[asterisk-users] CDR in case of CallForwarding
Hello users,
i am looking for a solution in terms of CDR for the outbound only call.
presently i have the following setup.
//extensions.conf
[from-outside]
exten = _X.,1,NoOp(IncomingCall)
exten = _X.,n,BackGround(choce.wav)
exten = _X.,n,WaitExten(5)
exten = _X.,n,Hangup
exten = _1XX.,n,NoOp(1XX series Dialing)
exten = _1XX.,n,Dial(SIP/${EXTEN},60,rg)
exten = _1XX.,n,NoOp(${DIALSTATUS})
exten = _1XX.,n,GotoIf($[ ${DIALSTATUS} = BUSY | ${DIALSTATUS} = CONGESTION
| ${DIALSTATUS} = HANGUP | ${DIALSTATUS} = CHANUNAVAIL ] ?dialmobile:end)
exten = _1XX.,n(dialmobile),Dial(SIP/${DBQUERY AND GET THE mobileNUMBER
FOR THE us...@ougoingprovider,60,r)
exten = _1XX.,n(end),Hangup()
exten = _2XX.,n,NoOp(2XX series Dialing)
exten = _2XX.,n,Dial(SIP/${EXTEN},60,rg)
exten = _2XX.,n,NoOp(${DIALSTATUS})
exten = _2XX.,n,GotoIf($[ ${DIALSTATUS} = BUSY | ${DIALSTATUS} = CONGESTION
| ${DIALSTATUS} = HANGUP | ${DIALSTATUS} = CHANUNAVAIL ] ?dialmobile:end)
exten = _2XX.,n(dialmobile),Dial(SIP/${DBQUERY AND GET THE mobileNUMBER
FOR THE us...@ougoingprovider,60,r)
exten = _2XX.,n(end),Hangup()
//sip.conf
[outgoingprovider]
username=X
secret=y
port=
host=dfdfddf
fromuser=
- i am planning to take the number of calls made and the minutes spent
incase of mobile call forwarding
as it uses my outbound trunk by giving the accountcode set to a
particular call.
- but i am getting the total call (sip call + mobile call) as a single
record in my cdr record for a given accountcode.
- i need to get something like SIP/mobilenumber either in lastdata or
dstchannel associated accountcode as a separate cdr entry.
i tried with disabling cdr using NoCDR for the SIP call but for the
mobile call if i use ResetCDR() also i am totally
losing the callrecord.
- i tried with the ForkCDR() too but of no use..
is my requirement can be fulfilled by tweaking some changes in the
extensions.conf functions/applications??
please advise as i need to bill the user for the outbound calls only...
any help is sincerely appreciated. thanks in advance.
srinivas
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[asterisk-users] Dialing OutBound SIP trunk using Dial() command
Hello users,
i am working on directly calling the numbers from the sip provider of my
choice from asterisk using Dial command as follows.
extensions.conf
[dial-out]
exten = _XX,1,NoOp(Dialing out)
exten =
_XX,n,Dial(SIP/1{EXTEN}:password:md5secret:authname:tarnsp...@host:port
, 20,r)
exten = _XX,n,Hangup()
//so i am trying to call the number using voip provider details i have
but i am getting the following error in asterisk CLI
SIP/408XXX:x::XXX:u...@xx
Called 140:x::XXX:u...@xx
-- SIP/xx-0a155070 is circuit-busy
when i try with other service provider i am getting a similar error in
asterisk CLI
SIP/1408X:y::YY:u...@yyy
Got SIP response 500 Nice try back from 64.xx.xx.xx
-- SIP/yyy-0a16ac20 is circuit-busy
my idea is to allow users to enter their own voip providers for outgoing
calls
so that customer can use his own voip provider
i am NOT LOOKING FOR A SOLUTION in /etc/sip.conf entries
like
register = username:passw...@myprovider
[myprovider]
username=
secret=
fromuser=
fromdomain=
host=
any help is appreciated.
Thanks
srinvias
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[asterisk-users] Free FaxForAsterisk ReceiveFAX not working
Hello users,
Recently i have installed the free version of FaxForAsterisk and trying to
work with it by sending a fax
on T38.
