI: 0 CALL REF: 246 Setup '500'
0 TEI: 0 CALL REF: 246 Setup acknowledge
0 TEI: 0 CALL REF: 246 Information 'c'
0 TEI: 0 CALL REF: 246 Information 'a'
0 TEI: 0 CALL REF: 246 Information 'l'
0 TEI: 0 CALL REF: 246 Information 'l'
0 TEI: 0 CALL REF: 246 Information 'e'
0 TEI: 0 CALL REF: 246 Information 'd'
0 TEI: 0 CALL REF: 246 Information 'n'
0 TEI: 0 CALL REF: 246 Information 'u'
0 TEI: 0 CALL REF: 246 Information 'm'
0 TEI: 0 CALL REF: 246 Information 'b'
0 TEI: 0 CALL REF: 246 Call proceeding
0 TEI: 0 CALL REF: 246 Progress
0 TEI: 0 CALL REF: 246 Progress
0 TEI: 0 CALL REF: 246 Disconnect 16 normal call clearing
0 TEI: 0 CALL REF: 246 Release
0 TEI: 0 CALL REF: 246 Release complete
--
Simone Cittadini
2K Elektronika
Tel +39.02.26265583
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Re: [asterisk-users] Connect to 'agi://blablabla' failed: Operation now in progress
Moises Silva ha scritto: AFAIK operation now in progress is a common status when you open a socket connection. When you use blocking sockets usually you dont see this because the connect call does not return until the connection is done. But when using non-blocking sockets, the connect call returns immediatly and if you try to connect again, you will get the operation now in progress message. I have seen this in my PHP Manager Proxy, but not sure what implications may have in FastAGI. May be it only tells that the connection stablishment takes a little longer, network congestion may be? We have a 'non blocking father' which spawns a 'blocking child' for each connection. So this can be the case, but I don't think it's related to network congestion, it's local on 127.0.0.1 and I see the messages even on low load. Oh well, since it works ... Regards On 7/13/06, Simone Cittadini [EMAIL PROTECTED] wrote: I get a lot of this warnings in my logs. Connect to 'agi://blablabla' failed: Operation now in progress What exactly 'operation now in progress means' ? is asterisk still trying so the call isn't lost ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk sending connects when it shouldn't (is there a q931-INFORMATION equivalent in IAX2 ?)
When asterisk receives those messages you hear when calling an
unreacheable cellular phone it sends a 'connect' over the terminating
PRI line (digium TE410P), making the call seen as billed from customer's
perspective.
I don't know if this behaviour is a bug or something I can resolve with
some fine tuning, so I'm sending to both lists.
this is the layout of machines :
|gsm gateway| - sip - |asterisk client| - iax2 - |asterisk server|
- zap - pri lines (nortel ?)
that's roughly the extension on the server :
exten = _X.,1,AGI(agi://127.0.0.1:54321/SomeAgiHere?someArgumentsHere)
exten = _X.,n,GotoIf($[${CALLABLE}=TRUE]?chkmax:hangup)
exten = _X.,n(chkmax),Set(GROUP()=${TECH_PRE})
exten = _X.,n,GotoIf($[${GROUP_COUNT(${TECH_PRE})} =
${MAX_CALLS}]?hangup:dial)
exten = _X.,n(dial),Dial(${STR_DIAL})
exten = _X.,n(hangup),Hangup
exten = h,1,Set(CDR(userfield)=${USERFIELD}-${HANGUPCAUSE})
Here the provider's trace of a call answered by asterisk :
/HDLU 4/Port
=== LAPD ===
--- ADDRESS ---
SAPI : 0 = call control procedures
CR : ..1.
EA0: ...0
TEI: 0 = non-automatic TEI assignment user equipment
EA1: ...1
--- CONTROL ---
--- I FRAME ---
I FORMAT : ...0
N(S) : 86
P : ...0
N(R) : 31
=== ETSI ISDN ===
PROT DISC : 08h = Q.931 user-network call control message
LEN CALL R : 2
SPARE : 0
FLAG : 1... = the message is sent to the side that
originates the call reference
CALL REF : 226
MESS TYPE : 07h = Connect
Here the complete trace :
/HDLU 4/Port
0 TEI: 0 CALL REF: 226 Setup '500' '[called number]'
0 TEI: 0 CALL REF: 226 Setup acknowledge
0 TEI: 0 CALL REF: 226 Call proceeding
0 TEI: 0 CALL REF: 226 Connect == should not
0 TEI: 0 CALL REF: 226 Connect acknowledge
0 TEI: 0 CALL REF: 226 Disconnect 16 normal call clearing
0 TEI: 0 CALL REF: 226 Release
0 TEI: 0 CALL REF: 226 Release complete
-
Here a trace from a correctly functioning non-voip system :
/HDLU 4/Port
0 TEI: 0 CALL REF: 246 Setup '500'
0 TEI: 0 CALL REF: 246 Setup acknowledge
0 TEI: 0 CALL REF: 246 Information 'c'
0 TEI: 0 CALL REF: 246 Information 'a'
0 TEI: 0 CALL REF: 246 Information 'l'
0 TEI: 0 CALL REF: 246 Information 'l'
0 TEI: 0 CALL REF: 246 Information 'e'
0 TEI: 0 CALL REF: 246 Information 'd'
0 TEI: 0 CALL REF: 246 Information 'n'
0 TEI: 0 CALL REF: 246 Information 'u'
0 TEI: 0 CALL REF: 246 Information 'm'
0 TEI: 0 CALL REF: 246 Information 'b'
0 TEI: 0 CALL REF: 246 Call proceeding
0 TEI: 0 CALL REF: 246 Progress
0 TEI: 0 CALL REF: 246 Progress
0 TEI: 0 CALL REF: 246 Disconnect 16 normal call clearing
0 TEI: 0 CALL REF: 246 Release
0 TEI: 0 CALL REF: 246 Release complete
On the asterisk client it seems that SIP correctly opens only a leg of
the call :
asterisk : 102 invite
- 100 Trying
- 200 OK
asterisk : ACK (now I hear the audio)
(I hangup)
asterisk : BYE
- 200 OK
Destroying call 'blabla'@ip
(with a normally answered call I see 183 Session progress instead of the
first 200 while ringing, and the the destroyed calls are two)
the iax debug : (still talking about the call that shouldn't send the
connect on isdn line)
-- Accepting AUTHENTICATED call from IP:
requested format = alaw,
requested prefs = (),
actual format = alaw,
host prefs = (alaw),
priority = mine
-- Executing Dial(IAX2/IP:4569-2, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACCEPT
Timestamp: 00014ms SCall: 2 DCall: 00188 [IP:4569]
FORMAT : 8
astegateway4*CLI
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK
Timestamp: 00014ms SCall: 00188 DCall: 2 [IP:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: VOICE Subclass: 8
Timestamp: 00090ms SCall: 00188 DCall: 2 [IP:4569]
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK
Timestamp: 00090ms SCall: 2 DCall: 00188 [IP:4569]
-- SIP/gateway4-20e0 answered IAX2/82.113.204.70:4569-2
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass:
ANSWER
Timestamp: 04698ms SCall: 2 DCall: 00188 [IP:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK
Timestamp: 04698ms SCall: 00188 DCall: 2 [IP:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 003 Type: VOICE Subclass: 8
Timestamp: 04764ms SCall: 2 DCall: 00188 [IP:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type:
Re: [asterisk-users] ooh323c - cdr
antonio ha scritto: I have a problem: when i make i call from a device h323 to sip, i have no cdr, and i don't see cdr variables for the channnel ooh323. If I remeber well, I had a similar problem and is something about setting the amaflags to billing in the h323 config files Anyone can help me ?? Thanx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] billed calls when cellullar phone is unreachable
We have a customer routing calls trough a pri (digium board), our system
then terminates the calls in various places (let say we offer LCR).
When we route a call to an unreachable cellular phone we know it cause
we get a particular ${HANGUPCAUSE} so we don't bill that call even if
billsec is 0 (the duration of the cellular is unreachable bla bla
message), but the customer says their system too records the call as
0 and their expected behaviour is to have the call recorded as duration
== 0.
(I'm supposing the customer is noticing the cellphones cause of the high
traffic, but probably this happens also with other kind of service
messages which aren't to be billed, have to try)
Now, is there some standard (we are in italy) which as to do with AMA
codes / PRI_CAUSE / HANGUPCAUSE whatever that non asterisk systems
expects to work right or have I to go trough try everything comes to
mind until it works ?
for example setting some PRI_CAUSE on the channel based on the type of
the HANGUPCAUSE I see ?
Why this seems to me a problem of the customer but I'm resolving it ?
