[asterisk-users] SIP hacked connection?
Hi Running 1.2.26 BRI stuffed. Calls made via PSTN via ISDN interface (Junghanns). SIP ports mapped through firewall as we often connect from outside, but all SIP accounts have good passwords. However our telecoms provider picked up a few suspicious calls to places we do not normally call at times we do not often call. Looking at Asterisk logs it shows SIP session from the internet connected in and making calls with account IDs we do not recognise - definitely none of ours. Very few calls have been made this way, trivial cost, but it is slightly worrying. Anyone any ideas on how this could be happening? Thank Paul ___ -- 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] UK experts only. BT Outgoing caller ID not showing
Hi guys Thanks for help on this so far. There was no typo - old exchange was System X and new one System Y. Also caller ID is enabled on the new DDI range so we get incoming caller ID. BT are looking at this - the guys I talked to is being very helpful and has referred this to a colleague (why do we find this so surprising in the UK - BT helpful!). Paul Redstone ___ --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] UK experts only. BT Outgoing caller ID not showing
Hi Just moved offices in the UK and moved our Asterisk box from old one to new one. Using idefisk softphones, Junghanns quadbri card for ISDN 2e interfaces. At both offices we had one standard number and a DDI range, routed with Asterisk. We'd set up the configuration so each idefisk set its own caller ID which then got sent by the extensions.conf script. Worked fine at old place but in new place the only number which is received is the central switchboard number. The conf files are unchanged except for the obvious number changes but nothing I can do sets the outgoing caller id. We're using the same version of idefisk and the same version of asterisk (1.2.4-bri stuffed). I found a wiki which said that the DDI numbers we want as caller IDs need to be flagged as allowed CallerID number - this is done by BT - but BT do not seem to understand this. Also our old local exchange was a System X but the new one is System Y. Anyone any ideas on this? Paul ___ --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] attended call transfer
Useful discussion on this. There are some other functions in this which need to be addressed. For example if doing an attended transfer and the recipient phone number goes to voicemail, you have to wait for the timeout to reconnect to the original caller - unless someone know differently. There should be a reconnect hot key. Again this is comparable to a conventional PABX where the attended transfer puts the caller on hold and pushing a button reconnects. Paul ___ --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] Attended transfer reconnect when goes to voicemail?
Hi Running bristuffed 0.3.0-PRE-1f which is 1.2.1. Using *2 in features.conf for attended transfer. Works well if someone answers. But the following sequence causes issue: 1. Receptionist takes call. 2. *2 then 123 to transfer to extension 123. 3. 123 is busy or does not answer so receptionist hears 123 voicemail 4. How can receptionist reconnect to calling user - could wait for voicemail to hang up which reconnects to caller but this takes a long time. Thanks Paul ___ --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] Preventing incoming calls from ringing SIP lines
Hi We're using three line SIP phones (X-lite), very nice, with Asterisk 1.2 But we want to prevent either direct incoming calls or calls from other extensions from ringing if the user is in another incoming call (i.e incoming into the extension), making an outgoing call or even checking their voicemail. In 1.0 the SetGroup and CheckGroup commands could do this but you have to build it into all parts of the dial plan. In 1.2 these do not exist and the Set(Group type commands with GotoIf are supposed to be used. But I still have not seen anywhere a full example of this. There is the call-limit setting in SIP - beautiful, works at the SIP level so easier than the dial plan. BUT with this you cannot do attended or blind transfers - not sensible. This must be a very common requirement, certainly is judging from the posts but in hours of searching I have not see the sort of complete solution which looks feasible. Thanks and sorry if I've missed it. Alternatively I'd be happy to use single line SIP softphones but cannot find one which feels good. TIA Paul 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
[Asterisk-Users] Re: IAX Devices Recommendation
Hi We purchased the AT320-EE IAXtalk phone from www.iaxtalk.com which ocnnects to our own asterisk server. Good value, a little tricky to set up - the instructions they supply to which they give you a link on their web site are OK, but their are some gaps which the asterisk wiki pages fill well - cannot find this at the moment but it explains how to do resets. IN summary you buy the phone and then upload the firmware for IAX2 protocol. Configuration is via web browser which works well. Automaticlaly logs in. Works well. Slightly slower to respond than (say) firefly softphone which we use for most users - the hardphone is for reception and as backup in case of computer failure. Paul Redstone ___ 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: Dell PowerEdge SC420 interrupt issue
We're using an SC420 and using BRI with a quadbri Junganns card, with IAX softphones and one hardphone. Working well except that we sometimes get dropped connections between IAX and the server with a max retries exceed message, which comes from the chan_iax driver code. The BRI side of things looks like it is fine. I had been thinking it might be a network issue but now wonder if it is an interrupt or other background process issue causing a timeout on the Dell - hence my post as it might be the same cause as yours. We're about to concentrate on this hypothesis. If it is then it could perhaps be due to: 1. Linux - we're running Debian 2.6.8 2. Something in the firmware - we have twin SATA drives, though not mirrored as we had orginally expected. 3. E-mail background process. Doubt it as it is only used for voicemail messages. 4. Windows networking/SAMBA share. We only use this for configuring the conf files from windows and backing up configuration etc. 5. Other background process. Perhaps moh? We're using madplay though I've just checked and noticed a few perhaps rogue mpg123 processes. 6. Overloading? We're only a 10 person office so figure the SC420 with 2.6 G Celeron should be enough. So no solutions here for you but using same platform with what looks like a timeout/background process type issue. Paul ___ 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 Retries Exceeded - IAX2. Network problem?
