[Asterisk-Users] Jingle support - can we test the feature ?
Hi, we would like to build IM-Voice community for our students around Asterisk, Jingle, Jabber. Can we already test those features ? Anyone already running such setup? Any more info ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial two extensions at the SAME time and connect them when possible
Hi, I want to start call between A and B. Currently call can be triggerred with either first calling A or B number and then the other number after fist picks up. I'd like to call A and B at the same time and connect them in call when possible... One way would probably be with putting both calls in conference, but maybe there is some more elegant way of doing it? Also is there any specific reason why calls are triggered so only one client is called and then the other ? Thanks , regards, Rob. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to stop Asterisk from picking up my phone?
Hi, I was able to configure (Incoming Calls) through AMP to make asterisk answer my line after 3 rings and forward it to an extension. However, I was unable to disable that feature? Any way? [EMAIL PROTECTED] 2.6, Asterisk 1.2.4 Thanks LinuxnizerBe the first to hear what's new at MSN - sign up to our free newsletters! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Where to buy Eicon DIVA cards
Hi Klaus, You can find a list of resellers for Eicon Diva cards at the following link: http://tinyurl.com/ka46q Kind regards David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus Darilion Sent: 19 April 2006 18:27 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Where to buy Eicon DIVA cards Hi! Can someone recommend a Eicon DIVA cards distributor in Austria/Europe? thanks klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to stop Asterisk picking up my incoming calls?
Hi, I was able to configure (Incoming Calls) through AMP to make asterisk answer my line after 3 rings and forward it to an extension. However, I was unable to disable that feature? Any way? [EMAIL PROTECTED] 2.6, Asterisk 1.2.4 Thanks LinuxnizerFed up with spam in your inbox? Find out how to deal with junk e-mail here! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Happy story
Hi, Couple of days ago I decided to clean up my storage room. I found couple of old server. I gave everything to charity organziations. But the oldest one was just too old to be used. (PIII 300Mhz, 64MB RAM, 1.6GB HD) :). So I kept it for testing. HoneyPot? not a good idea. Windows XP took 2 hours to intstall, never was able to make it boot. So I decided to install Asterisk. Took me 45 min to do so, but stopped because of no disk space. I plugged an old 10GB HD instead of 1.6G. Re-installed everything again without a problem. Logged through AMP created couple of extensions between my 3 PCs. Calls were very clear :) Decided to install an FXO card, looked around the net and found them around $100. But after some net searchfound that an old modem I already have would do the trick. So, I installed it and bingo! Now I can use my notebook to call anywhere at home when I'm traveling outside the country for FREE! Yes, Asterisk does work with PIII 300Mhz with 64MB RAM, I wouldn't recommend it for mission critical applications, but for me, it does the job. Thank you to everyone involved in the Asterisk project. Linuxnizer.Hotmail is evolving - be one of the first to try out the Windows Live Mail Beta ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Remember the incoming context?
EG == Edwin Groothuis [EMAIL PROTECTED] writes: EG Looks like I have to submit a patch! Or just store the original context in a variable? /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme codec translation and callerID library.
Jay Milk wrote: tom wrote: Also, does anyone know if there's a way to dynamically alter incoming Caller-IDs to add Caller ID text to them. ie. call comes in with ID 01234 567890 gets changed to A Company 01234 567890 ? Look at cid_rewrite, here: http://muware.com/asterisk Thanks, looking at the script I think I'll rewrite it myself. I was just wondering if there was a way of using DBPut/DBGet to reduce the number of connects. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly
This is a known issue with Asterisk's implementation of DTMF detection. There are two bug reports open up on bug tracker. Currently the best combination is to set DTMF TX method on Spa3k to INFO and auto on asterisk side. Works 95%, skips digits if you press buttons on the FXO end too fast. Until the DTMF stuff gets rewritten in Asterisk this gonna be this way, so far 2 years still no 100% dtmf detection, both detection and transmit parts are flawed and dont work 100%, even inband. Affected fxo gateways are (tested by myself): Sipura, Addpac, Planet, Wellgate. HTH, Vahan Dave Fullerton wrote: Greetings, I'm using asterisk to connect our three locations together with a sort of inter-company auto attendant connected like this: PBX (fxs) - Sipura 3k (fxo) - Asterisk -IAX- remote asterisk It works like this: Person picks up their phone and dials a number to get to the auto attendant (I don't have any FXO ports available on our PBX to do it the right way). The attendant answers and asks them the remote extension they want to dial. This setup has worked very well for several months. Last week I upgraded to 1.2.7.1 from 1.2.4 (I think). Since then I've been having trouble with the auto-attendant correctly detecting DTMF (missing digits). Some times it works flawlessly, others I have to try over and over before it is detected correctly. It isn't even consistently dropping the same digit from what I can see on the console. The only thing I've found is that I have a better chance of it working if I wait for the prompt to finish before dialing. I have changed the DTMF method from rfc2833 to info and finally inband with only a little change (inband seems to work the best). Has anyone else run into similar problems or have any more suggestions to try? This is the attendant portion of my extensions.conf: [inter-attendant] exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Set(TIMEOUT(response)=10) exten = s,4,Background(enter-ext-of-person) exten = i,1,Playback(invalid) exten = i,2,Goto(s,4) exten = i,3,Hangup exten = t,1,Playback(goodbye) exten = t,2,Hangup include = tests include = fullertonpbx include = intercompany Thank you for any insight you can provide. Dave Fullerton ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: sip.conf and jump from register to the extension
Am Thursday 20 April 2006 01:21 schrieb tom: Thomas Winter wrote: I have done additional tests, because the documentation sample was not 100 % identical to my register command. OK: register = 44198:[EMAIL PROTECTED]/200 This jumps to 200, s is also working NOT OK: user:[EMAIL PROTECTED]/200 It looks for extension user and is ignoring 200 or anythink else I think the non numeric username is the problem. Yes, I have done an restart of Asteriks after changing the sip.conf. Excuse me for sounding silly, but isn't the extension you mark at the end sent to your provider as the extension that they should use when calling you (ie. in the authentication statement, the remote server tries to connect as 200@yourhost-ip ) which is why some providers with broken sip implementations require you to have a specific extension after the /. ie. the extension they call on, is not neccesarily what is stated in the register statement, that's just the extension you've told them to call you on. Do a sip debug provider.com in the asterisk CLI to see what happens when the call comes in. I have used Ethereal. The initial call comes in: SIP/SDP Request Invite sip:[EMAIL PROTECTED] user is allways the username from the register , there is no information regards the user extension from the register command in. (or I didnt see them) If * is register at the sip provider there are contact bindings send to the SIP-proxy with the extension from the register command and some messages with bindings are comming back. I dont know how this is related to the call INVITE. Anyway, I have tested 4 different provider, two numeric and two alphanumeric username. numeric is working and alphanumeric is not working. If the provider software is broken it would be also good to know. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards,
2006/4/19, Lee Howard [EMAIL PROTECTED]: iaxmodem uses the spandsp library.So currently its modulation protocolsupport is limited to V.27ter and V.29.Partial (sending) V.17 supportis available.V.34 (Super G3) support is not.So those are fax speeds 2400 bps through 14400 bps for sending and 2400 bps through 9600 bps forreceiving.Most new fax machines that you would purchase off-the shelfwould support either 2400-14400 or 2400-33600 depending on whether or not it supported Super G3.So, the suggested design for a fax-enabled PBX solution could be :- for an average fax use (faxes from time to time, no fax mailings, telecom budget capable to bear longer durations for fax sendings), use an Asterisk-Hylafax-iaxmodem or your ITSP offering - for an intermediate fax use (fax mailings from time to time, sensitive telecom budget), use an Asterisk server with TDM passthru and a dedicated Hylafax fax modem equiped server or dedicate some PSTN lines to a fax server - for an intensive fax use (all day long fax mailings, high cost sensitivity), use a dedicated Fax Service Provider with competitive offering in the areas you are fax-mailing to.What do you think of that ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and siemens hipath 3500
Hello.