that's roughly the extension :
exten = _X.,1,AGI(agi://127.0.0.1:54321/SomeAgiHere?someArgumentsHere)
exten = _X.,n,GotoIf($[${CALLABLE}=TRUE]?chkmax:hangup)
exten = _X.,n(chkmax),Set(GROUP()=${TECH_PRE})
exten = _X.,n,GotoIf($[${GROUP_COUNT(${TECH_PRE})} =
${MAX_CALLS}]?hangup:dial)
exten = _X.,n(dial),Dial(${STR_DIAL})
exten = _X.,n(hangup),Hangup
exten = h,1,Set(CDR(userfield)=${USERFIELD}-${HANGUPCAUSE})
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Re: [asterisk-users] where the bottleneck lies ? (was: Serverredundancy)
Douglas Garstang ha scritto: -Original Message- From: Simone Cittadini [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 12, 2006 10:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] where the bottleneck lies ? (was: Serverredundancy) unplug ha scritto: I feel interested about you can support 16,000 users of your system. As I have tested using sipp in a dual CPU Xeon with 2G Ram, the maximum number of current call is about 160. In some forums, most of ppl claim the maximum current call is about 100-200. What do you expect the number of current call to handle in 16,000 users? I'm curious about what was limiting the number of calls in your tests. For every system I have in production/testing I see the only bottleneck is system load, cpu and memory usage is well beyond limits when things starts to fall apart. The unexplicable (at least by me) thing is that system load seems to be only partially influenced by the number of calls, for example sometimes there are 100/150 calls and the load is around 0.70, sometimes it skyrockets to 2.00 / 2.50 (when it is 2 calls quality is crippled, I think because of too many dropped packets). I see this behaviour no matter how simple/complex the system is, from just a terminator with a couple of digium in it and a five-lines extension to the central server with fastagi doing mysql queries and taking hundreds of concurrent calls in both sip and iax. Can it be something related to asterisk itself ? I'm thinking about installing oprofile on the various servers, someone by chance already did it ? Another consideration is if the phones have performed reinvites, and removed Asterisk from the RTP stream. If you can live without call recording, and other features where Asterisk has to remain in the RTP path, then I imagine that this would significanlty reduce load on the Asterisk systems. Could some of your phones be reinviting? This may explain the variation in load. Doug. no, all the traffic has to pass from the machine (and all the codec is g711 so no differences in transcoding either) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect to 'agi://blablabla' failed: Operation now in progress
I get a lot of this warnings in my logs. Connect to 'agi://blablabla' failed: Operation now in progress What exactly 'operation now in progress means' ? is asterisk still trying so the call isn't lost ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] where the bottleneck lies ? (was: Server redundancy)
unplug ha scritto: I feel interested about you can support 16,000 users of your system. As I have tested using sipp in a dual CPU Xeon with 2G Ram, the maximum number of current call is about 160. In some forums, most of ppl claim the maximum current call is about 100-200. What do you expect the number of current call to handle in 16,000 users? I'm curious about what was limiting the number of calls in your tests. For every system I have in production/testing I see the only bottleneck is system load, cpu and memory usage is well beyond limits when things starts to fall apart. The unexplicable (at least by me) thing is that system load seems to be only partially influenced by the number of calls, for example sometimes there are 100/150 calls and the load is around 0.70, sometimes it skyrockets to 2.00 / 2.50 (when it is 2 calls quality is crippled, I think because of too many dropped packets). I see this behaviour no matter how simple/complex the system is, from just a terminator with a couple of digium in it and a five-lines extension to the central server with fastagi doing mysql queries and taking hundreds of concurrent calls in both sip and iax. Can it be something related to asterisk itself ? I'm thinking about installing oprofile on the various servers, someone by chance already did it ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Running 40 act ive calls (too much för CPU?)
[EMAIL PROTECTED] ha scritto: Hi, We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server connected to the PSTN through two E1 pipes to a TE405P. This has been running just fine for several months... But yesturday we connected a large number of softphone SIP clients (50) and 25 of these where running simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio tracks on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. I think there's something other wrong with such an huge usage, meaning with other either a misconfiguration or another process running on the same machine conflicting with asterisk. What's the load ? in my experience a load 2 kills audio quality, I started with a system with AGI and mysql on the same dell machine as yours, had same problems (but not such an high cpu load). Now with mysql on a dedicated server, fastagi and dbpooling 50 calls gives an average load on the long run of 0.3, with lower bounds of 0.08, and the digiums are two TE410P. Why sometimes 50 calls will do 0.3 and sometimes 0.08 is still a mistery to me ... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] increase the volume ?
Noc Phibee ha scritto: anyone have a answer at this question ? I'm pretty sure the answer is you can't, it makes sense to adjust the gain only where the A/D magic occurs, so you have to tweak your ATAs, you can set levels in asterisk configs only when configuring devices, like in zapata.conf, but no general volume setting exists. I'll anyway be the first interested in a patch allowing to adjust the volume, maybe even realtime, that's because when you route international calls you have to consider different human gains. For example chinese speaks with very low voice and sud-american shouts quite always ... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 channel problems
Jon Schøpzinsky ha scritto: We have been having problems with our IAX2 channels for some time now. Our problems are jitter, and lost packets, resulting in bad audio quality. The weird thing is, that this mostly occurs on our local network. We have tested the network with pinging an hour, without any lost packets. One of our customers also has problems using IAX2, and he is only two networks away, according to traceroute. He is on a 100mbit dedicated connection. Is there a general problem in the IAX2 channel, which causes jitter? We are running 1.2.9.1, and have tried 1.2.7.1, 1.2.5 and 1.2.0, and we have the same problems with all of them. Our average system load is around 2-3, and we have 905 registered sip users, and around 60 calls running at all time, to queues, SIP and Zap channels. On our system when the load is around 3 we lose packet in iax, but due to excessive load, simply decreasing the number of calls solves the problem. fastagi and load balancing greatly mitigated the problem. Also forcing jitterbuffer on machines not at the edges was a problem. I leave it enabled only on the machines with digium cards and ata connected. I also use sip when possbile, a lot of people from previous posts reports sip being better than iax when jitter comes to play, but in my experience 1.2.7 works quite well ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chanspy Jitter?
Wes Baehr ha scritto: (Sometimes) When I’m monitoring calls, I hear a very bad jitter – usually only on one of the bridged channels. So at first I thought it was just the one end of the conversation actually causing the jitter – but it’s not. So I called in from another device to spy at the same time – and the other chanspy sounds perfectly normal. (And neither party is complaining of bad sound) So, periodically, chanspy seems to lose sync with its source – has anyone else had this problem? Running 1.2.7.1, calls are all SIP-IAX2 Me, same problem, same version, all possible combinations of SIP/IAX calls. Looking at iax2 netstats shows there's no real problem, jitter, delay, packet loss are well within acceptable limits, still what I hear is echoed robo-voice ! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] about billing realtime (maybe OT)
I've followed with interest the discussion about realtime billing, anyway, even if it could be a fascinating subject as a developer, I've always felt that from a project management point of view the problem is simply non-existent, because the money lost with wide-grained control is unimportant against the higher system complexity and time spent in research to achieve fine-grained control. I've tried to put a couple of formulae on paper to be less qualitative and more quantitative, be warned that : 1) I don't came from a TLC background, maybe there is something about call flows that I'm missing 2) I'm not a businessman, maybe there's something about money flows that I'm missing There are 3 ways to control customer's credit, different implementations have been proposed, but basically : - pre dial control, the simplest : if credit 0 do the Dial else redirect to the Playback(sorry, money makes the world turn) on hangup update credit - set maxtime per call, still easy to implement, a bit more heavy on the cpu, since you have to do some math before the call if credit 0 find the price for requested number and set maxtime, do the call else Playback(go away, loser) on hangup update credit - set maxtime per call flow, complex to implement, heavier on the cpu since it needs some kind of polling, heavier on db if you don't cache credit data set maxtime per call then during the call : every n minutes do : for each call in the flow (one customer, one flow): credit = credit - (n * call price per minute) update maxtime on hangup update credit Since this is becoming long I'll go quickly to the point : the max money you can loose with pre dial control, which is obviously the more wide-grained, is 2 * MaxConcurrentCalls * MaxCallLength * medium_price_per_minute I dont' take in account MaxPricePerMinute since high international prices are way too higher and less frequent than the average ones. Given MCL = 1 hour (if you don't set this value chances are your provider is setting it) and MCC = 20 (that's my biggest customer limit, and the average is 15/18) we loose around 100$, supposing high prices ( I'm actually loosing 72 euro ). This seems to support my feeling, just update the credit in a cronjob every 30 minutes or so and you should sleep safe. for the average case : 2 * average_calls_per_update_interval * average_call_length * medium_price_per_minute now we can infer average_calls_per_update_interval and average_call_length with select in the db, but I suppose there is some function like average_calls_per_update_interval = f(update_interval, hour-of-the-day) and my new feeling is that inserting different update_intervals in different hours-of-the-day in your crontab, so that the above formula stays below a value choosen with the investors, you can minimize the extra cpu usage while keeping the system simple and achieve results equivalent to the ones of more complex solutions. (as they say in books, I leave the Max/avg formulae for the other cases as an exercise for the reader) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hardware for new office suggestion
I want to thank you for the suggestions. The office is in the UK, so probably we will go for the ISDN30. I am trying to get a SDSL 2mbit for the line so that bandwidth should not be a problem, the internal LAN will be Gbit as said so the QoS as suggested will be only on the firewall (linux). I have lowered expenses for other equipment so I was thinking of buying a Dell 1800 or 2800 server 2x2,8Ghz 2gb ram to set up Asterisk, know this is a big server but they'll use the ISDN lines and VoIP so virtually there could be 20/25 simultaneous calls. I'll have a look at the wiki and the phones suggested, we'd definitely like phones with internal ethernet switch and PoE capable, I'll try to get an idea of what could work for us. Thanks again Simone Tim Panton wrote: On 14 Apr 2006, at 11:29, Simone wrote: Hi list, I am in the process of setting up Asterisk for a new office and since this is going to be my first real installation I'd appreciate some advice on the hardware from the real world. We will have 8 channels (still not sure if 4xISDN2 or ISDN30 8 channels, but I will definitely go for a Digium card with echo hw cancelation) and a DSL 2mbit line (QoS on the switch and firewall?), to be configured for both traditional and VoIP usage . I was looking at the Xorcom TS-1 server and I was wondering if you would recommend it for a 30 employees office or if you'd rather build it on a normal server (would a double PIII 1Ghz be enough), and also if you could give a suggestion on the phones (we will get an HP Gbit switch PoE). Thanks, any hint really appreciated Simone I can only base my advice on what we have done for a smaller office. If you want 8 lines it is probably as cheap to go for ISDN 30 as for 4xBRI at least it is here in the UK. We have a single span E1 card from digium without echo can in a small 1U rack mounted server (spec: 1Ghz Via processor and 512Mb ram). The Via might be a bit underpowered for 30 users, but unless you are transcoding, virtually any modern processor would be fine for 8 lines. You need to look out on the DSL line if it is ADSL, since they have low upstream bandwidth. Heavy outgoing mail messages (eg attachments sent to distribution lists) can easily fill the outgoing (256kbit/s) pipe degrading the voice quality. I'm very fond of the SNOM phones - elmeg are selling the old SNOM 190 model which is a decent office phone. For 30 you should be able to get them for less than £70 each. I've got 6 - 4 SNOMs and 2 elmegs - No problems with any of them, but they don't support PoE, so you may want to look at other models. Don't underestimate how much training/doc you will need to provide to get people going on the new system. They may have been using the old one for years and written little cribsheets about how to transfer etc. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk hardware for new office suggestion
Hi list, I am in the process of setting up Asterisk for a new office and since this is going to be my first real installation I'd appreciate some advice on the hardware from the real world. We will have 8 channels (still not sure if 4xISDN2 or ISDN30 8 channels, but I will definitely go for a Digium card with echo hw cancelation) and a DSL 2mbit line (QoS on the switch and firewall?), to be configured for both traditional and VoIP usage . I was looking at the Xorcom TS-1 server and I was wondering if you would recommend it for a 30 employees office or if you'd rather build it on a normal server (would a double PIII 1Ghz be enough), and also if you could give a suggestion on the phones (we will get an HP Gbit switch PoE). Thanks, any hint really appreciated Simone ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with chan_iax.c implimentationcausesbadaudio?