My version information is as follows
i)Asterisk 1.6.0.20
ii)res_fax-1.6.0.14_1.1.6-x86_32
iii)res_fax_digium-1.6.0.14_1.1.6-i686_32
sip.conf
[general]
t38pt_udptl=yes
extensions.conf
[default]
exten = _XX,1,NoOp(Fax Incoming Call)
exten = _XX,n,GoTo(faxin,${EXTEN},1)
[faxin]
exten = _XX,1,NoOp(This is ReceiveFAX application Testing)
exten = _XX,n,Wait(6)
exten = _XX,n,NoOp(*** SETTING FAXOPTS *)
exten = _XX,n,Set(FAXOPT(ecm)=yes)
exten = _XX,n,Set(FAXOPT(localstationid)= 1234567890)
exten = _XX,n,Set(FAXOPT(maxrate)=14400)
exten = _XX,n,Set(FAXOPT(minrate)=2400)
exten = _XX,n,Set(FAXOPT(modem)=V17)
exten = _XX,n,Wait(6)
exten = _XX,n,NoOp(* RECEIVING FAX *)
exten = _XX,n,ReceiveFAX(/root/receivefax.tif)
exten = h,1,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten = h,n,NoOp(FAXOPT(filename) : ${FAXOPT(filename)})
exten = h,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten = h,n,NoOp(FAXOPT(locastationid) : ${FAXOPT(localstationid)})
exten = h,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten = h,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten = h,n,NoOp(FAXOPT(pages) : ${FAXOPT(pages)})
exten = h,n,NoOp(FAXOPT(rate) : ${FAXOPT(rate)})
exten = h,n,NoOp(FAXOPT(remotestationid): ${FAXOPT(remotestationid)})
exten = h,n,NoOp(FAXOPT(resolution) : ${FAXOPT(resolution)})
exten = h,n,NoOp(FAXOPT(status) : ${FAXOPT(status)})
exten = h,n,NoOp(FAXOPT(statusstr) : ${FAXOPT(statusstr)})
exten = h,n,NoOp(FAXOPT(err) : ${FAXOPT(error)})
and my asterisk cli information is as follows
asterisk CLIfax show version
FAX For Asterisk Components:
Applications: 1.6.0.14_1.1.6
Digium FAX Driver: 1.6.0.14_1.1.6 (optimized for i686_32)
Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Executing [2016289...@default:1] NoOp(SIP/204.16.59.19-0014, Fax
Incoming Call) in new stack
-- Executing [2016289...@default:2] Goto(SIP/204.16.59.19-0014,
faxin,2016289913,1) in new stack
-- Goto (faxin,2016289913,1)
-- Executing [2016289...@faxin:1] NoOp(SIP/204.16.59.19-0014, This is
ReceiveFAX application Testing) in new stack
-- Executing [2016289...@faxin:2] Wait(SIP/204.16.59.19-0014, 6) in
new stack
-- Executing [2016289...@faxin:3] NoOp(SIP/204.16.59.19-0014, ***
SETTING FAXOPTS *) in new stack
-- Executing [2016289...@faxin:4] Set(SIP/204.16.59.19-0014,
FAXOPT(ecm)=yes) in new stack
-- Executing [2016289...@faxin:5] Set(SIP/204.16.59.19-0014,
FAXOPT(localstationid)= 1234567890) in new stack
-- Executing [2016289...@faxin:6] Set(SIP/204.16.59.19-0014,
FAXOPT(maxrate)=14400) in new stack
-- Executing [2016289...@faxin:7] Set(SIP/204.16.59.19-0014,
FAXOPT(minrate)=2400) in new stack
-- Executing [2016289...@faxin:8] Set(SIP/204.16.59.19-0014,
FAXOPT(modem)=V17) in new stack
-- Executing [2016289...@faxin:9] Wait(SIP/204.16.59.19-0014, 6) in
new stack
-- Executing [2016289...@faxin:10] NoOp(SIP/204.16.59.19-0014,
* RECEIVING FAX *) in new stack
-- Executing [2016289...@faxin:11] ReceiveFAX(SIP/204.16.59.19-0014,
/root/receivefax.tif) in new stack