Hi We're having some problems with max retries exceeded errors using IAX2 which causes dropped calls. Sometimes happens with Firefly softphone, now 1.9.9 (the current one) but has also happened with a hardphone we use (IAXtel). This is just for the internal connection between our desktops and our switch - these are calls which then go out ver ISDN/PSTN and the error is definitely an iax channel error, which means internal. So my guess is now that this is due to either a problem with network connectivity - our switch or with the network card in the Asterisk server. There is some suggestion it happens when the network is busy. Can anyone suggest a way of checking this out? I think I'm going to buy a different switch type and see if ti has an effect but a more systematic way would be better. Thanks Paul ___ 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] IAX2 Max Retries dropped calls Firefly
Hi We've recently set up and are using with success 1.0.7 using a Junghanns quadbri card to BRI ISDN, and Firefly with IAX2 protocol as softphones Works very well, however we're getting cases where sometimes the call just drops. From setting more verbose modes we get a log which is shown below. The problem seems to be the maxretries message which comes from chan_iax2. We are using Firefly 1.9.8 build 3945. However I cannot work out what this message means. There is some suggestion in when it occurs that it might be an IP connection issue from the softphone to the asterisk box. Connection is in one office via 100 M switches, very simple direct path. Firefly running Windows XP SP2. We're planning to try another softphone but quite like Firefly. Can anyone advise on this? Thanks Paul === Log extract -- Hungup 'Zap/1-1' == Spawn extension (bodiam, NN, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/ 10' -- Hungup 'IAX2/[EMAIL PROTECTED]/10' -- Registered '355' (AUTHENTICATED) at -- Registered '354' (AUTHENTICATED) at -- Accepting AUTHENTICATED call from requested format = 1024 , actual format = 1024 -- Executing Macro(IAX2/[EMAIL PROTECTED]/11, bodiam-iaxsip|352|IAX2/352) in new s tack -- Executing Dial(IAX2/[EMAIL PROTECTED]/11, IAX2/352|20|tT) in new stack -- Called 352 -- Call accepted by (format ilbc) -- Format for call is ilbc -- IAX2/352/15 is ringing -- IAX2/352/15 answered IAX2/[EMAIL PROTECTED]/11 -- Attempting native bridge of IAX2/[EMAIL PROTECTED]/11 and IAX2/352/15 May 17 11:47:56 WARNING[2763]: chan_iax2.c:1480 attempt_transmit: Max retries ex ceeded to host on IAX2/[EMAIL PROTECTED]/7 (type = 6, subclass = 2, ts=3800 76, seqno=66) May 17 11:47:56 WARNING[2763]: chan_iax2.c:1480 attempt_transmit: Max retries ex ceeded to host on IAX2/[EMAIL PROTECTED]/7 (type = 6, subclass = 11, ts=380 079, seqno=67) -- Hungup 'Zap/2-1' == Spawn extension (bodiam, NN, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/ 7' -- Hungup 'IAX2/[EMAIL PROTECTED]/7' -- Hungup 'IAX2/352/15' == Spawn extension (macro-bodiam-iaxsip, s, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/11' in macro 'bodiam-iaxsip' == Spawn extension (bodiam, 352, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/11' -- Hungup 'IAX2/[EMAIL PROTECTED]/11' ___ 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] English vs American voice files
Hi
In the end we found it easy to record our own using this section in
extensions.conf. This also meant that we could add our own company specific
ones in the same voice (not shown here). Basically you get someone to dial the
8NNN1 to record or 8NNN2 to playback. The prompts are shown below and we just
printed out this text. It was our intention to use festival to read these, but
this was easier. The text has been amended to reflect the UK (e.g. Hash instead
of pound).