Tim Panton ha scritto: I don't suppose you have an ethereal packet capture from a bad call ??? Or a description of the 'badness'? I have myself problems with iax2 sometimes, it drops a lot of packets even if there's no apparent reason to. For example two asterisk connected via iax2 on a local lan, one takes voip from the outside and the other terminates on a digium, suddendly iax2 show netstats show a dropped % of 15-20 for every call, even if the load is small (30 alaw calls on a 3.0 Ghz), then all come back to normal. During this period the load is abnormally high (3/3.5, see previous posts), but the idle cpu is around 85/90%. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Double Call Progress tones
Ron Wellsted ha scritto: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 This is slowly driving me nuts! I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk 1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls I get a double ring tone (UK style + US style). I also have a DECT phone on a Sipura SPA-3000 configured with UK tones. This gives me a double ring of UK + UK, so this suggests the call progress tones are being generated by the SIP device. As a result I have edited sip.conf to set progressinband=never but this has made no difference (even after a total restart). Previously I was running 1.0.7 without this problem, I upgraded to fix a problem with Monitor (the call stopped monitoring when transfered, 1.2.5 has fixed this). Does any one have any suggestions? Configure the ringing frequencies on the sip devices so that is something not udible by human hears (we did that as a quick fix before discovering progressinband some time ago, worked for linksys pap) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (unexplicable) peaks of machine load
Matt Florell ha scritto: I've noticed this as well from pre 1.0 versions through to 1.2.5 across 12 separate Asterisk servers. The severity seems to be random mostly. I still haven't figured out what is causing it. MATT--- Your file system is journaled ? this is another common thing that came to my mind (ext3) On 3/15/06, Simone Cittadini [EMAIL PROTECTED] wrote: I have strange peaks of machine load on my asterisk servers, looking at top the load is very high even if cpu usage is low and no swap memory is used. This happens on all the machines, some of them have asterisk, mysql, agi and digium cards on them, so I thought I was only asking too much, but yesterday I noticed the same behaviour on an asterisk machine with only two digium in it, no other service and a two line extension. I thought it can be a problem with digium cards but the interrupts aren't shared, and I have the same problem on a pure-voip server. Asterisk version varies from 1.2.1 to 1.2.5, the kernels are 2.4 or 2.6 (right ones for the installed cpu, not generic 386) The only things in common are : Linux debian, iax channels are used, with jitterbuffer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (unexplicable) peaks of machine load
Matt Florell ha scritto: Yep I use ext3, have you run test with any other file system? MATT--- No, I will do when I have time (and a server to test on) Your file system is journaled ? this is another common thing that came to my mind (ext3) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (unexplicable) peaks of machine load
I have strange peaks of machine load on my asterisk servers, looking at top the load is very high even if cpu usage is low and no swap memory is used. This happens on all the machines, some of them have asterisk, mysql, agi and digium cards on them, so I thought I was only asking too much, but yesterday I noticed the same behaviour on an asterisk machine with only two digium in it, no other service and a two line extension. I thought it can be a problem with digium cards but the interrupts aren't shared, and I have the same problem on a pure-voip server. Asterisk version varies from 1.2.1 to 1.2.5, the kernels are 2.4 or 2.6 (right ones for the installed cpu, not generic 386) The only things in common are : Linux debian, iax channels are used, with jitterbuffer When this ghost load becomes too high ( 3) asterisk starts losing packets, and the users starts losing patience ... Anyone experiencing a similar problem ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [OT?]SCCP image for cisco 7905g
Hi, I recently purchased a brand new 7905g with his SIP firmware (licensed). Now, I want to play a little with chan-sccp, but I'm unable to find the appropriate firmware for my phone. I know that I must get it directly from cisco, but before purchasing it will be very good for me to try a bit; does someone can provide me (or even tell me where can I find) the SCCP image (version doesn't matters as long it works with asterisk) for the phone? Thanks in advance, Simone. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can't call some numbers/providers , ani code missing in sip header (found the problem, not the solution)
With the help of one of the providers we terminate on, I've found the
source of the problem of getting busy even when the called isn't really
busy in the absence of ANI codes in sip headers generated by asterisk.
If I put a NoOp(${CALLINGANI2}) in the dialplan before the dial I can
see it holds the value '0', but seems that value won't find the way to
the sip header.
Is this an error for asterisk to not put the code or a misconfiguration
of the remote switches to drop calls without it ?
(Have I to open a bug or to request a feature ?)
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[Asterisk-Users] is there a variable for the calling IP ?
I know there's a variable for the IP of a SIP channel, but I can't find if such a variable is avaliable for a generic voip cahnnel, or at least h323 channels (ooh323) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can't dial some particular numbers (providers ?) with asterisk sip / zap channels
I have a strange problem when calling some numbers with asterisk, I get an hangup for busy condition even if the phone at the other end isn't busy. I can route the calls via SIP to another carrier and then I have a SIP code 486 or I can terminate them on digium cards (E1) and I have an Hangup code 17 I know for sure that one of the numbers is hosted by a different provider than the one that has the de-facto monopoly here, so maybe is a final-provider problem, even if I don't understand what kind of strange signalling can reach that provider from my asterisk, I don't see nothing unusual on the cli, is like any other call ended for a real busy condition. More weird is that with the SIP route the called phone rings once, than stops and I get the 486. What have I've already tried : Set(CALLERID(number)=[a real traditional phone number]) before the dial SetTransferCapability(SPEECH) as far as I know the route calls follow is : linksys pap --sip-- asterisk (1.2 or 1.0) --iax-- asterisk server (1.2) --zap-- ..?.. Hangup cause 17 linksys pap --sip-- asterisk (1.2 or 1.0) --iax-- asterisk server (1.2) --sip-- ..?.. - 486 Busy here (but the end phone ringed once) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
Adam Robins ha scritto: Thanks, but we already have the TOS bits set to 0xB8, which matches the QoS settings in our switches and routers. This is definitely something that changed in the 1.07 to 1.24 upgrade. We have a pair of identical 1.07 servers connected via the same network pipe that do not exhibit these issues. I might try recompiling with the old jitterbuffer to see if it makes a difference. I've not 1.24 in producton yet, still 1.21, anyway I've noticed that restarting asterisk every night dramatically reduces complaints about choppy calls (I think is something about a memory leak and not jitterbuffer, anyway is something easy to do so it's worth trying) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mysql phone number pattern match query
I am not a mySQL expert (obviously), my limited SQL experience is with MS SQL where stored procedures and views are an option. This is with mySQL 4.x, so no views. I'm no an expert too, but even if the algorithm is right and seems to bring some optimization I think mysql way of do things can't leverage such a method Select dialpattern from rates where left 5 match left 5 of dst this is a select of a substring, I don't think mysql can index a substring, so the query will be redone completely every time Order by length of dialpattern, descending I'm pretty sure mysql isn't so good at sorting, you're wasting a little more time Compare dialpattern to the first x number of digits from dst where x = the length of dial pattern here you have another substring The first match (when ordered by length descending) is the correct result (longest match) Now of course the performance issue is relative since we are searching between two little strings and not for some book with 'asterisk' and 'future' in the title on amazon. Since performance isn't probably an issue I suggest a simple price = None for (i=1, i++, ilen(dialstring)) price = select price from rates where prefix = dialstring[0:len(dialstring)-i] if price != None break if price == None we don't know how to bill this call else do stuff you have an O(len(dialstring)) search but the code is simple and cpus are fast If you know your system will never call numbers shorter than m you can substitue len(dialstring) with len(dialstring)-m If performance is an issue maybe (never tried myself) you can split the prefixes table in one table for the first 4 chars, like 0011 America1 0012 America2 ... 0020 Egypt ... 0086 China ... and one table for every destination with the remaining part of the code, so you first do a select on the first 4 chars of the dialed number, you know you'll always have one and only one match. the match is the name of the table where to do the O(n) search, but now n is even smaller and there is also a smaller number of rows to search from. (too bad international prefixes aren't all of the same length, so the numbers in the tables have less sense and you probably need a little more complex billing application) If you need to investigate what is the better query use EXPLAIN in front of them, and look at how mysql will do the query, what index uses and how many lines will it go through ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] lists problem, Gmail????????
C F ha scritto: Am I the only one having trouble with this list? Since the begining of the week I have not been receiving mail from the list like I used to, is this a gmail problem? or is it subscription problem? or is something wrong with the list? anybody else using gmail having any problems? Yes, I'm also getting some lag sometimes, one or two days without receiving mails ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: asterisk logger - urgent!!!
Dov Bigio ha scritto: I found the problem. Master.csv reached 2.0GB and since the moment this happened Asterisk went crazy! Since I am using cdr-mysql, how do I disable the use of csvs? Thank you Dov Why don't you simply rotate the logs with logrotate ? (no, I don't know how to disable them) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Obtaining billsecs in the dialplan after a call?
[EMAIL PROTECTED] ha scritto:
Hi,
I'm stuck on a silly thing. I need to get the billsec CDR value after a
call. But I'm finding its always 0.
Here's my test code:
exten = *244*,1,Dial(Local/[EMAIL PROTECTED]/n,,g)
exten = *244*,n,Noop(after dial duration is ${CDR(duration)} billsec is
${CDR(billsec)})
exten = *244*,n,Hangup
[custom-tests]
exten = test,1,Answer
exten = test,n,Playback(tt-somethingwrong)
exten = test,n,Hangup
The actual CDR record that gets posted in Master.csv looks like so:
,200,*244*,default,Exten 200 200,SIP/200-94dd,Local/[EMAIL PROTECTED],1,Hangup,,2006-02-10
11:57:42,2006-02-10 11:57:42,2006-02-10 11:57:45,3,3,ANSWERED,DOCUMENTATION
So the duration is there just fine. But ${CDR(billsec)} remains stubbonly
0.
Now I don't really understand the CDR code 100% - but it looks like
billsec is only worked out then the cdr is posted. But there is no way to
force the cdr to be posted from the dialplan, is there?
