[asterisk-users] Cannot call Alcatel PBX extn from SIP
Hi All
I have a TE110P card connected to Alcatel 4400 PBX thru PRI. Calls between SIP
extns are all okay but I cannot make or recieve calls between SIP and PBX. I
get :
WARNING[14759] app_dial.c: Unable to forward voice
in /var/log/asterisk/messages and following output on the CLI:
Executing Dial(SIP/shashi-08910350, Zap/g1/873|20) in new stack
-- Making new call for cr 32776
-- Requested transfer capability: 0x00 - SPEECH
Protocol Discriminator: Q.931 (8) len=45
Call Ref: len= 2 (reference 8/0x8) (Originator)
Message type: SETUP (5)
[04 03 80 90 a3]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability:
Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: A-Law (35)
[18 03 a9 83 81]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type: 3
Ext: 1 Channel: 1 ]
[28 0e 53 68 61 73 68 69 20 50 72 61 6b 61 73 68]
Display (len=14) $R[ Shashi Prakash ]
[6c 06 00 80 39 38 31 30]
Calling Number (len= 8) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown
Number Plan (0)
Presentation: Presentation permitted, user number
not screened (0) '9810' ]
[70 04 80 38 37 33]
Called Number (len= 6) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown
Number Plan (0) '873' ]
-- Called g1/873
Protocol Discriminator: Q.931 (8) len=10
Call Ref: len= 2 (reference 8/0x8) (Terminator)
Message type: CALL PROCEEDING (2)
[18 03 a9 83 81]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type: 3
Ext: 1 Channel: 1 ]
-- Processing IE 24 (cs0, Channel Identification)
-- Zap/1-1 is proceeding passing it to SIP/shashi-08910350
Protocol Discriminator: Q.931 (8) len=9
Call Ref: len= 2 (reference 8/0x8) (Terminator)
Message type: RELEASE (77)
[08 02 81 9c]
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user (1)
Ext: 1 Cause: Invalid number format (28), class = Normal
Event (1) ]
-- Processing IE 8 (cs0, Cause)
-- Channel 0/1, span 1 got hangup
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request
Protocol Discriminator: Q.931 (8) len=9
Call Ref: len= 2 (reference 8/0x8) (Originator)
Message type: RELEASE COMPLETE (90)
[08 02 81 90]
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user (1)
Ext: 1 Cause: Unknown (16), class = Normal Event (1) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/shashi-08910350' status is 'CHANUNAVAIL'
Hers's my extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
[sip]
exten = 9820,1,Dial(SIP/iyer)
exten = 9821,1,Dial(SIP/shweta)
exten = 9810,1,Dial(SIP/shashi)
exten = 873,1,Dial(Zap/g1/873)
[incoming]
exten = 9820,1,Dial(SIP/iyer)
exten = 9821,1,Dial(SIP/shweta)
exten = 9810,1,Dial(SIP/shashi)
exten = _XXX,1,Dial(Zap/g1/${EXTEN},20)
here's sip.conf
[general]
context=default ; Default context for incoming calls
bindport=5060 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
language=en ; Default language setting for all users/peers
[iyer]
type=friend
context=incoming
username=iyer
fromuser=iyer
callerid=K Y Iyer 9820
host=dynamic
nat=no
canreinvite=yes
disallow=all
allow=ulaw
alow=alaw
[shweta]
type=friend
context=incoming
username=shweta
fromuser=shweta
callerid=Shweta Jain 9821
host=dynamic
nat=no
canreinvite=yes
disallow=all
allow=gsm
allowguest=yes
allow=alaw
allow=ulaw
[shashi]
type=friend
context=incoming
username=shashi
fromuser=shashi
callerid=Shashi Prakash 9810
host=dynamic
nat=no
canreinvite=yes
disallow=all
allow=gsm
allow=alaw
allow=ulaw
here's zapata.conf
[trunkgroups]
[channels]
language=uk
context=default
switchtype=euroisdn
signalling=pri_cpe
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
overlapdial=yes
pridialplan=unknown
prilocaldialplan
[asterisk-users] Need help connecting Alcatel 4400 PBX to Asterisk
Title: Need help connecting Alcatel 4400 PBX to Asterisk Hi there I have a TE110P card fitted in my linux box running : Linux version 2.6.9-5.ELsmp ([EMAIL PROTECTED]) (gcc version 3.4.3 20041212 (Red Hat 3.4.3-9.EL4)) #1 SMP Wed Jan 5 19:30:39 EST 2005 I followed the installation steps on digium website...no errors reported. The modules seem to have loaded...here's what lsmod shows: Module Size Used by wcte11xp 30496 31 zaptel 196740 67 wcte11xp still the light on my card is offdoes that mean the card has not initialised properly? On loading Asterisk, I do not get any errors, but I do see these warnings: Parsing '/etc/asterisk/zapata.conf': Found Nov 1 11:57:21 WARNING[3454]: chan_zap.c:10874 setup_zap: Ignoring switchtype Nov 1 11:57:21 WARNING[3454]: chan_zap.c:10874 