[asterisk-users] Need Help in changing Voice message
Hi, Asterisk has a lot of customizable voice prompt in /var/lib/asterisk/sound but i want to change a very well known voice message which occurs when we try to dail a number against dial plan beep beep beep The person you are calling is unavaiable, please try again. I thought it would be availabe in the sound directory of asterisk but it is not there. When i dial such wrong number no log appears in the asterisk cli command just get this message so i am not getting any idea which macro or application generating this message. Anybody have any idea about how to change this? Thanks Regards Farooq -- ___ --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 disable DND feature key in Polycom Phone
Hi We have polycom 430,501 and 301 phones. Our customer does not need DND feature in any form. I can disable this feature from asterisk server but How can i disable this feature on phones. In the sip configuration file i found the parameter that change the phone behaviour during DND from busy to normal but still if the phone is in dnd mode the phone ringer would be off which is unacceptable. Any idea regarding this. Regards Farooq -- ___ --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] Polycom Phones Call Hold Reminder function problem
I want to enable on hold reminder function on polycom 430 phones. I have enabled it in sip.cfg using this setting hold localReminder call.hold.localReminder.enabled=1 call.hold.localReminder.period=60 call.hold.localReminder.startDelay=90/ /hold But still if the call is on hold the phones does not remind about the on hold call. Any idea? Regards Farooq -- ___ --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
Re: [asterisk-users] How to disable DND feature key in Polycom Phone
Thank you very much for help Regards Farooq Quoting Mojo with Horan Company, LLC [EMAIL PROTECTED]: to clarify what I'm talking about: I'm referring to the soundpoint ip admin guide for version 1.5 for example. The key/ wording is in section 4.6.1.15, or page 113. The key *numbers* referred to, however, are found in section 3.1.7, beginning on page 21. Moj Mojo with Horan Company, LLC wrote: I'm not sure of the correct wording in ipmid.cfg or sip.cfg, but I think you'd be most successful using the keys/ block. A probably wrong example might be: key.IP_500.9.function.prim=Null for a soundpoint 50x and 60x. or key.IP_300.7.function.prim=Null for a soundpoint 30x But it at least might get you pointed in the right direction. If Null isn't what you want you could map it to an arrow key or something else... Mojo Farooq Ahmed wrote: Hi We have polycom 430,501 and 301 phones. Our customer does not need DND feature in any form. I can disable this feature from asterisk server but How can i disable this feature on phones. In the sip configuration file i found the parameter that change the phone behaviour during DND from busy to normal but still if the phone is in dnd mode the phone ringer would be off which is unacceptable. Any idea regarding this. Regards Farooq ___ --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 ___ --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 -- ___ --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
Re: [asterisk-users] Need Help in changing Voice message
Thank you very much who answered to the questions. You have realy saved in wondering around the darkness. Yes it was related the phone not with the asterisk and i was looking in the asterisk yesterday. Because when i used xlite softphone i got the message which have stated in my mail and when i used other softphone i got differnt reply only warning beep . Regards Farooq -- ___ --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] Need Help in Asterisk BLF/Presence/Hints
Hi all,
I am working on
asterisk 1.2.18
zaptel 1.2.17
Polycom 650
polycom 430
SIP version 2.0.3.0131 for IP 650
SIP version for IP430 2.0.3.0127
freepbx 2.2.1
I am trying to configure BLF using asterisk but failed. I would be thankfull if
somebody help me.
Regards
FArooq
**
1
**
in my extension_additional.conf
[ext-local]
include = ext-local-custom
exten = 501,1,Macro(exten-vm,501,501)
exten = 501,n,Hangup
exten = 501,hint,SIP/501
exten = ${VM_PREFIX}501,1,Macro(vm,501,DIRECTDIAL)
exten = ${VM_PREFIX}501,n,Hangup
exten = 502,1,Macro(exten-vm,502,502)
exten = 502,n,Hangup
exten = 502,hint,SIP/502
exten = ${VM_PREFIX}502,1,Macro(vm,502,DIRECTDIAL)
exten = ${VM_PREFIX}502,n,Hangup
exten = 503,1,Macro(exten-vm,503,503)
exten = 503,n,Hangup
exten = 503,hint,SIP/503
exten = ${VM_PREFIX}503,1,Macro(vm,503,DIRECTDIAL)
exten = ${VM_PREFIX}503,n,Hangup
; end of [ext-local]
***
2
**
SIP_additional.conf
one of my extension is configured as
--
[507]
type=friend
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
mailbox=507 at device
host=dynamic
dtmfmode=rfc2833
dial=SIP/507
context=from-internal
canreinvite=no
subscribecontext = ext-local
notifyringing = yes
callerid=device 507
3
ext 501 phone is configured with complete contact directory.
