RE: [Asterisk-Users] How to allow multiple codecs in A@H
Hi, When adding codecs to the extension setup in [EMAIL PROTECTED], enter multiple codecs as gsmulawalaw on the same line. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of stevanus Sent: 15 June 2005 04:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to allow multiple codecs in [EMAIL PROTECTED] hi, just put those lines (allow=bla) in peer details box in AMP GUI. at section add sip trunk. best regards, stevanus Erdem HAK wrote: I wonder how to allow more then one codec in AMP ([EMAIL PROTECTED]) GUI? For example I want to configure like this allow=gsm allow=g729 ... I can add these by editing sip_additional.conf, but i want to add codecs using AMP, any suggestions? Thanks Erdem HAKI [EMAIL PROTECTED] ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Sipura SPA-841
Edwin Lam wrote: does anybody has experienece with Sipura SPA-841 phone unit? how's its sound quality especially speaker phone? i have several Grandstream phones and was getting fustrated about the quality and bugs of their firmware. As the other's have said, the speakerphone is useless. My only other complaint is the lack of a backlight. Oh the other hand, no backlight makes it nice for a bedroom phone. ___ 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] localize ${VM_DATE} ?
Hello,
I looked everywhere in the docs and in google but couldn't find an
answer.
Is it possible to localize the output of ${VM_DATE} (say, in french) ?
--
Only half the people in the world are above average intelligence.
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SV: [Asterisk-Users] Asterisk and Panasonic KX-TD1232
Thank you for response. So a standard quad bri card can serve the KX-TD1232, no need for anyt special fxs similar equipment. Would like not to invest much more money into the Panasonic, I wan't it out :P BR Amund -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Peter Svensson Sendt: 14. juni 2005 17:47 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Asterisk and Panasonic KX-TD1232 On Tue, 14 Jun 2005, Amund Nygaard wrote: We have around 50 phones in our company, and I am playing with the thought to gradually go over to using sip services and ip-phones internally. However at first I would liked the Asterisk just to sit between the phone line and the Panaosnic, so I can take out one lin/number at a time to use ip phones. I am new to Asterisk, and haven't done much configuring of the PBX either. So I also wonder how difficult such setup is. We use today 4 BRI lines that connects us to the telephone network, would I then need 2xTE410P to put the Asterisk between the Panasonic and the phone network? We use Asterisk in this exact way. You will either need two quad-bri cards in the asterisk box or 1 TE410P in the asterisk box and a Panasonic PRI card for the KX-TD1232. The TE410P is a quad PRI, not a quad BRI. We use PRI on all links. Peter ___ 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 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] TAPI for Dos (xbase / Clipper)
Hi list! Does anyone know of a TAPI that will work wil a Clipper application (MS DOS)? Alternatively I could recompile the lot and run it on a linux box but then again I would need a TAPI I guess? I just want clickdial from our CRM app. Alternatively, I could create a samba share for the asterisk call files and have our CRM app create Asterisk call files and just dump them on that share. Anyone ever tried this approach? Thanks!! ___ 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] newbie question..
hello all, the situation here is i want when user make outgoing call, asterisk will call 1800XX first then after 3 or 4 sec asterisk will insert the number that user want to call.. user don't know that the call is go to 1800XX first.. means user just insert the number that they want to call then asterisk will insert that number after 3 or 4 sec.. can i that in asterisk? i'll apreciate any help or advise.. regards, shahdan __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] Old but Gold
Title: Message
Everyone,
Im sure you've seen
this error a million times, but Ive looked everywhere I can think of still
haven't found a solution that works.
I'm trying to make
an outside call, I can call the physical phone from a xlite on another pc (and
vice versa) but whenever I try to make a call to the outside world, this
happens:
on the
CLI:
Jun 15 08:45:20
NOTICE[10390]: app_dial.c:972 dial_exec_full: Unable to create channel of type
'Zap' (cause 0) == Everyone is busy/congested at this time
(1:0/0/1)
This is my
zapata.conf:
[channels]
context=default
switchtype=national
signalling=fxo_ksrxwink=300
; Atlas seems to use long (250ms)
winksusecallerid=yescidsignalling=v23cidstart=polarityhidecallerid=nocallwaiting=yesusecallingpres=yessendcalleridafter=3callwaitingcallerid=yesthreewaycalling=notransfer=nocancelcallforward=yescallreturn=noechocancel=64echocancelwhenbridged=yesechotraining=yesechotraining=800rxgain=5.0txgain=6.5group=1callgroup=1pickupgroup=1immediate=nobusydetect=nobusycount=6callprogress=noprogzone=ukcallerid=01614830073
;I think this
refers to the telephone socket on the card; this will have something to do
with income or outgoing calls , probably incominggroup=1 ; in group
1callgroup=1pickupgroup=1context=defaultchannel=1 ; on
channel 1callerid=01614830073 dylan
This is the contents
of my zaptel.conf
# added by dylan as
per http://www.digium.com/index.php?menu=configuration#TDM11B
fxoks=1 # Make sure
that the FXS(green) module is closest to the bracket if you are looking at the
side of the card with all of $fxsks=4 # FXO
moduledefaultzone=ukloadzone=uk# end of dylan
added
This is the output
from ztcfg -vvv
[EMAIL PROTECTED]
asterisk]# ztcfg -vvv
Zaptel
Configuration==
Channel
map:
Channel 01: FXO Kewlstart (Default) (Slaves:
01)Channel 04: FXS Kewlstart (Default) (Slaves: 04)
2 channels configured.
[EMAIL PROTECTED] asterisk]#
And this is the
relavent part of my extentions.conf
; START OF DYLANS
MESSING AROUND SECTION
**[globals]OUTBOUND=Zap/4DYLAN=SIP/301JOHN=SIP/300EVERYONE=${DYLAN}${JOHN}
[nationalcalls]exten=
_01.,1,Dial(${OUTBOUND}/${EXTEN},20)exten=
_01.,2,Congestion
[from-sip]include =defaultinclude
=nationalcalls
exten =
301,1,Dial(SIP/301,5) ' if 301 is dialed, dial out on sip channel to extention
301 for 5 secondsexten = 301,2,Voicemail(u301)
exten =
500,1,VoicemailMainexten = 500,2,Hangup
exten =
300,1,Dial(SIP/300,10)exten =
300,2,Voicemail(u300)
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[Asterisk-Users] SIP REFER method.
Hi, It isn't maybe the best place to ask the question, but I don't know better :( Does anyone could tell me if sending REFER request virtually ends current call? I mean if one sends or receives REFER request, he should stop rending RTP, just as it is required for BYE request. In other words is is more or less equivalent to the BYE request? Regards, Marcin Okraszewski ___ 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] Portable USB headset for VoIP
The supplier is from www.broad-tel.com On 6/14/05, Jian Hong GUAN [EMAIL PROTECTED] wrote: That interests me. Can you send me the informations about products and suppliers? Best regards, --Hong ___ 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 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] How to restrict access to * for a specific soft/hard phone model?
Anobody gives me a tip how to recognize what soft/hard phone is in use for a call? I would like to allow access to * for those phones which have been tested and validated by me, e.g. calls allowed from X-lite but not from Linksys PAP2. I want to be sure that every user uses the same phone model, for example X-lite. Thanks! ___ 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] Calling on all Polycom Experts
Could not load time from 0.0.0.0(0.0.0.0). No dns server to resolve SNTP address Other than that I don't see a problem. You need to look at the logs/MAC-app.log for more clues. What I find useful is to download a free XML editor and load the config file into it, that will test for xml syntax errors you may not see by eye. Also, take everything out except the following and build from there. PHONE_CONFIG phone1 reg reg.1.address=xxx reg.1.auth.password=yyy reg.1.auth.userId=xxx reg.1.displayName=xxx reg.1.label=xxx reg.1.type=private reg.1.server.1.expires=3600 reg.1.server.1.address=ipaddress reg.1.server.1.expires.lineSeize=30 reg.1.server.1.port=5060 reg.1.server.1.register=1 reg.1.server.1.retryMaxCount=0 reg.1.server.1.retryTimeOut=0 reg.1.server.1.transport=2 call.serverMissedCall.1.enabled=1 msg.mwi.1.callBack=8500 msg.mwi.1.callBackMode=contact msg.mwi.1.subscribe= msg msg.bypassInstantMessage=1 / /phone1 /PHONE_CONFIG ___ 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] Stable Versions
Running stable, one can use 1.0.7 or 1.0.8 or just v1.0... Is that right? When v1 updates are made, what tree do they go to? So, there would really never be 1.0.7 updates or are there? -- respectfully, Joseph ___ 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] app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)
Hi,
Ive been struggling with asterisk for a few days now. I understand
pretty much how it works and how to tie things together (SIP - SIP
internally works fine etc), but just cannot get asterisk to work with an
X100P clone (its a Ambient MD3200, if that means anything to you guys).
I have tried (initially) asterisk 1.07 with zaptel 1.07, and now
Asterisk CVS-HEAD with zaptel cvs. Both give me the same error.
/etc/zaptel.conf
--
fxsks=1
loadzone=uk
defaultzone=uk
/etc/asterisk/zapata.conf
--
[channels]
language=en
context=incoming
signalling=fxs_ks
channel = 1
/etc/asterisk/extensions.conf
--
[general]
static=yes
writeprotect=no
[local]
exten = _11.,1,Dial(Zap/1,9w${EXTEN}:2)
[incoming]
exten = s,1,Answer()
exten = s,2,BackGround(demo-congrats) ; Play a congratulatory message
[outgoing]
exten = _9.,1,Ringing
exten = _9.,2,Wait,2
exten = _9.,3,Answer()
exten = _9.,4,Dial(ZAP/1/9w${EXTEN}:1)
[default]
include = outgoing
Loading zaptel modules:
--
asterisk zaptel # modprobe zaptel
asterisk zaptel # modprobe wcfxo
asterisk zaptel # lsmod
Module Size Used by
wcfxo 12576 0
zaptel222916 1 wcfxo
crc_ccitt 1952 1 zaptel
asterisk zaptel # ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
Running asterisk
--
asterisk zaptel # asterisk -gc
Registering builtin applications:
[AbsoluteTimeout]
== Registered application 'AbsoluteTimeout'
[Answer]
== Registered application 'Answer'
[BackGround]
== Registered application 'BackGround'
[Busy]
== Registered application 'Busy'
[Congestion]
== Registered application 'Congestion'
[DigitTimeout]
== Registered application 'DigitTimeout'
[Goto]
== Registered application 'Goto'
[GotoIf]
== Registered application 'GotoIf'
[GotoIfTime]
== Registered application 'GotoIfTime'
[ExecIfTime]
== Registered application 'ExecIfTime'
[Hangup]
== Registered application 'Hangup'
[NoOp]
== Registered application 'NoOp'
[Prefix]
== Registered application 'Prefix'
[Progress]
== Registered application 'Progress'
[ResetCDR]
== Registered application 'ResetCDR'
[ResponseTimeout]
== Registered application 'ResponseTimeout'
[Ringing]
== Registered application 'Ringing'
[SayNumber]
== Registered application 'SayNumber'
[SayDigits]
== Registered application 'SayDigits'
[SayAlpha]
== Registered application 'SayAlpha'
[SayPhonetic]
== Registered application 'SayPhonetic'
[SetAccount]
== Registered application 'SetAccount'
[SetAMAFlags]
== Registered application 'SetAMAFlags'
[SetGlobalVar]
== Registered application 'SetGlobalVar'
[SetLanguage]
== Registered application 'SetLanguage'
[Set]
== Registered application 'Set'
[SetVar]
== Registered application 'SetVar'
[ImportVar]
== Registered application 'ImportVar'
[StripMSD]
== Registered application 'StripMSD'
[Suffix]
== Registered application 'Suffix'
[Wait]
== Registered application 'Wait'
[WaitExten]
== Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
== Parsing '/etc/asterisk/modules.conf': Found
[res_features.so] = (Call Parking Resource)
== Parsing '/etc/asterisk/features.conf': Found
-- Registered extension context 'parkedcalls'
-- Added extension '700' priority 1 to parkedcalls
== Registered application 'ParkedCall'
== Registered application 'Park'
== Manager registered action ParkedCalls
[res_musiconhold.so] = (Music On Hold Resource)
== Registered application 'MusicOnHold'
== Registered application 'WaitMusicOnHold'
== Registered application 'SetMusicOnHold'
== Registered application 'StartMusicOnHold'
== Registered application 'StopMusicOnHold'
== Parsing '/etc/asterisk/musiconhold.conf': Found
[chan_zap.so] = (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1, FXS Kewlstart signalling
-- Automatically generated pseudo channel
== Registered channel type 'Zap' (Zapata Telephony Driver)
== Manager registered action ZapTransfer
== Manager registered action ZapHangup
== Manager registered action ZapDialOffhook
== Manager registered action ZapDNDon
== Manager registered action ZapDNDoff
== Manager registered action ZapShowChannels
[app_sayunixtime.so] = (Say time)
== Registered application 'SayUnixTime'
== Registered application 'DateTime'
[res_adsi.so] = (ADSI Resource)
== Parsing '/etc/asterisk/adsi.conf': Found
[res_crypto.so] = (Cryptographic Digital Signatures)
-- Loaded PUBLIC key 'iaxtel'
-- Loaded PUBLIC key 'freeworlddialup'
[res_indications.so] = (Indications Configuration)
== Parsing '/etc/asterisk/indications.conf': Found
-- Registered indication country 'cl'
-- Registered indication country 'tw'
-- Registered indication country 'us'
-- Registered indication country 'au'
-- Registered indication country 'fr'
[Asterisk-Users] Personalised Unavail / busy messages no longer play
We just upgraded our * to CVS head, and came across the following strange error: the personalised Busy and unavailable messages no longer play, we get the default Allison voice instead. I think that it is a problem with the wav49 format, because if I rerecord my personalised message, a new file with .wav is created, and this works. Anybody have any clues on this ? I also recently changed the voicemail.conf file to use wav instead of wav49, but I thought that was for attached messages - do I need to have both ? Julain ___ 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] Dial more then 9 digits
Could you kick me, I can't dial more then 9 digits. Is anyone some default length of extensions or dialed number. Thanks, Bob. ___ 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] How to restrict access to * for a specific soft/hard phone model?
I would like to allow access to * for those phones which have been tested and validated by me, e.g. calls allowed from X-lite but not from Linksys PAP2. I want to be sure that every user uses the same phone model, for example X-lite. mmmh... perhaps with sipgetheader you can get the useragent, and then drop/accept calls basing on some matching rules... Matteo -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ 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] Sipura SPA 3000 FXO Setting India
Hello all, I have a Sipura SPA 3000 and am trying to use the PSTN port on the unit to bring a POTS line into Asterisk. However, I am unable to find the localization settings to get the SPA 3000 to understand the local phone settings. The unit is located in India and I have not had any success in trying to find the correct settings. Anyone have any ideas? Thanks! ___ 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] Dial more then 9 digits
no I can? how is your dialout rules ? I have a client where you have to dial a 4 digit pin and then the rest of the number I simply have a exten = _1234.,1,Dail... On Wed, 2005-06-15 at 12:20 +0200, Bohuslav Coufal wrote: Could you kick me, I can't dial more then 9 digits. Is anyone some default length of extensions or dialed number. Thanks, Bob. ___ 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 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] Asterisk slow transferring calls
Hi,
Running Asterisk CVS-Head latest on a Dual P3 800 1Gb Ram.