Many sites may not need all of them and if you omit them the US voice will play
instead.
Paul
[EMAIL PROTECTED]
[macro-record-message]
;
; ARG1 file name of message, assumed to be in sounds folder, but if below has a
subfolder name prepended
; ARG2 text describing message
; Called with 8NNNX where NNN is the message and X is 1 to playback or 2 to
record.
exten = s,1,GotoIf($[${MACRO_EXTEN:4} = 2]?10:2) ; if fifth digit is 2
then go to record, otherwise playback
exten = s,2,Playback(/var/lib/asterisk/sounds/${ARG1}) ;playback here
exten = s,3,Wait(1)
exten = s,4,Hangup
exten = s,10,Wait(1) ;record here
exten = s,11,Record(/var/lib/asterisk/sounds/${ARG1}:gsm)
exten = s,12,Wait(1)
exten = s,13,Playback(/var/lib/asterisk/sounds/${ARG1})
exten = s,14,Wait(1)
exten = s,15,Hangup
[record-messages]
; Special context used to record voicemail messages
exten = _8001X,1,Macro(record-message,gb/hours,hours)
exten = _8002X,1,Macro(record-message,gb/minutes, minutes)
exten = _8003X,1,Macro(record-message,gb/auth-incorrect, Password incorrect.
Please enter your password followed by the hash key)
exten = _8004X,1,Macro(record-message,gb/auth-thankyou, Thank you. )
exten = _8005X,1,Macro(record-message,gb/invalid, 'I am sorry, that is not a
valid extension. Please try again' )
exten = _8006X,1,Macro(record-message,gb/pbx-invalid, 'I am sorry, that's not
a valid extension. Please try again. ')
exten = _8007X,1,Macro(record-message,gb/pbx-invalidpark, 'I am sorry, there
is no call parked on that extension. Please try again.')
exten = _8008X,1,Macro(record-message,gb/pbx-transfer, Transfer. )
exten = _8009X,1,Macro(record-message,gb/privacy-incorrect, 'I'm sorry, that
number is not valid. ')
exten = _8010X,1,Macro(record-message,gb/privacy-prompt, (Please enter your
ten-digit phone number, starting with the area code. )
exten = _8011X,1,Macro(record-message,gb/privacy-thankyou, Thank you. )
exten = _8012X,1,Macro(record-message,gb/privacy-unident, The party you are
trying to reach does not accept unidentified calls. )
exten = _8013X,1,Macro(record-message,gb/ss-noservice, The number you have
dialed is not in service. Please check the number and try again. )
exten = _8014X,1,Macro(record-message,gb/transfer, Please hold while I try
that extension. )
exten = _8015X,1,Macro(record-message,gb/tt-allbusy, All representatives of
the household are currently assisting other telemarketers. Please hold and your
call will be answered in the order it was received. )
exten = _8016X,1,Macro(record-message,gb/tt-monkeysintro, They have been
carried away by monkeys. )
exten = _8017X,1,Macro(record-message,gb/tt-somethingwrong, Something is
terribly wrong, )
exten = _8018X,1,Macro(record-message,gb/tt-weasels, Weasels have eaten our
phone system. )
exten = 80191,1,Playback(/var/lib/asterisk/sounds/gb/tt-allbusy)
exten = 80191,2,Playback(/var/lib/asterisk/sounds/gb/tt-monkeysintro)
exten = 80191,3,Playback(/var/lib/asterisk/sounds/tt-monkeys) ; Ho Ho
exten = _8020X,1,Macro(record-message,gb/dir-instr, 'If this is the person you
are looking for press 1 now, otherwise please press star now. )
exten = _8021X,1,Macro(record-message,gb/dir-intro, 'Welcome to the directory.