Buddywatch was enabled in the polycom contact directory
using config like below
item
lnDoe/ln
fnJohn/fn
ct507/ct
sd1/sd
rt1/rt
dc /
ad0/ad
ar0/ar
bw1/bw
bb0/bb
/item
**
Results
***
localhost*CLI show hints
localhost*CLI
-= Registered Asterisk Dial Plan Hints =-
507 : SIP/507 State:Unavailable Watchers 0
506 : SIP/506 State:Unavailable Watchers 0
505 : SIP/505 State:Unavailable Watchers 0
504 : SIP/504 State:IdleWatchers 0
503 : SIP/503 State:Unavailable Watchers 0
502 : SIP/502 State:IdleWatchers 0
501 : SIP/501 State:IdleWatchers 0
- 7 hints registered
localhost*CLI
localhost*CLI sip show subscriptions
Peer UserCall ID ExtensionLast state Type
0 active SIP subscriptions
localhost*CLI
--
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[asterisk-users] Configuring BLF or Asterisk presence feature
Hi all,
I am working on
asterisk 1.2.18
zaptel 1.2.17
Polycom 650
polycom 430
SIP version 2.0.3.0131 for IP 650
SIP version for IP430 2.0.3.0127
freepbx 2.2.1
I am trying to configure BLF using asterisk but failed. I would be thankfull if
somebody help me.
Regards
FArooq
**
1
**
in my extension_additional.conf
[ext-local]
include = ext-local-custom
exten = 501,1,Macro(exten-vm,501,501)
exten = 501,n,Hangup
exten = 501,hint,SIP/501
exten = ${VM_PREFIX}501,1,Macro(vm,501,DIRECTDIAL)
exten = ${VM_PREFIX}501,n,Hangup
exten = 502,1,Macro(exten-vm,502,502)
exten = 502,n,Hangup
exten = 502,hint,SIP/502
exten = ${VM_PREFIX}502,1,Macro(vm,502,DIRECTDIAL)
exten = ${VM_PREFIX}502,n,Hangup
exten = 503,1,Macro(exten-vm,503,503)
exten = 503,n,Hangup
exten = 503,hint,SIP/503
exten = ${VM_PREFIX}503,1,Macro(vm,503,DIRECTDIAL)
exten = ${VM_PREFIX}503,n,Hangup
; end of [ext-local]
***
2
**
SIP_additional.conf
one of my extension is configured as
--
[507]
type=friend
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
dial=SIP/507
context=from-internal
canreinvite=no
subscribecontext = ext-local
notifyringing = yes
callerid=device 507
3
ext 501 phone is configured with complete contact directory.