For some odd reason now that I have the asterisk box almost to the stage
I want it, I hit a problem.
I have a te405p in the system, Zap/g1 is connected to the telco as an
ISDN 30, Zap/g4 is connected by ISDN Primary Rate to an Ericcson BP250
phone system.
When calls come in on g1 they go straight through instantaneously to the
entensions on the ericsson or local sip phones no problems, if someone
on a sip phone calls an extension on the ericsson it goes straight
through no pause.
If someone on the ericsson system dials a sip phone it takes close to 3
full seconds before the sip phone rings, it takes that long just to get
to the asterisk box, although its not the ericsson phone system that is
the problem, if I dump a straight plain extensions.conf into the system
it works perfectly and is fast from the ericsson to the sip phone, if I
use the one I want to get running its slow again.
Can someone have a breeze through and let me know what they think might
be causing the problem.
I think I am not getting the right idea with out the contexts work and
it might be looping or something, te405p-in and sip need access to each
other and the ability to dialout, and voip, voip needs access to dial
the ericsson system and the sip phones (haven't added that part yet) but
not access to an outside line.
James
My extensions.conf
#include extensions_sip.conf
[globals]
EMERGENCY=0
EMERGENCY_TRUNK=Zap/10
[dialstring]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
[atp-out]
exten =
_9X.,1,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN:1})
exten = _9X.,2,Congestion
exten = _9X.,3,Hangup
[atp-in]
exten = 30182849,1,SetMusicOnHold(record)
exten = 30182849,2,Dial(SIP/bt-rlm,45,t)
exten = 30182849,3,Voicemail,u550
exten = 30182849,103,Voicemail,b550
[te405p-in]
exten = _2XX,1,Dial(Zap/g4/${EXTEN},60,r)
exten = _2XX,2,Hangup
exten = _73816592XX,1,Dial(Zap/g4/${EXTEN:-3},60,r)
exten = _73816592XX,2,Hangup
exten = _7XX,1,Dial(Zap/g4/${EXTEN},60,r)
exten = _7XX,2,Hangup
exten = _1XXX,1,Dial(Zap/g4/${EXTEN},60,r)
exten = _1XXX,2,Hangup
include = sip
include = parkedcalls
include = te405p-outgoing
include = transfer-record
[te405p-ext]
exten = s,1,SetMusicOnHold(random)
exten = s,2,Dial(SIP/bt-pavilion,45,t)
exten = s,4,VoiceMail,u500
exten = s,5,Hangup
exten = 38166400,1,SetMusicOnHold(random)
exten = 38166400,2,Dial(Zap/g4/211,600,t)
exten = 38166400,3,VoiceMail,u500
exten = 38166400,4,Hangup
exten = 38166444,1,DISA(1234|sip)
exten = _381664XX,1,SetMusicOnHold(random)
exten = _381664XX,2,Dial(Zap/g4/2${EXTEN:-2},600,t)
exten = _381664XX,3,VoiceMail,u500
exten = _381664XX,4,Hangup
[te405p-outgoing]
exten =
000,1,SetVar(CALLFILENAME=/mnt/asterisk/EMERGENCY_CALL-${CALLERID}-${TIM
ESTAMP})
exten = 000,2,Monitor(gsm,${CALLFILENAME},m)
exten = 000,3,Goto(emergency,s,1)
exten =
,1,SetVar(CALLFILENAME=/mnt/asterisk/EMERGENCY_CALL-${CALLERID}-${TI
MESTAMP})
exten = ,2,Monitor(gsm,${CALLFILENAME},m)
exten = ,3,Goto(emergency,s,1)
exten = _00011X.,1,AGI(blockintl.agi|${EXTEN:1})
exten = _01902X.,1,Hangup
exten =
_0X.,1,SetVar(CALLFILENAME=/mnt/asterisk/${CALLERID}-${EXTEN:1}-${TIMEST
AMP})
exten = _0X.,2,Monitor(gsm,${CALLFILENAME},m)
exten = _0X.,3,Dial(Zap/g1/${EXTEN:1})
exten = _0X.,4,Congestion
exten = _0X.,5,Hangup
include = phatphingers
[transfer-record]
exten =
_52XX,1,SetVar(CALLFILENAME=/mnt/asterisk/CallTo-${EXTEN:1}-${TIMESTAMP}
)
exten = _52XX,2,Monitor(gsm,${CALLFILENAME},m)
exten = _52XX,3,Dial(ZAP/g4/${EXTEN:1})
exten = _52XX,4,Congestion
exten = _52XX,104,Congestion
[voip]
exten = 589,1,Dial(IAX2/username:[EMAIL PROTECTED]/690)
exten = _2XX,1,Dial(Zap/g4/${EXTEN},60,r)
[parkedcalls]
exten = 590,1,playback(lm1/call_may_be_recorded)
exten =
590,2,ParkAndAnnounce(pbx-transfer:PARKED|7200|SIP/DNE|te405p-in,Zap/g4/
211,1)
[emergency]
exten = s,1,SetVar(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,SetGlobalVar(EMERGENCY=1)
exten = s,n,SetVar(SET_EMERG_FLAG=1)
exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten = s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress)
exten = s,n,SoftHangup(${EMERGENCY_TRUNK}-1)
exten = s,n,Wait(12)
exten = s,n,Goto(checkavail)
exten = s,s+2(inprogress),Congestion
exten = s,checkavail+101(notavail),Goto(trunkbusy)
exten = h,1,GotoIf($[${SET_EMERG_FLAG} = 1]?3)
exten = h,3,SetGlobalVar(EMERGENCY=0)
[phatphingers]
exten = _X.,1,answer
exten = _X.,2,wait(.5)
exten = _X.,3,playback(vm-extension)
exten = _X.,4,sayalpha(${EXTEN})
exten = _X.,5,playback(invalid)
exten = _X.,6,hangup
My extensions_sip.conf
[sip]
exten = 555,1,SetMusicOnHold(random)
exten = 555,2,Dial(ZAP/g4/211)
exten = 555,3,Voicemail,u555
exten = 555,103,Voicemail,b555
exten = 556,1,SetMusicOnHold(random)
exten = 556,2,Dial(SIP/js-softphone,30,Ttr)
exten = 556,3,Voicemail,u556
exten = 556,103,Voicemail,b556
exten = 557,1,SetMusicOnHold(random)
[Asterisk-Users] SIP call doesn't execute the 's'-extension
Hi, i have just started to configure access to the * over SIP-Phones. Therefore I have defined this SIP-Phone in sip.conf: [tobias] type=friend username=tobias secret=tobias auth=md5 host=dynamic reinvite=no dtmfmode=inband callerid=Tobias 1087006 allow=all context=javaAgi dtmfmode=rfc2833 As you can see i am directing calls from this user to the context [javaAgi] which is defined here in extension.conf: [javaAgi] exten = s,1,Answer() exten = s,2,Playback(code1000) exten = s,3,Hangup() exten = 1,1,Answer() exten = 1,2,Playback(code1000) exten = 1,3,Hangup() If i dial 1 on my SIP Phone everything works as suspected, the call is answered and the gsm-file is played. My understanding of the 's'-extension is, that it is executed then a call comes in an there is no extension wich matches the called number. But if i dial a random number i get an 404 Not found error. Here is an snippet of what * tells me on sip debug, but i can't get a clue out of it: 12 headers, 13 lines Using latest request as basis request Sending to 10.3.4.98 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 10.3.4.98:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x60e(GSM|ULAW|ALAW|SPEEX|ILBC)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user 'tobias' Looking for 2 in javaAgi Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.3.4.98:5060;branch=z9hG4bK102F7CD4855F4C4E927C3398E3C57BF4 From: Tobias sip:[EMAIL PROTECTED];tag=2760968676 To: sip:[EMAIL PROTECTED];tag=as396962de Call-ID: [EMAIL PROTECTED] CSeq: 58303 INVITE User-Agent: evision PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 Perhaps anyone can point me to the right direction ?? Tobias ___ 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] Dial more then 9 digits
my exten
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[default]
; If the number dialed by the calling party was 2000, then
; Dial the user 2000 via the SIP channel driver. Let the number
; ring for 20 seconds, and if no answer, proceed to priority 2.
; If the number gives a busy result, then jump to priority 102
;exten = s,1,Dial(SIP/${EXTEN})
;exten = s,1,Dial(SIP/7406100)
exten = 7406100,1,Dial(SIP/7406100)
exten = 7406101,1,Dial(H323/[EMAIL PROTECTED])
exten = 7406105,1,Dial(SIP/7406105)
exten = 7406106,1,Dial(SIP/7406106)
exten = 7406200,1,Dial(SIP/7406200)
exten = _74068XX,1,Dial(H323/[EMAIL PROTECTED])
exten = _OO.,1,Dial(H323/[EMAIL PROTECTED])
exten = _X,1,Dial(H323/[EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of altus
Sent: Wednesday, June 15, 2005 12:31 PM
To: asterisk
Subject: Re: [Asterisk-Users] Dial more then 9 digits
no
I can?
how is your dialout rules ?
I have a client where you have to dial a 4 digit pin and then the rest
of the number
I simply have a
exten = _1234.,1,Dail...
On Wed, 2005-06-15 at 12:20 +0200, Bohuslav Coufal wrote:
Could you kick me, I can't dial more then 9 digits. Is anyone some
default length of extensions or dialed number.
Thanks,
Bob.
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RE: [Asterisk-Users] Dial more then 9 digits
The
exten = _X,1,Dial(H323/[EMAIL PROTECTED])
sys any 9 digit number
try _X.,1
On Wed, 2005-06-15 at 13:23 +0200, Bohuslav Coufal wrote:
my exten
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[default]
; If the number dialed by the calling party was 2000, then
; Dial the user 2000 via the SIP channel driver. Let the number
; ring for 20 seconds, and if no answer, proceed to priority 2.
; If the number gives a busy result, then jump to priority 102
;exten = s,1,Dial(SIP/${EXTEN})
;exten = s,1,Dial(SIP/7406100)
exten = 7406100,1,Dial(SIP/7406100)
exten = 7406101,1,Dial(H323/[EMAIL PROTECTED])
exten = 7406105,1,Dial(SIP/7406105)
exten = 7406106,1,Dial(SIP/7406106)
exten = 7406200,1,Dial(SIP/7406200)
exten = _74068XX,1,Dial(H323/[EMAIL PROTECTED])
exten = _OO.,1,Dial(H323/[EMAIL PROTECTED])
exten = _X,1,Dial(H323/[EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of altus
Sent: Wednesday, June 15, 2005 12:31 PM
To: asterisk
Subject: Re: [Asterisk-Users] Dial more then 9 digits
no
I can?
how is your dialout rules ?
I have a client where you have to dial a 4 digit pin and then the rest
of the number
I simply have a
exten = _1234.,1,Dail...
On Wed, 2005-06-15 at 12:20 +0200, Bohuslav Coufal wrote:
Could you kick me, I can't dial more then 9 digits. Is anyone some
default length of extensions or dialed number.
Thanks,
Bob.
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Re: [Asterisk-Users] Dial more then 9 digits
On Wednesday 15 June 2005 12:40, altus wrote: exten = _OO.,1,Dial(H323/[EMAIL PROTECTED]) Sorry, I couldn't help but notice this... Is that really meant to be _OO (capital letter 'Oh') rather than _00 as the double-zero international prefix? Cheers, Gavin. ___ 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] Sipura SPA 3000 FXO Setting India
I have a Sipura SPA 3000 and am trying to use the PSTN port on the unit to bring a POTS line into Asterisk. However, I am unable to find the localization settings to get the SPA 3000 to understand the local phone settings. The unit is located in India and I have not had any success in trying to find the correct settings. Anyone have any ideas? Probably have a better chance of finding suggestions on the voxilla.com list. ___ 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] Dial more then 9 digits
Could you kick me, I can't dial more then 9 digits. Is anyone some default length of extensions or dialed number. That's likely the result of the dialplan within whatever phone that you're using. That dialplan was intended to be modified by you for whatever your needs happen to be. ___ 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] CDR's - ODBC and logging IP's
Hello.. I have configured asterisk to send CDR's to an ODBC datasource on IAX calls I can find the IP address of the caller in the 'channel' field For example: IAX2/username@ipaddr:4569-458 On SIP calls I never see the IP address of the caller For example: SIP/username-9d51 So on SIP calls there is not any possibility to log the ip adress of the caller? What can I do to enable logging of ip's? ___ 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] Dial more then 9 digits
This is double-zero international prefix. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Hamill Sent: Wednesday, June 15, 2005 1:46 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Dial more then 9 digits On Wednesday 15 June 2005 12:40, altus wrote: exten = _OO.,1,Dial(H323/[EMAIL PROTECTED]) Sorry, I couldn't help but notice this... Is that really meant to be _OO (capital letter 'Oh') rather than _00 as the double-zero international prefix? Cheers, Gavin. ___ 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 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] SIP call doesn't execute the 's'-extension
i have just started to configure access to the * over SIP-Phones. Therefore I have defined this SIP-Phone in sip.conf: [tobias] type=friend username=tobias secret=tobias auth=md5 host=dynamic reinvite=no dtmfmode=inband callerid=Tobias 1087006 allow=all context=javaAgi dtmfmode=rfc2833 As you can see i am directing calls from this user to the context [javaAgi] which is defined here in extension.conf: [javaAgi] exten = s,1,Answer() exten = s,2,Playback(code1000) exten = s,3,Hangup() exten = 1,1,Answer() exten = 1,2,Playback(code1000) exten = 1,3,Hangup() If i dial 1 on my SIP Phone everything works as suspected, the call is answered and the gsm-file is played. My understanding of the 's'-extension is, that it is executed then a call comes in an there is no extension wich matches the called number. But if i dial a random number i get an 404 Not found error. The s extension matches only when no digits are dialed. Dialing a 1 is a digit, so no match. Try playing around with exten=_.,1,Answer() and understand what the differences are. ___ 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] empty HDLC frame or bad CRC received
I notice to post, having the same problem (probably from the same person) I resently also switched from i4l to other (first tried capi but that didnt work out) Now I installed zaphfc and it works (I can call and I can receive calls) but from the moment I load zaptel (modprobe zaptel) I get al lot of empty HDLC frame or bad CRC received messages (and the never stop) I also noticed that sometimes asterisk doesnt pick up (doesnt notice there is being called) When I do have a call I frequently (every 2 3 seconds) hear a crack on both sides (only when connected to via isdn) I tryied chaning the signalling thing, didnt change anything, but I think it is not related as I get the messages whereter or not asterisk is running (they start appearing as soon as I load zaptel) Anybody any idea? Bart Seresia ___ 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] Anyone noticed Voipjet voice quality problems?