-- Channel 'SIP/204.16.59.19-0014' receiving FAX '/root/receivefax.tif'
[Dec 31 17:39:55] NOTICE[23578]: res_fax.c:712 generic_fax_exec: Negotiating
T.38 for receive on SIP/204.16.59.19-0014
[Dec 31 17:39:55] NOTICE[23578]: res_fax.c:779 generic_fax_exec: Negotiated
T.38 for receive on SIP/204.16.59.19-0014
-- Channel 'SIP/204.16.59.19-0014' FAX session '14' started
-- FAX handle 0: [ 000.000103 ], STAT_EVT_STRT_RX st: IDLE rt: IDLENSRX
-- FAX handle 0: [ 000.000165 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY rt:
RRDYNHRY
-- FAX handle 0: [ 000.000194 ], STAT_INFO_CSI
-- FAX handle 0: [ 000.000246 ], STAT_INFO_DIS
-- FAX handle 0: [ 005.252355 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt:
WDSRNT21
-- FAX handle 0: [ 008.502165 ], STAT_EVT_T4_EXP st: WT_DIS_RSP rt: NT4X
-- FAX handle 0: [ 008.502187 ], STAT_EVT_FSC_ERR st: WT_DIS_RSP rt:
RXXXNFRX
-- FAX handle 0: [ 008.502203 ], STAT_INFO_CSI
-- FAX handle 0: [ 008.502254 ], STAT_INFO_DIS
-- FAX handle 0: [ 010.671745 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt:
WDSRNT21
-- FAX handle 0: [ 014.001042 ], STAT_EVT_T4_EXP st: WT_DIS_RSP rt: NT4X
-- FAX handle 0: [ 014.001060 ], STAT_EVT_FSC_ERR st: WT_DIS_RSP rt:
RXXXNFRX
-- FAX handle 0: [ 014.001072 ], STAT_INFO_CSI
-- FAX handle 0: [ 014.001116 ], STAT_INFO_DIS
-- FAX handle 0: [ 016.172621 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt:
WDSRNT21
-- FAX handle 0: [ 019.502918 ], STAT_EVT_T4_EXP st: WT_DIS_RSP rt: NT4X
-- FAX handle 0: [ 019.502934 ], STAT_EVT_FSC_ERR st: WT_DIS_RSP rt:
RXXXNFRX
-- FAX handle 0: [ 019.502944 ], STAT_INFO_CSI
-- FAX handle 0: [ 019.502987 ], STAT_INFO_DIS
-- FAX handle 0: [ 021.671498 ], STAT_EVT_TX_V21_DONE
[asterisk-users] Asterisk Manager API - DTMF issues
Hello users, i have been testing the DTMF tone detection using originate command both from Asterisk CLI and java API. but my DTMF entry at the originate user is not getting detected by the asterisk in both the cases what i should do to make it work any help will be appreciated. my versions i)asterisk 1.4.25 ii)SkyepeForAsterisk 1.4_1.0.6 asterisk CLI originate Skype/xx extension 1...@testing #extensions.conf [testing] exten = 1000,1,NoOp() exten = 1000,n,BackGround(welcome) exten = 1,1,Dial(SIP/101,20,r) exten = 1,n,Hangup() exten = 2,1,Dial(SIP/102,20,r) exten = 2,n,Hangup() Thanks in advance srinivas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SkypeForAsterisk
Hello users, i am planning to forward my skype calls from skype to the asterisk registerd skype. The scenario is as follows. i)SkypeUserA calls SkypeUserB ii)SkypeUserB forwards his calls to SkypeUserC iii)SkypeUserC sees he got call from SkypeUserA. do i have a way to extract the SkypeUserB's details so that i can control who can forward the calls to my asterisk box. Thanks in advance Srinivas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gtalk Asterisk integration
Hello users,
I am trying to integrate asterisk and gtalk.