You have to read that variable after the hangup, use the h extension
and / or
ResetCDR([options])
Causes the Call Data Record to be reset, optionally storing the current
CDR before zeroing it out (if 'w' option is specifed).
A CDR record *will* be stored for any activity following this command.
using h is cleaner in my opinion
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Re: [Asterisk-Users] double ringing tone on asterisk 1.2 ((better) workaround)
Matteo Piazza ha scritto: You must change in the indication.conf the country [general] country=it ; default location After reading a description of apparently the same problem by Juan J. Sierralta more detailed than mine tuuu tuuu instead of tuuu we've solved the problem changing the call progress tone of sip phones to something not udible. ___ No, we've solved the problem on the server setting progressinband=no in sip.conf, now you get a 'real' tone when the other endpoint starts ringing, instead of one tone generated by the client while the call is in progress and one when the endpoint is ringing. the country setting was already 'it' Now the problem remains only in h323 channels, but it seems there isn't such a variable to configure (ooh323) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can't hear 'service messages' when iax is in the middle
If I call a cellular phone while it's off, I can't hear the voice saying called number is unreachable, but only if I'm passing trough a iax channel. SIP client --- Asterisk --- SIP gateway, works SIP client --- Asterisk client --- Asterisk server --- SIP gateway, doesn't work (I can't put an explicit Answer in the extension for billing purposes) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Gateway and Context Issues
same problem here, made a workaround with an agi
Hi,
We are a service provider using Asterisk for our softswitch. We offer
SIP connections via IP phones as well as PRI and POTS replacements for
our customers. However, i am having problems with incoming calls from
a Cisco IAD2431 and its dialing context. When a call comes from the
PBX through the IAD2431 to Asterisk, the calls are not in the
customer1 context as they should be. Our customer is not able to dial
4 digit extensions to their other IP phones. See config examples
below.
Has anyone else experienced the problems I am having?
Thanks,
Sum Ding Wong
;---
; sip.conf
;---
[general]
context=default
srvlookup=yes
dtmfmode=inband
qualify=yes
nat=yes
host=dynamic
canreinvite=no
pedantic=no
disallow=all
allow=ulaw
allow=g729
allow=g723
allow=alaw
[ 12.34.43.3]
context=customer1
type=friend
qualify=200
host= 12.34.43.3
canreinvite=no
;--
; Extensions.conf
;--
[macro-voicemail]
;usage Macro(voicemail,extension,mailbox)
exten = s,1,Dial(${ARG1},15,rt)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG2})
exten = s-BUSY,1,Voicemail(b${ARG2})
exten = _s-.,1,Goto(s-NOANSWER,1)
[customer1]
include = voicemail
include = customer1-did
include = customer1-internal
include = outgoing-unlimited
[customer1-internal]
exten = _*.,1,Pickup(${EXTEN:1})
exten = 1272,1,Macro(voicemail,SIP/2125551272,2125551272)
exten = 1273,1,Macro(voicemail,SIP/2125551273,2125551273)
exten = 1274,1,Macro(voicemail,SIP/2125551274,2125551274)
exten = 1275,1,Macro(voicemail,SIP/2125551275,2125551275)
exten = 1276,1,Macro(voicemail,SIP/2125551276,2125551276)
[customer1-did]
exten = 2125551285,1,Dial( SIP/[EMAIL PROTECTED],30,rt )
exten = 2125551272,1,Macro(voicemail,SIP/2125551272,2125551272)
exten = 2125551273,1,Macro(voicemail,SIP/2125551273,2125551273)
exten = 2125551274,1,Macro(voicemail,SIP/2125551274,2125551274)
exten = 2125551275,1,Macro(voicemail,SIP/2125551275,2125551275)
exten = 2125551276,1,Macro(voicemail,SIP/2125551276,2125551276)
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___
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[Asterisk-Users] double ringing tone on asterisk 1.2 (workaround)
After reading a description of apparently the same problem by Juan J. Sierralta more detailed than mine tuuu tuuu instead of tuuu we've solved the problem changing the call progress tone of sip phones to something not udible. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT?: International number parsing
Ron hotmail ha scritto: The short answer is no, you will never have a situation where the 'local' part of the term number is mistaken for part of the dialcode. for example, your customer dials 0119647701773352 (Iraq mobile number) Iraq011964 Iraq-Baghdad 0119641 Iraq-Mobile 0119647701 this would cause a match on Iraq, and Iraq-Mobile, but not on baghdad, the 'most' accurate match would be the dialcode with the most digits... That's the way I'm doing it : let's MAX_PRE_LENGHT be the maximum lenght for a prefix (as today it's 10, for 0061891006, Australia Christmas Island) and DST_LENGTH the lenght of the called number (DST) for i in range(min(MAX_PRE_LENGTH, DST_LENGTH)): probablePrefix = DST[0:min(MAX_PRE_LENGTH, DST_LENGTH)-i] select probablePrefix from a table with all the prefixes (and other info you can need) if we found something that's the prefix, break to the application else continue with a smaller try From the original post it seems there are two tables, one for the country and one for the city, like having one table with 0011964 - Iraq and one with Iraq - 1 - Bagdad Iraq - 7701 - Mobile I don't know if this speed up things, in my case it surely won't since I have a large-grained detail for locating the call (I'm not interested in city codes, so for example I've only one entry for Italy, and not a lot of entries like 'Italy Milan', 'Italy Rome'...) so a join would slow the benefit of smaller values for MAX_PRE_LENGTH, it depends on the application. Seems that when you need to have fine-grained detail the search is made in reverse, for example message boxes for cellular phones : black box understanding warning if I call (not a real number, but I know a real example I won't post for obvious reasons) 345 - 333444555 while the cell is off I get a voice : answer 333444555, the phone is off, leave a message if I call 345 - 333444555 the message is the same : answer 333444555, the phone is off, leave a message so the search is made backwards, and the application starts as long as only one possible match is found. I don't even think we are talking about relational db here, probably some directory to speed up things with a tree-search, anyone working in the large who can confirm ? /black box understanding warning ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fast AGI Options. Eeeek!
Sig Lange ha scritto: I have successfully written FastAGI applications in python, and it was a good experience. Do you have some template code you can share ? or references to point us to ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan ooh323 choppy sound
I terminate some calls on a h323 device (a quescom gsmgateway) from asterisk 1.2.3 with ooh323, the customer is complayining about choppy sound on most of the calls, the only warning message I can see is : src/chan_h323.c:944 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_102 (the calls sounds perfectly with iax/zap termination and the quescom seems to work fine with other h323 devices like cisco iad 2400) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no nat, but one way only audio
I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no nat, but one way only audio (more info)
I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ? There's one configuration working : lynksys pap -sip- asterisk server -sip- quescom this way both sides can hear voice but with : lynksys pap connected to a switch -sip- asterisk client (on the same switch with one eth and on internet with another one) -iax asterisk server -sip- quescom voice can only be heard ---this way-- the client is 1.0 and the server 1.2 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: ignore me, just a test
sorry, just a test, seems I'm no more receiving mails ... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] double ringing tone on asterisk 1.2
Rich Adamson ha scritto:
the problem appears no matter where I terminate the call (IAX or
Zap), and I don't have that problem on a 1.0.7 connected to the same
PRI lines and IAX servers , what I have to check ? looked in confif
files but appears to be the same (indications, modules loaded, iax,
zapata, the dial in extensions is the same)
Are you by chance including an r option in the dialplan entries
associated with these calls?
If so, try removing it.
no, I'm including it on client asterisk boxes to partially solve the
problem (since it stops the double tuuu and sends a fake ringtone till
the call is answered), but it's not a nice solution since you have no
real call indication
Since there does not seem to be anyone else complaining about the same
problem, there must be something in your config that is causing it.
Without specific copy/paste samples of what you've configured, no one
is going to be able to guess at what you are doing.
Given the issue is happening with both PRI's and IAX links, I'd have to
guess that you've got something wrong in extensions.conf.