I have asterisk with an old avm b1 v3.0 configured and working with
capi channel . The isdn card is connected to an S0 isdn bus of a
siemens hipath 3000 version 4.0 It is possible to make
outgoing calls and receive too, but I think that there is some kind of
signaling problem when i call from an ip phone or soft ip phone because
i do not get ring or busy tone calling to any PBX phone, but if i get
outside line (i.e. calling a cell phone at PSTN) then i can hear the
busy or ring tones.
So, if i call to any pbx phone i do not hear anything until someone
picks up the phone and if it is busy i do not know and after a while i
get the normal call clearing and the call is finished.
IP PHONE - B1 (ISDN) chan capi AT * -- S0 BUS HIPATH - PSTN
Does anybody know any useful trick to solve this?
I know that may be it is a signaling problem and isdn related question, but if someone can help me it would be great.
Pardon my bad English and thanks.
Some configured parameters:
;
;capi.conf
;
; general section
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=es
; interface sections ...
[ISDN1]
;ntmode=yes
isdnmode=msn
incomingmsn=620
controller=1
group=1 ;dialout group
;prefix=0
softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards
accountcode= ;Asterisk accountcode to use in CDRs
context=capi-in ;context for incoming calls
holdtype=hold ;
;immediate=yes
;echosquelch=1
;echocancel=yes
;echotail=64
bridge=yes
callgroup=1
language=es
***
;
;indications.conf
;
[general]
country=es
[es]
description = Spain
ringcadence =1500,3000
dial = 423
busy =425/200,0/200
congestion = 425/200,0/200,425/200,0/200,425/200,0/600
callwaiting = 425/175,0/175,425/175,0/3500
dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425
record = 1400/500,0/15000
info = 950/330,0/1000
dialout = 500
*
;
; part of dial extensions.conf for dialing outisde
;RDSI=620 is the number for the isdn S0 bus
;
[capi-out]
exten = _0.,1,NoOp(Salida a la calle ${CALLERID})
exten = _0.,2,Dial(CAPI/g1/${RDSI},20)
...
**
Thanks.
Ricardo
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[Asterisk-Users] avm b1with chan capi and siemens hipath
Hello.
I have asterisk with an old avm b1 v3.0 configured and working with capi
channel . The isdn card is connected to an S0 isdn bus of a siemens
hipath 3000 version 4.0 It is possible to make outgoing calls and
receive too, but I think that there is some kind of signaling problem
when i call from an ip phone or soft ip phone because i do not get ring
or busy tone calling to any PBX phone, but if i get outside line (i.e.
calling a cell phone at PSTN) then i can hear the busy or ring tones.
So, if i call to any pbx phone i do not hear anything until someone
picks up the phone and if it is busy i do not know and after a while i
get the normal call clearing and the call is finished.
IP PHONE - B1 (ISDN) chan capi AT * -- S0 BUS HIPATH
- PSTN
Does anybody know any useful trick to solve this?
I know that may be it is a signaling problem and isdn related question,
but if someone can help me it would be great.
Pardon my bad English and thanks.
Some configured parameters:
;
;capi.conf
;
; general section
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=es
; interface sections ...
[ISDN1]
;ntmode=yes
isdnmode=msn
incomingmsn=620
controller=1
group=1 ;dialout group
;prefix=0
softdtmf=on ;enable/disable software dtmf detection, recommended
for AVM cards
accountcode= ;Asterisk accountcode to use in CDRs
context=capi-in ;context for incoming calls
holdtype=hold ;
;immediate=yes
;echosquelch=1
;echocancel=yes
;echotail=64
bridge=yes
callgroup=1
language=es
***
;
;indications.conf
;
[general]
country=es
[es]
description = Spain
ringcadence =1500,3000
dial = 423
busy =425/200,0/200
congestion = 425/200,0/200,425/200,0/200,425/200,0/600
callwaiting = 425/175,0/175,425/175,0/3500
dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425
record = 1400/500,0/15000
info = 950/330,0/1000
dialout = 500
*
;
; part of dial extensions.conf for dialing outisde
;RDSI=620 is the number for the isdn S0 bus
;
[capi-out]
exten = _0.,1,NoOp(Salida a la calle ${CALLERID})
exten = _0.,2,Dial(CAPI/g1/${RDSI},20)
...
**
Thanks.
Ricardo
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Re: [Asterisk-Users] avm b1with chan capi and siemens hipath
Did you try the /b option of Dial() with capi?
This enables early-b3, whcih gives you progress tones from the ISDN line.
Armin
On Thu, 20 Apr 2006, Ricardo wrote:
Hello.
I have asterisk with an old avm b1 v3.0 configured and working with capi
channel . The isdn card is connected to an S0 isdn bus of a siemens hipath
3000 version 4.0 It is possible to make outgoing calls and receive too, but I
think that there is some kind of signaling problem when i call from an ip
phone or soft ip phone because i do not get ring or busy tone calling to any
PBX phone, but if i get outside line (i.e. calling a cell phone at PSTN) then
i can hear the busy or ring tones.
So, if i call to any pbx phone i do not hear anything until someone picks up
the phone and if it is busy i do not know and after a while i get the normal
call clearing and the call is finished.
IP PHONE - B1 (ISDN) chan capi AT * -- S0 BUS HIPATH -
PSTN
Does anybody know any useful trick to solve this?
I know that may be it is a signaling problem and isdn related question, but if
someone can help me it would be great.
Pardon my bad English and thanks.
Some configured parameters:
;
; capi.conf
;
; general section
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=es
; interface sections ...