Buddywatch was enabled in the polycom contact directory
using config like below
item
lnDoe/ln
fnJohn/fn
ct507/ct
sd1/sd
rt1/rt
dc /
ad0/ad
ar0/ar
bw1/bw
bb0/bb
/item
**
Results
***
localhost*CLI show hints
localhost*CLI
-= Registered Asterisk Dial Plan Hints =-
507 : SIP/507 State:Unavailable Watchers 0
506 : SIP/506 State:Unavailable Watchers 0
505 : SIP/505 State:Unavailable Watchers 0
504 : SIP/504 State:IdleWatchers 0
503 : SIP/503 State:Unavailable Watchers 0
502 : SIP/502 State:IdleWatchers 0
501 : SIP/501 State:IdleWatchers 0
- 7 hints registered
localhost*CLI
localhost*CLI sip show subscriptions
Peer UserCall ID ExtensionLast state Type
0 active SIP subscriptions
localhost*CLI
--
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[asterisk-users] Need Advice/Suggestion
Hi all, As we know we can configure in astersik like before 5:00pm calls go to reception and after 5:00 pm calls go to some mobile no. One of my client requested that he wants to manually shift the dial plan like above as he has flexiable timing sometime he finishes at 3:00pm some time 8pm. I can not give him freepbx access. Any idea or solution. Regards Farooq -- ___ --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] Need Advice/Suggestion
Hi all, As we know we can configure in astersik like before 5:00pm calls go to reception and after 5:00 pm calls go to some mobile no. One of my client requested that he wants to manually shift the dial plan like above as he has flexiable timing sometime he finishes at 3:00pm some time 8pm. I can not give him freepbx access. Any idea or solution. Regards Farooq ___ --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] Configuring BLF or Asterisk presence/Hints feature
Hi all,
I am working on
asterisk 1.2.18
zaptel 1.2.17
Polycom 650
polycom 430
SIP version 2.0.3.0131 for IP 650
SIP version for IP430 2.0.3.0127
freepbx 2.2.1
I am trying to configure BLF using asterisk but failed. I would be thankfull if
somebody help me.
Regards
FArooq
**
1
**
in my extension_additional.conf
[ext-local]
include = ext-local-custom
exten = 501,1,Macro(exten-vm,501,501)
exten = 501,n,Hangup
exten = 501,hint,SIP/501
exten = ${VM_PREFIX}501,1,Macro(vm,501,DIRECTDIAL)
exten = ${VM_PREFIX}501,n,Hangup
exten = 502,1,Macro(exten-vm,502,502)
exten = 502,n,Hangup
exten = 502,hint,SIP/502
exten = ${VM_PREFIX}502,1,Macro(vm,502,DIRECTDIAL)
exten = ${VM_PREFIX}502,n,Hangup
exten = 503,1,Macro(exten-vm,503,503)
exten = 503,n,Hangup
exten = 503,hint,SIP/503
exten = ${VM_PREFIX}503,1,Macro(vm,503,DIRECTDIAL)
exten = ${VM_PREFIX}503,n,Hangup
; end of [ext-local]
***
2
**
SIP_additional.conf
one of my extension is configured as
--
[507]
type=friend
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
dial=SIP/507
context=from-internal
canreinvite=no
subscribecontext = ext-local
notifyringing = yes
callerid=device 507
3
ext 501 phone is configured with complete contact directory.
Buddywatch was enabled in the polycom contact directory
using config like below
item
lnDoe/ln
fnJohn/fn
ct507/ct
sd1/sd
rt1/rt
dc /
ad0/ad
ar0/ar
bw1/bw
bb0/bb
/item
**
Results
***
localhost*CLI show hints
localhost*CLI
-= Registered Asterisk Dial Plan Hints =-
507 : SIP/507 State:Unavailable Watchers 0
506 : SIP/506 State:Unavailable Watchers 0
505 : SIP/505 State:Unavailable Watchers 0
504 : SIP/504 State:IdleWatchers 0
503 : SIP/503 State:Unavailable Watchers 0
502 : SIP/502 State:IdleWatchers 0
501 : SIP/501 State:IdleWatchers 0
- 7 hints registered
localhost*CLI
localhost*CLI sip show subscriptions
Peer UserCall ID ExtensionLast state Type
0 active SIP subscriptions
localhost*CLI
--
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Re: RE : [asterisk-users] Asterisk is not showing the correctIncomming CallerID