is something like :
exten = _X.,1,AGI(agidial.py|${EXTEN})
exten = _X.,n,Dial(${STR_DIAL})
where STR_DIAL is given the right value by the agi, something like
'/zap/g15/00511265'
and on the cli I can see
-- Accepting AUTHENTICATED call from xxx :
requested format = alaw,
requested prefs = (),
actual format = alaw,
host prefs = (alaw|g726),
priority = mine
-- Executing AGI(IAX2/xxx-39, agidial.py|00511265) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agidial.py
-- AGI Script agidial.py completed, returning 0
-- Executing Dial(IAX2/xxx-39, Zap/g15/00511265) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g15/00511265
the complete call path is :
lynksys pap -sip asterisk client -iax asterisk server -zap||iax
somewhere outside my scope
or :
lynksys pap -sip asterisk server -zap||iax somewhere outside my scope
the problem occurs in both cases
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[Asterisk-Users] CPU load (was: dimensioning: Where is the CPU vs Asterisk load table)
Erick Perez ha scritto: -And the most important I read was: Keep load under 5 in single CPUs and 10 in dual CPUs (didn't mention dual cores in the article). That seemed to me a lot, so i googled around a little trying to understand the true meaning of those numbers : I'll sum up here what I've found, sparing you the formulae (look for linux load average neil gunther) First of all the sampling of cpu load gives more weight to recent samples, so is better to look at the third value, average in the last 15 minutes, without being scared by high punctual values. Following what the gurus says the value should be kept below 3, or below the number of cpus, given what we are measuring (the number of process ready and waiting to be executed), those values means to me a rule of thumb and make no one wait to do his job. It's not a lot of meaning, is it ? What I suppose we want to say is when I start hearing the calls bad ?, like gamers don't care about FPS but want to know which graphic card I have to buy to frag aliens smoothly ?. I'm not a C programmer so I don't know asterisk internals, what I'll say now maybe is totally nonsense, I leave the sensate replies to the community. If I have an asterisk process waiting, is sensate to state that if it waits too long, when his turn comes he'll drop the packets as the timestamp on them is too old and sound quality will start decreasing ? If this is the case, isn't the important measure not how many are waiting but how long are they waiting ? Since the upper bound to load should be low enough so they don't have to drop . (as a fast reply I can say that I made some calls while my dual 3.0 Ghz was under load 5, and they sounded good, alaw no transcoding) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] double ringing tone on asterisk 1.2
While I wait for the call to be answered I hear a double ringing tone, like : expected tone : tuuu tuuu tuuu tuuu what I hear : tuuu tuuu tuuu tuuu tuuu tuuu tuuu tuuu the second tuuu I think is generated somewhere and not true, since it sounds slightly different and the lambda between the first and the second is always different. the problem appears no matter where I terminate the call (IAX or Zap), and I don't have that problem on a 1.0.7 connected to the same PRI lines and IAX servers , what I have to check ? looked in confif files but appears to be the same (indications, modules loaded, iax, zapata, the dial in extensions is the same) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] double ringing tone on asterisk 1.2
Rich Adamson ha scritto: Simone Cittadini wrote: the problem appears no matter where I terminate the call (IAX or Zap), and I don't have that problem on a 1.0.7 connected to the same PRI lines and IAX servers , what I have to check ? looked in confif files but appears to be the same (indications, modules loaded, iax, zapata, the dial in extensions is the same) Are you by chance including an r option in the dialplan entries associated with these calls? If so, try removing it. no, I'm including it on client asterisk boxes to partially solve the problem (since it stops the double tuuu and sends a fake ringtone till the call is answered), but it's not a nice solution since you have no real call indication ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR problem - incorrect time
Chris Mason (Lists) ha scritto: We have a billing system that depends on the CDRs. We had a guest that made a one minute call to a local cellphone, this call went out Zap channel through our channel bank. The CDR recorded a 200 minute call, but I checked with the Telco's records and it had terminated after one minute. What can cause this and what can I do to prevent it? happened to me once, I've noticed that the txt cdr (under /var/log/asterisk) was missing the line for that call, so probably something went wrong writing that record, it's not a solution but at least double checking the db cdr with the txt ones is a way to look for such errors. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nested MySQL Commands
Douglas Garstang ha scritto: Peter, I assume you mean something like this in extensions.conf: exten = _X.,1,AGI(master-dial-logic.pl) and then there's only one call. All logic would be performed by the perl script. This has many advantages. One disadvantage however is that potentially, there could be 120 simultaneous instances of this script running (one per call). Douglas. but you can use fastagi, it will be maybe a little more complex to write the server code but it should scale better, shouldn't it ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nested MySQL Commands
Douglas Garstang ha scritto: So I really wish there was some way to measure how well the worst case scenario would perform. This would be 120 simultaneous calls (don't know how many per second) on a Dual 3.8Ghz Dell PowerEdge 1850 with 2GB RAM. Asterisk would call an AGI script, written in perl, to route all calls. The script would have to perform multiple database queries in order to route a call. It will work if you need no transcoding, I tested a python agi doing something like 6 query to accept / instradate the call and it works for 150 / 200 simultaneous calls, the machine starts sweating of course, but the voice quality is still good, no drops. Mine is just a quick prototype, using fastagi or writing the agi in C is surely the way to go, imho fastagi will let you have a more configurable / customizable system since you can write the application in a object oriented language. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco as5400, sip, asterisk. cisco won't detect that the call is answered
We've got this configuration : Cisco as5400 --- asterisk main server asterisk for cells gsm gateway cisco and the gsm gateway are connected to asterisk via sip, the two asterisk servers are connected via iax. On a succesful call the cisco (not always, 60% of the times) will keep sending a ringtone to the connected phone, even if the call is answered, actually if the user behind the cisco talks the one after the gsm gateway will hear him, but not the contrary. (like when you have a problem with nat, plus the I'm still hearing the ringingtone problem) ((no, cisco is on a public IP, also the two asterisk servers, and all sip is canreinvite=no) the dial chain is something like : asterisk main server: [cisco context] X.,1,Dial(iax/[EMAIL PROTECTED] for cells) asterisk for cells: [cisco context] X.,1,Dial(sip/[EMAIL PROTECTED] gateway) If the main server dialplan becomes like : [cisco context] X.,1,Answer X.,n,Dial(iax/[EMAIL PROTECTED] for cells) the problem is solved, but all the calls are seen as answered by the cisco (well, they are) and this is not good for billing purposes. (the 'asterisk for cell' server writes the cdr duration / billsec correctly, but trust is not of the business world ) (there are some lynksis paps connected to the asterisk main server, and they work perfectly) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the best Dell Machine for Asterisk?
Zoa ha scritto: Something is using up way too much memory, are you sure asterisk is using 800mb of ram ? it should be ten times less. Zoa You're right, I forgot there are also huge mysql tables on the same machine (with no agi and transcoding) 80 alaw concurrent calls , cdr_mysql, terminating on one TE410 Mem: 3105772k total, 733928k used, 2371844k free,8k buffers Cpu(s): 5.0% user, 5.5% system, 0.0% nice, 89.5% idle load average: 0.37, 0.39, 0.41 So that is ~80 calls per GB of ram which is 20% of 400 users so that should be 5 or 6GB to handle 100% usage. The load avg is the most important here. You want to keep it under 1.00 or you have processes waiting which increases jitter. Your system will be at 80% usage with 160 calls, assuming linear scaling. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P E1 Red Alarm
Olivier Perrin ha scritto: Hi, You could only take timing from one E1 per card. So you should use : span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 instead of : span=1,1,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 span=3,1,0,ccs,hdb3,crc4 span=4,1,0,ccs,hdb3,crc4 Anyway it always worked for me with timing = 1 for all spans, if I unplug one span I see a nessage about changing the timing source and all keeps working ... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] machine load (was best dell a long time ago)
(with no agi and transcoding) 80 alaw concurrent calls , cdr_mysql, terminating on one TE410 Mem: 3105772k total, 733928k used, 2371844k free,8k buffers Cpu(s): 5.0% user, 5.5% system, 0.0% nice, 89.5% idle load average: 0.37, 0.39, 0.41 So that is ~80 calls per GB of ram which is 20% of 400 users so that should be 5 or 6GB to handle 100% usage. The load avg is the most important here. You want to keep it under 1.00 or you have processes waiting which increases jitter. Your system will be at 80% usage with 160 calls, assuming linear scaling. What are the specs for processor, memory and chipset that you pulled this stat from? ___ xeon 3 Ghz, kernel 2.4 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the best Dell Machine for Asterisk?
Mike Fedyk ha scritto: Hiu Yen Onn wrote: How big of RAM for Asterisk server? My production environment will be about 400 users in the office. In one server? 4GB. And more if you can. I'd suggest you use several servers for 400 users unless the percentage of active phones is ~10%. Mike (with no agi and transcoding) 80 alaw concurrent calls , cdr_mysql, terminating on one TE410 Mem: 3105772k total, 733928k used, 2371844k free,8k buffers Cpu(s): 5.0% user, 5.5% system, 0.0% nice, 89.5% idle load average: 0.37, 0.39, 0.41 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
Douglas Garstang ha scritto: The word from Kevin Fleming and Digium is that the use of realtime to support multiple Asterisk boxes sharing sip is not supported or even known to work at this point. What about IAX ? If I connect two asterisk servers to a common mysql backend (only iaxusers, no sip or extensions) will it : a) work smoothly, don't waste time optimizing your agi b) definitively will not work, you're doomed c) we don't know, try it and let us know ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem on ZAP channel
[EMAIL PROTECTED] wrote: Hello group members, This is my first mail to this list. I am having one problem. When I dial a number from zap channel, there's 5-6 seconds delay. Is there any way to reduce/remove this delay? First of all try to find where the delay stands. Dial the number with the CLI open, if the delay is after the last pressed button and the channel coming up in the cli is a phone problem, look for timeouts in the configuration (on my lynksys I can force the sending of the number with #, dunno if it is a standard or a feature). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] select codec based on extension
Leandro Rzezak ha scritto:
I'm having same problem. Were you able to solve it?
No, codecs became a secondary problem later in our project so we ended
up with 711 on all servers and more bandwidth, anyway the post refers
to asterisk 1.0.something and I never investigated the problem in more
detail... I think it's possible, usually when you receive no answers (as
the case of that post) you have made a really silly question :)
On 10/18/05, *Simone Cittadini* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I've the following installation :
|asterisk client| --- |asterisk server| --- |other asterisk
server|
all the connections are made in IAX, the client and first server
allows
711 and 729
the other server only allows 729 since it has low bandwidth at
disposal
all the numbers but a few are routed to a digium card in the first
server, the others are routed to the other server, this way :
[default]
exten = _123X.,1,Dial(IAX2/otherserver/${EXTEN})
exten = _123X.,2,Hangup
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Hangup
when I call 123456 from the client box ...
on the client :
Call accepted by asterisk server (format alaw)
on the server :
Call accepted by other asterisk server (format g729)
on the other server :
Called [EMAIL PROTECTED]
and then on the server in the middle :
Oct 18 18:00:37 NOTICE[2846]: channel.c:1724 ast_set_write_format:
Unable to find a path from alaw to g729
Oct 18 18:00:37 NOTICE[2846]: channel.c:1757 ast_set_read_format:
Unable
to find a path from g729 to alaw
since that something at the end of the call and the paps which sits
before the first asterisk server both have g729, I don't like too
much
having to pay to translate something which need not translation.
Is there a clever combination of sip.conf, iax.conf and
extensions.conf
I'm missing to solve my problem ?