Couple of days. Apparently the new US carrier has some changes that needs to be made. On 6/14/05, Wiley Siler [EMAIL PROTECTED] wrote: Did they say when it would be corrected? W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Tuesday, June 14, 2005 9:22 AM To: Matt Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems? Caller ID is still not working to certain areas. This problem was confirmed by voipjet tech support in their last e-mail to me. On 6/13/05, Matt [EMAIL PROTECTED] wrote: I never noticed any problems.. so I can't comment :) hehe On 6/11/05, Pedro [EMAIL PROTECTED] wrote: Finally got a response from voipjet support and they say they have switched to a new provider for US termination. I have yet to test this out as I have not had a chance to build them back into our routes but will report my findings once I do. Anyone else notice any improvements? On 6/9/05, Moody [EMAIL PROTECTED] wrote: We have been having serious quality problems using the westcoast server - been using the East coast server with increased success but seeing some issues related to going cross continent. Voipjet, you listening? ___ 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 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 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 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] Nasty little incident ...
We have a te410p, with the following connections: span 1 connected to a 32 Channel EuroISDN span 2 connected to a card in a legacy pbx (Meridian) span 3 connected to a 10 Channel EuroISDN span 4 connected to a card in a legacy pbx (Meridian) We have no need for the meridian now, and decided to turn it off. I did not change the zaptel.conf settings, nor the zapata.conf settings. When the meridian was turned off, * would no longer allow any outbound or inbound calls through spans 1 and 3 (although these are connected to the pstn). When I turned the meridian back on - in a hurry I might add ;) (had no time to play with configurations) and restarted *, then everything was ok again ... Should I comment out span 2 and 4, run a ztcfg, unplug the cables in 2 and 4, and then turn off the meridian ? Julian. /* zaptel.conf */ span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 span=3,2,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78 span=4,0,0,ccs,hdb3,crc4 bchan=94-108,110-124 dchan=109 loadzone=uk defaultzone=uk ___ 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] Sipura SPA 3000 FXO Setting India
Thanks didn't find anything there, so I thought I'd try here On 6/15/05, Rich Adamson [EMAIL PROTECTED] wrote: I have a Sipura SPA 3000 and am trying to use the PSTN port on the unit to bring a POTS line into Asterisk. However, I am unable to find the localization settings to get the SPA 3000 to understand the local phone settings. The unit is located in India and I have not had any success in trying to find the correct settings. Anyone have any ideas? Probably have a better chance of finding suggestions on the voxilla.com list. ___ 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] asterisk security
Hello, I would like to have some advices about security, securing asterisk server Already : - configured asterisk to run as non-root user (http://www.voip-info.org/tiki-index.php?page=Asterisk+non-root) - fw config (http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules) Would like to know what are the things I have to be carefull with - prevent anyone to use my asterisk srv to call anywhere in the world, some alert to put in place ? - prevent to listen my conversation, or other one using my asterisk srv - other advices ??? thanks for help G ___ 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] Load problems
hello, from time to time my cpu load grows up to 100% and the system becomes unusable (calls get disconnected, quality is VERY poor, etc.) the only solution i found so far is to restart the asterisk service, but it's definitely not a way of solving the problem. My * version is Asterisk CVS-HEAD-05/18/05-00:15:59 i have two quad-span E1 cards in this system and it acts as a PBX for 7-to-1 isdn concentrator. thanks, Calin. ___ 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] newbie question..
the situation here is i want when user make outgoing
call, asterisk will call 1800XX first then after 3
or 4 sec asterisk will insert the number that user
want to call..
user don't know that the call is go to 1800XX
first..
means user just insert the number that they want to
call then asterisk will insert that number after 3 or
4 sec..
can i that in asterisk?
i'll apreciate any help or advise..
Might try something like this:
exten = _9XXX,1,Dial(Zap/4/1800XXw${EXTEN})
where each w adds a some delay.
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[Asterisk-Users] RE: Calling on all Polycom Experts
From: Ryan Stark [EMAIL PROTECTED] Subject: [Asterisk-Users] Calling on all Polycom Experts To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Hey all, I'll give my reseller a call for support in the morning, but I usually have better/faster luck on the list. I've got a SoundPoint IP500 that I upgraded to BootROM 2.6.2 and SIP image 1.5.2 on someone elses advice, I forgot to change out the old config for the new when I loaded the image up (I guess the config changed a bunch between 1.5.2 and 1.3.1) I was prompted with an error message: There was an error proccessing the config file, Error of type 0x4020. Then I used the config file that came with the new release to write a new config for that phone, rebooted, same error. I did the 468* reset and it did the same thing again. Any ideas on what that error is and how I fix it? (Polycom logs quoted bellow sig.) Thanks, -Ryan I wouldn't call myself an expert, but I don't see in the logs where the phone successfully requested the config files. We had the same problem when upgrading. It had to do with our FTP server's firewall. They changed the way the FTP stuff is done when requesting the phone's cfg file. Hope that helps get you on the right track. I didn't discover what the root problem was until I moved the FTP files to an un-firewalled box all together to see if the FTP server itself was whack. ___ 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] Sipura SPA 3000 FXO Setting India
The only other approach that I can think of is looking at the asterisk code for India definitions to see what the TDM card is using, tones, etc. Then find a match to those specs in the spa screens. Thanks didn't find anything there, so I thought I'd try here On 6/15/05, Rich Adamson [EMAIL PROTECTED] wrote: I have a Sipura SPA 3000 and am trying to use the PSTN port on the unit to bring a POTS line into Asterisk. However, I am unable to find the localization settings to get the SPA 3000 to understand the local phone settings. The unit is located in India and I have not had any success in trying to find the correct settings. Anyone have any ideas? Probably have a better chance of finding suggestions on the voxilla.com list. ___ 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] Port Inquiry
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have been learning asterisk for approx 3 months. On my asterisk server when I do a TCPDump I am getting messages like: 23:44:18.401093 voip.glcomputers.com vlan-157-game-40.comnet.bg: icmp: voip.glcomputers.com udp port 1026 unreachable [tos 0xc0] Most of the time it just gives me an ip address like 61.152.158.126, 61.152.158.101, 218.83.153.58, 147.135.12.6, 220.168.156.71 and either port 1026 or 1027. Anyone share information on this? Other Ports which are unreachable are 33473-33475, 334676, 33438, 33479 which from what I can tell is apart of my SIP Provider which is Broadvoice at the present time. Does anyone know how to change the NOAA Location upon the *61. When I dial it, I am getting weather for New York City and would like to get weather info for South Bend, IN - -- - -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) iD8DBQFCsCbGJYguHL5xYBARAiJ5AJ9L0dw7HPx+/2GnGO4uyKJLvN5sXACffLtE sp4x6Xaz16nz+XwG7fT/9lc= =6Smz -END PGP SIGNATURE- ___ 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] Nasty little incident ...
We have a te410p, with the following connections: span 1 connected to a 32 Channel EuroISDN span 2 connected to a card in a legacy pbx (Meridian) span 3 connected to a 10 Channel EuroISDN span 4 connected to a card in a legacy pbx (Meridian) We have no need for the meridian now, and decided to turn it off. I did not change the zaptel.conf settings, nor the zapata.conf settings. When the meridian was turned off, * would no longer allow any outbound or inbound calls through spans 1 and 3 (although these are connected to the pstn). When I turned the meridian back on - in a hurry I might add ;) (had no time to play with configurations) and restarted *, then everything was ok again ... Should I comment out span 2 and 4, run a ztcfg, unplug the cables in 2 and 4, and then turn off the meridian ? Julian. /* zaptel.conf */ span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 span=3,2,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78 span=4,0,0,ccs,hdb3,crc4 bchan=94-108,110-124 dchan=109 loadzone=uk defaultzone=uk Just a wild guess When the two meridian links disappeared, the channel numbers probably changed. Instead of channels 1 through 124, you probably have channels 1 through 62 and your supporting dialplan (and other channel specific items) likely don't match. ___ 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] Asterisk and Panasonic KX-TD1232
Amund, I can't speak for the Digium ISDN card, but in Europe the Eicon Diva Server card and AVM C4 server cards work well for connection to to exchange lines. For what you want to do you need ISDN cards that operate in NT and/or TE mode. Assuming that you are using all 4 x BRI (8 channels) in the 1232 then you need a 4xBRI card facing your Panasonic exmulating the exchange and a 4xBRI card facing the exchange. From recollection the AVM C4 cannot emulate the exchange, so you probably need theb Eicon Diva Server card: http://www.eicon.com/worldwide/products/MediaGateways/disv4bri.htm Mike - Original Message - From: Amund Nygaard [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, June 14, 2005 1:56 PM Subject: [Asterisk-Users] Asterisk and Panasonic KX-TD1232 Hello We have around 50 phones in our company, and I am playing with the thought to gradually go over to using sip services and ip-phones internally. However at first I would liked the Asterisk just to sit between the phone line and the Panaosnic, so I can take out one lin/number at a time to use ip phones. I am new to Asterisk, and haven't done much configuring of the PBX either. So I also wonder how difficult such setup is. We use today 4 BRI lines that connects us to the telephone network, would I then need 2xTE410P to put the Asterisk between the Panasonic and the phone network? BR Amund Nygaard IT-Manager A NOVO Norway AS ___ 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] asterisk security
I would like to have some advices about security, securing asterisk server Already : - configured asterisk to run as non-root user (http://www.voip-info.org/tiki-index.php?page=Asterisk+non-root) - fw config (http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules) Would like to know what are the things I have to be carefull with - prevent anyone to use my asterisk srv to call anywhere in the world, some alert to put in place ? - prevent to listen my conversation, or other one using my asterisk srv - other advices ??? Next thing I'd suggest is to use an external sip phone (or * system) to try to access your asterisk system without the appropriate userid and password entries (or use entries that don't match your current asterisk definitions. Same with iax if you're allowing that. Seems there are a fair number of people that think they understand asterisk, its use of contexts, etc, but really don't. If I were going to try and hack your asterisk system from a remote location, what would I try to do? Place calls through your system without you knowing it (amoung other things). Using port scanners (like nessus, nmap, etc) will only tell you what tcp/udp ports are open, but will not give you a clue whether your sip, iax, or other I/O channels are defined in a reasonably secure way. ___ 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] [Help] ZT_CHANCONFIG failed on channel 25
Hi, I a new user of asterisk, I'm trying to in install zaptel drivers on my ISDN card Digium Tiger 3xx TE110P. And my configuration is # # Zaptel Configuration File # span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = it ; ; ; Zapata Configuration file ; [channels] immediate=no switchtype=EuroISDN signalling=pri_cpe pridialplan=unknown context=incoming usecallerid=yes group=1 channel = 1-15,17-31 when I lunch the zaptel sevice I got this problem. Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules:Running ztcfg: ZT_CHANCONFIG failed on channel 25: No such device or address (6) [FAILED] I need help ___ 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] Making Asterisk NOT Pickup a Line when Ringing?
On Tuesday 14 June 2005 20:10, Rich Adamson wrote: The wait(s) isn't needed either; in this case it adds no value to the solution whatsoever other then making you think its waiting around. Actually they are necessary. So what do you think happens after the specified x seconds? Nothing, unless you have more statements. So why burn cpu cycles to calculate the end of the wait period, and _then_ do nothing? If there isn't a sufficiently (60s?) long wait in there then Asterisk will finish the dialplan and the next ring indication will make it start the dialplan over again. And again. And again. The Wait() prevents this. -A. ___ 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] Old but Gold
On Wednesday 15 June 2005 04:47, [EMAIL PROTECTED] wrote: Jun 15 08:45:20 NOTICE[10390]: app_dial.c:972 dial_exec_full: Unable to create channel of type 'Zap' (cause 0) == Everyone is busy/congested at this time (1:0/0/1) Execute zap show channels and make sure asterisk sees them. channel=1 ; on channel 1 callerid=01614830073 dylan The callerid line won't be part of channel 1's declaration since it's after the channel declaration but other than that it seems to look good. fxoks=1 # Make sure that the FXS(green) module is closest to the bracket if you are looking at the side of the card with all of $ fxsks=4 # FXO module defaultzone=uk loadzone=uk # end of dylan added OUTBOUND=Zap/4 I didn't see a channel = 4 in your zapata.conf. Confidentiality Notice: The information contained in this e-mail is for the intended recipient(s) alone. It may contain privileged and confidential information that is exempt from disclosure under English law and if you are not an intended recipient, you must not copy, distribute or take any action in reliance on it. If you have received this e-mail in error, please notify us immediately either by using the reply facility on your e-mail system or by contacting us at [EMAIL PROTECTED] . If this message is being transmitted over the Internet, be aware that it may be intercepted by third parties. Is this really necessary for a mailing list? I mean Christ it's even 10 lines long... -A. ___ 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] 488 Not Acceptable Here
The message is generated directly by the called Sipura/PAP2. No, if you read the sip debug carefully, you would see Asterisk is transmitting 488 Not Acceptable Here. If you mean the destination device, that's not possible since the user was calling an echo test. This message is very likely due to a codec issue (ie, the called unit was instructed to use G279 but it had already one call setup with G729), or the called unit was in the process of setting up a call and had no available G279 codec for the second call. ( the Sipuras/PAP2 reserve G729 during call setup even though it might end up using G711). The codec is only released once the call is set up. I know that, which is why the sip.conf entry is set to allow=g729 and allow=ulaw. Any other ideas? -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 FWD: 46990 ___ 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] Should I choose DSL @ 1.5 or a full T1?
On Tuesday 14 June 2005 18:47, Barton Fisher wrote: So if I understand correctly, a full T1 should be 1.5Mbps full duplex. And it should support 22 SIP Users at once - Right? Depends on the codec and VOIP technology used and what else is going out over the line.With the right technology and conditions, and with the right codec, you could easily fit over 130 conversations in that same pipe. But to keep the discussion short: yes. 22 is perhaps stretching it with ulaw but 18-19 is about the right answer. -A. ___ 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] Caller ID
Hi, First, you must ensure that the your Telco is sending you the caller ID with incoming calls. In some countries this is an aditional service you have to pay for, upon request. If you already have the service from the Telco, check in your zapata.conf that you have callerid=asreceived on your channels and group definitions. Hope this will help. Regards, Juan Manuel Coronado Z. On mar, 2005-06-14 at 14:50 +0200, Stojan Sljivic - GDS wrote: Hi, I'm using TDM04B and Asterisk 1.0.5. How can I setup the Asterisk so that I get caller ID? I do not get caller ID currently. Regards, Stojan Sljivic ___ 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 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] AVAYA Asteris H323 chanel
Yes. I configured it for a former employer. We had an S8700 talking to * via h.323 with no problems. oh323 did need to have it's rtp frame size adjusted initially for some sound quality issues, and we needed to dbl check that oh323 wasn't trying to negotiate for codecs that * didn't want to handle. Aside from that, it's been working flawlessly since. On 6/14/05, Bohuslav Coufal [EMAIL PROTECTED] wrote: I'm trying to make H.323 trunk between AVAYAAsterisk. But call from AVAYA is terminated inmediatelly when apps DIAL on Asterisk is started. Does any one use AVAYA and h.323 channel? Thanks Bob. ___ 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 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] Broadvoice and Inbound DTMF
I have Broadvoice set up with dtmfmode=inband. All was working just fine. Suddenly today I noticed that if someone calls in to my Asterisk box thru the Broadvoice number, the system no longer recognizes the DTMF tones. I also tried rfc2833 and info. Any ideas? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ 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] HT-488 vs. SPA-3000?
Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... I've not played with the ht488, but I believe others have posted this device does not provide access to the pstn-fxo port. The spa3k does provide that access (if you want it). Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. The spa3k works fine with asterisk as many have posted. However, once in awhile it does act a little strange in two different ways: 1. the spa3k will sometimes interpret some voices as tones which cause a little disturbance to any conversation going on. It is sort of like the old telephony talk off that existed years ago. Doesn't happen all that often and seems to be more sensitive to female voices based on my one-year of experience. 2. sometimes it seems to operate in half-duplex mode, where if you try to talk at the same time as the other end is talking, the other end won't hear you. Neither one of those have been all that objectionable to me, but they happen and others have posted roughly the same issues. I've not heard of anyone that has found a way to minimize those two issues. The down side of the spa3k right now is that Cisco bought the company and there likely won't be much advancement of the code until after the ownership (and development efforts) are sorted out by both companies. (The same kind of product delays has been seen with their Linksys purchase, as well as when other companies are bought/sold.) Its fairly common knowledge that ex-Cisco folks started Sipura for the sole purpose of selling the company for a hugh profit. Their success in accomplishing that objective could only be measured in terms of producing Sipura products that had at least some acceptance of those products by end users. With those previous objectives accomplished, how will Cisco handle the Sipura products in the future? (It's any- one's guess at this point since Cisco also has at least some track record of mismanaging purchased companies for whatever reason.) From an internal Cisco strategic perspective, they now own the assets that can make a major dent in the mass-market end-user voip product arena, and hopefully they'll take that in a positive direction. Given the price of the spa3k, I don't have any issue with purchasing more of them right now. Excellent choice for the one-to-three pstn-fxo market space. ___ 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: Call being answered, but no audio on either end
Thanks Gene. Here is my localnet: localnet=172.16.64.0/255.255.240.0 Which matches our subnets network address and subnet mask. Are you recommending that I make it more restrictive? Thanks, Geoff -Original Message- From: Gene Willingham [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 14, 2005 9:13 PM To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Subject: RE: Call being answered, but no audio on either end I think I found the source of this. Been tracing it for a week. Look in sip.conf. It appears the definition of localnet has a bearing on how some sip devices handle invites and NAT. I had changed the localnet to 192.168.3.0, but did not change the netmask. localnet=192.168.3.0/255.255.0.0; All RFC 1918 addresses are local networks When I changed the netmask to 255.255.255.0 the problem appeared to go away. It appears the more restrictive localnet the better results at handling sip devices behind NAT devices. Gene 19. Call being answered, but no audio on either end (Intermittent) (Geoff Manning) -- Message: 19 Date: Tue, 14 Jun 2005 17:30:31 -0400 From: Geoff Manning [EMAIL PROTECTED] Subject: [Asterisk-Users] Call being answered, but no audio on either end (Intermittent) To: Asterisk Users (E-mail) asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 The best type of error possible, intermittent. We have PSTN numbers being switched to SIP then forwarded to our Asterisk server which sits inside our LAN Every once and a while (maybe 1 out of every 20 calls) goes like this: -- Executing Answer(SIP/213.199.36.50-0818e3e8, ) in new stack -- Executing Ringing(SIP/213.199.36.50-0818e3e8, ) in new stack -- Executing Dial(SIP/213.199.36.50-0818e3e8, ZAP/g1/:8213) in new stack -- Called g1/:8213 -- Zap/1-1 answered SIP/213.199.36.50-0818e3e8 -- Hungup 'Zap/1-1' == Spawn extension (from-gv-uk, 441252580625, 3) exited non-zero on 'SIP/213.199.36.50-0818e3e8' Looks normal right? During this whole exchange, neither side can hear the other. Not even a ringing sound. The above looks no different than the successful calls. Has anyone seen this type of behavior before? Thanks! -- ___ 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] Making Asterisk NOT Pickup a Line when Ringing?
The wait(s) isn't needed either; in this case it adds no value to the solution whatsoever other then making you think its waiting around. Actually they are necessary. So what do you think happens after the specified x seconds? Nothing, unless you have more statements. So why burn cpu cycles to calculate the end of the wait period, and _then_ do nothing? If there isn't a sufficiently (60s?) long wait in there then Asterisk will finish the dialplan and the next ring indication will make it start the dialplan over again. And again. And again. Not on my cvs-head system that has been in use for over a year, and updated to current cvs-head on a regular basis. Used that same approach (until recently) for a fax pstn line. The analog fax machine always answered the incoming calls, and zapata.conf always pointed to a context that never did anything (not even a wait). We still used that same analog pstn line for outgoing calls. I think the key issue in this thread is the OP's wish to have some external analog device answer incoming calls. When that external device answers the incoming call, ringing disappears from asterisk's perspective and asterisk treats it just like an abandoned call. If the incoming call context doesn't have any statements to answer that call, its not going to do anything with it. (In other words, the call disappeared within a second or two after the first ring. Waiting around for another 10 to 60 seconds, or whatever, doesn't do anything useful.) ___ 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] Caller ID
Hi Juan, I have Caller Id service enabled. When I connect the line to the phone I see the caller Id on the phone's display. I have callerid=asreceived. I have also played with various combinations of cidsignalling and cidstart, but with no success. Regards, Stojan Sljivic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Manuel Coronado Z. Sent: Wednesday, June 15, 2005 15:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Caller ID Hi, First, you must ensure that the your Telco is sending you the caller ID with incoming calls. In some countries this is an aditional service you have to pay for, upon request. If you already have the service from the Telco, check in your zapata.conf that you have callerid=asreceived on your channels and group definitions. Hope this will help. Regards, Juan Manuel Coronado Z. On mar, 2005-06-14 at 14:50 +0200, Stojan Sljivic - GDS wrote: Hi, I'm using TDM04B and Asterisk 1.0.5. How can I setup the Asterisk so that I get caller ID? I do not get caller ID currently. Regards, Stojan Sljivic ___ 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 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 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] Broadvoice and Inbound DTMF
Nevermind. It is now working. Must be Broadvoice. Surprise! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: Wednesday, June 15, 2005 9:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Broadvoice and Inbound DTMF I have Broadvoice set up with dtmfmode=inband. All was working just fine. Suddenly today I noticed that if someone calls in to my Asterisk box thru the Broadvoice number, the system no longer recognizes the DTMF tones. I also tried rfc2833 and info. Any ideas? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ 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 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] Nasty little incident ...
Thanks for the help, comments inline: Rich Adamson wrote: We have a te410p, with the following connections: span 1 connected to a 32 Channel EuroISDN span 2 connected to a card in a legacy pbx (Meridian) span 3 connected to a 10 Channel EuroISDN span 4 connected to a card in a legacy pbx (Meridian) We have no need for the meridian now, and decided to turn it off. I did not change the zaptel.conf settings, nor the zapata.conf settings. When the meridian was turned off, * would no longer allow any outbound or inbound calls through spans 1 and 3 (although these are connected to the pstn). When I turned the meridian back on - in a hurry I might add ;) (had no time to play with configurations) and restarted *, then everything was ok again ... Should I comment out span 2 and 4, run a ztcfg, unplug the cables in 2 and 4, and then turn off the meridian ? Julian. /* zaptel.conf */ span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 span=3,2,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78 span=4,0,0,ccs,hdb3,crc4 bchan=94-108,110-124 dchan=109 loadzone=uk defaultzone=uk Just a wild guess When the two meridian links disappeared, the channel numbers probably changed. Instead of channels 1 through 124, you probably have channels 1 through 62 and your supporting dialplan (and other channel specific items) likely don't match. I thought that the definitions in the zaptel.conf and zapata.conf (see below) defined the channel numbers, not the physical channels themselves ? I use Dial(zap/g3) to call on the zap channels. /* zapata.conf */ context=isdn32-b prilocaldialplan=national internationalprefix = 00 nationalprefix = 0 localprefix = 01702 group=1 signalling=pri_cpe switchtype=euroisdn channel=1-15,17-31 context=meridian-b group=2 signalling=pri_net switchtype=euroisdn channel=32-46,48-62 context=isdn32-a pridialplan=unknown group=3 signalling=pri_cpe switchtype=euroisdn channel=63-77,79-93 context=meridian-a group=4 signalling=pri_net switchtype=euroisdn channel=94-108,110-124 ___ 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 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] AVAYA Asteris H323 chanel
Thanks, now it works. Problem was in CVS and libraries versions. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Wednesday, June 15, 2005 3:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AVAYA Asteris H323 chanel Yes. I configured it for a former employer. We had an S8700 talking to * via h.323 with no problems. oh323 did need to have it's rtp frame size adjusted initially for some sound quality issues, and we needed to dbl check that oh323 wasn't trying to negotiate for codecs that * didn't want to handle. Aside from that, it's been working flawlessly since. On 6/14/05, Bohuslav Coufal [EMAIL PROTECTED] wrote: I'm trying to make H.323 trunk between AVAYAAsterisk. But call from AVAYA is terminated inmediatelly when apps DIAL on Asterisk is started. Does any one use AVAYA and h.323 channel? Thanks Bob. ___ 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 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 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] SIP call doesn't execute the 's'-extension
Hi, Rich Adamson schrieb: The s extension matches only when no digits are dialed. Dialing a 1 is a digit, so no match. Oh, ok, i think now i understood. So the s extension is mainly the starting point for contexes which i reaches from other contexes, eg. because of a goto. When I receive a call there are naturally some digits dialed and with the pattern matching, you have suggested, i am able to react on them. Thanks a lot :- Tobias ___ 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] Making Asterisk NOT Pickup a Line when Ringing?
On Wednesday 15 June 2005 10:56, Rich Adamson wrote: I think the key issue in this thread is the OP's wish to have some external analog device answer incoming calls. When that external device answers the incoming call, ringing disappears from asterisk's perspective and asterisk treats it just like an abandoned call. If the incoming call context doesn't have any statements to answer that call, its not going to do anything with it. (In other words, the call disappeared within a second or two after the first ring. Waiting around for another 10 to 60 seconds, or whatever, doesn't do anything useful.) I realize he's waiting for something else to pick up, but I thought he was wanting to do something in addition to not answering (updating CDR, running a script of some kind) -- In that case I believe the Wait()'s necessary to prevent Asterisk from recording multiple CDR entries or running the script multiple times. You're absolutely right if you really do want to do nothing, just do that and it'll work just fine. :-) -A. ___ 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] Asterisk and Max TNT
Hello, I'm currently testing Asterisk over a T1 cross connect to a MaxTNT chassis that we have. It is working fine switching the calls through, but there is about a 10 second delay from the time Asterisk initiates the call until the TNT accepts it. It appears to be a ANI issue, I've changed several settings and formatting options on the T1 between the two, as well as turning on/off the callerid options in Zapata.conf, it's very strange. I'm pretty sure this is an interoperability issue between the two devices, I'm looking for a magic setting. The TNT doesn't have this problem via SIP. Regards Michael Baird ___ 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] user web interface
Hi, I am using asterisk strictly as a voicemail server. I know there are a number of web interfaces available-- I have looked at couple like AMP, etc... Is there one in particular that is generally considered better than the others? Is there one that is most feature rich with regard to managing voice mailboxes? Thank you for your input. Dan ___ 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] SIP call doesn't execute the 's'-extension
You only have a 1 in the javaAgi context and you aren't point the javaAgi to any other contexts, pressing anyting else but 1 will get a not found error because you only have 1 defined. If you want the call to continue you need to send it to another context or add more to the javaAgi context. Tobias Wolf wrote: Hi, i have just started to configure access to the * over SIP-Phones. Therefore I have defined this SIP-Phone in sip.conf: [tobias] type=friend username=tobias secret=tobias auth=md5 host=dynamic reinvite=no dtmfmode=inband callerid=Tobias 1087006 allow=all context=javaAgi dtmfmode=rfc2833 As you can see i am directing calls from this user to the context [javaAgi] which is defined here in extension.conf: [javaAgi] exten = s,1,Answer() exten = s,2,Playback(code1000) exten = s,3,Hangup() exten = 1,1,Answer() exten = 1,2,Playback(code1000) exten = 1,3,Hangup() If i dial 1 on my SIP Phone everything works as suspected, the call is answered and the gsm-file is played. My understanding of the 's'-extension is, that it is executed then a call comes in an there is no extension wich matches the called number. But if i dial a random number i get an 404 Not found error. Here is an snippet of what * tells me on sip debug, but i can't get a clue out of it: 12 headers, 13 lines Using latest request as basis request Sending to 10.3.4.98 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 10.3.4.98:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x60e(GSM|ULAW|ALAW|SPEEX|ILBC)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user 'tobias' Looking for 2 in javaAgi Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.3.4.98:5060;branch=z9hG4bK102F7CD4855F4C4E927C3398E3C57BF4 From: Tobias sip:[EMAIL PROTECTED];tag=2760968676 To: sip:[EMAIL PROTECTED];tag=as396962de Call-ID: [EMAIL PROTECTED] CSeq: 58303 INVITE User-Agent: evision PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 Perhaps anyone can point me to the right direction ?? Tobias ___ 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 -- - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] ___ 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] Nasty little incident ...