Yes national and internation prefix was 1 and +1 there. Thank you very much Francois BERGERET. Regards Farooq Quoting [EMAIL PROTECTED]: Hi Farook and the list, You have may be forgotten to input that in the misdn.conf file : nationalprefix=0 internationalprefix=00 dialplan=0 localdialplan=0 cpndialplan=0 Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Farooq Ahmed Envoyé : mercredi 16 mai 2007 06:14 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Asterisk is not showing the correctIncomming CallerID I forgot to give the asterisk logs pbx*CLI -- Executing Set(mISDN/2-2, FROM_DID=3722) in new stack -- Executing Gosub(mISDN/2-2, app-blacklist-check|s|1) in new stack -- Executing LookupBlacklist(mISDN/2-2, ) in new stack -- Executing GotoIf(mISDN/2-2, 0?blacklisted) in new stack -- Executing Return(mISDN/2-2, ) in new stack -- Executing Goto(mISDN/2-2, ext-group|1|1) in new stack -- Goto (ext-group,1,1) -- Executing Macro(mISDN/2-2, user-callerid|) in new stack -- Executing NoOp(mISDN/2-2, user-callerid: 1416222888) in new stack -- Executing GotoIf(mISDN/2-2, 0?report) in new stack -- Executing GotoIf(mISDN/2-2, 0?start) in new stack -- Executing Set(mISDN/2-2, REALCALLERIDNUM=1416222888) in new stack -- Executing NoOp(mISDN/2-2, REALCALLERIDNUM is 1416222888) in new stack -- Executing Set(mISDN/2-2, AMPUSER=) in new stack -- Executing Set(mISDN/2-2, AMPUSERCIDNAME=) in new stack -- Executing GotoIf(mISDN/2-2, 1?report) in new stack -- Goto (macro-user-callerid,s,11) -- Executing NoOp(mISDN/2-2, TTL: ARG1: ) in new stack -- Executing GotoIf(mISDN/2-2, 0?continue) in new stack -- Executing Set(mISDN/2-2, _TTL=64) in new stack -- Executing GotoIf(mISDN/2-2, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(mISDN/2-2, Using CallerID 1416222888) in new stack -- Executing Set(mISDN/2-2, modifiedcallerid=1416222888) in new stack -- Executing Set(mISDN/2-2, CALLERID(number)=1416222888) in new stack -- Executing GotoIf(mISDN/2-2, 1?skipdb) in new stack -- Goto (ext-group,1,4) -- Executing Set(mISDN/2-2, __NODEST=) in new stack -- Executing Set(mISDN/2-2, __BLKVM_OVERRIDE=BLKVM/1/mISDN/2-2) in new stack -- Executing Set(mISDN/2-2, __BLKVM_BASE=1) in new stack -- Executing Set(mISDN/2-2, DB(BLKVM/1/mISDN/2-2)=TRUE) in new stack -- Executing Set(mISDN/2-2, RRNODEST=) in new stack -- Executing Set(mISDN/2-2, __NODEST=1) in new stack -- Executing GotoIf(mISDN/2-2, 1?REPCID) in new stack -- Goto (ext-group,1,14) -- Executing NoOp(mISDN/2-2, CALLERID(name) is ) in new stack -- Executing Set(mISDN/2-2, RecordMethod=Group) in new stack -- Executing Macro(mISDN/2-2, record-enable|903-909|Group) in new stack -- Executing GotoIf(mISDN/2-2, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI(mISDN/2-2, recordingcheck|20070516-140757|1179288477.1037) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(mISDN/2-2, No recording needed) in new stack -- Executing Set(mISDN/2-2, RingGroupMethod=hunt) in new stack -- Executing Macro(mISDN/2-2, dial|10||903-909) in new stack -- Executing DeadAGI(mISDN/2-2, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'unknown' number is '1416222888' dialparties.agi: Methodology of ring is 'hunt' dialparties.agi: USE_CONFIRMATION: 'FALSE' dialparties.agi: RINGGROUP_INDEX: '' -- dialparties.agi: Added extension 903 to extension map -- dialparties.agi: Added extension 909 to extension map -- dialparties.agi: Extension 903 cf is disabled -- dialparties.agi: Extension 909 cf is disabled -- dialparties.agi: Extension 903 do not disturb is disabled -- dialparties.agi: Extension 909 do not disturb is disabled dialparties.agi: extnum: 903 dialparties.agi: exthascw: 1 dialparties.agi: exthascfb: 0 dialparties.agi: extcfb: dialparties.agi: exthascfu: 0 dialparties.agi: extcfu: dialparties.agi: extnum: 909 dialparties.agi: exthascw: 1 dialparties.agi: exthascfb: 0 dialparties.agi: extcfb: dialparties.agi: exthascfu: 0 dialparties.agi: extcfu: dialparties.agi: NODEST: 1 adding M(auto-blkvm) to dialopts: M(auto-blkvm) -- AGI Script