___
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[Asterisk-Users] IAX media path, forcing server to stay in the middle
I can't find how to force an asterisk server to stay in the middle between two asterisk clients, the iax2 reinvite pulls the call out of the cdr, which is no good ... suppose A calls B for 10 minutes clientA --- server ---clientB in the server cdr I see an A-B call of some seconds and if I enable cdr in clientB I see the correct 10minutes billsec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX media path, forcing server to stay in the middle
Simone Cittadini ha scritto: I can't find how to force an asterisk server to stay in the middle between two asterisk clients, the iax2 reinvite pulls the call out of the cdr, which is no good ... suppose A calls B for 10 minutes clientA --- server ---clientB in the server cdr I see an A-B call of some seconds and if I enable cdr in clientB I see the correct 10minutes billsec ok, panic moment ended, found the answer in the wiki : With *notransfer=yes* you can prohibit Asterisk from stepping out of the media path and connecting the two endpoints directly to each other. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: how to make contribution in asterisk
Tejas Shah ha scritto: hi all, I am a newbie in asterisk. I am doing my project on implementing VoIP gateway.I installed asterisk 1.0.7 on Debian. This package was available in Debian-Sarge. For this implementation i choose asterisk.I just bought digitnetworks X100P PSTN card. I have some queries : Compile and install 1.2.1, it's a bit different (in a better way, of course) and there's no sense in learning something that will change soon. 1)For this project purpose, Is this card suitable and enough? i m just going to download 3-4 soft IP phones. Since this card has only one FXO port, I think with this i can get PSTN call on my soft IP phones and also i can make call from any soft IP phones to analog phone. whether i m thiking in right direction or not? Yes, if you want to assign a different number to every softphone and have the external dialer select the phone with a number placed after the did be warned that the call will be answered even in the softphone isn't, so the caller will pay just to wait for you to answer. (not sure on this, maybe there's a solution) 2) After installation of this card i will go for simple dialplan structure to confirm how this VoIP gateway works.Since i m new to asterisk, By doing this i will get better idea abt asterisk. Am i doing right? I usually go with : sip registration, registered sip calling Echo app (most useful to test nat issues), internal softphones calling each others, registered sip calling outside (to a cell, so I can look at the given did), outside call routed to an internal sip phone. 3) Since i m doing my project work, i hav e to show some implementation which should be my own. I heard about Asterisk Gateway Interface (AGI). So by using AGI what can i develop? since it uses PERL,PYTHON,PHP for development, which shd i go for. As all three are new for me. Which will be fast and easy to learn? Python, and learn a bit of object oriented programming too, it will come in hand if the project becomes complex 4)I think other option available for me is to do some modifeications in the source code? How much time it will require to analyse and understand the asterisk code? I m not so much comfortable with C programming. So whether it will be be suitable to go for this modification? how much time will be reuired to understand the code? (probable time in days). Or i shd go for AGI? Go for AGI. 5) Are there some other options available with which i can show that i have worked with asterisk and developed something new, so that i can showit as my project work? Actually I miss the exact meaning of project work, are you a student and is something like a pratical exam ? Are you totally free in what functionalities to implement ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unplugging E1 cables while asterisk running
Yesterday I've had to unplug one cable coming from a TE410 card to plug it in another hole, due to provider's changes in the patch panel. The calls on that span stopped working (can't create zap channel), the problem was solved restarting asterisk. Note that the PRI termination hasn't changed, only moved the cables connecting the card to it from one patch panel to another. The cable's guy told me that unplugging and quickly replugging E1 cables isn't a problem on traditional systems, anyone konws the reason or if it is a bug that should be reported ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unplugging E1 cables while asterisk running
C F ha scritto: What version are you running? In 1.0.9 and CVS HEAD of the 1.2 branch I do it all the time and I don't have to restart. 1.2.1, on a debian, on a dell. Dunno what it plugs into, some strange big machine with a lot of colored wires and a warning, lethal voltage written on it, maybe the brand was nortel ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Some values ignored when using static realtime
Hi, I'm experiencing a strange issue with static realtime. Seems that some values belonging to 'general' category (like, for example, rtptimeout, rtpholdtimeout, realm) are ignored. Running asterisk 1.2.1, I've tried both res_odbc and res_mysql (that one from asterisk-addons tarball) without luck. Should I file a bug in mantis? TIA, Simone. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] screen safe_asterisk does'nt spawn asterisk
Tzafrir Cohen ha scritto: On Thu, Dec 15, 2005 at 01:45:24PM +0100, Simone Cittadini wrote: screen -d -m asterisk -vvvcng works well for me, but I'd prefer to run safe_asterisk in production Any reason you need to run asterisk in a console? asterisk -r allows you to view the current console. First of all, I think my complain about 'screen safe_asterisk' not working was a nonsense, even if I'd get it work it would detach the safe_asterisk script and not asterisk's process Anyway, screen seems the only way to see agi's output (old discussion in the list, and some lines in the wiki), for example : agy.py : [...] def Write(self,data): Write unbuffered line output to STDERR. Ensures data is flushed out. sys.stderr.write(str(data) + \n) sys.stderr.flush() [...] myhagi.py : import agy.py import hgsm.py agiDo = agi.AGI() hGsm = hgsm.HGSM() dst = sys.argv[1] gatDst = hGsm.getGatewayFromDst(dst) agiDo.Write(gatDst: +str(gatDst)) this last line will print on the CLI with 'asterisk -vvvc' nothing is printed with 'safe_asterisk' - 'asterisk -r' so I must 'screen -d -m asterisk -vvvcng' - 'screen -r' ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the best Dell Machine for Asterisk?
Matt Florell ha scritto: The best Dell for a production environment Asterisk server is no Dell at all. They make some great workstations, but I've had many problems with their servers(as have many others on this list) when trying to use them in production for Asterisk. Take a look at the Digium compatibility list: http://www.digium.com/index.php?menu=compatibility *I've installed a PowerEdge 2850 - Xeon 3.0GHz/2MB, 800FSB with two TE410P in it, the cards didn't worked out of the box, but they worked after a couple of hours googling around, and it is in production since 3 months, never gone down. * *(I'm not advocating dell, actually I don't even like dell as a society, only sharing my experience) * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] screen safe_asterisk does'nt spawn asterisk
screen -d -m asterisk -vvvcng works well for me, but I'd prefer to run safe_asterisk in production anyway 'screen -d -m safe_asterisk' spawns no asterisk processes, anyone knows the reason ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] outgoing calls that last an unreasonably long time
Warren Burstein ha scritto: What is frustrating is that the cdr file shows the dst as T rather than as the phone number dialed. I realize that AbsoluteTimout causes it to jump to the T extension, but it would help to know who the user dialed (asking a week later isn't going to get any useful information out of the user). It's not in the log file, either - would increasing the log level help here? I don't know how this AbsolutTimeout works, anyway I put all the info I need in variables before the actual Dial, then in the h extension I call SetUserField() (or whatever is called), helps me keeping track of reasons for non-terminated calls ... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HOW TO: CDR Customer IP address where call came in from
Rehan Ahmed ha scritto: I dont see the ip in the Master.csv but you can view the IP when the call comes in on the CLI Window. I am guessing there must be a command or a way to record this ip in your CDR using AGI, we are using agi to make our own CDR but i would apreciate if some one can tell how to record the IP address of the caller. If you install the iaxusers/sipusers mysql backend (which everyone seems to call realtime) the ip will be stored in a 'ipaddr' column. You can put a select in some agi to retrieve the IP of the peer. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call simulators
Use asterisk itself to build a box which generates the calls. Maybe what some people misses (call simulators are quite a recurrent query on the list) is that you can move a text file with the equivalent of a manager API action Originate in the spool/asterisk/outgoing/ directory and the call will be placed, so it's quite simple to do some intensive test. http://www.asteriskguru.com/tutorials/astertest.html seems nice, never used and I read somewhere it wont compile out of the box with 1.2, but you have the source ... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to restric user to call only specified country
ram ha scritto:
i have local extensions
and i have connected sip provider account to call out side
but i have account can call any part of the world
how to restrict some of users should call only USA or any Other
In a hundred of ways, I think the most straightforward is making a table
in some database with two columns, 'user' and 'unpermitted_prefix',
then before the dial put an agi which sets a variable looked by a gotoif
which makes the difference between Dial and
Playback(you-are-not-important-enough-for-our-company-to-let-you-call-everywhere).
european-guy
I don't know how it works with prefixes in usa, maybe you can simply
chek for 00 at the begginging of the ${EXTEN} so no need for agi, anyway
the db solution is quite simple and adds little overhead, giving you
flexibility which can come in hand for future requests ...
/european-guy
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Re: [Asterisk-Users] logging performance, important impact?
Moises Silva ha scritto: How important is the impact i could have if I have a single entry log file in /etc/asterisk/logger.conf wich loggs everything, even debug level. This is sometimes important to us because it helps us to make a track of the issues some times we have with the system. I just want to know if there is a considerable impact in performance because of the writing of the logs. I haven't made benchmarks, but speaking out of my experience and knowing that asterisk debug level is very verbose I think it will have a sensible impact. I can remember a very slow samba installation due to the sysadmin forgetting to turn off the debug level of logging, it made the difference between we can use it and we switch back to windows, and I'm talking about a dozen of users, not big numbers. Are you sure debug level will help you tracking the issues ? Usually debug level info is for debug like what is the bottleneck ?, why my prepaid agi isn't doing the update on hangup ?, nothing you need to keep tracking once you are in production. imho Is better to log as few expected stuff as possible and as much unexpected stuff as possible. /imho Anyway autoanswering your question is pretty simple, put an agi which timestamps the first line of each extension and one for the last one, send a lot of calls in the system with and without debugging and look at the results. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A rather big setup.
Vedran Dakic ha scritto: How does Asterisk handle this kind of setup with one-two/cluster central server(s) and a bunch of other servers connected with IAX(2)? If you have local calls, do they go directly from phone to phone, do they go from phone to per-floor-Asterisk server, or they have to be interconnected via the main Asterisk server(s)/cluster? With SIP the default is to directly connect the phones once the call is setup (I think also in IAX), investigate canreinvite / nat. Of course you can't do call detail record for calls which aren't forced to pass from the server, see if it's a problem ... imho As for maintenance we have a dozen of pcs with asterisk installed, each of them is server for 8/10 sip phones and client to a central asterisk server which then connects to E1. Asterisk pcs are scattered around, they pass trough at least one natted network, usually two. Never a problem. Connecting all the SIP phones to SER load balancing to more than one asterisk server will make you learn a lot about sip internals, proxy, domains, authentication and other interesting stuff, but if you need to have a working sistems and can start from zero go iax and spare yourself a lot of frustration. /imho ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A rather big setup.