[ISDN1] ;ntmode=yesisdnmode=msn incomingmsn=620 controller=1
group=1 ;dialout group
;prefix=0 softdtmf=on ;enable/disable software dtmf detection,
recommended for AVM cards
accountcode= ;Asterisk accountcode to use in CDRs
context=capi-in ;context for incoming calls
holdtype=hold ;
; immediate=yes echosquelch=1 echocancel=yes echotail=64
bridge=yes callgroup=1 language=es
***
;
; indications.conf
;
[general]
country=es
[es]
description = Spain
ringcadence =1500,3000
dial = 423
busy =425/200,0/200
congestion = 425/200,0/200,425/200,0/200,425/200,0/600
callwaiting = 425/175,0/175,425/175,0/3500
dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425
record = 1400/500,0/15000
info = 950/330,0/1000
dialout = 500
*
;
; part of dial extensions.conf for dialing outisde
; RDSI=620 is the number for the isdn S0 bus
;
[capi-out]
exten = _0.,1,NoOp(Salida a la calle ${CALLERID})
exten = _0.,2,Dial(CAPI/g1/${RDSI},20)
...
**
Thanks.
Ricardo
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[Asterisk-Users] why DUNDi ${IPADDR} can not transfer to 127.0.0.1?
in the documents, it said that
; 'dest' is the destination to supply for reaching that number. The; following variables can be used in the destination string and will; be automatically substituted:; ${NUMBER}: The number being requested
; ${IPADDR}: The IP address to connect to; ${SECRET}: The current rotating secret key to be used
but in my experiences.. i need to change the${IPADDR} to the ip address of theasterisk server..
If this is the bug?
thanks,-- Jefferyiaxtel Num: 1-700-576-1311fwdnet Num: 728150
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[Asterisk-Users] SPA-3000 Bug? Dropped calls while registering.
Hello All!
I am in the process of assembling an asterisk-based
phone system for my office - using SPA-3000s to connect the network
to the PSTN. I am wondering if anybody else canget (or has
already seen) the same behaviour out of their 3000.
The short version: Send multiple Calls
to the SPA's FXO port at the same time it is re-registering with
Asterisk.
SPA HTTP
Configuration: PSTN Line - Register
Expires:5 (to
ensure it is registering all the time)
Dial one number through the
SPA's FXO port- establish a conversation
Dial another number through the
same FXO port (SPA3000/NXY).
What SHOULD happen is the second caller receives a
'504 - Service Unavailable' error while the first caller happily continues the
established conversation. What happens here: the
already established call gets dropped, AND the second caller gets a 504
error.
I did send a note to Linksys - and will see what
kind of reponse they have.
With longer "Register Expires:" times (10, 30, 60
seconds) it took more attempts to make the call
drop.
I have my Register Expires time cranked up to 86400
(1 day) now - and am hoping I don't see another repeat.
---
There are three SPA-3000s in the
system. I looked at some more complicated
dialplan layouts, and decided to keep it simple:
exten = s,1,Dial(${PSTN2}/${ARG1},,n)exten
= s,2,Dial(${PSTN3}/${ARG1},,n)exten =
s,3,Dial(${PSTN1}/${ARG1},,n)exten = s,4,Wait(1)exten =
s,5,Playback(all-circuits-busy-now)exten =
s,6,Congestion()
PSTN1,2,3 are 3 SPA-3000s registered with
Asterisk.
This approach relies on the SPA denying a call if
it is already in use.
Looking through the logs, the SIP packets
seem to be in order. INVITE, 100-Trying, 504-Service
Unavailable, ACK.
I'm at the end of my technical limit (ever
increasing as I play in the open-source world) - but my best guess
is:
During the Register process, something is
temporarily reset (such as a variable indicating that the line is in use)
such that when the second call comes in - it is actually connected to the
existing conversation for a brief period before the SPA realizes that the line
is actually already in use. As part of a cleanup
procedure - a hangup procedure is run: disconnecting the call.
The Equipment my trials were done on:
SPA3000 Hardware Version:
2.0.1(7376), Software Version: 3.1.10(GWd), and
also tried Software 3.1.7.
Nothing plugged into the FXS
port.
Asterisk1.2.4 running on FreeBSD 5.4
(i386), AMD Athlon 64 3200+, 1GB RAM.
SNOM 320. Application-Version: snom320-SIP
5.3.6 Rootfs: snom320 jffs2 v3.36
Polycom IP501 don't have access to the
software/hardware version from where I am right now
Cellphone
All SIP equipment is running on a dedicated
LAN. Network "splitters" were used to run two parallel LANs through the
existing cabling. (cat5e has 4 twisted pairs, only 2 twisted pairs
are needed for a 100BASET connection) The only computers on the LAN are the
asterisk box, and my workstation (2 NICs each).
Regards,
Dana Harding
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[Asterisk-Users] Asterisk MGCP reinvite
Hi all, I'm having troubles with the MGCP canreinvite option. I have two eyep media and Asterisk on the same network, same codecs, the option line canreinvite=yes properly set in the mgcp.conf file... but it won't do. Any idea anyone? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk + mobicents
Hello, I look at the mobicents project. Somebody has experience within both projects ? Regards Harry ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] still some moh troubles
Hi, After following the suggestions on the mailing lists and the wiki I'm still experiencing choppy moh. The song plays but with frequent noise parts. - I'm using asterisk 1.2.4 on our production server and 1.2.7 on the test server. - native moh with .gsm and .pcm formats (according to http://astrecipes.net/?n=152) - compiled ztdummy as a timing source any pointers on how to dig deeper into the problem or remedy it are very much appreciated regards, Bart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] transforming g729 files to wav files
Hi! The GX::Transcoder can convert G.729 to wav files. http://www.germanixsoft.de/ Greets Christian Tofik Suleymanov schrieb: Hello list, is there any open-source software that recodes g729 sound files to wav sound files ? The only way (at least) to do such transformation is with interactive form on: http://www.asteriskguru.com/audio_conversion.php Tofik Suleymanov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Christian Wengel n:Wengel;Christian org;quoted-printable:coXorange networXervice Bendig Dohrmann GbR;Technische Unterst=C3=BCtzung und Entwicklung adr:;;Stadtring 4;Cottbus;Brandenburg;03042;Deutschland email;internet:[EMAIL PROTECTED] tel;work:+49-355-3812740 tel;fax:+49-355-3812717 tel;cell:+49-160-96442842 x-mozilla-html:TRUE url:http://www.coxorange.net version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best Fax send through Asterisk plan?
Hi, Any idea how I can send fax through Asterisk by my old fax? Some guys suggested FAX-ATA(T.38 detect)-Asterisk-ATA(T.38 detect)-FAX. With plan is best? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: why DUNDi ${IPADDR} has been transfered to 127.0.0.1?