We have a te410p, with the following connections: span 1 connected to a 32 Channel EuroISDN span 2 connected to a card in a legacy pbx (Meridian) span 3 connected to a 10 Channel EuroISDN span 4 connected to a card in a legacy pbx (Meridian) We have no need for the meridian now, and decided to turn it off. I did not change the zaptel.conf settings, nor the zapata.conf settings. When the meridian was turned off, * would no longer allow any outbound or inbound calls through spans 1 and 3 (although these are connected to the pstn). When I turned the meridian back on - in a hurry I might add ;) (had no time to play with configurations) and restarted *, then everything was ok again ... Should I comment out span 2 and 4, run a ztcfg, unplug the cables in 2 and 4, and then turn off the meridian ? Julian. /* zaptel.conf */ span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 span=3,2,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78 span=4,0,0,ccs,hdb3,crc4 bchan=94-108,110-124 dchan=109 loadzone=uk defaultzone=uk Just a wild guess When the two meridian links disappeared, the channel numbers probably changed. Instead of channels 1 through 124, you probably have channels 1 through 62 and your supporting dialplan (and other channel specific items) likely don't match. I thought that the definitions in the zaptel.conf and zapata.conf (see below) defined the channel numbers, not the physical channels themselves ? I use Dial(zap/g3) to call on the zap channels. /* zapata.conf */ context=isdn32-b prilocaldialplan=national internationalprefix = 00 nationalprefix = 0 localprefix = 01702 group=1 signalling=pri_cpe switchtype=euroisdn channel=1-15,17-31 context=meridian-b group=2 signalling=pri_net switchtype=euroisdn channel=32-46,48-62 context=isdn32-a pridialplan=unknown group=3 signalling=pri_cpe switchtype=euroisdn channel=63-77,79-93 context=meridian-a group=4 signalling=pri_net switchtype=euroisdn channel=94-108,110-124 I'm sure there are others on this list that can add to this, but when the card drivers are loaded and ztfg run, the channels that are discovered have to be mapped to what's in zaptel.conf one way or another. (Moving card driver load around changes the discovered order and one must manually modify zaptel.conf to match.) Then each zap channel is defined in zapata.conf, and those definitions have to match the channel numbers resulting from the above zaptel.conf stuff. So, what happens when two E1s disappear? Do the avaiable channel numbers change at the zaptel.conf level? My best guess is they do, but I don't have E1s around to play with to prove it. So, that's my best guess and it certainly can be an incorrect guess on my part. ___ 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] Error installing Asterisk with zaptel and libpri
Hi all, I have a vergin Linux box (FC3) and Digium's TDM20B card installed in it. I followed the Digium's quick intallation guide. donwloded CVS successfully, installed in this order zaptellibpriAsterisk and got this error without installing asterisk in my box. * (this is the end of the screen dump) /usr/bin/ld: cannot find -lidn collect2: ld returned 1 exit status make[1]: *** [app_curl.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk]# * I repeated the installation by cleaning the previous installations 2,3 times and got the same error. I saw sometimes my TDM20B card's LEDs were lit. (I was happy). but now it is off all the time and I have another problem there, i.e, after any reboot I have to restart the netwrok manually everytime. it is not automatically finding and setting up the network.? please help me kumara ___ 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] Handling -1 in dialplans
Hi,
How do you handle the case where a module returns -1 ?
eg consider this:
exten = 123,1,Answer
exten = 123,n,Playback(some-message)
exten = 123,n,etc ...
exten = 123,n,etc ...
exten = 123,n,etc ...
exten = 123,n,Command(${SOME_PARAMETER})
Now what if command returns -1 here ? I would like to branch
accordingly. Also how do you handle jumping to n+101 here - you don't
know what n is ?
Thanks,
Thomas
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Re: [Asterisk-Users] SIP call doesn't execute the 's'-extension
The s extension matches only when no digits are dialed. Dialing a 1 is a digit, so no match. Oh, ok, i think now i understood. So the s extension is mainly the starting point for contexes which i reaches from other contexes, eg. because of a goto. When I receive a call there are naturally some digits dialed and with the pattern matching, you have suggested, i am able to react on them. In the most general case, the exten=s is for incoming analog pstn lines (fxo ports) where the central office sends a call to your asterisk box by ringing the line. There are no digits sent to asterisk at all, therefore exten=s is used to handle that incoming call. Or, if you have an account with an itsp that sends incoming calls to your asterisk via iax/sip and doesn't send any digits to you, then exten=s is used for those as well. It really has nothing to do with 'other' contexts. ___ 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] app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)
No Ideas? This seems like quite a common issue but I have searched and
searched for a solution and not found any?
Cheers.
Sandy.
Sandy Thomson wrote:
Hi,
Ive been struggling with asterisk for a few days now. I understand
pretty much how it works and how to tie things together (SIP - SIP
internally works fine etc), but just cannot get asterisk to work with an
X100P clone (its a Ambient MD3200, if that means anything to you guys).
I have tried (initially) asterisk 1.07 with zaptel 1.07, and now
Asterisk CVS-HEAD with zaptel cvs. Both give me the same error.
/etc/zaptel.conf
--
fxsks=1
loadzone=uk
defaultzone=uk
/etc/asterisk/zapata.conf
--
[channels]
language=en
context=incoming
signalling=fxs_ks
channel = 1
/etc/asterisk/extensions.conf
--
[general]
static=yes
writeprotect=no
[local]
exten = _11.,1,Dial(Zap/1,9w${EXTEN}:2)
[incoming]
exten = s,1,Answer()
exten = s,2,BackGround(demo-congrats) ; Play a congratulatory message
[outgoing]
exten = _9.,1,Ringing
exten = _9.,2,Wait,2
exten = _9.,3,Answer()
exten = _9.,4,Dial(ZAP/1/9w${EXTEN}:1)
[default]
include = outgoing
Loading zaptel modules:
--
asterisk zaptel # modprobe zaptel
asterisk zaptel # modprobe wcfxo
asterisk zaptel # lsmod
Module Size Used by
wcfxo 12576 0
zaptel222916 1 wcfxo
crc_ccitt 1952 1 zaptel
asterisk zaptel # ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
Running asterisk
--
asterisk zaptel # asterisk -gc
Registering builtin applications:
[AbsoluteTimeout]
== Registered application 'AbsoluteTimeout'
[Answer]
== Registered application 'Answer'
[BackGround]
== Registered application 'BackGround'
[Busy]
== Registered application 'Busy'
[Congestion]
== Registered application 'Congestion'
[DigitTimeout]
== Registered application 'DigitTimeout'
[Goto]
== Registered application 'Goto'
[GotoIf]
== Registered application 'GotoIf'
[GotoIfTime]
== Registered application 'GotoIfTime'
[ExecIfTime]
== Registered application 'ExecIfTime'
[Hangup]
== Registered application 'Hangup'
[NoOp]
== Registered application 'NoOp'
[Prefix]
== Registered application 'Prefix'
[Progress]
== Registered application 'Progress'
[ResetCDR]
== Registered application 'ResetCDR'
[ResponseTimeout]
== Registered application 'ResponseTimeout'
[Ringing]
== Registered application 'Ringing'
[SayNumber]
== Registered application 'SayNumber'
[SayDigits]
== Registered application 'SayDigits'
[SayAlpha]
== Registered application 'SayAlpha'
[SayPhonetic]
== Registered application 'SayPhonetic'
[SetAccount]
== Registered application 'SetAccount'
[SetAMAFlags]
== Registered application 'SetAMAFlags'
[SetGlobalVar]
== Registered application 'SetGlobalVar'
[SetLanguage]
== Registered application 'SetLanguage'
[Set]
== Registered application 'Set'
[SetVar]
== Registered application 'SetVar'
[ImportVar]
== Registered application 'ImportVar'
[StripMSD]
== Registered application 'StripMSD'
[Suffix]
== Registered application 'Suffix'
[Wait]
== Registered application 'Wait'
[WaitExten]
== Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
== Parsing '/etc/asterisk/modules.conf': Found
[res_features.so] = (Call Parking Resource)
== Parsing '/etc/asterisk/features.conf': Found
-- Registered extension context 'parkedcalls'
-- Added extension '700' priority 1 to parkedcalls
== Registered application 'ParkedCall'
== Registered application 'Park'
== Manager registered action ParkedCalls
[res_musiconhold.so] = (Music On Hold Resource)
== Registered application 'MusicOnHold'
== Registered application 'WaitMusicOnHold'
== Registered application 'SetMusicOnHold'
== Registered application 'StartMusicOnHold'
== Registered application 'StopMusicOnHold'
== Parsing '/etc/asterisk/musiconhold.conf': Found
[chan_zap.so] = (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1, FXS Kewlstart signalling
-- Automatically generated pseudo channel
== Registered channel type 'Zap' (Zapata Telephony Driver)
== Manager registered action ZapTransfer
== Manager registered action ZapHangup
== Manager registered action ZapDialOffhook
== Manager registered action ZapDNDon
== Manager registered action ZapDNDoff
== Manager registered action ZapShowChannels
[app_sayunixtime.so] = (Say time)
== Registered application 'SayUnixTime'
== Registered application 'DateTime'
[res_adsi.so] = (ADSI Resource)
== Parsing '/etc/asterisk/adsi.conf': Found
[res_crypto.so] = (Cryptographic Digital Signatures)
-- Loaded PUBLIC key 'iaxtel'
-- Loaded PUBLIC key 'freeworlddialup'
[res_indications.so] = (Indications Configuration)
== Parsing '/etc/asterisk/indications.conf': Found
-- Registered indication country 'cl'
--
Re: [Asterisk-Users] [Help] ZT_CHANCONFIG failed on channel 25
I got the same error ona TDM04B... Comment out this line on zaptel/zconfig.h and recompile zaptel. /* * Uncomment if you happen have an early TDM400P Rev H which * sometimes forgets its PCI ID to have wcfxs match essentially all * subvendor ID's */ /* #define TDM_REVH_MATCHALL */ Hope it helps. Denis Galvão AsteriskBrasil.org On 15 de jun de 2005, at 10:17, Yousef Herzallah wrote: Hi, I a new user of asterisk, I'm trying to in install zaptel drivers on my ISDN card Digium Tiger 3xx TE110P. And my configuration is # # Zaptel Configuration File # span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = it ; ; ; Zapata Configuration file ; [channels] immediate=no switchtype=EuroISDN signalling=pri_cpe pridialplan=unknown context=incoming usecallerid=yes group=1 channel = 1-15,17-31 when I lunch the zaptel sevice I got this problem. Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules:Running ztcfg: ZT_CHANCONFIG failed on channel 25: No such device or address (6) [FAILED] I need help ___ 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 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] HT-488 vs. SPA-3000?
I have only had experience with the Sipura 3000 and I would agree with the voice volume problems. I have given up on it working properly (adjusted gains, impedences, firmware, etc), the voice quality is just to low to actually use. I actually purchased a second one thinking that the first might be defective. Would not recommend it because of the low sound volume problem. Talking on the phone is actually the point of the device so who cares how configurable it is if you cannot hear anything. I purchased a Digium TDM400P and have had very good luck with it. Dan On 6/15/05, Rich Adamson [EMAIL PROTECTED] wrote: Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... I've not played with the ht488, but I believe others have posted this device does not provide access to the pstn-fxo port. The spa3k does provide that access (if you want it). Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. The spa3k works fine with asterisk as many have posted. However, once in awhile it does act a little strange in two different ways: 1. the spa3k will sometimes interpret some voices as tones which cause a little disturbance to any conversation going on. It is sort of like the old telephony talk off that existed years ago. Doesn't happen all that often and seems to be more sensitive to female voices based on my one-year of experience. 2. sometimes it seems to operate in half-duplex mode, where if you try to talk at the same time as the other end is talking, the other end won't hear you. Neither one of those have been all that objectionable to me, but they happen and others have posted roughly the same issues. I've not heard of anyone that has found a way to minimize those two issues. The down side of the spa3k right now is that Cisco bought the company and there likely won't be much advancement of the code until after the ownership (and development efforts) are sorted out by both companies. (The same kind of product delays has been seen with their Linksys purchase, as well as when other companies are bought/sold.) Its fairly common knowledge that ex-Cisco folks started Sipura for the sole purpose of selling the company for a hugh profit. Their success in accomplishing that objective could only be measured in terms of producing Sipura products that had at least some acceptance of those products by end users. With those previous objectives accomplished, how will Cisco handle the Sipura products in the future? (It's any- one's guess at this point since Cisco also has at least some track record of mismanaging purchased companies for whatever reason.) From an internal Cisco strategic perspective, they now own the assets that can make a major dent in the mass-market end-user voip product arena, and hopefully they'll take that in a positive direction. Given the price of the spa3k, I don't have any issue with purchasing more of them right now. Excellent choice for the one-to-three pstn-fxo market space. ___ 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 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] Asterisk SMS configuration
Tony, I'm havin a similar issue i'm in the UK using x100p with the patch for CID and get the following. Any ideas Executing Goto(Zap/1-1, sms-in|s|1) in new stack -- Goto (sms-in,s,1) -- Executing SMS(Zap/1-1, default|a) in new stack -- SMS TX 93 00 6D -- SMS RX 93 00 6D -- SMS TX 94 00 6C -- SMS RX 94 00 6C -- Executing Wait(Zap/1-1, 1) in new stack -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (sms-in, s, 3) exited non-zero on 'Zap/1-1' Jon Tony Hoyle wrote: Tony Hoyle wrote: response). Has anyone got this working in the UK? Do I have to set a country specific setting? OK I got it working... there's a timeout in app_sms.c that just isn't long enough for the BT implementation - the app gives up long before the message centre has had time to respond. I got the fix from http://projects.codefidence.com/asterisk.html eventually (specifically http://projects.codefidence.com/src/sms-il.diff). The important bit is the pause: --- app_sms.c 21 Jan 2005 07:06:24 - 1.17 +++ app_sms.c 1 Apr 2005 00:15:29 - @@ -1240,7 +1240,7 @@ h-obyte = 1; h-opause = 200; if (h-omsg[0] == 0x93) - h-opause = 2400; /* initial message delay 300ms (for BT) */ + h-opause = 6000; /* initial message delay 300ms (for BT) */ h-obytep = 0; h-obitp = 0; h-osync = 80; Tony ___ 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 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] asterisk gsm gateway hardware recommendation?
Hello, I would like to implement a home GSM gateway using asterisk. What would you recommend me as a low-cost hardware for creating a gsm channel? I found 2n gsm gateway, that supports sip and chan_blue for bluetooth connections. Any recommendations? Basically, I want to end calls to some GSM number in my sip telephone and for some prefixes dial out using that same sip telephone. Also sending and receiving SMS will be a plus. I have a friend living in luxembourg, which would like a slovak phone number to communicate with friends. It would end on my server at home and all calls to his sim card will be routed to his ip telephone in luxembourg (and vice versa). Support for more than one sim card is a plus. Since it's a home/hobby use, I would prefer a low-cost solution. Any ideas (may be off-list) are welcome). Thanks, Juraj. ___ 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] app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)
I think you have a bad ZAP dialplan.
You have for instance: exten = _11.,1,Dial(Zap/1,9w${EXTEN}:2)
when you should have something like:
exten = _11.,1,Dial(Zap/1-1,9w${EXTEN:2})
Hope this help.
Sebas
Sandy Thomson wrote:
No Ideas? This seems like quite a common issue but I have searched and
searched for a solution and not found any?
Cheers.
Sandy.
Sandy Thomson wrote:
Hi,
Ive been struggling with asterisk for a few days now. I understand
pretty much how it works and how to tie things together (SIP - SIP
internally works fine etc), but just cannot get asterisk to work with an
X100P clone (its a Ambient MD3200, if that means anything to you guys).
I have tried (initially) asterisk 1.07 with zaptel 1.07, and now
Asterisk CVS-HEAD with zaptel cvs. Both give me the same error.