[asterisk-users] Asterisk is not showing the correct Incomming CallerID
Hi Everyone, I have an asterisk box in my office. It does not display the correct Incomming Caller id. For incomming we are using ISDN Bri line which is terminated in a Digium 4 port bri card (B410P). Like if a number say 02 12345678 calls to our line asterisk displays it 12 12345678. Similarlay if a mobile number say 0416 123456 dials us , asterisk displays 1416 123456. I am not sure where the problem is either in misdn.conf or extension.conf. Would be kind enough if some give any Pointer or help. Regards Farooq -- ___ --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 is not showing the correct Incomming CallerID
is ringing == Spawn extension (macro-dial, s, 42) exited non-zero on 'mISDN/2-2' in macro 'dial' == Spawn extension (macro-dial, s, 42) exited non-zero on 'mISDN/2-2' -- Executing Macro(mISDN/2-2, hangupcall) in new stack -- Executing ResetCDR(mISDN/2-2, w) in new stack -- Executing NoCDR(mISDN/2-2, ) in new stack -- Executing GotoIf(mISDN/2-2, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing GotoIf(mISDN/2-2, 0?theend) in new stack -- Executing NoOp(mISDN/2-2, Cleaning Up Block VM Flag: BLKVM/1/mISDN/2-2) in new stack -- Executing DBdel(mISDN/2-2, BLKVM/1/mISDN/2-2) in new stack -- DBdel: family=BLKVM, key=1/mISDN/2-2 -- Executing Wait(mISDN/2-2, 5) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'mISDN/2-2' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'mISDN/2-2' pbx*CLI Quoting Farooq Ahmed [EMAIL PROTECTED]: Hi Everyone, I have an asterisk box in my office. It does not display the correct Incomming Caller id. For incomming we are using ISDN Bri line which is terminated in a Digium 4 port bri card (B410P). Like if a number say 02 12345678 calls to our line asterisk displays it 12 12345678. Similarlay if a mobile number say 0416 123456 dials us , asterisk displays 1416 123456. I am not sure where the problem is either in misdn.conf or extension.conf. Would be kind enough if some give any Pointer or help. Regards Farooq -- ___ --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] Digium B410P Need Help
Hi All Trying to install Digium B410P on Trixbox 2. After initializing card driver and asterisk i m getting follow message asterisk shows no port. Would be kind enough if somebody help me. Regards Farooq #misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX Port 2: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX Port 3: NT-mode BRI S/T interface port (for phones) - Layer 2 protocol 0x0202 is detected, but not allowed for NT lib. * Port NOT useable for PBX Port 4: NT-mode BRI S/T interface port (for phones) - Layer 2 protocol 0x0202 is detected, but not allowed for NT lib. * Port NOT useable for PBX -- ___ --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] Digium B410P Need Help
Yes i tried but nothing change Regards Farooq Quoting yusuf [EMAIL PROTECTED]: Farooq Ahmed wrote: Hi All Trying to install Digium B410P on Trixbox 2. After initializing card driver and asterisk i m getting follow message asterisk shows no port. Would be kind enough if somebody help me. Regards Farooq #misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX Port 2: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX Port 3: NT-mode BRI S/T interface port (for phones) - Layer 2 protocol 0x0202 is detected, but not allowed for NT lib. * Port NOT useable for PBX Port 4: NT-mode BRI S/T interface port (for phones) - Layer 2 protocol 0x0202 is detected, but not allowed for NT lib. * Port NOT useable for PBX Hi, in /etc/misdn-init.conf, switch the mode to te_ptmp= or something. -- thanks, Yusuf ___ --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] Two or More Bri Cards