Please enter the first three letters of your party's last name using your touch
tone keypad. Use the 7 key for Q, and the 9 key for Zed.')
exten = _8022X,1,Macro(record-message,gb/dir-nomatch, No directory entries
match your search. )
exten = _8023X,1,Macro(record-message,gb/dir-nomore, There are no more
compatible entries in the directory. )
;; Not needed - blank exten = _8024X,1,Macro(record-message,gb/dir-intro-fn,
TO BE FILLED IN)
exten = _8031X,1,Macro(record-message,gb/conf-getchannel, Please enter the
channel number followed by the hash key. )
exten = _8032X,1,Macro(record-message,gb/conf-getconfno, Please enter your
conference number followed by the hash key. )
exten = _8033X,1,Macro(record-message,gb/conf-getpin, Please enter the
conference pin number. )
exten = _8034X,1,Macro(record-message,gb/conf-invalid, That is not a valid
conference number. Please try again. )
exten = _8035X,1,Macro(record-message,gb/conf-invalidpin, That pin is invalid
for this conference. )
exten = _8036X,1,Macro(record-message,gb/conf-onlyperson, You are currently
the only person in this conference. )
exten = _8037X,1,Macro(record-message,gb/conf-adminmenu, 'Please press 1 to
mute or unmute yourself. Or press 2 to lock or unlock the conference. )
exten =
[Asterisk-Users] Re: IAX Firefly config (Jeromy Grimmett)
Hi
This works for us:
In IAX.conf
[NNN]
notransfer=yes
secret=**
context=contextname
host=dynamic
type=friend
callerid=John Doe0
mailbox=NNN
qualify=no
language=gb
The mailbox is for the mailbox number.
In extensions.conf we use a macro
[macro-iaxext]
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20,tT)
; if drops out here means unavailable
exten = s,2,NoOp(State 2 ${DIALSTATUS})
exten = s,3,Voicemail(b${ARG1})
with lines like:
exten =NNN,1,Macro(iaxext,NNN,IAX2/NNN)
Nothing very clever but works.
Remember to open port 4569 in your firewall if you use from outside on internet
to server and to map to asterisk server in firewall.
Paul
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[Asterisk-Users] IAX2 attended transfer on 1-0-6 Stable
Hi Guys
I'm still wrestling with trying to make IAX2 softphones do attended transfer.
My iax.conf section has:
[XXX]
secret=mysecret
context=mycontext
host=dynamic
type=friend
callerid=Paul Redstone
mailbox=NNN ; notifies if mailbox has something in
qualify=no ; checks for connection every 10 seconds
notransfer=yes
All my dial statements in extensions.conf are like:
Dial(ZAP/g1/${ARG1},20,tTr) with t and T so that I should be able to transfer.
Unattended transfer works OK using # but attended does not.
Features.conf has
[featuremap]
blindxfer = #1; Blind transfer key - this simple let syou
transfer a call and immediately hangup
disconnect = *0 ; Disconnect
automon = *1 ; One Touch Record
atxfer = *2 ; Attended transfer which lets you transfer and
talk to someone else then hangup to transfer
Not sure if this matters as I running 1-0-6 stable.
Tried every iax2 softphone around. I'm missing something here but cannot see
what it is. Google comes up with statements about attended transfers being
difficult but that is all.
Anyone definitive experience of softphones doing attended transfers?
TIA
Paul
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[Asterisk-Users] Asterisk 1.0.6 stable IAX2 Firefly supervised call transfer?
Hi I see various discussions on this but cannot get it to work, and is not clear that anyone resolved this. This seems pretty fundamental so I am missing something, but I cannot find it anywhere. # does work for blind transfers - no problem. But the various * commands given in features.conf do not. OK, I've picked up that this may not be in the released one but also I've found that chan_iax2.c does talk about attended transfers. Also iax2 debug shows that the * key is being recognised and passed back. Can anyone help on this - IAX2 is so much better than SIP which may not have this problem. We're using Firefly phone - neat, simple, seems reliable. Paul ___ 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: UK (english) sound files (Paul R)
So now that they are done how about you post the files for us? Share the wealth. Mark Will be happy to do so once macro refined a little, but it is rather long (about 600 lines) and I thought long posts were bad manners. Otherwise this will be odne by the end of the weekend/ Paul ___ 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] UK (english) sound files (Paul R)
To add something to a post of a few
days ago on this:
We're just putting in an asterisk
system and wanted to have our own messages.
We're Asterisk and are not yet live but the
following works.