setup_zap: Ignoring signalling on running asterisk -cvvv . I do see Aterisk Ready at the end ...then What do these warnings mean? Also, I DO NOT get these lines on asterisk startup:- channel 0/1 successfully restarted on span 1 -- B-channel 0/2 successfully restarted on span 1 -- B-channel 0/3 successfully restarted on span 1 -- B-channel 0/4 successfully restarted on span 1 -- B-channel 0/5 successfully restarted on span 1 -- B-channel 0/6 successfully restarted on span 1 does that mean my channels are not available? *CLI zap show status Description Alarms IRQ bpviol CRC4 Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 *CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 --- here's my extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes [sip] exten = 9820,1,Dial(SIP/iyer) exten = 9821,1,Dial(SIP/shweta) exten = 9810,1,Dial(SIP/shashi) exten = 9851,1,Dial(Zap/g1/851,20) [incoming] exten = s,1,Answer() exten = s,2,Playback(hello-world) exten = s,3,Hangup() exten = 9821,1,Dial(SIP/shashi) exten = 9851,n,Dial(Zap/g1/851) --- here's zapata.conf [trunkgroups] trunkgroup = 1,16 spanmap =1,1,1 [channels] switchtype=euroisdn signalling=pri_cpe context=incoming language=uk group=1 callgroup=1 pickupgroup=1 echocancel=yes immediate=no channel = 1-15,17-31 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancelwhenbridged=yes musiconhold=default --- here's zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone = us defaultzone=us --- Now the problem I can call and talk SIP to SIP...here's what I see on asterisk CLI -- Executing Dial(SIP/iyer-09326480, SIP/shweta) in new stack -- Called shweta -- SIP/shweta-0932b9c0 is ringing But when I call zap extension, here's what I get: Executing Dial(SIP/iyer-09326480, Zap/g1/851|20) in new stack Nov 1 12:07:55 NOTICE[3513]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/iyer-09326480' status is 'CONGESTION' I have connected the PBX to digium card with the specified cable and done the settings in PBX specified at: http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI What am I doing wrong? I'd like to mention that on the Alcatel PX rack on the PRA2 card, the NO-SIGNAL (NOS) light comes on when I shut down my linux box but it's off when I load zapteldoesn't that mean that PBX is able to sync to my asterisk server? Any help would be greatly appreciated. Thanks in advance Kind Regards Shweta ___ --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] channel.c: Unable to request channel ZAP
Hi there I also get this error: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) whenever I try to call my Alcatel 4400 PBX etxn from SIP using TE110P. The output of zap show channels is: Chan Extension Context Language MusicOnHold pseudoincominguk default 1incominguk 2incominguk 3incominguk ...so on till 31incominguk except channel 16 which is delta All 30 channels show status FREE on the exchange end. Any help would be greatly appreciated. Regards Shweta -Original Message- From: [EMAIL PROTECTED] on behalf of Forrest Beck Sent: Thu 11/2/2006 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] channel.c: Unable to request channel ZAP What does zap show channels show? Are all the channels shown as in use? What is set in zapata.conf for resetinterval= ? If anything. I believe that resetinterval is used to reset unused channels for any channels that are left open. On 10/31/06, Asterisk [EMAIL PROTECTED] wrote: Hi All, I have one rather annoying problem...my PBX can work great for weeks, when suddenly I start receiving these messages when I try to make a zaptel call: Oct 31 13:52:47 NOTICE[15636] app_dial.c: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) Oct 31 13:52:49 NOTICE[15648] channel.c: Unable to request channel ZAP/g1/247 I'm using Sangoma A104 card (with four E1 spans), and these problems are only occurring on the first two spans (which are connected to a legacy PBX) - the second two spans, which are connected to the Telco, work perfectly. Even more: when these messages start to occur, I can hardly initiate any call via problematic two spans (1st and 2nd), where I can with no problem initiate a new call thru the unproblematic two spans (3rd and 4th). Restart of the Asterisk is the only cure so far. Does anyone know what could possibly be the cause, or how could I troubleshot this problem? Regards. Alex ___ --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 winmail.dat___ --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