hi all we want to use Two single port Bri cards in Trixbox. Any idea which card is having good support and performance repotation especially when using two or more in Trixbox. Regards farooq -- ___ --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: RE : [asterisk-users] Two or More Bri Cards
Two farooq Quoting [EMAIL PROTECTED]: Hi ! Prefer to have only one card with how many ports you want. Always better for IRQ flow. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Farooq Ahmed Envoyé : lundi 26 mars 2007 09:11 À : asterisk-bsd@lists.digium.com Cc : asterisk-users@lists.digium.com Objet : [asterisk-users] Two or More Bri Cards hi all we want to use Two single port Bri cards in Trixbox. Any idea which card is having good support and performance repotation especially when using two or more in Trixbox. Regards farooq -- ___ --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 -- ___ --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] how to check and set D-channel status
Hi All, Can anybody guide me to check the D-channel info of my netjet ISDN card I am trying to configure it with asterisk using misdn_capi and chan_capi. How can set differnt protocol at D-channel. Thanks Regards Farooq -- ___ --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] Test Message
Hi. I am using second account to test the message to this user list. I sent the problem from my [EMAIL PROTECTED] but it was not appeared in list. I am recieving messages from this list but when trying to send email to list it disappears somewhere. Regards Farooq ___ --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 in using Two BRi Cards in Asterisk
Hi,
I have done my best and tired of searching the net about the problem. If
anybody could help
would be a great favour.
Description of Problem
I am trying to install two Netpci cards(Traverse Technology Netjet ISDN-s) on
Trixbox 2 and aim
is to use in Asterisk as dailin and dialout. I compliled the driver as directed
in the manufacture
manual. After installation dmesg shows that both of the cards have installed
successfully.
When testing two problem is there
1) Card one receiving call normally but when dialing out .. it dials the number
when other person
picks the call no voice of either side can be heard.
2)Secound card is neither receiving nor dialing error in asterisk is comming
like ISDN2#02: CAPI
INFO 0x3303: Protocol error layer 3.
I am trying to use ISDN channels in asterisk using Chan_capi.
For the detail i have added the following outputs in this mail.
#Netjet Card was compiled using these options
#DMESG RESULTS
#RESULTS of CAPIINFO
#Output of etc/CAPI.conf
#output of etc/asterisk/capi.conf
#output capi info from Asterisk CLI #Output from ASTERISK CLI terminal ..when
outgoing call
was rejected (error output)
Thanks and Regards
Farooq
Netjet Card was compiled using these options
---
Device Drivers ---
ISDN subsystem ---
M ISDN support
--- CAPI subsystem
M CAPI2.0 support
[ ] Verbose reason code reporting (kernel size +=7K)
[*] CAPI2.0 Middleware support (EXPERIMENTAL)
M CAPI2.0 /dev/capi support
[*] CAPI2.0 filesystem support
CAPI2.0 capidrv interface support
--- CAPI hardware drivers
Active AVM cards ---
Active Eicon DIVA Server cards ---
Modular ISDN driver ---
M Support modular ISDN driver
[ ] Enable memory leak debug for mISDN (NEW)
[*] Support for AVM Fritz!Cards
[*] Support for NETJet cards
DMESG RESULTS
---
CAPI Subsystem Rev 1.1.2.8
capifs: Rev 1.1.2.3
capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs) Modular ISDN
Stack core
$Revision: 1.37 $
mISDNd: kernel daemon started (current:ca4d8680)
mISDNd: test event done
ISDN L1 driver version 1.18
ISDN L2 driver version 1.31
mISDN: DSS1 Rev. 1.42
mISDN Capi 2.0 driver file version 1.20
ISAC module $Revision: 1.17 $
mISDN_dsp: Audio DSP Rev. 1.24 (debug=0x0) EchoCancellor MG2
dtmftreshold(100)
mISDN_dsp: DSP clocks every 64 samples. This equals 8 jiffies.