Our PA simply has a list of the
extract from extensions.conf as shown below,
calls a number e.g. 8004, then hears a beep,
says the message andthen clicks #. It
then plays back the message. It took her
about 2 hours to do all the messages. The
benefit of your own voice is that company
specific messages sound consistent. We've
also changed some of the prompts a little
to Anglicise them - e.g. hash instead of
pounds key.
The messages go immediately into the
gb sounds folder so are active immediately.
We've also done the same with digits,
letters etc - the scripts put the message
straight into the correct folder.
The macro is intended to be enhanced
so as to allow the message to be just played
and also, in the future perhaps to use festival
to read what should be said. The message
list is taken from the asterisk wiki page
on sounds and we'd adding in a few company
specific ones.
Send an e-mail to [EMAIL PROTECTED]
with subject "Sounds Request for Paul" and
we'll end back the script once we have it
done.
The benefit of this is it is easy to
add new scripts and to amend one. The [record-messages]
context can be included just for some users
and even disabled most of the time.
[macro-record-message];;
ARG1 file name of message, assumed to be
in sounds folder, but if below has a subfolder
name prepended; ARG2 text describing
message (NT YET USED);exten =
s,1,Wait(1)exten = s,2,Record(/var/lib/asterisk/sounds/gb/${ARG1}:gsm)exten
= s,3,Wait(1)exten = s,4,Playback(/var/lib/asterisk/sounds/gb/${ARG1})exten
= s,5,Wait(1)exten = s,6,Hangup
[record-messages]; Special context
used to record voicemail messagesexten
= 8001,1,Macro(record-message,hours,hours)exten
= 8002,1,Macro(record-message,minutes,
minutes)exten = 8003,1,Macro(record-message,auth-incorrect,
Password incorrect. Please enter your password
followed by the hash key)exten =
8004,1,Macro(record-message,auth-thankyou,
Thank you. )exten = 8005,1,Macro(record-message,invalid,
'I am sorry, that is not a valid extension.
Please try again' )exten = 8006,1,Macro(record-message,pbx-invalid,
'I am sorry, that's not a valid extension.
Please try again. ')exten = 8007,1,Macro(record-message,pbx-invalidpark,
'I am sorry, there is no call parked on that
extension. Please try again.')exten =
8008,1,Macro(record-message,pbx-transfer,
Transfer. )
etc
Paul
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Re: [Asterisk-Users] UK (english) sound files (Paul R
Sorry - still forget to clear the HTML flag - thought I'd better post again.
To add something to a post of a few days ago on this:
We're just putting in an asterisk system and wanted to have our own messages.
We're Asterisk and are not yet live but the following works.
Our PA simply has a list of the extract from extensions.conf as shown below,
calls a number e.g. 8004, then hears a beep, says the message and
then clicks #. It then plays back the message. It took her about 2 hours to do
all the messages. The benefit of your own voice is that company specific
messages sound consistent. We've also changed some of the prompts a little to
Anglicise them - e.g. hash instead of pounds key.
The messages go immediately into the gb sounds folder so are active
immediately.
We've also done the same with digits, letters etc - the scripts put the message
straight into the correct folder.
The macro is intended to be enhanced so as to allow the message to be just
played and also, in the future perhaps to use festival to read what should be
said. The message list is taken from the asterisk wiki page on sounds and we'd
adding in a few company specific ones.
Send an e-mail to [EMAIL PROTECTED] with subject Sounds Request for
Paul and we'll end back the script once we have it done.
The benefit of this is it is easy to add new scripts and to amend one. The
[record-messages] context can be included just for some users and even disabled
most of the time.
[macro-record-message]
;
; ARG1 file name of message, assumed to be in sounds folder, but if below has a
subfolder name prepended
; ARG2 text describing message (NT YET USED)
;
exten = s,1,Wait(1)
exten = s,2,Record(/var/lib/asterisk/sounds/gb/${ARG1}:gsm)
exten = s,3,Wait(1)
exten = s,4,Playback(/var/lib/asterisk/sounds/gb/${ARG1})
exten = s,5,Wait(1)
exten = s,6,Hangup
[record-messages]
; Special context used to record voicemail messages
exten = 8001,1,Macro(record-message,hours,hours)
exten = 8002,1,Macro(record-message,minutes, minutes)
exten = 8003,1,Macro(record-message,auth-incorrect, Password incorrect. Please
enter your password followed by the hash key)
exten = 8004,1,Macro(record-message,auth-thankyou, Thank you. )
exten = 8005,1,Macro(record-message,invalid, 'I am sorry, that is not a valid
extension. Please try again' )
exten = 8006,1,Macro(record-message,pbx-invalid, 'I am sorry, that's not a
valid extension. Please try again. ')
exten = 8007,1,Macro(record-message,pbx-invalidpark, 'I am sorry, there is no
call parked on that extension. Please try again.')