-- Forwarded message --From: 陈帆 [EMAIL PROTECTED]Date: Apr 20, 2006 5:39 PMSubject: why DUNDi ${IPADDR} can not transfer to
127.0.0.1?To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
in the documents, it said that
; 'dest' is the destination to supply for reaching that number. The; following variables can be used in the destination string and will; be automatically substituted:; ${NUMBER}: The number being requested
; ${IPADDR}: The IP address to connect to; ${SECRET}: The current rotating secret key to be used
but in my experiences.. i need to change the${IPADDR} to the ip address of theasterisk server..
If this is the bug?
thanks,-- Jefferyiaxtel Num: 1-700-576-1311fwdnet Num: 728150 -- Jefferyiaxtel Num: 1-700-576-1311fwdnet Num: 728150
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RE: [Asterisk-Users] ANNOUNCE: Asterisk Jobs and Consulting Site
Very underutilized at the moment. Once my schedule clears, expect some really asterisk specific quesitonaires pre-qualification screens that really defines the scope of work and expectations priort to bidding. Nothing worse that putting together a proposal for a customer that is not really looking to purchase, just looking for ideas. Same holds true for Gurus that don't deliver according to expectations. Thanks, Steve -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Thu 4/20/2006 12:58 AM To: asterisk-users@lists.digium.com Cc: Subject: Re: [Asterisk-Users] ANNOUNCE: Asterisk Jobs and Consulting Site On Thursday 20 April 2006 00:13, Matt Gibson wrote: I would like to announce the availability of a new site dedicated to finding and creating jobs in the Asterisk VOIP field. I've created this site, after noticing there are no sites dedicated to providing quality job postings and hiring abilities to people in the field. You mean like http://www.asteriskhelpdesk.com? I'm sure there are a couple of others too I'm missing. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk + mobicents
Hi Harry, I look at the mobicents project. Somebody has experience within both projects ? I dont have any real experience with mobicents but I now that some guys from mobicents built a resource adapter for asterisk about a year ago. Here are some notes about it: http://wiki.java.net/bin/view/Communications/MobicentsAsteriskRA =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Playback(something,noanswer) on Zap?
Hi! Our telco routes multiple numbers through PRI to our Asterisk. Not all of these numbers are in use. I have noticed recently that someone keeps calling unused phone number from outside world. I called them and asked why do they call dead number. The person on the far end explained that she keeps calling this number because she hears busy tone every time... Most telcos these days provide verbal in-band notification in case if number does not exist. Those nice female voices. People do expect this behaviour from their phones. People no longer accept beeps as number does not exist signal. First thing I've tried was Playback(invalid). The problem was that asterisk answered incoming call. This should not happen when caller does not reach his/her destination. Next, I tried Playback(invalid,noanswer). This time, Asterisk did not answer the call. But there was no sound! Apparently, Playback(invalid,noanswer) does not work with Zap/PRI. Is this bug? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDRs and billing
Hello I configured Asterisk to put CDRs in the database like it was explained in: www.voip-info.org/wiki/view/Asterisk+cdr+pgsql What I want to know is how do the billing solutions (like Asterisk2Billing) work with Asterisk. The billing system just use the information that Asterisk puts in the CDR table? Or they connect directly to Asterisk? Or is Asterisk that has, before the Dial command, to put the information on the Asterisk2Billing tables? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] aterisk+h323 trunk?!
Good day to you all! I have been reading this mailing list for quite some time, and now, i do have a question. I have a working asterisk server with VOIP telephone number connected to it via SIP, and it works just fine. Now I am installing new server on new VoIP provider and provider only supports H323 trunks, but I havo no ide how to make it work. Unfortunatly they are not very open about shareing information, so I need someone who would be so nice to explain in short how it works and how to make it work. Thank you, Tic Pavlin, Neosystems d.o.o., Ljubljana, Slovenia ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] attended transfer issue
2006/4/20, John Novack (port) [EMAIL PROTECTED]: There should be no need forTWO feature codes.I fully second that : what matters most is to satisfy users.Unified transfer method offer :- simplicity,- hardware independance (think about mobile phones, or people occasionnaly using foreign language configured phones when visiting a sister company abroad) - and above all, it keeps calls from being lost.So it should be implemented in Asterisk and it's up to Polycom, Snom and others to design phones that at least do not prevent people to use # sign based unified transfer method if they wish to. For the sake of behaviour consistency, maybe :- this unified transfer method (let's say U for unified) should be introduced in features.conf independently of previous t or T methods and it's up developpers to reuse, factorize or rewrite existing transfer code and as long as those 3 methods as supported, - and previous t or T methods should be droped sometimes later on to simplify code support.Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using ISDN MSNs for dialing out of Asterisk
On Tuesday 18 April 2006 14:58, Christian Gröger wrote:
Hi,
I am using Asterisk with misdn connected to an ISDN Line, so I have
several numbers I can use...
I know that I can use misdn like this in my extensions.conf:
exten = _0.,1,Dial(mISDN/1/${EXTEN:1})
But how can I use another number/MSN of my ISDN connection... it
always uses the default number, but i'd like to use another MSN for
calling... Can somebody help me please?
Not sure about mISDN... But we have E1 PRI and 100 numbers. We can send
any of these 100 numbers as caller id and it will reach remote end. If
we send anything else then our telco sets caller id to our default
number (first of 100). I use AGI to set caller id for outgoing calls.
If you want to use other available numbers as caller id, try to set
caller id in your extensions.conf.
Set(CALLERID(number)=1234567)
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Re: [Asterisk-Users] CDRs and billing
Joao Pereira wrote: Hello I configured Asterisk to put CDRs in the database like it was explained in: www.voip-info.org/wiki/view/Asterisk+cdr+pgsql What I want to know is how do the billing solutions (like Asterisk2Billing) work with Asterisk. The billing system just use the information that Asterisk puts in the CDR table? Or they connect directly to Asterisk? Or is Asterisk that has, before the Dial command, to put the information on the Asterisk2Billing tables? Asterisk2Billing requires you route the calls to its AGI, and it keeps its own database, so what you did is of no use for billing. I haven't found an application that bills from the CDRs, everything I found wanted to create the database entries. I think ASTPP can read your CDR, though. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Internet connection
Hi all, just a questionhope quite simple! Why if my internet connection goes down, my clients and sip phones stop to work? (go in logedoff)? I have aregister with a sip server ? It could be it ? If my * box do not register with the server it exclude the rest of the file ? :/ Thanks Giordano Le informazioni contenute nella presente e-mail e nei documenti eventualmente allegati possono essere confidenziali e sono comunque riservate al destinatario della stessa. La loro diffusione, distribuzione e/o copiatura da parte di terzi è proibita. Se avete ricevuto questa comunicazione per errore, Vi preghiamo di informare immediatamente il mittente del messaggio e di distruggere questa e-mail. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] still some moh troubles
Bart van Daal wrote: Hi, After following the suggestions on the mailing lists and the wiki I'm still experiencing choppy moh. The song plays but with frequent noise parts. - I'm using asterisk 1.2.4 on our production server and 1.2.7 on the test server. - native moh with .gsm and .pcm formats (according to Actually, you'll want to use ulaw for Native MOH. CUT #!/bin/sh for filename in *mp3 do eval filename=`echo $filename | cut -f1 -d.` echo Converting $filename sox -V $filename.mp3 -t au -r 8000 -U -b -c 1 $filename.ulaw resample -ql done CUT Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-3000 Bug? Dropped calls while registering.
i just got a SPA3000 but still not using it on production, and i
havent tested deeply. However, have you tried using incominglimit=1
in the register context of the SPA?? i guess that would limit in the
PBX rather that sending the call to the SPA.