my configuration is as follows
OS:centos
asterisk-1.6.0
asterisk-addons-1.6.0
dahdi-linux-2.2
dahdi-tools-2.2
libpri-1.4 share
iksemel-1.2
#/etc/asterisk/jabber.conf
[general]
debug=yes
autoprune=no
autoregister=no
[google]
type=client
serverhost=talk.google.com
username=x...@gmail.com
secret=x
port=5222
usetls=yes
usesasl=yes
statusmessage=Invox Google Talk
timeout=100
# /etc/asterisk/gtalk.conf[general]
context=google-in
bindaddr=192.168.1.74
allowguest=yes
[guest]
;disallow=all
allow=ulaw
context=google-in
[test]
username=y...@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=google
# /etc/asterisk/extensions.conf
[google-in]
;Incoming
exten = s,1,NoOp(call from Google Talk)
exten = s,n,Set(CALLERID(name)=From Google Talk)
exten = s,n,Dial(SIP/1000,20,r)
exten = s,n,Hangup()
;Outgoing
exten = 100,1,JabberStatus(google,y...@gmail.com,STATUS)
exten = 100,n,NoOp(Jabber Status=${STATUS})
exten = 100,n,Dial(Gtalk/google/invoxgt...@gmail.com/Talk)
exten = 100,n,Hangup()
# /etc/asterisk/rtp.conf
rtpstart=1650
rtpend=4560
ports opened on the router
tcp 443 -incoming, outgoing
tcp 5222-incoming,outgoing
udp- all open incoming, outgoing
- i am able to call from my external gtalk client to the server configured
user
# this case is working fine
Executing [...@google-in:1] NoOp(Gtalk/Y-49af, call from Google
Talk) in new stack
Executing [...@google-in:2] Set(Gtalk/Y-49af, CALLERID(name)=From
Google Talk) in new stack
Executing [...@google-in:3] Dial(Gtalk/Y-49af, SIP/1000,20,r) in new
stack
Using SIP RTP CoS mark 5
-- Called 1000
- when i try to call from asterisk to the external client
# this case is not working and throwing following error
Executing [...@google-in:1] JabberStatus(SIP/1000-000e, google,
yy...@gmail.com,STATUS) in new stack
Executing [...@google-in:2] NoOp(SIP/1000-000e, Jabber
Status=1) in new stack
Executing [...@google-in:3] Dial(SIP/1000-000e, Gtalk/google/
yy...@gmail.com/Talk) in new stack
[Nov 30 16:22:25] ERROR[16255]: chan_gtalk.c:932 gtalk_alloc: no gtalk
capable clients to talk to.
[Nov 30 16:22:25] WARNING[16255]: app_dial.c:1518 dial_exec_full:
Unable to create channel of type 'Gtalk' (cause 0 - Unknown)
Everyone is busy/congested at this time (1:0/0/1)
Executing [...@google-in:4] Hangup(SIP/1000-000e, ) in new
stack
Spawn extension (google-in, 100, 4) exited non-zero on
'SIP/1000-000e'
###
asterisk cli out put
###
jabber show connected
User: xx...@gmail.com - Connected
jabber show buddies
yy...@gmail.com
localhost*CLI Resource: Talk.v104C77D0BCE
localhost*CLI node: http://www.google.com/xmpp/client/caps
localhost*CLI version: 1.0.0.104
localhost*CLI Jingle capable: yes
gtalk show channels
Channel Jabber ID Resource
Read Write
0 active gtalk channels
i am getting error no gtalk capable clients to talk to
i tried with both asterisk 1.4.25 version and asterisk 1.6.0 but no
difference
anybody can help me out finding a forward move on this???
Thanks in advance
srinivas
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[asterisk-users] Swift application and DTMF
Hello users,
i have successfully installed the cepstral voice and in the text only mode
its working fine
when i swift applicaiton in dtmf mode like
exten =111,1,Swift(hello user| 5000|1)
exten =111,n,NoOp(dtmf is ${SWIFT_DTMF})
exten = 111,n,Hangup()
case1:
when i am listening to the hello user prompt if i press any key
1,2,3,4,5,6,7,8,9,0,*,#
i am getting the ${SWIFT_DTMF } value as
1xx -- if i press 1
2xx -- if i press 2
and this is the same for all other digits including 0,#,* keys
and the prompt stops
i am getting following in my asterisk CLI
app_swift.c:453 engine:DTMF=#147987812 --when i press #
app_swift.c:453 engine:DTMF=7147987812 --when i press 7
case2:
when i pressed the above digits after the prompt finishes
i get 1 -- if i press 1
2 --if i press 2
etc.