DTMF modul version 1.16
Traverse Tech. NETjet-S driver, revision 1.6
nj_probe(mISDN): found adapter NETJet S at :00:0b.0 NETJet setup_instance:
protocol is 2
layermask is 0 NETJet card ca611740 dch ca611894 bch1 ca6119f8 bch2 ca611b5c
NETJet1 ISAC STAR 4a
NETJet1 ISAC MODE 0
NETJet1 ISAC ADF2 0
NETJet1 ISAC ISTA 0
NETJet1 ISAC CIR0 7c
mISDN_isac_init: ISAC version (0): 2086/2186 V1.1
NETJet1 B1 tiger: send buf ca0da000 - ca0da7fc
NETJet1 B1 tiger: rec buf ca0db000 - ca0db1fc
NETJet1 B1 tiger: dmacfg a0db000/a0da000 pulse=0 NETJet 1 cards installed
kcapi: Controller 1: mISDN1 attached
contr-addr(01) cnr(01) st(0100)
nj_probe(mISDN): found adapter NETJet S at :00:0d.0 NETJet setup_instance:
protocol is 2
layermask is 0 NETJet card cad0e340 dch cad0e494 bch1 cad0e5f8 bch2 cad0e75c
kcapi: card 1 mISDN1 ready.
NETJet2 ISAC STAR 4a
NETJet2 ISAC MODE 0
NETJet2 ISAC ADF2 0
NETJet2 ISAC ISTA 0
NETJet2 ISAC CIR0 7c
mISDN_isac_init: ISAC version (0): 2086/2186 V1.1
NETJet2 B1 tiger: send buf ca30e000 - ca30e7fc
NETJet2 B1 tiger: rec buf ca315000 - ca3151fc
NETJet2 B1 tiger: dmacfg a315000/a30e000 pulse=0 NETJet 2 cards installed
kcapi: Controller 2: mISDN2 attached
contr-addr(02) cnr(02) st(0200)
kcapi: card 2 mISDN2 ready.
RESULTS of CAPIINFO
[EMAIL PROTECTED] etc]# capiinfo
Number of Controllers : 2
Controller 1:
Manufacturer: mISDN CAPI controller NETJet1 CAPI Version: 2.0 Manufacturer
Version: 1.0 Serial
Number: 0002
BChannels: 2
Global Options: 0x0018
DTMF supported
Supplementary Services supported
B1 protocols support: 0x0003
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
B2 protocols support: 0x0043
ISO 7776 (X.75 SLP)
Transparent
Transparent (ignoring framing errors of B1 protocol)
B3 protocols support: 0x0001
Transparent
0100
0200
1800
0300
4300
0100
Supplementary services support: 0x0033
Hold / Retrieve
Terminal Portability
Call Forwarding
Call Deflection
Controller 2:
Manufacturer: mISDN CAPI controller NETJet2 CAPI Version: 2.0 Manufacturer
Version: 1.0 Serial
Number: 0002
BChannels: 2
Global Options: 0x0018
DTMF supported
Supplementary Services supported
B1 protocols support: 0x0003
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
B2
Re: [asterisk-users] Problem in using Two BRi Cards in Asterisk
And any idea about the issue on card one... means why outgoing is not working. Farooq Quoting Paul Hales [EMAIL PROTECTED]: From memory, to get more than 1 single port BRI card running in a machine you need to make changes to the source code of the driver. :( PaulH On Fri, 2007-03-23 at 11:12 +1100, Farooq Ahmed wrote: Hi, I have done my best and tired of searching the net about the problem. If anybody could help would be a great favour. Description of Problem I am trying to install two Netpci cards(Traverse Technology Netjet ISDN-s) on Trixbox 2 and aim is to use in Asterisk as dailin and dialout. I compliled the driver as directed in the manufacture manual. After installation dmesg shows that both of the cards have installed successfully. When testing two problem is there 1) Card one receiving call normally but when dialing out .. it dials the number when other person picks the call no voice of either side can be heard. 