Regards
On 4/20/06, Dana Harding [EMAIL PROTECTED] wrote:
Hello All!
I am in the process of assembling an asterisk-based phone system for my
office - using SPA-3000s to connect the network to the PSTN. I am
wondering if anybody else can get (or has already seen) the same behaviour
out of their 3000.
The short version: Send multiple Calls to the SPA's FXO port at the same
time it is re-registering with Asterisk.
SPA HTTP Configuration: PSTN Line - Register Expires: 5
(to ensure it is registering all the time)
Dial one number through the SPA's FXO port - establish a conversation
Dial another number through the same FXO port (SPA3000/NXY).
What SHOULD happen is the second caller receives a '504 - Service
Unavailable' error while the first caller happily continues the established
conversation. What happens here: the already established call gets
dropped, AND the second caller gets a 504 error.
I did send a note to Linksys - and will see what kind of reponse they have.
With longer Register Expires: times (10, 30, 60 seconds) it took more
attempts to make the call drop.
I have my Register Expires time cranked up to 86400 (1 day) now - and am
hoping I don't see another repeat.
---
There are three SPA-3000s in the system. I looked at some more
complicated dialplan layouts, and decided to keep it simple:
exten = s,1,Dial(${PSTN2}/${ARG1},,n)
exten = s,2,Dial(${PSTN3}/${ARG1},,n)
exten = s,3,Dial(${PSTN1}/${ARG1},,n)
exten = s,4,Wait(1)
exten = s,5,Playback(all-circuits-busy-now)
exten = s,6,Congestion()
PSTN1,2,3 are 3 SPA-3000s registered with Asterisk.
This approach relies on the SPA denying a call if it is already in use.
Looking through the logs, the SIP packets seem to be in order. INVITE,
100-Trying, 504-Service Unavailable, ACK.
I'm at the end of my technical limit (ever increasing as I play in the
open-source world) - but my best guess is:
During the Register process, something is temporarily reset (such as a
variable indicating that the line is in use) such that when the second call
comes in - it is actually connected to the existing conversation for a brief
period before the SPA realizes that the line is actually already in use.
As part of a cleanup procedure - a hangup procedure is run: disconnecting
the call.
The Equipment my trials were done on:
SPA3000 Hardware Version: 2.0.1(7376), Software Version: 3.1.10(GWd),
and also tried Software 3.1.7.
Nothing plugged into the FXS port.
Asterisk 1.2.4 running on FreeBSD 5.4 (i386), AMD Athlon 64 3200+, 1GB RAM.
SNOM 320. Application-Version: snom320-SIP 5.3.6 Rootfs: snom320 jffs2
v3.36
Polycom IP501 don't have access to the software/hardware version from
where I am right now
Cellphone
All SIP equipment is running on a dedicated LAN. Network splitters were
used to run two parallel LANs through the existing cabling. (cat5e has 4
twisted pairs, only 2 twisted pairs are needed for a 100BASET connection)
The only computers on the LAN are the asterisk box, and my workstation (2
NICs each).
Regards,
Dana Harding
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Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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RE: [Asterisk-Users] still some moh troubles
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: donderdag 20 april 2006 14:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] still some moh troubles Bart van Daal wrote: Hi, After following the suggestions on the mailing lists and the wiki I'm still experiencing choppy moh. The song plays but with frequent noise parts. - I'm using asterisk 1.2.4 on our production server and 1.2.7 on the test server. - native moh with .gsm and .pcm formats (according to Actually, you'll want to use ulaw for Native MOH. CUT #!/bin/sh for filename in *mp3 do eval filename=`echo $filename | cut -f1 -d.` echo Converting $filename sox -V $filename.mp3 -t au -r 8000 -U -b -c 1 $filename.ulaw resample -ql done CUT Doug Thanks for you suggestion Doug, I've converted the files using your script to ulaw but experience the same problem. A thing I forgot to mention is that it only happens on calls passing the trunks to the cisco-routers that terminate to pstn so not on internal sip-sip calls. Normal voice communication runs smoothly over the trunks it's only the moh that causes some problems. again, any pointers like those of Doug are very much appreciated thanks! Bart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly
I too am experiencing DTMF problems with 1.2.7.1 that I did not experience with recent prior versions. I've backed up to version 1.2.6 and so far DTMF detection is working reliably (but that's only with about 10 calls worth of testing). I've only had problems over SIP channels. Zap channels did not have problems with 1.2.7.1. I do not have any IAX channels, so cannot comment on that. I know others tend to discount DTMF problems because of known problems with how Asterisk handles DTMF, but there does seem to be enough anecdotal evidence that something bad has recently happened to make things worse. Dave, would you mind trying version 1.2.6 to see if that also resolves your problems? Dave Fullerton wrote: Greetings, I'm using asterisk to connect our three locations together with a sort of inter-company auto attendant connected like this: PBX (fxs) - Sipura 3k (fxo) - Asterisk -IAX- remote asterisk It works like this: Person picks up their phone and dials a number to get to the auto attendant (I don't have any FXO ports available on our PBX to do it the right way). The attendant answers and asks them the remote extension they want to dial. This setup has worked very well for several months. Last week I upgraded to 1.2.7.1 from 1.2.4 (I think). Since then I've been having trouble with the auto-attendant correctly detecting DTMF (missing digits). Some times it works flawlessly, others I have to try over and over before it is detected correctly. It isn't even consistently dropping the same digit from what I can see on the console. The only thing I've found is that I have a better chance of it working if I wait for the prompt to finish before dialing. I have changed the DTMF method from rfc2833 to info and finally inband with only a little change (inband seems to work the best). Has anyone else run into similar problems or have any more suggestions to try? This is the attendant portion of my extensions.conf: [inter-attendant] exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Set(TIMEOUT(response)=10) exten = s,4,Background(enter-ext-of-person) exten = i,1,Playback(invalid) exten = i,2,Goto(s,4) exten = i,3,Hangup exten = t,1,Playback(goodbye) exten = t,2,Hangup include = tests include = fullertonpbx include = intercompany Thank you for any insight you can provide. Dave Fullerton ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCE: Asterisk Jobs and Consulting Site
Yeah, I've had a project listed at asteriskhelpdesk.com for over a month and it still has 0 bids. I wouldn't waste my time redesigning the pages, they won't come... Bart - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, April 19, 2006 9:58 PM Subject: Re: [Asterisk-Users] ANNOUNCE: Asterisk Jobs and Consulting Site On Thursday 20 April 2006 00:13, Matt Gibson wrote: I would like to announce the availability of a new site dedicated to finding and creating jobs in the Asterisk VOIP field. I've created this site, after noticing there are no sites dedicated to providing quality job postings and hiring abilities to people in the field. You mean like http://www.asteriskhelpdesk.com? I'm sure there are a couple of others too I'm missing. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ring a grop of extension, then playback a file, then transfer to external number
Andre Courchesne - Consultant wrote: A call comes in from a Zap line. 5 SIP extension ring if nobody picks up, the call is transfered to a cell phone number. That works. I not want to add a playback of a file (Please waite while you are being transfered) before transfering the call to the cell phone. I did not completely understand what you are asking but have you tried something like: exten = 662,1,Dial(SIP/123Sip/124,24) exten = 662,n,playback(wait-while-try-cell-phone) exten = 662,n,Dial(Zap/g1/CellPhoneNumber) This would try ringing your sip devices for 24 seconds and if no one picked up would then play the message 'wait-while-try-cell-phone' and then would dial the cell phone. Don Pobanz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MGCP reinvite
2006/4/20, Nicolas TOUSSAINT [EMAIL PROTECTED]: I'm having troubles with the MGCP canreinvite option. I have two eyep media and Asterisk on the same network, same codecs, the option line canreinvite=yes properly set in the mgcp.conf file... but it won't do. Any idea anyone? I had many problems with MGCP protocol in Asterisk. My enterprise dropped out the asterisk solution in one site because mgcp didn't work. Phones stopped working without visible reason (no dialtone) having to restart asterisk to get them working again, and strange things like this. I never could find a way to download a sip firmware to update the mgcp devices. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] aterisk+h323 trunk?!