except there is no DTMF detection
for numbers 0(zero),#,*
and i am getting the following in my asterisk CLI
app_swift.c:453 engine:DTMF=7 --when i press 7
app_swift.c:482 engine: No DTMF --when i press #,0,*
Please help me out
Thanks in advance
srinivas antarvedi
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[asterisk-users] X100P FXO PCI card not receiving calls
Hello users, i have recently purchased Authentica x100p Fxo card for asterisk 1.4 i have following settings # /etc/zaptel.conf fxsks=1 loadzone=in defaultzone=in # /etc/asterisk/zapata.conf [channels] context=from-pstn usecallerid=no hidecallerid=yes immediate=no signalling=fxs_ks echocancel=yes channel = 1 #/etc/asterisk/extensions.conf [from-pstn] exten = s,1,Answer() exten = s,n,Playback(vm-intro) exten = s,n,Hangup() # lspci -vv 11:00.0 Communication controller: Motorola Wildcard X100P Subsystem: Efar Microsystems: Unknown device 0001 Flags: bus master, medium devsel, latency 32, IRQ 193 I/O ports at 2000 [size=256] Memory at ed10 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 # cat /proc/interrupts CPU0 CPU1 0: 921447 931323IO-APIC-edge timer 1: 5 5IO-APIC-edge i8042 8: 0 1IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 12: 34 33IO-APIC-edge i8042 14: 140339 90649IO-APIC-edge ide0 15: 0 0IO-APIC-edge libata 169: 38 25 IO-APIC-level ehci_hcd, uhci_hcd, uhci_hcd 177: 0 0 IO-APIC-level uhci_hcd, uhci_hcd 185: 9688 9482 IO-APIC-level libata, ehci_hcd, uhci_hcd, uhci_hcd 193:116 36 IO-APIC-level wcfxo 209: 15304 0 PCI-MSI eth0 NMI: 1 0 LOC:18441221843838 ERR: 0 MIS: 0 #cat /proc/zaptel/1 Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER) 1 WCFXO/0/0 FXSKS (In use) i am unable to make incoming calls route to the asterisk. i didnt see anything in my asterisk CLI. can anybody advise?? Thanks in advance Srinivas Antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] DUNDi Errors (ENCREJ)
Hello users.
i am planning to implement the dundi protocol among 3 servers
where the real channels residing in 2 servers and the remaining one
is only for routing purpose..
here is how my config files
#Routing_server
routing server -192.168.1.11
node1-192.168.1.21
node2-192.168.1.31
i)dundi.conf
dundi=dundicontext,0,IAX2,priv:${secr...@192.168.1.11/${NUMBER},nopartial
[MACaddress of node1]
model=symmetric
host = 192.168.1.21
inkey = priv
outkey = priv
include = priv
permit = priv
qualify = yes
order=primary
;[MAC oF system node2];
;model=symmetric
;host = 192.168.1.31
;inkey = priv
;outkey = priv
;include = priv
;permit = priv
;qualify = yes
;order=secondary
2)extension.conf
[dundicontext]
include = lookupdundi
[lookupdundi]
switch = DUNDi/dundi
3)iax.conf
[priv]
dbsecret=dundi/secret
type=friend
context=dundicontext
- when i tested the dundi show peers in my server the 2 nodes
information i was able to see
- when i used dundi lookup 2...@dundi
i am getting this error
Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: ENCREJ (Response)
Flags: 00 STrans: 16791 DTrans: 30106 [192.168.1.11:4520] (Final)
Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: ACK (Response)
Flags: 00 STrans: 30106 DTrans: 16791 [192.168.1.11:4520] (Final)
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: ENCRYPT (Command)
Flags: 00 STrans: 26692 DTrans: 0 [192.168.1.11:4520]
ENTITY IDENT: 00:23:7d:93:f7:5e
KEYCRC32: 4234245369
ENCDATA : [IV 11dc622321b7c86ae1e7016c4fe4101d] 4 encrypted blocks
Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: ENCREJ (Response)
Flags: 00 STrans: 22476 DTrans: 26692 [192.168.1.11:4520] (Final)
Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: ACK (Response)
Flags: 00 STrans: 26692 DTrans: 22476 [192.168.1.11:4520] (Final)
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: ENCRYPT (Command)
Flags: 00 STrans: 00299 DTrans: 0 [192.168.1.11:4520]
ENTITY IDENT: 00:23:7d:93:f7:5e
SHAREDKEY : [ 66 df 0a c5 75 59 2f 75 bc 48 bd c8 39 6c 33 df
73 37 85 10 86 ed b5 da 4c 88 a7 c5 00 f0 ab d0 2a 9b b3 71 86 7a c4
53 dd dc 4f 29 fb 43 b4 17 d9 91 e1 df 5b 6f 6d c2 b0 f7 d2 f9 f0 b8
3b 0c 0e 7d af ef 8e 4f cf 9f 7e ca 50 b2 04 97 60 2b cb df fd 97 82
d4 bf a0 cf 9a 66 60 11 19 bc 6b 63 30 a5 05 2f 9e a7 63 1d 90 f6 ac
13 23 39 30 33 1d 29 7a 0 6 da 52 5d b0 d7 e7 3f e7 ef 2d a1 ]
SIGNATURE : [ 65 da f2 7a 0f e1 ea 40 73 56 bc 78 d0 05 c0 c3
ec 4c 97 53 cc 2f 2f 97 01 1a 0d ee 8f 21 8f 5a c2 65 91 6d 32 16 dc
27 75 f6 12 9f 3e f3 bd 34 29 9e c9 af 8d 03 ef 43 7c f4 4d 48 e6 cc
70 af 86 89 ef 24 78 3e c3 71 be cb 55 2c e3 79 19 61 2b 34 d4 8f 62
f6 99 8d 27 9f af 56 a3 8b 30 c6 a1 42 de e5 92 4b f0 8f a2 90 91 86
27 fd 0f 7f 1d b6 4a f7 7 2 53 95 d9 d2 14 03 c6 fd b9 9e 5a ]
ENCDATA : [IV 11dc622321b7c86ae1e7016c4fe4101d] 4 encrypted blocks
- To resolve this i tried to remove all keys in all servers and once
again created and
distributed the loaded in each system with keys init command but
stilll i am
getting the same error
can anybody help me out???