Vedran Dakic ha scritto: I can only guess that I should have the ability to deliver a solution that can do some 100/500 simultaneously. The only question is how powerful should be a machine (or machines) that could do around 100/500 simultaneously. And, just for the sake of knowing, what should the setup be alike if it was 240/1000 simultaneously? My suggestion is to buy the E1 cards first of all and put them in a test server, equipped with asterisk and all the relevant agi / db connections / moh etc.. Then loop the card with a crossover cable and run some test script to generate the medium and upper bound call flows. That should give you an idea of your cpu/ram requirements. In the second case there's no need for a cluster, a good server will do, (obviously a second server for backup is a good idea ). I'm assuming you can use a/ulaw to transmit the data, if bandwidth is a problem and you must compress cpu usage becomes a boottleneck to keep in mind. A/ulaw? I saw some reports that G.729 uses very little bandwidth and has a quality part granted (audio quality). It's not a question of hardware and/or CPU power, I have two dual Opteron configurations and could install some more, it's just the question of that setup running with quality audio and no unwanted events. G729 has a very good quality -considered the bandwidth used-, but if your customers are used to conventional telephony they will no doubt notice the difference, so go with G711 (probably alaw, since you use E1 I suppose you are in europe) Anyway if bandwidth is a problem consider ilbc / speex which are free and have good audio qualities also. Lastly a lot of the quality comes from a well configured phone, tweak with volumes and timeouts. I presume that I should have all of the phones using the same codec (so, no transcoding), and preferrably the same VoIP protocol. I have a choice there so everything's possible. Let's say - IP10s has H.323, SIP and MGCP firmwares, although I'd like to leave H.323 out of the story. Yes, leaving H323 out of the story is a good way to start the project :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A rather big setup.
Vedran Dakic ha scritto: I have been asked by the customer to deilver a big PBX-system based on Asterisk. The requirements are approximately: - up to 240 lines for making outside calls from the building - up to 1000 internal phone conversations (within the building) - scalable up to 300/1500 calls You mean 240 / 1000 simultaneous calls or 240 outside lines and 1000 internal phones ? In the second case there's no need for a cluster, a good server will do, ( obviously a second server for backup is a good idea ) I'm assuming you can use a/ulaw to transmit the data, if bandwidth is a problem and you must compress cpu usage becomes a boottleneck to keep in mind. I'm having ~80 concurrent calls from iax/sip to pri in alaw from an userbase of ~150 clients and the cpu is around 6% on a dual 2.8 Ghz. 1000 phones are a lot, and sip sometimes is an hassle (mostly nat), I don't know your network topology, but maybe you can consider to connect every group of phones to an asterisk pc and the pcs to the server via iax, which uses a little less bandwidth and most of all works out of the box. A pentium 400 can handle ~8 calls with ilbc, so every modern pc will do. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Asterisk combination to get around NAT
Stuart Hirst ha scritto: Has anyone successfully used SER and Asterisk together on the same server to get around NAT traversal issues. I have looked at many of the NAT traversal topics which either involve commercial products and significant costs or solutions such as STUN or proprietary systems such as xten. I've installed ser + mediaproxy + asterisk without much trouble following the docs you find at www.onsip.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Channel and jitter buffer
Hi, what's the current status of jb implementation in chan_sip? Are there any patches out there available to be applied to the brand new 1.2-release? Cheers, Simone. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dazed and Confused
Matt ha scritto: Hi, Just yesterday I got an amber light on my PowerEdge 2850 saying PCI Parity Error EB113 The on-screen message says: Uhhuh. NMI received. Dazed and confused, but trying to continue You probably have a hardware problem with your RAM chips I solved it putting the digium card in another pci slot (actually the first one) I think it also happened once when the card got too much red alarms for the pri coming down from provider's side, but can't be sure as the server is in housing and I don't know the exact moment when the screen went amber ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Large Implementation
Sixto Diaz ha scritto: I think that if you store the Dial Plan in a database instead of a flat file, there is no problem with the amount of extensions. Is this Ok? Sixto Diaz - Original Message - From: Dario M. Colombo [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, November 17, 2005 10:06 AM Subject: [Asterisk-Users] Large Implementation Hi, somebody has implemented Asterisk in one organizacion with amount of extenciones in the order of 20.000? Thanks. are you sure you really need that much extensions ? (I assume you mean the number of lines in extensions.conf) Probably (imho) you are missing the use of AGI/macros and regular expressions ... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with 827-4v and asterisk as a pstn GW
Hi, I've a problem with a cisco 827-4v and asterisk (1.0.9) acting as sip-to-pstn GW. The issue is that when a call comes in from the pstn, asterisk correctly contacts the router, which in turns send a 183 Session progress. Obviously, asterisk thinks that the telephone is not ringing (because it expects a 180 Ringing) and we have no ringback on the pstn side. Putting a ringing() in the dialplan is not an option. Anyone has suggestions? Cheers, Simone. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can't add zap channels to a group
I've asterisk 1.0.7 (debian package) with zaptel 1.2-beta1 (to avoid the
rmmod hangs the server problem already discussed here).
The card is a digium TE410P, configured in this way :
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=3,1,0,ccs,hdb3,crc4
bchan=63-77
dchan=78
bchan=79-93
span=4,1,0,ccs,hdb3,crc4
bchan=94-108
dchan=109
bchan=110-124
loadzone=it
defaultzone=it
(span 2 has problems at the physical level, so I've disabled it,
enabling it gives the same results and a lot of red alarms)
I want to group spans number 1, 2 and 3 and leave span 4 in a separate
group, so :
/etc/asterisk/zapata.conf
[channels]
language=it
context=default
switchtype=euroisdn
pridialplan=national
prilocaldialplan=national
signalling=pri_cpe
callerid=asreceived
usecallerid=yes
hidecallerid=no
usecallingpres=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
channel = 1-15
channel = 17-31
channel = 63-77
channel = 79-93
group=2
channel = 94-108
channel = 110-124
and in /etc/asterisk/extensions.conf :
exten = _1001X.,1,NoOp(EXTEN: ${EXTEN}, SIPCALLID: ${SIPCALLID})
exten = _1001X.,2,SetAccount(N01)
exten = _1001X.,3,Dial(Zap/G1/${EXTEN:4})
exten = _1001X.,4,Hangup
But when the first span is full, no more dials are made on the other
channels, and if I use g2 (tied to 1002 prefix in the same way) I get a
can't create zap chan, everyone is busy/congested)
If I Dial(Zap/3-63/${EXTEN}) for test I get an unknown option - ...
isn't that the syntax to dial a specific chan on a specific span ?
I looked everywere in the wiki and all seems to confirm the correctness
of my config files, but clearly something must be wrong ...
(when I start asterisk it shows the setup for all the channels, also zap
show channels shows them all)
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[Asterisk-Users] Getting output from agi scripts (python)
I don't get output in the cli from agi scripts when connecting to a running instance of asterisk. And that is all well and known : This is a known problem. Asterisk will only send STDERR from AGI scripts to the actual console Asterisk is running on I can't, don't want, to do the /usr/bin/screen -L -d -m -S asterisk /usr/sbin/asterisk -vgc trick So I putted in my python scripts some logging to file, it doesn't work. logger = logging.getLogger() logger.setLevel(logging.DEBUG) hdlr = logging.FileHandler(agi_log.txt) logger.addHandler(hdlr) logger.debug(foobar) hdlr.flush() hdlr.close() writes foobar in a file when called from shell, just creates the file if integrated in a agi. (I can't understand how It's a minor issue for most people. btw) suggestions ? tricks ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] select codec based on extension
I've the following installation :
|asterisk client| --- |asterisk server| --- |other asterisk server|
all the connections are made in IAX, the client and first server allows
711 and 729
the other server only allows 729 since it has low bandwidth at disposal
all the numbers but a few are routed to a digium card in the first
server, the others are routed to the other server, this way :
[default]
exten = _123X.,1,Dial(IAX2/otherserver/${EXTEN})
exten = _123X.,2,Hangup
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Hangup
when I call 123456 from the client box ...
on the client :
Call accepted by asterisk server (format alaw)
on the server :
Call accepted by other asterisk server (format g729)
on the other server :
Called [EMAIL PROTECTED]
and then on the server in the middle :
Oct 18 18:00:37 NOTICE[2846]: channel.c:1724 ast_set_write_format:
Unable to find a path from alaw to g729
Oct 18 18:00:37 NOTICE[2846]: channel.c:1757 ast_set_read_format: Unable
to find a path from g729 to alaw
since that something at the end of the call and the paps which sits
before the first asterisk server both have g729, I don't like too much
having to pay to translate something which need not translation.
Is there a clever combination of sip.conf, iax.conf and extensions.conf
I'm missing to solve my problem ?