Tic Pavlin wrote: Good day to you all! I have been reading this mailing list for quite some time, and now, i do have a question. I have a working asterisk server with VOIP telephone number connected to it via SIP, and it works just fine. Now I am installing new server on new VoIP provider and provider only supports H323 trunks, but I havo no ide how to make it work. Unfortunatly they are not very open about shareing information, so I need someone who would be so nice to explain in short how it works and how to make it work. Thank you, Tic Pavlin, Neosystems d.o.o., Ljubljana, Slovenia ___ Hi, in asterisk there are 3 different h323 technologies. 1. h323 included with asterisk in asterisk-1.2.4/channels/h323 2. ooh323 included in asterisk-addons 3. oh323 (www.inaccessnetworks.com) there is various differences btween the 3. h323 uses asterisk's rtp stack, oh323,ooh323 uses its own install either 1. Then configure the conf(oh323.conf or h323.conf or ooh323.conf) . Will the Voip Provider be a h323 gatekeeper ?? Then in extensions.conf : if no gatekeeper: Dial(H323/[EMAIL PROTECTED]) if gatekeeper: Dial(H323/[EMAIL PROTECTED]) depending on what you install your Dial is Dial(H323...) or Dial (OOH323) -- thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to stop Asterisk picking up my incoming calls?
I was able to configure (Incoming Calls) through AMP to make asterisk answer my line after 3 rings and forward it to an extension. However, I was unable to disable that feature? In AMP, go on Maintenance-Config Edit In you zapata-auto.conf (assuming you used genzaptelconf), change the context for this line context=from-pstn to context=from-pstn-noanswer. Then, in extension_custom.conf add this context [from-pstn-noanswer] exten = s,1,Wait,3 ; Wait 3 seconds, to get callerid exten = s,2,Hangup this will make Asterisk ignore the incoming calls but at the same time will give you the list of callers in your CDR N.B.: you have to restart Asterisk to make those changes apply hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] avm b1with chan capi and siemens hipath
2006/4/20, Armin Schindler [EMAIL PROTECTED]: Did you try the /b option of Dial() with capi?This enables early-b3, whcih gives you progress tones from the ISDN line.Armin That was the reason! I though that i tested that option before but may be i made some mistake. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Jingle support - can we test the feature ?
we would like to build IM-Voice community for our students around Asterisk, Jingle, Jabber. Can we already test those features ? Anyone already running such setup? Any more info ? Have you looked at Wildfire ? http://www.jivesoftware.org/wildfire/ There is an Asterisk-plugin that update your status automagically when you're on the phone hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk service crashes
The service just crashed again. This time I ran asterisk cvvv. It looks like ogg_vorbis is where the problem is. Here are the last few lines from the dump: [app_system.so] = (Generic System() application) == Registered application 'TrySystem' == Registered application 'System' [app_nbscat.so] = (Silly NBS Stream Application) == Registered application 'NBScat' [app_md5.so] = (MD5 checksum applications) == Registered application 'MD5Check' == Registered application 'MD5' [app_macro.so] = (Extension Macros) == Registered application 'MacroExit' == Registered application 'MacroIf' == Registered application 'Macro' [format_ogg_vorbis.so] = (OGG/Vorbis audio) == Registered file format ogg_vorbis, extension(s) ogg Segmentation fault Does anyone have experience with this or know how to fix it? The only thing that seems to work is rebooting the server. Thanks, William From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josué Conti Sent: Wednesday, April 19, 2006 7:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk service crashes Try asterisk -g Regards Josué 2006/4/19, Gareth Blades [EMAIL PROTECTED]: Enter the 'dmesg' command. It displays a log of kernel messages etc... and may show up a problem. On Wed, 2006-04-19 at 03:03, [EMAIL PROTECTED] wrote: List, The past few days the asterisk service on my server has crashed several times. I have had it running for months and have made no changes to it. When it crashes, I am unable to make calls or gain access to the CLI. The service has been stopped. If I try to start it again (service asterisk start), it will start and run for a few seconds then crash again. After a reboot, it will run successfully for several hours before doing it again. Here is a ps aux of the services while the server is crashed.Does anyone see any service that would have a conflict with the asterisk service? FYI, the only cron I have running is a reboot scheduled once a week. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 21, Issue 113
Hi List!!Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support.I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf:[general]context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes ;domain=mydomain.tld ;domain=mydomain.tld,mydomain-incoming;domain=1.2.3.4 ;allowexternalinvites=no ;autodomain=yes ;pedantic=yes ;tos=184 ;tos=lowdelay ;maxexpiry=3600 ;defaultexpiry=120 ;notifymimetype=text/plain ;checkmwi=10 ;vmexten=voicemail ;videosupport=yes ;recordhistory=yesdisallow=all allow=g729allow=gsmallow=ulaw jitterbuffer=yes maxjitterbuffer=1500 ;allow=ilbc ;musicclass=default ;language=en ;relaxdtmf=yes rtptimeout=60 ;rtpholdtimeout=300 ;trustrpid = no ;sendrpid = yes ;progressinband=never ;useragent=Asterisk PBX ;promiscredir = no ;usereqphone = no dtmfmode = rfc2833 ;compactheaders = yes ;sipdebug = yes ;subscribecontext = default ;notifyringing = yes And these are the extensions:[]type=friendhost=dynamic dtmfmode=rfc2833username=secret=[2]type=friendhost=dynamicdtmfmode=rfc2833username=secret=As you can see I put the jitterbuffer, maxjitterbuffer and rtptimeout options. I think with this, the call has a huge improvement and I still reading about it. This is the CLI output with different commands: sip show peersName/username Host Dyn Nat ACL Port Statususuario2/usuario2 10.xxx.xxx.xxx D 5060 Unmonitoredusuario1/usuario1 10.xxx.xxx.xxx D 5060 Unmonitored2 sip peers [2 online , 0 offline]sip show usersUsername Secret Accountcode Def.Context ACL NATusuario2 usuario2 default No RFC3581 usuario1 usuario1 default No RFC3581--- (8 headers 0 lines)---Looking for 200.xxx..xxx in default (domain )Transmitting (no NAT) to 10.xxx.xxx.xxx :5060:SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 10.xxx.xxx.xxx;rport;branch=z9hG4bK0a0101e200104447938870d300d4;received=10.xxx.xxx.xxxFrom: sip:[EMAIL PROTECTED] ;tag=312051512495To: sip:200.xxx.xxx.xxx;tag=as767ed6bbCall-ID: [EMAIL PROTECTED]CSeq: 150 OPTIONS User-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYMax-Forwards: 70Contact: sip:200.xxx.xxx.xxxAccept: application/sdpContent-Length: 