exten = 8008,1,Macro(record-message,pbx-transfer, Transfer. )
etc
Paul
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[Asterisk-Users] Fritz Card ISDN in UK - Unable to dial. 0x3301/0x3302 errors.
Hi
Newbie asterisk guy here and forgive
this slightly long mail, but I'm stuck on
this for a week.
I'm having major problems getting a
Fritz card to dial out in the UK (or indeed
answer, but I've been concentrating on dialing
out). I'm getting the 0x3301 or 0x3302 error.
My capi.conf file is:--
[general]nationalprefix=0internationalprefix=00rxgain=0.8txgain=0.8
[interfaces]msn=1580XX
(the x's are replaced with our full msn)incomingmsn=*controller=1softdtmf=1accountcode=context=bodiam
(and the bodiam context includes bodiam-out);echosquelch=1;echocancel=yes;echotail=64;callgroup=1;deflect=12345678devices=2
The significant part of my extensions.conf
is:--[bodiam-out]exten
= _9.,1,StripMSD,1exten = _.,2,NoOp("Test1")exten
= _.,3,Dial,CAPI/1580xx:BYEXTENSION,30,r
(x's replaced with full msn)exten =
_.,4 NoOp("test2")exten = _.,5,Congestion
The output when enabling CAPI is
-
-- Executing StripMSD("SIP/202-5a30", "1")
in new stack -- Executing
NoOp("SIP/202-5a30", ""Test1"") in new stack
-- Executing Dial("SIP/202-5a30", "CAPI/1580xx:BYEXTENSION")
in new stack -- data
= "">
-- capi request omsn = 1580xx
== found capi with omsn = 1580xx
== CAPI Call CAPI[contr1/1580xx]/0
-- Called 1580xx:01580yy
(01580xx has msn; 01580yy is number
being called) -- CONNECT_CONF
ID=001 #0x0004 LEN=0014 Controller/PLCI/NCCI
= 0x101 Info
= 0x0
== received CONNECT_CONF PLCI
= 0x101 INFO = 0 --
DISCONNECT_IND ID=001 #0x0005 LEN=0014
Controller/PLCI/NCCI
= 0x101 Reason
= 0x3301
== DISCONNECT_IND PLCI=0x101
REASON=0x3301 == No one is available
to answer at this timeApr 6 07:51:34
WARNING[576]: pbx.c:1291 pbx_extension_helper:
No application '' for extension (bodiam,
01580yy, 4) == Spawn extension
(bodiam, 01580830902, 4) exited non-zero
on 'SIP/202-5a30'
Sometimes
I get 0x3302.
The ISDN line being used is one normally
used by our conventional PABX so we know
it works - we just plug it into the Fritz
card instead.
capi info shows the channels are recognised
and free in Asterisk.
capiinfo in command prompt shows:
Controller 1:Manufacturer: AVM GmbHCAPI
Version: 2.0Manufacturer Version: 3.17-02
(49.18)Serial Number: 101BChannels:
2Global Options: 0x0039
internal controller supported
DTMF supported Supplementary
Services supported channel
allocation supported (leased lines)B1
protocols support: 0x411f
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
V.110 asynconous operation with start/stop
byte framing V.110 synconous
operation with HDLC framing
T.30 modem for fax group 3
Modem asyncronous operation with start/stop
byte framingB2 protocols support: 0x0b1b
ISO 7776 (X.75 SLP) Transparent
LAPD with Q.921 for D channel X.25 (SAPI
16) T.30 fro fax group 3
ISO 7776 (X.75 SLP) with V.42bis compression
V.120 asyncronous mode V.120
bit-transparent modeB3 protocols support:
0x80bf Transparent
T.90NL, T.70NL, T.90 ISO
8208 (X.25 DTE-DTE) X.25
DCE T.30 for fax group 3
reserved
0100 0200
3900 1f010040 1b0b
bf80
0101
0002
Supplementary services support: 0x03ff
Hold / Retrieve Terminal
Portability ECT
3PTY Call Forwarding
Call Deflection MCID
CCBS--The 0x3301
and 3302 errors seem to be protocol ones
and it 'feels' like some sort of country
configuration issue or something like that.