2)Secound card is neither receiving nor dialing error in asterisk is comming like ISDN2#02: CAPI INFO 0x3303: Protocol error layer 3. I am trying to use ISDN channels in asterisk using Chan_capi. For the detail i have added the following outputs in this mail. #Netjet Card was compiled using these options #DMESG RESULTS #RESULTS of CAPIINFO #Output of etc/CAPI.conf #output of etc/asterisk/capi.conf #output capi info from Asterisk CLI #Output from ASTERISK CLI terminal ..when outgoing call was rejected (error output) Thanks and Regards Farooq Netjet Card was compiled using these options --- Device Drivers --- ISDN subsystem --- M ISDN support --- CAPI subsystem M CAPI2.0 support [ ] Verbose reason code reporting (kernel size +=7K) [*] CAPI2.0 Middleware support (EXPERIMENTAL) M CAPI2.0 /dev/capi support [*] CAPI2.0 filesystem support CAPI2.0 capidrv interface support --- CAPI hardware drivers Active AVM cards --- Active Eicon DIVA Server cards --- Modular ISDN driver --- M Support modular ISDN driver [ ] Enable memory leak debug for mISDN (NEW) [*] Support for AVM Fritz!Cards [*] Support for NETJet cards DMESG RESULTS --- CAPI Subsystem Rev 1.1.2.8 capifs: Rev 1.1.2.3 capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs) Modular ISDN Stack core $Revision: 1.37 $ mISDNd: kernel daemon started (current:ca4d8680) mISDNd: test event done ISDN L1 driver version 1.18 ISDN L2 driver version 1.31 mISDN: DSS1 Rev. 1.42 mISDN Capi 2.0 driver file version 1.20 ISAC module $Revision: 1.17 $ mISDN_dsp: Audio DSP Rev. 1.24 (debug=0x0) EchoCancellor MG2 dtmftreshold(100) mISDN_dsp: DSP clocks every 64 samples. This equals 8 jiffies. DTMF modul version 1.16 Traverse Tech. NETjet-S driver, revision 1.6 nj_probe(mISDN): found adapter NETJet S at :00:0b.0 NETJet setup_instance: protocol is 2 layermask is 0 NETJet card ca611740 dch ca611894 bch1 ca6119f8 bch2 ca611b5c NETJet1 ISAC STAR 4a NETJet1 ISAC MODE 0 NETJet1 ISAC ADF2 0 NETJet1 ISAC ISTA 0 NETJet1 ISAC CIR0 7c mISDN_isac_init: ISAC version (0): 2086/2186 V1.1 NETJet1 B1 tiger: send buf ca0da000 - ca0da7fc NETJet1 B1 tiger: rec buf ca0db000 - ca0db1fc NETJet1 B1 tiger: dmacfg a0db000/a0da000 pulse=0 NETJet 1 cards installed kcapi: Controller 1: mISDN1 attached contr-addr(01) cnr(01) st(0100) nj_probe(mISDN): found adapter NETJet S at :00:0d.0 NETJet setup_instance: protocol is 2 layermask is 0 NETJet card cad0e340 dch cad0e494 bch1 cad0e5f8 bch2 cad0e75c kcapi: card 1 mISDN1 ready. NETJet2 ISAC STAR 4a NETJet2 ISAC MODE 0 NETJet2 ISAC ADF2 0 NETJet2 ISAC ISTA 0 NETJet2 ISAC CIR0 7c mISDN_isac_init: ISAC version (0): 2086/2186 V1.1 NETJet2 B1 tiger: send buf ca30e000 - ca30e7fc NETJet2 B1 tiger: rec buf ca315000 - ca3151fc NETJet2 B1 tiger: dmacfg a315000/a30e000 pulse=0 NETJet 2 cards installed kcapi: Controller 2: mISDN2 attached contr-addr(02) cnr(02) st(0200) kcapi: card 2 mISDN2 ready. RESULTS of CAPIINFO [EMAIL PROTECTED] etc]# capiinfo Number of Controllers : 2 Controller 1: Manufacturer: mISDN CAPI controller NETJet1 CAPI Version: 2.0 Manufacturer Version: 1.0 Serial Number: 0002 BChannels: 2 Global Options: 0x0018 DTMF supported Supplementary Services supported B1 protocols support: 0x0003 64 kbit/s with HDLC framing 64 kbit/s bit-transparent operation B2 protocols support: 0x0043 ISO 7776 (X.75 SLP) Transparent Transparent (ignoring framing errors of B1 protocol) B3 protocols support: 0x0001 Transparent 0100 0200 1800