0But I have another question. Our users surf the Internet by cable modems and we have a CMTS Motorola BSR 1000 with QoS options. I know I can configure it to manage QoS but I don't know very well how to do it. If somebody knows any tutorial or experiences administrating this device, please let me know Thanks againCarlos Bernat Message: 8Date: Wed, 19 Apr 2006 15:46:21 -0500From: Cavanna, Richard [EMAIL PROTECTED]Subject: [Asterisk-Users] RE: Delayed voice for 10 secs To: asterisk-users@lists.digium.comMessage-ID:[EMAIL PROTECTED] Content-Type: text/plain; charset=us-asciiPlease post pertinent config files and a CLI output so the list can helpwith the 10 sec delayYou set codec selection in SIP.conf. This selects preferred codec from top to bottom as well as jitter buffer settings and the RTP timeout.Sip.confdisallow=allallow=g729allow=gsmallow=ulawjitterbuffer=yes;forcejitterbuffer=yesmaxjitterbuffer=1500rtptimeout=60 As for the DTMF issue try to use rfc2833in sip.conf define your extention[]username=type=friendsecret=Xqualify=noport=5060nat=yes[EMAIL PROTECTED]host=dynamic dtmfmode=rfc2833context=from-internalcanreinvite=nocallerid=device Rich ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDRs and billing
Ok, no problem, Ill do it with the AGI. Do I need to re-compile asterisk to support the AGI writing? or it goes by default? Thank you Joao Pereira Chris Mason (Lists) wrote: Joao Pereira wrote: Hello I configured Asterisk to put CDRs in the database like it was explained in: www.voip-info.org/wiki/view/Asterisk+cdr+pgsql What I want to know is how do the billing solutions (like Asterisk2Billing) work with Asterisk. The billing system just use the information that Asterisk puts in the CDR table? Or they connect directly to Asterisk? Or is Asterisk that has, before the Dial command, to put the information on the Asterisk2Billing tables? Asterisk2Billing requires you route the calls to its AGI, and it keeps its own database, so what you did is of no use for billing. I haven't found an application that bills from the CDRs, everything I found wanted to create the database entries. I think ASTPP can read your CDR, though. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDRs and billing
One billing solution that Works with your CDRs is AsterBill (www.cybexdev.com) for postpaid. (not opensource, you have to buy) It runs a cronjob to get the latest CDRs and bill each accountcode. For prepaid of course it does it with AGI. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Chris Mason (Lists) Enviado el: Thursday, April 20, 2006 5:54 AM Para: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] CDRs and billing Joao Pereira wrote: Hello I configured Asterisk to put CDRs in the database like it was explained in: www.voip-info.org/wiki/view/Asterisk+cdr+pgsql What I want to know is how do the billing solutions (like Asterisk2Billing) work with Asterisk. The billing system just use the information that Asterisk puts in the CDR table? Or they connect directly to Asterisk? Or is Asterisk that has, before the Dial command, to put the information on the Asterisk2Billing tables? Asterisk2Billing requires you route the calls to its AGI, and it keeps its own database, so what you did is of no use for billing. I haven't found an application that bills from the CDRs, everything I found wanted to create the database entries. I think ASTPP can read your CDR, though. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly
Bryan Boatright wrote: I too am experiencing DTMF problems with 1.2.7.1 that I did not experience with recent prior versions. I've backed up to version 1.2.6 and so far DTMF detection is working reliably (but that's only with about 10 calls worth of testing). I've only had problems over SIP channels. Zap channels did not have problems with 1.2.7.1. I do not have any IAX channels, so cannot comment on that. I know others tend to discount DTMF problems because of known problems with how Asterisk handles DTMF, but there does seem to be enough anecdotal evidence that something bad has recently happened to make things worse. Dave, would you mind trying version 1.2.6 to see if that also resolves your problems? I hate to say me too, but I have been experiencing some DTMF issues since 1.2.4. Have tried with 1.2.4, 1.2.6, and 1.2.7.1; all with the same result. No DTMF, regardless of SIP INFO, RFC2833, or inband(ulaw). This is on a inbound SIP trunks from Level3. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly
I have reverted back to 1.2.6 and set my sipuras to tx dtmf as info so I can see them with sip debug. I'll see if there is a difference and report on my findings in a couple days. -Dave Bryan Boatright wrote: I too am experiencing DTMF problems with 1.2.7.1 that I did not experience with recent prior versions. I've backed up to version 1.2.6 and so far DTMF detection is working reliably (but that's only with about 10 calls worth of testing). I've only had problems over SIP channels. Zap channels did not have problems with 1.2.7.1. I do not have any IAX channels, so cannot comment on that. I know others tend to discount DTMF problems because of known problems with how Asterisk handles DTMF, but there does seem to be enough anecdotal evidence that something bad has recently happened to make things worse. Dave, would you mind trying version 1.2.6 to see if that also resolves your problems? Dave Fullerton wrote: Greetings, I'm using asterisk to connect our three locations together with a sort of inter-company auto attendant connected like this: PBX (fxs) - Sipura 3k (fxo) - Asterisk -IAX- remote asterisk It works like this: Person picks up their phone and dials a number to get to the auto attendant (I don't have any FXO ports available on our PBX to do it the right way). The attendant answers and asks them the remote extension they want to dial. This setup has worked very well for several months. Last week I upgraded to 1.2.7.1 from 1.2.4 (I think). Since then I've been having trouble with the auto-attendant correctly detecting DTMF (missing digits). Some times it works flawlessly, others I have to try over and over before it is detected correctly. It isn't even consistently dropping the same digit from what I can see on the console. The only thing I've found is that I have a better chance of it working if I wait for the prompt to finish before dialing. I have changed the DTMF method from rfc2833 to info and finally inband with only a little change (inband seems to work the best). Has anyone else run into similar problems or have any more suggestions to try? This is the attendant portion of my extensions.conf: [inter-attendant] exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Set(TIMEOUT(response)=10) exten = s,4,Background(enter-ext-of-person) exten = i,1,Playback(invalid) exten = i,2,Goto(s,4) exten = i,3,Hangup exten = t,1,Playback(goodbye) exten = t,2,Hangup include = tests include = fullertonpbx include = intercompany Thank you for any insight you can provide. Dave Fullerton ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk service crashes