Thanks and regards
srinivas antarvedi
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[asterisk-users] Problems with ReceiveFAX (asterisk 1.6.0.3 and t38)
Hello users, have been facing problems with t38 passthrough using asterisk 1.6.0.3. observed also that in case of SendFAX we are not having major issues, its almost successfull. ReceiveFAX has problems most of the time. we have been using a ringcentral account for testing this receivefax. so ringcentral is trying for 3 times if the sending fax failed for the first time. what i observed is that for the first two attempts it failing with the UNEXPECTED MESSAGE RECEIVED and the last time it was successfull . and the above scenario is not always replicatable and some times its failing completely(3rd attempt also fails) i have the tcpdump's .cap files so if anybody want to look at them too i can send.i tried to send along with this mail but the mail was rejected may be because of exceeding the attachment size. Any help is appreciable Thanks and regards Srinivas Antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FaxIn problems
Asterisk-1.6.0.3 OS-2.6.24.2.dn.p4 kernel-CentOS release 4.6 (Final) libpri-1.6 compatable zaptel-1.6 compatible I have been using the accounts for faxin for faxing. For some of the numbers when i send fax it went through successfully. For some numbers the following error is occuring in asterisk CLI app_fax.c:173 phase_e_handler: Error transmitting fax. result=13: Unexpected message received. app_fax.c:621 transmit: Transmission failed app_fax.c:173 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely. app_fax.c:618 transmit: Transmission error What might be the problem??? how to debug this issues with the fax in asterisk CLI?? any help is appreciated.. Thanks in advance Srinivas Antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to bind a SIP channel to an IP
Hello members, Mysetup: Asterisk 1.4 Phones:Polycom501 I wanted to register my polycom phones only from a fixed IP(on LAN ) i tried following scenarios and my results are described as follows 1)sip.conf [xxx] host=192.168.0.15 result is after some time the registration expires and i was unable to receive calls on my channel... 2)sip.conf [xxx] defaultip=192.168.0.15 i) result is after some time the registration expires and i was unable to receive calls on my channel ii)it is even allowing me to register from another ip address say 192.168.0.16 3)sip.conf [xxx] host=dynamic defaultip=192.168.0.15 in this case i dont have any problems and it was working fine... can anybody helpme out to bind the phones to a particular ip if not is it possible to do at all just give me a hint so that i will work on Thanks in advances Srinivas Antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMeAdmin() working problem
Hello users, This is regarding MeetMeAdmin() administration from DialPlan exten = 12345,1,MeetMe(123|MX) ; Enter conference number 123 ;Exit conference by pressing a single digit exten = 12345,2,Hangup() exten = 1,1,MeetMeAdmin(123|M|1) ;mute the user 1 exten = 2,1,MeetMeAdmin(123|m|1) ;un-mute the user 1 exten = 3,1,MeetMeAdmin(123|k|1) ;kick the user 1 actually i supposed to give the user values from the usernumber field of meetme list confnumber command at CLI i cannot give a channel name (ex: 1000 as in SIP/1000) in the above MeetMeAdmin() command under user and the application storing the first channel in user number 1 and so on... so from the dialplan how can i control the users for management purpose(single user mute,single user unmute ,single user kickout) can it be done??? or cannot?? waiting for valuable suggestions thanks in advance regards srinivas antarvedi Srinivas Antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Listen And Talk mode differentiation of meetme() conference
Hello users, i am trying to setup a conference system and i have following requirement 1)some users are only in listen mode 2)some users are only in talk mode 3)some users are able to do both talk and listen how to diffrentiate them when they enter into a particular mode? meaning do i have to give a separate access number in my extensions.conf file so that i will bridge them all together in once coference using meetme() or is there any separate way to do that my idea is like this one 1)all listen only users can call on 123 exten = 123,1,MeetMe(|Mm) exten = 123,2,Hangup() 2)all talkers can call on 456 exten = 456,1,MeetMe(|Mt) exten = 456,2,Hangup() 3)both talk and listen users can call on 789 exten = 789,1,MeetMe(|M) exten = 789,2,Hangup() does this setup only works? or is there any other method of doing the things just enlighten me so that i can finalize my setup thanks in advance regards srinvias antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMeAdmin() working problem