Can any one give me any advice/help on this
as I'm been stuck on this for a week.
Many thanksPaul
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[Asterisk-Users] (no subject)
Hi
Repeated e-mail as I forgot to make plain text - sorry. Newbie asterisk guy
here and forgive this slightly long mail, but I'm stuck on this for a week.
I'm having major problems getting a Fritz card to dial out in the UK (or indeed
answer, but I've been concentrating on dialing out). I'm getting the 0x3301 or
0x3302 error.
My capi.conf file is:
--
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=1580XX (the x's are replaced with our full msn)
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=bodiam (and the bodiam context includes bodiam-out)
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2
The significant part of my extensions.conf is:
--
[bodiam-out]
exten = _9.,1,StripMSD,1
exten = _.,2,NoOp(Test1)
exten = _.,3,Dial,CAPI/1580xx:BYEXTENSION,30,r (x's replaced with full
msn)
exten = _.,4 NoOp(test2)
exten = _.,5,Congestion
The output when enabling CAPI is
-
-- Executing StripMSD(SIP/202-5a30, 1) in new stack
-- Executing NoOp(SIP/202-5a30, Test1) in new stack
-- Executing Dial(SIP/202-5a30, CAPI/1580xx:BYEXTENSION) in new
stack
-- data = 1580xx:01580yy
-- capi request omsn = 1580xx
== found capi with omsn = 1580xx
== CAPI Call CAPI[contr1/1580xx]/0 -- Called 1580xx:01580yy
(01580xx has msn; 01580yy is number being called)
-- CONNECT_CONF ID=001 #0x0004 LEN=0014
Controller/PLCI/NCCI= 0x101
Info= 0x0
== received CONNECT_CONF PLCI = 0x101 INFO = 0
-- DISCONNECT_IND ID=001 #0x0005 LEN=0014
Controller/PLCI/NCCI= 0x101
Reason = 0x3301
== DISCONNECT_IND PLCI=0x101 REASON=0x3301
== No one is available to answer at this time
Apr 6 07:51:34 WARNING[576]: pbx.c:1291 pbx_extension_helper: No application
'' for extension (bodiam, 01580yy, 4)
== Spawn extension (bodiam, 01580830902, 4) exited non-zero on 'SIP/202-5a30'
Sometimes I get 0x3302.
The ISDN line being used is one normally used by our conventional PABX so we
know it works - we just plug it into the Fritz card instead.
capi info shows the channels are recognised and free in Asterisk.
capiinfo in command prompt shows:
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.17-02 (49.18)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
internal controller supported
DTMF supported
Supplementary Services supported
channel allocation supported (leased lines)
B1 protocols support: 0x411f
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
V.110 asynconous operation with start/stop byte framing
V.110 synconous operation with HDLC framing
T.30 modem for fax group 3
Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
ISO 7776 (X.75 SLP)
Transparent
LAPD with Q.921 for D channel X.25 (SAPI 16)
T.30 fro fax group 3
ISO 7776 (X.75 SLP) with V.42bis compression
V.120 asyncronous mode
V.120 bit-transparent mode
B3 protocols support: 0x80bf
Transparent
T.90NL, T.70NL, T.90
ISO 8208 (X.25 DTE-DTE)
X.25 DCE
T.30 for fax group 3
reserved
0100
0200
3900
1f010040
1b0b
bf80
0101 0002
Supplementary services support: 0x03ff
Hold / Retrieve
Terminal Portability
ECT
3PTY
Call Forwarding
Call Deflection
MCID
CCBS
--
The 0x3301 and 3302 errors seem to be protocol ones and it 'feels' like some
sort of country configuration issue or something like that. Can any one give me
any advice/help on this as I'm been stuck on this for a week.
Many thanks
Paul
___
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[Asterisk-Users] Fritz Card ISDN in UK - Unable to dial. 0x3301/0x3302 errors
Hi
Repeated e-mail as I forgot to make plain text - sorry and then forgot subject
line. Newbie asterisk guy here and forgive this slightly long mail, but I'm
stuck on this for a week.