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Re: [Asterisk-Users] compiling Asterisk 1.2 with zaptel and h.323
Lenz ha scritto: Hello list, I have prepared a small recipe on how to compile Asterisk 1.2 beta 1 with a TDM400 card and H.323. You can find it at http://www.oinko.net/astrecipes/index.php?n=102 Any comment / suggestion / modification /bugfix is welcome! I've found that when you compile zaptel in debian you must link /usr/src/kernel-headers-2.4.whatever to /usr/src/linux and zaptel-1.2 dir to /usr/src/zaptel, and make zaptel from there or it won't find a lot of stuff ... where kernel-headers-2.4.whatever is from the package specific to your architecture, generic deb won't do no need to modprobe zaptel and modprobe wctdm since zaptel is required by wc, just modprobe wc ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unloading TE110P bristuffed module cause kernel panic
Francesco Angi ha scritto: Hi folks, I've already searched the mailing list but no one else seems to have my same problem. I'm using Asterisk with the following configuration: Fedora Core 4 (but I also tried Fedora 3) 1 Digium TE110P 1 TDM40B 1 HFC-S 'Cologne' bristuff 0.2.0-RC8o (zaptel 1.0.9.2) I compiled right, I can load kernel modules but when I try to unload wcte11xp module (the one for TE110P card) I get a kernel panic: Kernel panic - not syncing: /usr/src/bristuff-0.2.0-RC8o/zaptel-1.0.9.2/wcte11xp.c:333: spin_lock(usr/src/zbristuff-0.2.0-RC8o/zaptel-1.0.9.2/wcte11xp.c:d9640004) already locked by /usr/src/bristuff-0.2.0-RC8o/zaptel-1.0.9.2/wcte11xp.c/887. (Not tainted) This happens if I load and unload by zaptel script or if modprobe or insmod 'by hand', then run ztcfg and the unload the module. No bristuffed zaptel works right and bristuffed zaptel module for TDM40B works right. The card does not share IRQ with other devices, anyway I tried to have only TE110P mounted on PCI slot and to change PCI slot where card is mounted. Nothing to do. I really don't know what else I can try. Thanks for help, _fangi_ Same problem with debian sarge on a dell and asterisk 1.0.7 from packages, unloading the module freezes the system, (rebooting the machine worked right), I installed zaptel 1.2beta and it seems to work, but I haven't really tested it, only loaded/unloaded/loaded and placed a couple of calls. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing/SPA-841/CDR Log
Dinesh Nair ha scritto: On 10/10/05 22:30 Waldo Rubinstein said the following: 1) When are asterisk CDR logs _normally_ generated? When the call arrives, when the call hangs up, or both? I have looked at the records when the call hangs up. But if you use a h extension, at the end of that extension ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing/SPA-841/CDR Log
Waldo Rubinstein ha scritto: You mean to say that it will ONLY log if I have an h extension or if I don't? Shouldn't it be logged no matter what? No, of course it logs no matter whats, I was meaning that if you have exten = h,1,... exten = h,2, ecc ... don't expect the h extension to have at disposal the cdr line in the db, the actual INSERT is done at the end of all extension processing (lost a day trying to figure out what's wrong with an agi before understanding that) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: TE410P not working (autoanswer)
Simone Cittadini ha scritto: I'm trying to install a TE410P this is what happens with compiled zaptel 1.0.9, 1.2-beta and 1.0.9 from http://updates.xorcom.com/iso/ this is my zaptel.conf (checked with the provider the values): span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=it defaultzone=it then I modprobe wct4xxp debug=1 t1e1override=15 and the kernel says : and zttool blinks : YEL/RED/REC T4XXP (PCI) Card 0 Span 1 starting asterisk changes a lot 'cause you get much more RED/REC than YEL/RED/REC Solved putting the digium in another pci slot ... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P not working
I'm trying to install a TE410P this is what happens with compiled zaptel 1.0.9, 1.2-beta and 1.0.9 from http://updates.xorcom.com/iso/ this is my zaptel.conf (checked with the provider the values): span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=it defaultzone=it then I modprobe wct4xxp debug=1 t1e1override=15 and the kernel says : Sep 30 16:12:40 localhost kernel: Zapata Telephony Interface Registered on major 196 Sep 30 16:12:40 localhost kernel: Found TE4XXP at base address df5ffc00, remapped to f8aa2c00 Sep 30 16:12:40 localhost kernel: TE4XXP version c01a0164, burst ON, slip debug: OFF Sep 30 16:12:40 localhost kernel: FALC version: 0005, Board ID: 00 Sep 30 16:12:40 localhost kernel: Reg 0: 0x17f2f400 Sep 30 16:12:40 localhost kernel: Reg 1: 0x17f2f000 Sep 30 16:12:40 localhost kernel: Reg 2: 0x Sep 30 16:12:40 localhost kernel: Reg 3: 0x Sep 30 16:12:40 localhost kernel: Reg 4: 0x0001 Sep 30 16:12:40 localhost kernel: Reg 5: 0x Sep 30 16:12:40 localhost kernel: Reg 6: 0xc01a0164 Sep 30 16:12:40 localhost kernel: Reg 7: 0x1000 Sep 30 16:12:40 localhost kernel: Reg 8: 0x Sep 30 16:12:40 localhost kernel: Reg 9: 0x00ff Sep 30 16:12:40 localhost kernel: Reg 10: 0x Sep 30 16:12:40 localhost kernel: TE4XXP: Launching card: 0 Sep 30 16:12:40 localhost kernel: TE4XXP: Setting up global serial parameters Sep 30 16:12:40 localhost kernel: Successfully initialized serial bus for unit 0 Sep 30 16:12:40 localhost kernel: Successfully initialized serial bus for unit 1 Sep 30 16:12:40 localhost kernel: Successfully initialized serial bus for unit 2 Sep 30 16:12:40 localhost kernel: Successfully initialized serial bus for unit 3 Sep 30 16:12:40 localhost kernel: Found a Wildcard: Wildcard TE410P (2nd Gen) so I do /sbin/ztcfg -vvv, which tells me : Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. while the kernel logs : Sep 30 16:21:07 localhost kernel: About to enter spanconfig! Sep 30 16:21:07 localhost kernel: TE4XXP: Configuring span 1 Sep 30 16:21:07 localhost kernel: Done with spanconfig! Sep 30 16:21:07 localhost kernel: TE4XXP: Configured channel 1 (TE4/0/1/1) sigtype 128 Sep 30 16:21:07 localhost kernel: Unassigning channel 0/1! Sep 30 16:21:07 localhost kernel: TE4XXP: Configured channel 2 (TE4/0/1/2) sigtype 128 Sep 30 16:21:07 localhost kernel: Unassigning channel 0/2! Sep 30 16:21:07 localhost kernel: TE4XXP: Configured channel 3 (TE4/0/1/3) sigtype 128 Sep 30 16:21:07 localhost kernel: Unassigning channel 0/3! Sep 30 16:21:07 localhost kernel: TE4XXP: Configured channel 4 (TE4/0/1/4) sigtype 128 Sep 30 16:21:07 localhost kernel: Unassigning channel 0/4! Sep 30 16:21:07 localhost kernel: TE4XXP: Configured channel 5 (TE4/0/1/5) sigtype 128 Sep 30 16:21:07 localhost kernel: Unassigning channel 0/5! Sep 30 16:21:07 localhost kernel: TE4XXP: Configured channel 6 (TE4/0/1/6) sigtype 128 Sep 30 16:21:07 localhost kernel: Unassigning channel 0/6! Sep 30 16:21:07 localhost kernel: TE4XXP: Configured channel
Re: [Asterisk-Users] Play sound on connect
Mir ha scritto: Thanks for your answer. This is not what the customer wants, they answer +500 calls a day, and dont want to say Welcome to BigCorp every time. They want a personal welcome file to be played to the caller every time they pick up the ringing phone. Maybe you can do a quick trick with the L option of dial : # *L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. The following special variables are optional for limit calls: (pasted from app_dial.c) * *LIMIT_PLAYAUDIO_CALLER* - yes|no (default yes) - Play sounds to the caller. * *LIMIT_PLAYAUDIO_CALLEE* - yes|no - Play sounds to the callee. * *LIMIT_TIMEOUT_FILE* - File to play when time is up. * *LIMIT_CONNECT_FILE* - File to play when call begins. * *LIMIT_WARNING_FILE* - File to play as warning if 'y' is defined. If *LIMIT_WARNING_FILE* is not defined, then special sound macro to auto-say the time left is used (You have [XX minutes] YY seconds). setting x to a very large value, y to something small, z to null, playing sounds also to the callee, and setting a personalized *LIMIT_CONNECT_FILE with a little agi script called before the Dial command ... * ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: passing variables to h extension
Tony Mountifield ha scritto:
It works for me (using CVS HEAD, but I'm sure it's worked in the past for
me on Stable too). I think there must be some other reason it's not working
for you.
Just done a little test for it, as follows...
My extensions.conf:
[vartest]
exten = _X.,1,SetVar(FRED=hello)
exten = _X.,2,NoOp(FRED=${FRED})
exten = _X.,3,Playback(demo-congrats)
exten = _X.,4,Hangup
exten = h,1,NoOp(FRED=${FRED})
Yes it always worked also for me, using 1.2-beta1, typing error in noops
used for debug was having me look in the wrong place to set the vars !
Sorry for the rtfm question then
anyway I now wonder even more why accounting is done via cron jobs in
php-agi apps you find around isn't that only a waste of resources,
since you have to tag in some way calls already accounted ?
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[Asterisk-Users] passing variables to h extension
Is there a way to pass variables/arguments to the h extension ?
for example :
[default]
exten = _1098933X.,1,NoOp(CARRIER TWT-TIM, EXTEN: ${EXTEN}},
SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN})
exten = _1098933X.,2,SetVar(_PROVA=bla)
[lot of stuff, agi, goto, tricks and magic that happens]
exten = _1098933X.,10,Dial(${CHAN_DEST},,L(360:3599900)) - don't
mind L, a quick hack for dtmf not working with sip
exten = _1098933X.,11,Hangup
exten = _1098933X.,12,Playback(no-credit)
exten = _1098933X.,13,Hangup
exten = h,1,NoOp(${PROVA})
When the calls hangup, no bla is printed on screen, I think it's fine,
since the variable is automatically trashed when the channel is
hungup., sigh ...
But I need to pass some variables from the calling extension to an AGI,
like :
exten = h,1,DeadAGI(update-credit.py|${CALLER}|${CALLED}|${CARRIER})
in order to decrement the amount of credit for each customer after every
call.
I've seen that in others prepaid systems built over asterisk the
updating of available credit is done in a cron job, have I to take it as
a sign that real-time billing is impossible ? Hope I haven't to
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Re: [Asterisk-Users] Re: MAX PRI for single server
Yes, you missed something: 4 PRIs = 92 Lines per Card * 4 Cards = 368 Lines Isn't that just in North America? I believe most of the world uses E1 PRIs with 30 lines per PRI. right, we are in italy here, 1 PRI == 30 lines (calls) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk pri heavy load testing (was MAX PRI for single server)
I have test 3.0GHz systems - Intel Desktop board. I've been testing with a TE405P with looped ports - 1 to 3, 2 to 4. My test is 20 second long calls with one side playing music on hold, the other playing gsm prompts. All channels full (60 calls out, 60 in). Niiice, can I ask what software/extension/script did you used to do such a test ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)
that still leaves me with a need for 30 ISDN lines. As far as I can tell most of the Digicom cards have 4 FXS ports and I've read on this list that at most two could coincide in a box simultaneously without causing an interupt flood. Is it true ? My boss is just asking me if it is possible to stuck 4* TE411P in a single server, for a total of 480 lines, someone can assure me it is possible/impossible (manageable/unmanageable) from real-life experience ? thanks * ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing - Disable accounts when balance gets 0 value
This billing is also able to set accounts balance and for each call. Now I need to disable accounts which balance gets a determined value. I was thinking on changing account pass for that specif account which we need to disable. And then in the sip.com reload info. Can you help me with new (new ways for doing so) or programing ideas too once billing server has not the same public IP than Asterisk server. I ll appreciate your comments ok. I use ser+radius to do authentication, this way I can disable users or groups of users in a standard way, without using tricks like changing passwords. (when your customer pays he expect to have the same password as before, have you saved it ? where ? in a safe way ?) radius has a mysql backend, so also no need to reload config files. Asterisk and radius share the same db, with some not-too-complex agi before the actual Dial you can do stuff like setting the call timeout based on the remaining credit, blocking the call if the credit is too much in the red, and so on... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