Hello users, Actually i am planning to setup a conference system i have following dialplan [default] exten = 12345,1,MeetMe(1234|X) exten = 12345,2,Hangup() exten = 1,1,MeetMeAdmin(1234|M|user1) exten = 1,2,GoTo(12345|1) exten = 2,1,MeetMeAdmin(1234|m|user1) exten = 2,2,GoTo(12345|1) exten = 3,1,MeetMeAdmin(1234|k|user1) exten = 3,2,GoTo(12345|1) exten = 4,1,MeetMeAdmin(1234|N) exten = 4,2,GoTo(12345|1) exten = 5,1,MeetMeAdmin(1234|n) exten = 5,2,GoTo(12345|1) exten = 6,1,MeetMeAdmin(1234|K) exten = 6,2,GoTo(12345|1) Actually users login into the conference system by dialing in 12345 to enter into conference 1234 and the admin presses 1,2,3,4,5,6 to implement features of conference respectively Mute single user unMute single user Kick single user Mute total conference unMute total conference Kick total conferece while extensions 4,5,6 working fine but individual users mute,unmute,kick(1,2,3 options) not working and the CLI showing specified user not found can anybody helpme out not using any zaptel drivers using only ztdummy Thanks in advance Srinivas Antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_sip.c:2918 auto congestion
Hello users, actually we are tyring to setup a dialer to test outbound autodialer and we are uanble to bridge the answered outbound calls to the local agents and the debug in asterisk is showing the follwoing error message: chan_sip.c:2918 auto congestion can anybody have any idea where might be the problem? give a hint Thanks and regards Srinivas Antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Failure of Sending Voicemail As an attachment in E-mail
Hello all, I am struggling with sending voicemail as an attachement in Email. When i have given the email like [EMAIL PROTECTED] it is delivering to my gamil account perfectly(of course to spam folder). But when i given the email like [EMAIL PROTECTED] it is not delivering to my company email account.. What should i do ? Actually my company is using a third party email server.. Just give me a hint Thanks in advance for your reply Regards Srinivas Antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] service provider connection problem
Hello all, Can anyone have any experience working with service provider like Talkfree . They are giving the user accounts based on the single user accounts and those needs to be directly register to the service provider not to the local system i have taken a connection which when configured to service providers domain direclty ,xlite can make calls without any problem but if i want to use it using my asterisk server (for a simulation to call center) the service provider is asking for 407 proxy authentication and i am unable to resolve this issue can anyone have any circumventing ideas to this solution thanks and regards srinivas antarvedi ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk.NET API --help required
Hello all, Here is the requirement from my side to use Asterisk.NET API to generate an automated call (outgoing) from asterisk and then link to one of the extensions which plays a sound file for the callee. For this i have worked out in the follwing way 1)modified manager.conf to facilitate this API to talk to asterisk 2)used the command Originate to call a Registered user under asterisk and when the user answers the phone it plays whatever i put against the extension.. But my exact requirement is like this 1)Call to the user 2)if answers connect him to the extension provided in the extensions.conf 3)if the user didnt lift the phone within the deault timeout period(30 sec) 4)if the user cancels the phone (Congestion case) 5)if the user not registerd to the(unreachable case) to trace the cases of 3, 4, 5 how should i follow the API I got confused with originate action,orginate sucess event , originate failure event can anybody give me a hint so that i can proceed further thanks in advance for the kind suggestions. regards srinivas antarvedi -- Srinivas Antarvedi ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Text-To-Speech synthesizer--help required
Hello users, Actually i wanted to implement Text-To-Speech engine from cepstral voice using swift application i tried the documentation of doing this and i was unsuccessful at doing this work with asterisk can anybody please help me out finding the solution to installation thanks in advacnce srinivas Antarvedi ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial application response code--help required
Hello all, I am testing the Dial application with the fall through priorities for different cases what i want is the flow after failure of the Dial application which simulates response codes like 1)404 -- Not found 2)480 --Temporarily Unavailable 3)486 --User busy i did manipulate the priority flow like the following for the case 2 and 3 ... exten = _XX,1,Dial(SIP/extension) exten = _XX,2,VoiceMail([EMAIL PROTECTED]|u) --for 480 case exten = _XX,3,Hangup() exten = _XX,102,VoiceMail([EMAIL PROTECTED]|b)--for 486 case exten = _XX,103,Hangup() that was my understanding if the flow is correct? just correct me if i am wrong and what about the case 1 (404 Not Found)? Thanks and regards srinivas antarvedi -- Srinivas Antarvedi ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MediaHandling--Help Required
