Shariq Khan
2008-Aug-30 17:17 UTC
[asterisk-users] Congestion in Outgoing call through PRI
When i dial out any number through PRI it gives the following error every
time, while incoming calls works fine
I have sangoma E1 PRI card.
-- Executing Dial("SIP/2000-081b9938",
"Zap/g0/03333501125||") in new
stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/03333501125
-- Zap/1-1 is proceeding passing it to SIP/2000-081b9938
-- Zap/1-1 is making progress passing it to SIP/2000-081b9938
-- Channel 0/1, span 1 got hangup request
-- Zap/1-1 is circuit-busy
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("SIP/2000-081b9938", "") in new
stack
== Spawn extension (default, 9203333501125, 2) exited non-zero on
'SIP/2000-081b9938'
Zaptel.conf
----------------
loadzone=us
defaultzone=us
#Sangoma A101 port 1 [slot:4 bus:5 span:1] <wanpipe1>
span=1,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
Zapata.conf
-----------------
[trunkgroups]
[channels]
context=default
usecallerid=no
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;Sangoma A101 port 1 [slot:4 bus:5 span:1] <wanpipe1>
switchtype=euroisdn
context=from-pstn
group=0
signalling=pri_cpe
channel =>1-15,17-31
extensions.conf
-----------------------
[globals]
;CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=Zap/g0 ; Trunk interface
[from-pstn]
exten => 4392839,1,Answer
exten => 4392839,2,Wait(1000)
exten => 4392839,3,Goto(default,1000,1)
[default]
exten => 1000,1,Playback(transfer)
exten => 1000,2,Hangup
exten => _92X.,1,Dial(${TRUNK}/${EXTEN:2},,)
exten => _92X.,2,Hangup
sip.conf
-----------
[1000]
type=friend
secret=1000
host=dynamic
disallow=all
allow=alaw
allow=ulaw
Where i m on the mistake............
Shariq
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Grygoriy Dobrovolskyy
2008-Aug-30 18:00 UTC
[asterisk-users] Congestion in Outgoing call through PRI
2008/8/30 Shariq Khan <shariqrazakhan at gmail.com>> When i dial out any number through PRI it gives the following error every > time, while incoming calls works fine > I have sangoma E1 PRI card. > > -- Executing Dial("SIP/2000-081b9938", "Zap/g0/03333501125||") in new > stack > -- Requested transfer capability: 0x00 - SPEECH > -- Called g0/03333501125 > -- Zap/1-1 is proceeding passing it to SIP/2000-081b9938 > -- Zap/1-1 is making progress passing it to SIP/2000-081b9938 > -- Channel 0/1, span 1 got hangup request > -- Zap/1-1 is circuit-busy > -- Hungup 'Zap/1-1' > == Everyone is busy/congested at this time (1:0/1/0) > -- Executing Hangup("SIP/2000-081b9938", "") in new stack > == Spawn extension (default, 9203333501125, 2) exited non-zero on > 'SIP/2000-081b9938' > > Zaptel.conf > ---------------- > loadzone=us > defaultzone=us > > #Sangoma A101 port 1 [slot:4 bus:5 span:1] <wanpipe1> > span=1,0,0,ccs,hdb3,crc4 > bchan=1-15,17-31 > dchan=16 > > > Zapata.conf > ----------------- > [trunkgroups] > > [channels] > context=default > usecallerid=no > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > canpark=yes > cancallforward=yes > callreturn=yes > echocancel=yes > echocancelwhenbridged=yes > relaxdtmf=yes > rxgain=0.0 > txgain=0.0 > group=1 > callgroup=1 > pickupgroup=1 > > immediate=no > > ;Sangoma A101 port 1 [slot:4 bus:5 span:1] <wanpipe1> > switchtype=euroisdn > context=from-pstn > group=0 > signalling=pri_cpe > channel =>1-15,17-31 > > extensions.conf > ----------------------- > > [globals] > ;CONSOLE=Console/dsp ; Console interface for > demo > TRUNK=Zap/g0 ; Trunk interface > > [from-pstn] > > exten => 4392839,1,Answer > exten => 4392839,2,Wait(1000) > exten => 4392839,3,Goto(default,1000,1) > > [default] > > exten => 1000,1,Playback(transfer) > exten => 1000,2,Hangup > > exten => _92X.,1,Dial(${TRUNK}/${EXTEN:2},,) > exten => _92X.,2,Hangup > > sip.conf > ----------- > [1000] > type=friend > secret=1000 > host=dynamic > disallow=all > allow=alaw > allow=ulaw > > Where i m on the mistake............ > > > Shariq > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >Update to latest libpri and tell us if it still demonstrates the problem, use HEAD version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080830/f390603b/attachment.htm
Octavio Ruiz
2008-Sep-03 02:56 UTC
[asterisk-users] Congestion in Outgoing call through PRI
On Sat, Aug 30, 2008 at 12:17 PM, Shariq Khan <shariqrazakhan at gmail.com> wrote:> When i dial out any number through PRI it gives the following error every > time, while incoming calls works fine > I have sangoma E1 PRI card.The output of a CLI> pri intese debug at Asterisk CLI before make a test call would be very useful, libPRI 1.4.7 is just fine. -- Octavio H. Ruiz Cervera Tel.: (+52 55) 8590-9000 Ext. 7016 Mobile: (+52 1 55) 14-087790 Mobile: (+52 1 55) 41-351242
Richard Lyman
2008-Sep-03 15:33 UTC
[asterisk-users] Congestion in Outgoing call through PRI
Octavio Ruiz wrote:> On Sat, Aug 30, 2008 at 12:17 PM, Shariq Khan <shariqrazakhan at gmail.com> wrote: > >> When i dial out any number through PRI it gives the following error every >> time, while incoming calls works fine >> I have sangoma E1 PRI card. >> > > The output of a > > CLI> pri intese debug > > at Asterisk CLI before make a test call would be very useful, libPRI > 1.4.7 is just fine. > >I am amazed no one else have suggested trying a different phone type like an IAX2 softphone. (if i am right, this will work) I get the feeling you are using a SIP based phone (hard or soft) and are having issues with your router. (i suggest you use port triggering, not port forwarding (if you stick with SIP)) Note: i am going from memory as to your initial CLI output.