I'm having major problems getting a Fritz card to dial out in the UK (or indeed
answer, but I've been concentrating on dialing out). I'm getting the 0x3301 or
0x3302 error.
My capi.conf file is:
--
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=1580XX (the x's are replaced with our full msn)
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=bodiam (and the bodiam context includes bodiam-out)
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2
The significant part of my extensions.conf is:
--
[bodiam-out]
exten = _9.,1,StripMSD,1
exten = _.,2,NoOp(Test1)
exten = _.,3,Dial,CAPI/1580xx:BYEXTENSION,30,r (x's replaced with full
msn)
exten = _.,4 NoOp(test2)
exten = _.,5,Congestion
The output when enabling CAPI is
-
-- Executing StripMSD(SIP/202-5a30, 1) in new stack
-- Executing NoOp(SIP/202-5a30, Test1) in new stack
-- Executing Dial(SIP/202-5a30, CAPI/1580xx:BYEXTENSION) in new
stack
-- data = 1580xx:01580yy
-- capi request omsn = 1580xx
== found capi with omsn = 1580xx
== CAPI Call CAPI[contr1/1580xx]/0 -- Called 1580xx:01580yy
(01580xx has msn; 01580yy is number being called)
-- CONNECT_CONF ID=001 #0x0004 LEN=0014
Controller/PLCI/NCCI= 0x101
Info= 0x0
== received CONNECT_CONF PLCI = 0x101 INFO = 0
-- DISCONNECT_IND ID=001 #0x0005 LEN=0014
Controller/PLCI/NCCI= 0x101
Reason = 0x3301
== DISCONNECT_IND PLCI=0x101 REASON=0x3301
== No one is available to answer at this time
Apr 6 07:51:34 WARNING[576]: pbx.c:1291 pbx_extension_helper: No application
'' for extension (bodiam, 01580yy, 4)
== Spawn extension (bodiam, 01580830902, 4) exited non-zero on 'SIP/202-5a30'
Sometimes I get 0x3302.
The ISDN line being used is one normally used by our conventional PABX so we
know it works - we just plug it into the Fritz card instead.
capi info shows the channels are recognised and free in Asterisk.
capiinfo in command prompt shows:
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.17-02 (49.18)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
internal controller supported
DTMF supported
Supplementary Services supported
channel allocation supported (leased lines)
B1 protocols support: 0x411f
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
V.110 asynconous operation with start/stop byte framing
V.110 synconous operation with HDLC framing
T.30 modem for fax group 3
Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
ISO 7776 (X.75 SLP)
Transparent
LAPD with Q.921 for D channel X.25 (SAPI 16)
T.30 fro fax group 3
ISO 7776 (X.75 SLP) with V.42bis compression
V.120 asyncronous mode
V.120 bit-transparent mode
B3 protocols support: 0x80bf
Transparent
T.90NL, T.70NL, T.90
ISO 8208 (X.25 DTE-DTE)
X.25 DCE
T.30 for fax group 3
reserved
0100
0200
3900
1f010040
1b0b
bf80
0101 0002
Supplementary services support: 0x03ff
Hold / Retrieve
Terminal Portability
ECT
3PTY
Call Forwarding
Call Deflection
MCID
CCBS
--
The 0x3301 and 3302 errors seem to be protocol ones and it 'feels' like some
sort of country configuration issue or something like that. Can any one give me
any advice/help on this as I'm been stuck on this for a week.
Many thanks
Paul
___
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Re: [Asterisk-Users] Fritz Card ISDN in UK - Unable to dial.
I'm having major problems getting a Fritz card to dial out in the UK (or indeed answer, but I've been concentrating on dialing out). I'm getting the 0x3301 or 0x3302 error.Not a direct answer, but are you _absolutely sure_ the card works? Ihad the exact same thing late last year here in Australia, there wasnothing wrong with the config but once we got around to testing it atanother known installation it was found to be faulty. Replaced itunder warranty, and everything worked as we had it.Might save you some head-banging, at least...Andrew Good point Andrew, this card has not been used for two years. But it does seem to recognise the ISDN msn - if I change the msn settings to be an invalid number it gives a message to indicate that, so my guess is that it is functioning if it can pick this up. Unfortunately I do not have any other cards to try. I did wonder whether the Fritz card has any settings which reflect national or other protocols but I think that all of Europe uses the same now and think this is the default. Paul ___ 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