/etc/zaptel.conf
--
fxsks=1
loadzone=uk
defaultzone=uk
/etc/asterisk/zapata.conf
--
[channels]
language=en
context=incoming
signalling=fxs_ks
channel = 1
/etc/asterisk/extensions.conf
--
[general]
static=yes
writeprotect=no
[local]
exten = _11.,1,Dial(Zap/1,9w${EXTEN}:2)
[incoming]
exten = s,1,Answer()
exten = s,2,BackGround(demo-congrats) ; Play a congratulatory message
[outgoing]
exten = _9.,1,Ringing
exten = _9.,2,Wait,2
exten = _9.,3,Answer()
exten = _9.,4,Dial(ZAP/1/9w${EXTEN}:1)
[default]
include = outgoing
Loading zaptel modules:
--
asterisk zaptel # modprobe zaptel
asterisk zaptel # modprobe wcfxo
asterisk zaptel # lsmod
Module Size Used by
wcfxo 12576 0
zaptel222916 1 wcfxo
crc_ccitt 1952 1 zaptel
asterisk zaptel # ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
Running asterisk
--
asterisk zaptel # asterisk -gc
Registering builtin applications:
[AbsoluteTimeout]
== Registered application 'AbsoluteTimeout'
[Answer]
== Registered application 'Answer'
[BackGround]
== Registered application 'BackGround'
[Busy]
== Registered application 'Busy'
[Congestion]
== Registered application 'Congestion'
[DigitTimeout]
== Registered application 'DigitTimeout'
[Goto]
== Registered application 'Goto'
[GotoIf]
== Registered application 'GotoIf'
[GotoIfTime]
== Registered application 'GotoIfTime'
[ExecIfTime]
== Registered application 'ExecIfTime'
[Hangup]
== Registered application 'Hangup'
[NoOp]
== Registered application 'NoOp'
[Prefix]
== Registered application 'Prefix'
[Progress]
== Registered application 'Progress'
[ResetCDR]
== Registered application 'ResetCDR'
[ResponseTimeout]
== Registered application 'ResponseTimeout'
[Ringing]
== Registered application 'Ringing'
[SayNumber]
== Registered application 'SayNumber'
[SayDigits]
== Registered application 'SayDigits'
[SayAlpha]
== Registered application 'SayAlpha'
[SayPhonetic]
== Registered application 'SayPhonetic'
[SetAccount]
== Registered application 'SetAccount'
[SetAMAFlags]
== Registered application 'SetAMAFlags'
[SetGlobalVar]
== Registered application 'SetGlobalVar'
[SetLanguage]
== Registered application 'SetLanguage'
[Set]
== Registered application 'Set'
[SetVar]
== Registered application 'SetVar'
[ImportVar]
== Registered application 'ImportVar'
[StripMSD]
== Registered application 'StripMSD'
[Suffix]
== Registered application 'Suffix'
[Wait]
== Registered application 'Wait'
[WaitExten]
== Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
== Parsing '/etc/asterisk/modules.conf': Found
[res_features.so] = (Call Parking Resource)
== Parsing '/etc/asterisk/features.conf': Found
-- Registered extension context 'parkedcalls'
-- Added extension '700' priority 1 to parkedcalls
== Registered application 'ParkedCall'
== Registered application 'Park'
== Manager registered action ParkedCalls
[res_musiconhold.so] = (Music On Hold Resource)
== Registered application 'MusicOnHold'
== Registered application 'WaitMusicOnHold'
== Registered application 'SetMusicOnHold'
== Registered application 'StartMusicOnHold'
== Registered application 'StopMusicOnHold'
== Parsing '/etc/asterisk/musiconhold.conf': Found
[chan_zap.so] = (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1, FXS Kewlstart signalling
-- Automatically generated pseudo channel
== Registered channel type 'Zap' (Zapata Telephony Driver)
== Manager registered action ZapTransfer
== Manager registered action ZapHangup
== Manager registered action ZapDialOffhook
== Manager registered action ZapDNDon
== Manager registered action ZapDNDoff
== Manager registered action ZapShowChannels
[app_sayunixtime.so] = (Say time)
== Registered application 'SayUnixTime'
== Registered application 'DateTime'
[res_adsi.so] = (ADSI Resource)
== Parsing '/etc/asterisk/adsi.conf': Found
[res_crypto.so] = (Cryptographic Digital Signatures)
-- Loaded PUBLIC key 'iaxtel'
-- Loaded PUBLIC key 'freeworlddialup'
[res_indications.so] = (Indications Configuration)
== Parsing
Re: [Asterisk-Users] app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)
I don't have clone card to verify this, but I think you'll find the
chipset on that particular card is not the same chipset used on the
digium card. Since the asterisk drivers are written for specific
chipsets, I'd have to suggest you've got an almost zero chance of
making the clone work.
No Ideas? This seems like quite a common issue but I have searched and
searched for a solution and not found any?
Cheers.
Sandy.
Sandy Thomson wrote:
Hi,
Ive been struggling with asterisk for a few days now. I understand
pretty much how it works and how to tie things together (SIP - SIP
internally works fine etc), but just cannot get asterisk to work with an
X100P clone (its a Ambient MD3200, if that means anything to you guys).
I have tried (initially) asterisk 1.07 with zaptel 1.07, and now
Asterisk CVS-HEAD with zaptel cvs. Both give me the same error.
/etc/zaptel.conf
--
fxsks=1
loadzone=uk
defaultzone=uk
/etc/asterisk/zapata.conf
--
[channels]
language=en
context=incoming
signalling=fxs_ks
channel = 1
/etc/asterisk/extensions.conf
--
[general]
static=yes
writeprotect=no
[local]
exten = _11.,1,Dial(Zap/1,9w${EXTEN}:2)
[incoming]
exten = s,1,Answer()
exten = s,2,BackGround(demo-congrats) ; Play a congratulatory message
[outgoing]
exten = _9.,1,Ringing
exten = _9.,2,Wait,2
exten = _9.,3,Answer()
exten = _9.,4,Dial(ZAP/1/9w${EXTEN}:1)
[default]
include = outgoing
Loading zaptel modules:
--
asterisk zaptel # modprobe zaptel
asterisk zaptel # modprobe wcfxo
asterisk zaptel # lsmod
Module Size Used by
wcfxo 12576 0
zaptel222916 1 wcfxo
crc_ccitt 1952 1 zaptel
asterisk zaptel # ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
Running asterisk
--
asterisk zaptel # asterisk -gc
Registering builtin applications:
[AbsoluteTimeout]
== Registered application 'AbsoluteTimeout'
[Answer]
== Registered application 'Answer'
[BackGround]
== Registered application 'BackGround'
[Busy]
== Registered application 'Busy'
[Congestion]
== Registered application 'Congestion'
[DigitTimeout]
== Registered application 'DigitTimeout'
[Goto]
== Registered application 'Goto'
[GotoIf]
== Registered application 'GotoIf'
[GotoIfTime]
== Registered application 'GotoIfTime'
[ExecIfTime]
== Registered application 'ExecIfTime'
[Hangup]
== Registered application 'Hangup'
[NoOp]
== Registered application 'NoOp'
[Prefix]
== Registered application 'Prefix'
[Progress]
== Registered application 'Progress'
[ResetCDR]
== Registered application 'ResetCDR'
[ResponseTimeout]
== Registered application 'ResponseTimeout'
[Ringing]
== Registered application 'Ringing'
[SayNumber]
== Registered application 'SayNumber'
[SayDigits]
== Registered application 'SayDigits'
[SayAlpha]
== Registered application 'SayAlpha'
[SayPhonetic]
== Registered application 'SayPhonetic'
[SetAccount]
== Registered application 'SetAccount'
[SetAMAFlags]
== Registered application 'SetAMAFlags'
[SetGlobalVar]
== Registered application 'SetGlobalVar'
[SetLanguage]
== Registered application 'SetLanguage'
[Set]
== Registered application 'Set'
[SetVar]
== Registered application 'SetVar'
[ImportVar]
== Registered application 'ImportVar'
[StripMSD]
== Registered application 'StripMSD'
[Suffix]
== Registered application 'Suffix'
[Wait]
== Registered application 'Wait'
[WaitExten]
== Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
== Parsing '/etc/asterisk/modules.conf': Found
[res_features.so] = (Call Parking Resource)
== Parsing '/etc/asterisk/features.conf': Found
-- Registered extension context 'parkedcalls'
-- Added extension '700' priority 1 to parkedcalls
== Registered application 'ParkedCall'
== Registered application 'Park'
== Manager registered action ParkedCalls
[res_musiconhold.so] = (Music On Hold Resource)
== Registered application 'MusicOnHold'
== Registered application 'WaitMusicOnHold'
== Registered application 'SetMusicOnHold'
== Registered application 'StartMusicOnHold'
== Registered application 'StopMusicOnHold'
== Parsing '/etc/asterisk/musiconhold.conf': Found
[chan_zap.so] = (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1, FXS Kewlstart signalling
-- Automatically generated pseudo channel
== Registered channel type 'Zap' (Zapata Telephony Driver)
== Manager registered action ZapTransfer
== Manager registered action ZapHangup
== Manager registered action ZapDialOffhook
== Manager registered action ZapDNDon
== Manager registered action ZapDNDoff
== Manager registered action
Re: [Asterisk-Users] Asterisk SMS configuration
Jon Creasey wrote: Tony, I'm havin a similar issue i'm in the UK using x100p with the patch for CID and get the following. Any ideas Did you change line as mentioned earlier? Without that patch incoming SMS won't work at all. You might need to use an even higher pause... might be worth experimenting. Tony ___ 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] Port Inquiry
Graham, this question would be better placed on the [EMAIL PROTECTED] user list but to answer your question, you need to ftp to the weather service and find out your city/state code and insert it into the appropriate place in the script so it will download the correct text file. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Graham Pearson Sent: Wednesday, 15 June 2005 9:02 AM To: Asterisk Users Mailing List Subject: [Asterisk-Users] Port Inquiry -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have been learning asterisk for approx 3 months. On my asterisk server when I do a TCPDump I am getting messages like: 23:44:18.401093 voip.glcomputers.com vlan-157-game-40.comnet.bg: icmp: voip.glcomputers.com udp port 1026 unreachable [tos 0xc0] Most of the time it just gives me an ip address like 61.152.158.126, 61.152.158.101, 218.83.153.58, 147.135.12.6, 220.168.156.71 and either port 1026 or 1027. Anyone share information on this? Other Ports which are unreachable are 33473-33475, 334676, 33438, 33479 which from what I can tell is apart of my SIP Provider which is Broadvoice at the present time. Does anyone know how to change the NOAA Location upon the *61. When I dial it, I am getting weather for New York City and would like to get weather info for South Bend, IN - -- - -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) iD8DBQFCsCbGJYguHL5xYBARAiJ5AJ9L0dw7HPx+/2GnGO4uyKJLvN5sXACffLtE sp4x6Xaz16nz+XwG7fT/9lc= =6Smz -END PGP SIGNATURE- ___ 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 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] HT-488 vs. SPA-3000?
On Wed, 15 Jun 2005 10:55:09 -0500, Dan Littlejohn wrote: I have only had experience with the Sipura 3000 and I would agree with the voice volume problems. I have given up on it working properly (adjusted gains, impedences, firmware, etc), the voice quality is just to low to actually use. I actually purchased a second one thinking that the first might be defective. Would not recommend it because of the low sound volume problem. Talking on the phone is actually the point of the device so who cares how configurable it is if you cannot hear anything. I purchased a Digium TDM400P and have had very good luck with it. Dan I had exactly the same experience with the SPA-3000. Too bad too since it's nice device...if it were 6 db hotter. I also installed a TDM-400, which was better in a lot of ways but not perfect. When I rebuild my server I ended up simply call forwarding my POTS lines to a DID provided by an ITSP. This has been the best as far as quality is concerned. If my DSL line goes down I simply defeat the call forwarding on the main line and answer an analog phone for a while, or call forward to me cell. Michael On 6/15/05, Rich Adamson [EMAIL PROTECTED] wrote: Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... I've not played with the ht488, but I believe others have posted this device does not provide access to the pstn-fxo port. The spa3k does provide that access (if you want it). Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. The spa3k works fine with asterisk as many have posted. However, once in awhile it does act a little strange in two different ways: 1. the spa3k will sometimes interpret some voices as tones which cause a little disturbance to any conversation going on. It is sort of like the old telephony talk off that existed years ago. Doesn't happen all that often and seems to be more sensitive to female voices based on my one-year of experience. 2. sometimes it seems to operate in half-duplex mode, where if you try to talk at the same time as the other end is talking, the other end won't hear you. Neither one of those have been all that objectionable to me, but they happen and others have posted roughly the same issues. I've not heard of anyone that has found a way to minimize those two issues. The down side of the spa3k right now is that Cisco bought the company and there likely won't be much advancement of the code until after the ownership (and development efforts) are sorted out by both companies. (The same kind of product delays has been seen with their Linksys purchase, as well as when other companies are bought/sold.) Its fairly common knowledge that ex-Cisco folks started Sipura for the sole purpose of selling the company for a hugh profit. Their success in accomplishing that objective could only be measured in terms of producing Sipura products that had at least some acceptance of those products by end users. With those previous objectives accomplished, how will Cisco handle the Sipura products in the future? (It's any- one's guess at this point since Cisco also has at least some track record of mismanaging purchased companies for whatever reason.) From an internal Cisco strategic perspective, they now own the assets that can make a major dent in the mass-market end-user voip product arena, and hopefully they'll take that in a positive direction. Given the price of the spa3k, I don't have any issue with purchasing more of them right now. Excellent choice for the one-to-three pstn-fxo market space. ___ 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 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] HT-488 vs. SPA-3000?