Hello Users, My Setup is like this openser --Registrar asterisk --Callflow using asterisk-b2bua + radius for accounting My Intention was to generate a Acct-Stop Packet when there is a failure of RTP media from one of the UAC's( callee or caller) who is in dialog. so that the Caller will not be charged for Meaning less network problems Because there is no way asterisk knows about failed UAC as he may not send a BYE Packet . i used the following parameters set canreinvite=no; rtptimeout=60 seconds; Still there is no Acct-Stop packet generated until the session expires timer fires which is equal to Session-Timeout value from radius? Can anybody have any idea of handling network problem of his type? Looking forward for suggestions Thanks in advance srinivas antarvedi ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Users location --help required
Hello all, i am Presently working on integration of asterisk and openser i have a doubt regarding the asterisk . if you take openser when users register it stores the users in location table whether the users running behind NAT or on global ips and when comes to asterisk where does it store ? because i have seen the documentation of integration of asterisk and openser realtime and content there talked about realtime integration of subscriber and sip.conf tables . and i dont want to register users under asterisk so it should fetch the location of users from location table of openser can above fetching mechanism from openser to asterisk using database views be possible? Thanks in advance Srinivas Antarvedi ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime Agents.conf
Hello, i have a small setup which requries that agents should be added dynamically, means their usernames and passwords using a database (MySql). can anybody have idea please give me a hint thanks in advance -- Srinivas Antarvedi ___ --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] voicemail advanced options problem with mysql datbase
Hello all i have an asterisk setup integrated with mysql via odbc driver myproblem is: when i reading my voicemails, in advanced options the following functions not working with realtime asterisk but working with flat files. 1. Reply to the message(option no:1) 2.Leave a message(option no:5) i have following settings in my general section _ searchcontexts=yes _sendvoicemail=yes [test1] 1001 = ,, [test2] 2001 = yyy,,, Error Message showing: No mailbox number '2001' in context 'test1', no reply sent The above problem occuring when i was reading my mailbox and when i try to send a reply to the person who sent me the message using advanced options no1 Can anybody plaease help me out? Thanks in advace Srinivas Antarvedi ___ --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] voicemail advanced options problem with mysql datbase
Hello all i have an asterisk setup integrated with mysql via odbc driver myproblem is: when i reading my voicemails, in advanced options the following functions not working with realtime asterisk but working with flat files. 1. Reply to the message(option no:1) 2.Leave a message(option no:5) i have following settings in my general section _ searchcontexts=yes _sendvoicemail=yes [test1] 1001 = ,, [test2] 2001 = yyy,,, Error Message showing: No mailbox number '2001' in context 'test1', no reply sent The above problem occuring when i was reading my mailbox and when i try to send a reply to the person who sent me the message using advanced options no1 Can anybody plaease help me out? Thanks in advace Srinivas Antarvedi ___ --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] Dialout option problem in voicemail.conf
hello all, i have a set up of 2 contexts with ivr features and it works fine with voicemail also using callback=somecontext i can callback persons on that context but problem is if i included third context i can only callback any one context users not all users how can i solve this issue ! plz help me out ! thanks in advance srinivas antarvedi ___ --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] problem with dialout option in voicemail.conf
hello all, i have a small setup in my office which can just send voicemails and retrive them on a LAN now we wanted to go for a nat with the 2 different contexts with entirely different environement the problem i have faced is: when one of the local guy leaves a message i can call him back using his extension as callback property in the voicemail.conf if some outside guy leaves a message means i need to include his context separately using a separate mailboxid and password if the no of users increses and if they are not listed as users in my asterisk box means how can i callback them when i review my voicemails using callback property in voicemail.conf thanks in advance regards asima ___ --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