We have 6 SPA3000s. The device is extremely configurable and works inbound/outbound with Asterisk with the latest firmware update with little trouble. However, we've yet to resolve sound volume and quality issues. The PSTN to SPA gain and SPA to PSTN gain along with FXS Port Input Gain and Output Gain settings have had no positive effect. The problem is entirely with the analog line adapter. VoIP calls from the analog phone to other VoIP destinations are perfect. We also have several SPA-1001s and SPA-2000s that have been running perfect since day 1. Also Sipura support is nonexistant. Just our experience. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dan Littlejohn Sent: Wednesday, June 15, 2005 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] HT-488 vs. SPA-3000? I have only had experience with the Sipura 3000 and I would agree with the voice volume problems. I have given up on it working properly (adjusted gains, impedences, firmware, etc), the voice quality is just to low to actually use. I actually purchased a second one thinking that the first might be defective. Would not recommend it because of the low sound volume problem. Talking on the phone is actually the point of the device so who cares how configurable it is if you cannot hear anything. I purchased a Digium TDM400P and have had very good luck with it. Dan On 6/15/05, Rich Adamson [EMAIL PROTECTED] wrote: Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... I've not played with the ht488, but I believe others have posted this device does not provide access to the pstn-fxo port. The spa3k does provide that access (if you want it). Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. The spa3k works fine with asterisk as many have posted. However, once in awhile it does act a little strange in two different ways: 1. the spa3k will sometimes interpret some voices as tones which cause a little disturbance to any conversation going on. It is sort of like the old telephony talk off that existed years ago. Doesn't happen all that often and seems to be more sensitive to female voices based on my one-year of experience. 2. sometimes it seems to operate in half-duplex mode, where if you try to talk at the same time as the other end is talking, the other end won't hear you. Neither one of those have been all that objectionable to me, but they happen and others have posted roughly the same issues. I've not heard of anyone that has found a way to minimize those two issues. The down side of the spa3k right now is that Cisco bought the company and there likely won't be much advancement of the code until after the ownership (and development efforts) are sorted out by both companies. (The same kind of product delays has been seen with their Linksys purchase, as well as when other companies are bought/sold.) Its fairly common knowledge that ex-Cisco folks started Sipura for the sole purpose of selling the company for a hugh profit. Their success in accomplishing that objective could only be measured in terms of producing Sipura products that had at least some acceptance of those products by end users. With those previous objectives accomplished, how will Cisco handle the Sipura products in the future? (It's any- one's guess at this point since Cisco also has at least some track record of mismanaging purchased companies for whatever reason.) From an internal Cisco strategic perspective, they now own the assets that can make a major dent in the mass-market end-user voip product arena, and hopefully they'll take that in a positive direction. Given the price of the spa3k, I don't have any issue with purchasing more of them right now. Excellent choice for the one-to-three pstn-fxo market space. ___ 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 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 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] Caller ID
What standard does your telco send the caller-id in ? ETSI FSK, bellcore... ? On Wed, 2005-06-15 at 16:26 +0200, Stojan Sljivic - GDS wrote: Hi Juan, I have Caller Id service enabled. When I connect the line to the phone I see the caller Id on the phone's display. I have callerid=asreceived. I have also played with various combinations of cidsignalling and cidstart, but with no success. Regards, Stojan Sljivic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Manuel Coronado Z. Sent: Wednesday, June 15, 2005 15:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Caller ID Hi, First, you must ensure that the your Telco is sending you the caller ID with incoming calls. In some countries this is an aditional service you have to pay for, upon request. If you already have the service from the Telco, check in your zapata.conf that you have callerid=asreceived on your channels and group definitions. Hope this will help. Regards, Juan Manuel Coronado Z. On mar, 2005-06-14 at 14:50 +0200, Stojan Sljivic - GDS wrote: Hi, I'm using TDM04B and Asterisk 1.0.5. How can I setup the Asterisk so that I get caller ID? I do not get caller ID currently. Regards, Stojan Sljivic ___ 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 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 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 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] Asterisk SMS configuration
Yes i've changed the line mentioned to 6000 but i'm not sure thats the problem as from abrief look at the code it appears that only applies to a certain response that app_sms send which seems to be fine. I'll maybe try with a higher value later Just for interest what hardware areyou running. Jon Tony Hoyle wrote: Jon Creasey wrote: Tony, I'm havin a similar issue i'm in the UK using x100p with the patch for CID and get the following. Any ideas Did you change line as mentioned earlier? Without that patch incoming SMS won't work at all. You might need to use an even higher pause... might be worth experimenting. Tony ___ 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 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] app_dial.c:977 dial_exec_full: Unable to createchannel of type 'Zap' (cause 0)
The clone work good, Try to change your dialplan for zap channel and it will
work.
Try this exten =_11.,1,Dial(Zap/1/${EXTEN:2},90,Tt).
It work for mi with my clone card and my x100p (original card) for long
time.
saludos
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, June 15, 2005 7:48 PM
Subject: Re: [Asterisk-Users] app_dial.c:977 dial_exec_full: Unable to
createchannel of type 'Zap' (cause 0)
I don't have clone card to verify this, but I think you'll find the
chipset on that particular card is not the same chipset used on the
digium card. Since the asterisk drivers are written for specific
chipsets, I'd have to suggest you've got an almost zero chance of
making the clone work.
No Ideas? This seems like quite a common issue but I have searched and
searched for a solution and not found any?
Cheers.
Sandy.
Sandy Thomson wrote:
Hi,
Ive been struggling with asterisk for a few days now. I understand
pretty much how it works and how to tie things together (SIP - SIP
internally works fine etc), but just cannot get asterisk to work with an
X100P clone (its a Ambient MD3200, if that means anything to you guys).
I have tried (initially) asterisk 1.07 with zaptel 1.07, and now
Asterisk CVS-HEAD with zaptel cvs. Both give me the same error.
/etc/zaptel.conf
--
fxsks=1
loadzone=uk
defaultzone=uk
/etc/asterisk/zapata.conf
--
[channels]
language=en
context=incoming
signalling=fxs_ks
channel = 1
/etc/asterisk/extensions.conf
--
[general]
static=yes
writeprotect=no
[local]
exten = _11.,1,Dial(Zap/1,9w${EXTEN}:2)
[incoming]
exten = s,1,Answer()
exten = s,2,BackGround(demo-congrats) ; Play a congratulatory message
[outgoing]
exten = _9.,1,Ringing
exten = _9.,2,Wait,2
exten = _9.,3,Answer()
exten = _9.,4,Dial(ZAP/1/9w${EXTEN}:1)
[default]
include = outgoing
Loading zaptel modules:
--
asterisk zaptel # modprobe zaptel
asterisk zaptel # modprobe wcfxo
asterisk zaptel # lsmod
Module Size Used by
wcfxo 12576 0
zaptel222916 1 wcfxo
crc_ccitt 1952 1 zaptel
asterisk zaptel # ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
Running asterisk
--
asterisk zaptel # asterisk -gc
Registering builtin applications:
[AbsoluteTimeout]
== Registered application 'AbsoluteTimeout'
[Answer]
== Registered application 'Answer'
[BackGround]
== Registered application 'BackGround'
[Busy]
== Registered application 'Busy'
[Congestion]
== Registered application 'Congestion'
[DigitTimeout]
== Registered application 'DigitTimeout'
[Goto]
== Registered application 'Goto'
[GotoIf]
== Registered application 'GotoIf'
[GotoIfTime]
== Registered application 'GotoIfTime'
[ExecIfTime]
== Registered application 'ExecIfTime'
[Hangup]
== Registered application 'Hangup'
[NoOp]
== Registered application 'NoOp'
[Prefix]
== Registered application 'Prefix'
[Progress]
== Registered application 'Progress'
[ResetCDR]
== Registered application 'ResetCDR'
[ResponseTimeout]
== Registered application 'ResponseTimeout'
[Ringing]
== Registered application 'Ringing'
[SayNumber]
== Registered application 'SayNumber'
[SayDigits]
== Registered application 'SayDigits'
[SayAlpha]
== Registered application 'SayAlpha'
[SayPhonetic]
== Registered application 'SayPhonetic'
[SetAccount]
== Registered application 'SetAccount'
[SetAMAFlags]
== Registered application 'SetAMAFlags'
[SetGlobalVar]
== Registered application 'SetGlobalVar'
[SetLanguage]
== Registered application 'SetLanguage'
[Set]
== Registered application 'Set'
[SetVar]
== Registered application 'SetVar'
[ImportVar]
== Registered application 'ImportVar'
[StripMSD]
== Registered application 'StripMSD'
[Suffix]
== Registered application 'Suffix'
[Wait]
== Registered application 'Wait'
[WaitExten]
== Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
== Parsing '/etc/asterisk/modules.conf': Found
[res_features.so] = (Call Parking Resource)
== Parsing '/etc/asterisk/features.conf': Found
-- Registered extension context 'parkedcalls'
-- Added extension '700' priority 1 to parkedcalls
== Registered application 'ParkedCall'
== Registered application 'Park'
== Manager registered action ParkedCalls
[res_musiconhold.so] = (Music On Hold Resource)
== Registered application 'MusicOnHold'
== Registered application 'WaitMusicOnHold'
== Registered application 'SetMusicOnHold'
== Registered application 'StartMusicOnHold'
== Registered application 'StopMusicOnHold'
== Parsing '/etc/asterisk/musiconhold.conf': Found
[chan_zap.so] = (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1, FXS Kewlstart
[Asterisk-Users] Question on cdr_odbc
As per the wiki: http://www.voip-info.org/wiki-Asterisk+cdr+odbc#comments What do I have to do to get asterisk to recognize to use the odbc? I see I edit the cdr_odbc.conf file.. but do I have to compile asterisk with any flag or anything? ___ 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] Re: Re: Digium Website Update: Asterisk Busi ness Edition
David Brodbeck [EMAIL PROTECTED] writes: You can't follow the religion of free software and still run a company that pays the bills. I believe that you can. -- Esben Stien is [EMAIL PROTECTED] s a http://www. s tn m irc://irc. b - i . e/%23contact [sip|iax]: e e jid:b0ef@n n ___ 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] SIP transfer/REFER to voicemail problem
I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't answer. Caller (A) hears the Asterisk voicemail prompts, but the voicemail application doesn't hear any audio or DTMF. Easy to duplicate: 1.) A - B (INVITE) 2.) B - C (REFER A to C) 3.) A - C More descriptive: 1.) Caller (A) calls intermediary (B). (B can be any SIP user agent) 2.) Intermediary (B) REFERs caller (A) to callee (C) 3.) C is either a SIP UA which times-out and Asterisk takes to Voicemail, or an extension tied to VoicemailMain. I've come across a thread saying that the Asterisk voicemail system only uses the GSM codec, but if this were the problem, then how can the caller (using mu-law) hear the voicemail prompts? Would Asterisk be doing a half duplex protocol conversion? Any insight would be greatly appreciated!! Current configuration: Fedora Core 1 Asterisk - 1.0.7 (had same problem on 1.0.6) SJPhone - 1.50.271d, Mar 11 2005 (WinXP) XLite - 1103m build stamp 14262 (WinXP) Zultys Zip2 - ZUTS 3.52 sip.conf exerpt: [6003] ; (A) type=friend regexten=6003 username=6003 host=dynamic disallow=all ;allow=gsm allow=ulaw [6004] ; (C) type=friend regexten=6004 username=6004 host=dynamic disallow=all ;allow=gsm allow=ulaw [2101] ; (B) type=friend regexten=2101 username=2101 host=dynamic disallow=all ;allow=gsm allow=ulaw extensions.conf exerpt: exten = 6003,1,Dial(SIP/1003,15) exten = 6003,2,Voicemail(u1003) exten = 6003,102,Voicemail(b1003) exten = 6004,1,Dial(SIP/1004,5) exten = 6004,2,Voicemail(u1004) exten = 6004,102,Voicemail(b1004) exten = 2101,1,Dial(SIP/2101) exten = 8500,1,VoicemailMain exten = 8500,2,Hangup Asterisk (-dvvgc) with sip debug on (REFER-ing caller to VoicemailMain) : -- No username but # key pressed. Using CID '6003' -- Playing 'vm-password' (language 'en') Urgent handler -- Incorrect password '' for user '6003' (context = ,any) -- Playing 'vm-incorrect-mailbox' (language 'en') Urgent handler __ Discover Yahoo! Find restaurants, movies, travel and more fun for the weekend. Check it out! http://discover.yahoo.com/weekend.html ___ 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] HT-488 vs. SPA-3000?
The majority of the audio level issues seem to be on the fxo port and setting the transmission levels (gain) to compensate for the cable loss to the central office. Eg, setting the pstn gain values to what should be appropriate causes echo, etc, not unlike the TDM card. (I have both in use.) In other words, the further the spa3000 (or TDM card) is from the central office, the more difficult it seems to be to set gain values that are acceptable. That's apparently why many people find its use is okay while others seem to think its objectionable. We have 6 SPA3000s. The device is extremely configurable and works inbound/outbound with Asterisk with the latest firmware update with little trouble. However, we've yet to resolve sound volume and quality issues. The PSTN to SPA gain and SPA to PSTN gain along with FXS Port Input Gain and Output Gain settings have had no positive effect. The problem is entirely with the analog line adapter. VoIP calls from the analog phone to other VoIP destinations are perfect. We also have several SPA-1001s and SPA-2000s that have been running perfect since day 1. Also Sipura support is nonexistant. Just our experience. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dan Littlejohn Sent: Wednesday, June 15, 2005 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] HT-488 vs. SPA-3000? I have only had experience with the Sipura 3000 and I would agree with the voice volume problems. I have given up on it working properly (adjusted gains, impedences, firmware, etc), the voice quality is just to low to actually use. I actually purchased a second one thinking that the first might be defective. Would not recommend it because of the low sound volume problem. Talking on the phone is actually the point of the device so who cares how configurable it is if you cannot hear anything. I purchased a Digium TDM400P and have had very good luck with it. Dan On 6/15/05, Rich Adamson [EMAIL PROTECTED] wrote: Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... I've not played with the ht488, but I believe others have posted this device does not provide access to the pstn-fxo port. The spa3k does provide that access (if you want it). Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. The spa3k works fine with asterisk as many have posted. However, once in awhile it does act a little strange in two different ways: 1. the spa3k will sometimes interpret some voices as tones which cause a little disturbance to any conversation going on. It is sort of like the old telephony talk off that existed years ago. Doesn't happen all that often and seems to be more sensitive to female voices based on my one-year of experience. 2. sometimes it seems to operate in half-duplex mode, where if you try to talk at the same time as the other end is talking, the other end won't hear you. Neither one of those have been all that objectionable to me, but they happen and others have posted roughly the same issues. I've not heard of anyone that has found a way to minimize those two issues. The down side of the spa3k right now is that Cisco bought the company and there likely won't be much advancement of the code until after the ownership (and development efforts) are sorted out by both companies. (The same kind of product delays has been seen with their Linksys purchase, as well as when other companies are bought/sold.) Its fairly common knowledge that ex-Cisco folks started Sipura for the sole purpose of selling the company for a hugh profit. Their success in accomplishing that objective could only be measured in terms of producing Sipura products that had at least some acceptance of those products by end users. With those previous objectives accomplished, how will Cisco handle the Sipura products in the future? (It's any- one's guess at this point since Cisco also has at least some track record of mismanaging purchased companies for whatever reason.) From an internal Cisco strategic perspective, they now own the assets that can make a major dent in the mass-market end-user voip product arena, and hopefully they'll take that in a positive direction. Given the price of the spa3k, I don't have any issue with purchasing more of them right now. Excellent choice for the one-to-three pstn-fxo market space. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options
RE: [Asterisk-Users] HT-488 vs. SPA-3000?
I'm curious what other standalone FXO adapters work with Asterisk. At everything from the default to the maximum in positive and negative values, and combination of gain settings, we still get unacceptable distortion and echo. I've checked the phone lines, they work normally with a regular phone. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: Wednesday, June 15, 2005 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] HT-488 vs. SPA-3000? The majority of the audio level issues seem to be on the fxo port and setting the transmission levels (gain) to compensate for the cable loss to the central office. Eg, setting the pstn gain values to what should be appropriate causes echo, etc, not unlike the TDM card. (I have both in use.) In other words, the further the spa3000 (or TDM card) is from the central office, the more difficult it seems to be to set gain values that are acceptable. That's apparently why many people find its use is okay while others seem to think its objectionable. We have 6 SPA3000s. The device is extremely configurable and works inbound/outbound with Asterisk with the latest firmware update with little trouble. However, we've yet to resolve sound volume and quality issues. The PSTN to SPA gain and SPA to PSTN gain along with FXS Port Input Gain and Output Gain settings have had no positive effect. The problem is entirely with the analog line adapter. VoIP calls from the analog phone to other VoIP destinations are perfect. We also have several SPA-1001s and SPA-2000s that have been running perfect since day 1. Also Sipura support is nonexistant. Just our experience. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dan Littlejohn Sent: Wednesday, June 15, 2005 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] HT-488 vs. SPA-3000? I have only had experience with the Sipura 3000 and I would agree with the voice volume problems. I have given up on it working properly (adjusted gains, impedences, firmware, etc), the voice quality is just to low to actually use. I actually purchased a second one thinking that the first might be defective. Would not recommend it because of the low sound volume problem. Talking on the phone is actually the point of the device so who cares how configurable it is if you cannot hear anything. I purchased a Digium TDM400P and have had very good luck with it. Dan On 6/15/05, Rich Adamson [EMAIL PROTECTED] wrote: Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... I've not played with the ht488, but I believe others have posted this device does not provide access to the pstn-fxo port. The spa3k does provide that access (if you want it). Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. The spa3k works fine with asterisk as many have posted. However, once in awhile it does act a little strange in two different ways: 1. the spa3k will sometimes interpret some voices as tones which cause a little disturbance to any conversation going on. It is sort of like the old telephony talk off that existed years ago. Doesn't happen all that often and seems to be more sensitive to female voices based on my one-year of experience. 2. sometimes it seems to operate in half-duplex mode, where if you try to talk at the same time as the other end is talking, the other end won't hear you. Neither one of those have been all that objectionable to me, but they happen and others have posted roughly the same issues. I've not heard of anyone that has found a way to minimize those two issues. The down side of the spa3k right now is that Cisco bought the company and there likely won't be much advancement of the code until after the ownership (and development efforts) are sorted out by both companies. (The same kind of product delays has been seen with their Linksys purchase, as well as when other companies are bought/sold.) Its fairly common knowledge that ex-Cisco folks started Sipura for the sole purpose of selling the company for a hugh profit. Their success in accomplishing that objective could only be measured in terms of producing Sipura products that had at least some acceptance of those products by end users. With those previous objectives accomplished, how will Cisco handle the Sipura products in the future? (It's any- one's guess at this point since Cisco also has at least some track record of mismanaging purchased companies for whatever reason.) From an internal Cisco
[Asterisk-Users] Voip-info.org
Site down again?? Voip-info.org? or maybe really slow? ___ 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] Help with Cron and Reload
This will sound weird but the command 'asterisk -r -x reload' fails to work when issued by Cron. But it works when I issue it from a bash session. What is not configured correctly? I need to refresh the configuration every a short amount of time. rom [EMAIL PROTECTED] Wed Jun 15 18:42:00 2005 Date: Wed, 15 Jun 2005 18:42:00 -0400 From: [EMAIL PROTECTED] (Cron Daemon) To: [EMAIL PROTECTED] Subject: Cron [EMAIL PROTECTED] asterisk -r -x reload X-Cron-Env: SHELL=/bin/sh X-Cron-Env: HOME=/root X-Cron-Env: PATH=/usr/bin:/bin X-Cron-Env: LOGNAME=root /bin/sh: line 1: asterisk: command not found Any ideas? ___ 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] Unable to open master device '/dev/zap/ctl'
Hi all, when I try to load asterisk I get this error [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Jun 15 10:41:29 WARNING[3615]: chan_zap.c:869 zt_open: Unable to open '/dev/zap/channel': No such file or directory Jun 15 10:41:29 ERROR[3615]: chan_zap.c:6572 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp-channel = 1, channel = 1 Jun 15 10:41:29 ERROR[3615]: chan_zap.c:9910 setup_zap: Unable to register channel '1' Jun 15 10:41:29 WARNING[3615]: loader.c:402 __load_resource: chan_zap.so: load_module failed, returning -1 Jun 15 10:41:29 WARNING[3615]: loader.c:523 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED] asterisk]# if I try to run asterisk I get this error [EMAIL PROTECTED] ~]# cd /etc/asterisk [EMAIL PROTECTED] asterisk]# ztcfg -vv Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' line 204: Cannot get number of tones chanel 1 line 204: Cannot init tones chanel 1 line 204: Cannot get number of tones chanel 2 line 204: Cannot init tones chanel 2 5 error(s) detected [EMAIL PROTECTED] asterisk]# please help me kumara ___ 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] Help with Cron and Reload
Federico Alves wrote: This will sound weird but the command 'asterisk -r -x reload' fails to work when issued by Cron. But it works when I issue it from a bash session. What is not configured correctly? I need to refresh the configuration every a short amount of time. rom [EMAIL PROTECTED] Wed Jun 15 18:42:00 2005 Date: Wed, 15 Jun 2005 18:42:00 -0400 From: [EMAIL PROTECTED] (Cron Daemon) To: [EMAIL PROTECTED] Subject: Cron [EMAIL PROTECTED] asterisk -r -x reload X-Cron-Env: SHELL=/bin/sh X-Cron-Env: HOME=/root X-Cron-Env: PATH=/usr/bin:/bin X-Cron-Env: LOGNAME=root /bin/sh: line 1: asterisk: command not found Any ideas? Login in as root, type in: type asterisk I get /usr/sbin/asterisk Change crontab (crontab -e) to use the full path. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] Help with Cron and Reload
On 6/15/05, Federico Alves [EMAIL PROTECTED] wrote: This will sound weird but the command 'asterisk -r -x reload' fails to work when issued by Cron. But it works when I issue it from a bash session. What is not configured correctly? I need to refresh the configuration every a short amount of time. Use the full path when calling asterisk. The cron environment is not like a standard shell in all respects. Chris ___ 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] WiFi IP Phones
Guys. I know there are wifi sip phones out there but I have a question, are any of these phones anti explosive? By that I mean, there are certain regulations about phones or cel phones that are not recommended to operate in environments like gas stations due to sparks and the chance of ingiting gas fumes. Are there any wifi sip phones out here that have complaince with regulations to operate in hazardous environments like Oil Platforms, etc? phones denominated anti explosive or something? ___ 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] Voip-info.org
On Wed, 15 Jun 2005 14:37:42 -0400 Huddleston, Robert [EMAIL PROTECTED] wrote: Site down again?? Voip-info.org? or maybe really slow? Up here for me at 15:00 EDT... ___ 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] phantom answer
Title: Message
People,
My goal is to get
asterisk dialing out via my landline (POTS) from a sip softphone. Ive got the
phone, The TDM400p is installed and working. (See below) When ever I dial a
number that is directed to the outgoing port on my card (fxs/fxo?) I get no
ringing, then it claims its been answered. the CLI reports the
following:
Executing
Dial("SIP/301-f97a", "Zap/4/01614299100|20") in new stack
-- Called 4/01614299100 -- Zap/4-1 answered
SIP/301-f97aJun 15 17:57:38 NOTICE[11121]: rtp.c:277 process_rfc3389:
Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off
on client if possible. Client IP: 192.168.0.7 -- Hungup
'Zap/4-1'
Anyone Any Ideas?
BTW Apologies for the disclaimer at the bottom, but the mail server adds it on
by default and there's nothing I can do about it.
*CLI zap show
channels Chan Extension
Context Language
MusicOnHoldpseudo
default
1
default
default
4
incoming
default*CLI
This is the
important bit from zapata.conf
; DYLAN ADDED FROM DIGIUM.COM
echocancel=yes ; You can set this to 32, 64,
or 128, tweak to your needs.echocancelwhenbridged=yesechotraining=yes ;
Asterisk trains to the beginning of the call, number is in
millisecondscallerid=01614830073signalling=fxo_ksgroup=1context=default
; Points to the default context of your extensions.confchannel = 1
signalling=fxs_ks;callerid=asreceivedgroup=2context=incomingchannel=
4; END OF DYLAN ADDED FROM DIGIUM.COM *
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[Asterisk-Users] Config files under CVS versioning system
Hi all, I'm about to start building a bunch of asterisk servers with a team of developers and I thought it would be a good idea to put each server's config files under CVS so that we can keep track of changes, revert back etc... Which files do you think I should include in the cvs modules? just the .conf files? AND would it be possible to have the actual server copies be the versioned copesi so all I'd have to do is a cvs update to install new changes? Thoughs and suggestions appreciated, Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ 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] WiFi IP Phones
Anton - if you had a large opportunity and wanted a manufacturer to certify the phones as anti-explosive, I know a few that would probably attest to their phones being anti explosive as long as there was no major liability involved. I do not see anti explosive listed in any of the technical specifications of WLAN phones made by Zyxel Hitachi UTStarCom Uniden Cisco Net2Com Cory Andrews Purchasing / EVP VOIPSupply.com v 716.630.1555 X22 e [EMAIL PROTECTED] Anton Krall wrote: Guys. I know there are wifi sip phones out there but I have a question, are any of these phones anti explosive? By that I mean, there are certain regulations about phones or cel phones that are not recommended to operate in environments like gas stations due to sparks and the chance of ingiting gas fumes. Are there any wifi sip phones out here that have complaince with regulations to operate in hazardous environments like Oil Platforms, etc? phones denominated anti explosive or something? ___ 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 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] Help with Cron and Reload
i think that is because asterisk is intstalled in /usr/sbin/ and you dont have that un your PATH env variable for cron. best regards On 6/15/05, Federico Alves [EMAIL PROTECTED] wrote: This will sound weird but the command 'asterisk -r -x reload' fails to work when issued by Cron. But it works when I issue it from a bash session. What is not configured correctly? I need to refresh the configuration every a short amount of time. rom [EMAIL PROTECTED] Wed Jun 15 18:42:00 2005 Date: Wed, 15 Jun 2005 18:42:00 -0400 From: [EMAIL PROTECTED] (Cron Daemon) To: [EMAIL PROTECTED] Subject: Cron [EMAIL PROTECTED] asterisk -r -x reload X-Cron-Env: SHELL=/bin/sh X-Cron-Env: HOME=/root X-Cron-Env: PATH=/usr/bin:/bin X-Cron-Env: LOGNAME=root /bin/sh: line 1: asterisk: command not found Any ideas? ___ 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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ 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] Unable to open master device '/dev/zap/ctl'
Seems like your linux doesn't see your zaptel hardware. You can try with lspci -vvv and watch for Network controller: Tiger Jet Network. Hope this help Sebas Kumara Jayaweera wrote: Hi all, when I try to load asterisk I get this error [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Jun 15 10:41:29 WARNING[3615]: chan_zap.c:869 zt_open: Unable to open '/dev/zap/channel': No such file or directory Jun 15 10:41:29 ERROR[3615]: chan_zap.c:6572 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp-channel = 1, channel = 1 Jun 15 10:41:29 ERROR[3615]: chan_zap.c:9910 setup_zap: Unable to register channel '1' Jun 15 10:41:29 WARNING[3615]: loader.c:402 __load_resource: chan_zap.so: load_module failed, returning -1 Jun 15 10:41:29 WARNING[3615]: loader.c:523 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED] asterisk]# if I try to run asterisk I get this error [EMAIL PROTECTED] ~]# cd /etc/asterisk [EMAIL PROTECTED] asterisk]# ztcfg -vv Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' line 204: Cannot get number of tones chanel 1 line 204: Cannot init tones chanel 1 line 204: Cannot get number of tones chanel 2 line 204: Cannot init tones chanel 2 5 error(s) detected [EMAIL PROTECTED] asterisk]# please help me kumara ___ 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 -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Libertador 6250 4 piso Tl.: 4 706- (int. 121) [EMAIL PROTECTED] ___ 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] CellPhone BlueTooth adapater with Wireless Profile ??
All Any body know of a generic bluetooth adpater for the universal 2.5mm headset jack on a cell phone that supports the wireless profile *NOT* the headset profile I know jabra has the A210 http://www.jabra.com/JabraCMS/NA/EN/MainMenu/Products/Accessories/JabraA210/ JabraA210 but it only support the headset profile .. I am trying to shoe horn my current braindead cell into DocknTalk http://www.phonelabs.com/prd05.asp, and the BlueTooth interface requires a WirelessProile ___ 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] E911 Interface
Does anyone know of a E911 interface I can get? Regards, Chris ___ 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] WiFi IP Phones
Cory, the word he is looking for is Intrinsically Safe. And yes there are some around (I know from when I used to work for Nira now Ascom Nira that there is a big market for safe versions of the pagers and Dect handsets). Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Wednesday, 15 June 2005 3:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi IP Phones Anton - if you had a large opportunity and wanted a manufacturer to certify the phones as anti-explosive, I know a few that would probably attest to their phones being anti explosive as long as there was no major liability involved. I do not see anti explosive listed in any of the technical specifications of WLAN phones made by Zyxel Hitachi UTStarCom Uniden Cisco Net2Com Cory Andrews Purchasing / EVP VOIPSupply.com v - 716.630.1555 X22 e - [EMAIL PROTECTED] Anton Krall wrote: Guys. I know there are wifi sip phones out there but I have a question, are any of these phones anti explosive? By that I mean, there are certain regulations about phones or cel phones that are not recommended to operate in environments like gas stations due to sparks and the chance of ingiting gas fumes. Are there any wifi sip phones out here that have complaince with regulations to operate in hazardous environments like Oil Platforms, etc? phones denominated anti explosive or something? ___ 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 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 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: Asterisk-Users Digest, Vol 11, Issue 100
Jon, thanks for your help, but I'd rather not do it using agents and
queues, ideally what would happen is it would simply play the message
and wait for the person to press a button, if nothing is pressed, it
just keeps going down the list. Any other suggestions?
[EMAIL PROTECTED] wrote:
Date: Wed, 15 Jun 2005 00:53:14 -0500
From: Jon Gabrielson [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie question about pressing a key to
be connected to the caller
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
Check out ackcall=yes in agents.conf
It allows them to press # to accept, or press * to not accept.
then you can do something like:
exten = 101,1,Dial(Agent/101,20,A(presspoundtoanswer))
or if you want to get more fancy, check out queues.conf
where you can set ring orders and answer penalties.
Hope this helps,
Jon.
On Tuesday 14 June 2005 09:18 pm, Jason wrote:
I have a newbie question about the dialplan. I have a main menu that
picks up on a certain DID number, and gives a list of options. When an
option is selected, for instance 1 for sales, it rings a number of users
in succession until one picks up and is connected to the caller,
otherwise it goes to voicemail. This is all working well. However, I
would like to have the system play a message to the user when they pick
up, saying There is a call for sales, press 1 to accept the call or 2
to ignore. If the user pressed 1 they would then be connected to the
person calling in, if 2 it would just go to the next person in the
group. Any help would be appreciated, my current context is below for
reference:
[ext-sales]
exten = 1,1,Answer();
exten = 1,2,SetCIDName(Sales);
exten = 1,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?ext-cs-vm,1,1)
exten = 1,4,Playback(custom/PleaseWait)
exten = 1,5,Macro(dial,${RINGTIMER},tm,717-6197585949#)
exten = 1,6,Macro(dial,${RINGTIMER},tm,707-6199208398#)
exten = 1,7,Macro(dial,${RINGTIMER},tm,8323686410#)
exten = 1,8,Macro(dial,${RINGTIMER},tm,717-6197585949#)
exten = 1,9,Macro(dial,${RINGTIMER},tm,707-6199208398#)
exten = 1,10,Macro(dial,${RINGTIMER},tm,8323686410#)
exten = 1,11,Goto(ext-sales-vm,1,1)
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