The service just crashed again. This time I ran asterisk cvvv. It looks like ogg_vorbis is where the problem is. Here are the last few lines from the dump: [app_system.so] = (Generic System() application) == Registered application 'TrySystem' == Registered application 'System' [app_nbscat.so] = (Silly NBS Stream Application) == Registered application 'NBScat' [app_md5.so] = (MD5 checksum applications) == Registered application 'MD5Check' == Registered application 'MD5' [app_macro.so] = (Extension Macros) == Registered application 'MacroExit' == Registered application 'MacroIf' == Registered application 'Macro' [format_ogg_vorbis.so] = (OGG/Vorbis audio) == Registered file format ogg_vorbis, extension(s) ogg Segmentation fault Does anyone have experience with this or know how to fix it? The only thing that seems to work is rebooting the server. Thanks, William From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josué Conti Sent: Wednesday, April 19, 2006 7:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk service crashes Try asterisk -g Regards Josué 2006/4/19, Gareth Blades [EMAIL PROTECTED]: Enter the 'dmesg' command. It displays a log of kernel messages etc... and may show up a problem. On Wed, 2006-04-19 at 03:03, [EMAIL PROTECTED] wrote: List, The past few days the asterisk service on my server has crashed several times. I have had it running for months and have made no changes to it. When it crashes, I am unable to make calls or gain access to the CLI. The service has been stopped. If I try to start it again (service asterisk start), it will start and run for a few seconds then crash again. After a reboot, it will run successfully for several hours before doing it again. Here is a ps aux of the services while the server is crashed. Does anyone see any service that would have a conflict with the asterisk service? FYI, the only cron I have running is a reboot scheduled once a week. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cubix Softphone + Asterisk 1.2.6
I've tried Idefisk and Cubix Softphones, and they both work fine, except for two issues: 1. Idefisk seems to have a longer delay between the time I can hit tones, and 2. Cubix, while can send DTMF faster, never actually connects to an Asterisk-dialed call -- I can't hear the party who answers. #2 has been asked but unanswered here: http://lists.digium.com/pipermail/asterisk-users/2006-February/139240.html I've got a weird problem with both Firefly iaxLite (both IAX softphones). They don't seem to stop ringing when an incoming call is make to them. If the call is answered the conversation starts both ways but the ringing sound still keeps going and the softphones keep displaying that a call is coming in (but they do not display that the call is answered). I read on the voip-info website that the fix for this with Firefly is to set jitterbuffer to no which I tried but it didn't work. Because the problem is with two IAX softphones I'm not sure whether its a configuration problem with the asterisk server or, by change, the same bug with both softphones. Has anyone else come up against this? Can you change the amount of time between DTMF in Idefisk? Can you modify a config to get Cubix to actually connect to a Dial()ed call? Beckman --- Peter Beckman Internet Guy [EMAIL PROTECTED] http://www.purplecow.com/ --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] does anyone know anything about chan_btp or btpd?
I've tried posting a simple message twice, regarding using chan_btp to dial a phone number through a Zap interface, but I've received no answers, and I can't seem to figure out how to do what I want (which seems to be a pretty typical use of chan_btp - I mean, isn't it used to dial peoples' phone numbers when their bluetooth device is not present?). I was wondering if anyone could point me in the direction of anyone who may know a bit about the chan_btp driver (developers, users, whatever). My original question is posted below in case anyone can help: Can anyone tell me how me to get asterisk to dial out a phone number using BTP when a bluetooth device is not detected? I can get BTP to dial to a SIP phone, but I can't get it to dial through a POTS phone line using the Zap interface.. I've tried putting the following under the clients section in /etc/asterisk/btp.conf: client =user,00:12:34:56:78:90,Zap/4/1234567890 and in extensions.conf: exten = 222,1,Playback(pls-hold-while-try) exten = 222,2,Dial(BTP/user,60,m) exten = 222,3,Hangup but asterisk doesn't dial the phone number 1234567890, it simply does: Zap/4-1 answered SIP/304-fc8a and then gives me a dial tone.. From btp.conf, it says: ;If a default channel is specified, we ; use that channel if nobody has found the bluetooth device. so it seems as though I can connect to a channel (in this case Zap/4), but I can't actually get the channel to dial the given phone number.. If anyone can tell me what I'm doing wrong, I would very much appreciate it. Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Jingle support - can we test the feature ?
- Original Message - From: Time Bandit [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 20, 2006 4:18 PM Subject: Re: [Asterisk-Users] Jingle support - can we test the feature ? we would like to build IM-Voice community for our students around Asterisk, Jingle, Jabber. Can we already test those features ? Anyone already running such setup? Any more info ? Have you looked at Wildfire ? http://www.jivesoftware.org/wildfire/ There is an Asterisk-plugin that update your status automagically when you're on the phone -- Hi, thanks for pointer. I know for that project, but reading about Jingle, Jabber and Asterisk integration it seems not so interesting for me at the moment... Regards, Rob. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel and zapata configuration
Hi I am trying to use asterisk with an Aculab card using ss7 protocol. i have a problem when configuring zaptel and zapata files. could you give me the right configuration of this files to get asterisk functionning with ss7 protocol? I hope that you could help me! thanks and best regards _ MSN Hotmail sur i-mode : envoyez et recevez des e-mails depuis votre téléphone portable ! http://www.msn.fr/hotmailimode/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MeetMe: lots of buffer overruns/underruns when connecting over IAX
Hello, Situation: I've got two asterisk 1.2.4 servers, connected to each other over the internet with IAX2 with about 20msec delay. One of the servers is hosting MeetMe. It's working fine as long as only SIP phones connected to the meetme server participate in the conference. As soon as a participant using IAX2 is connecting, lots and lots of buffer overruns and underruns are generated: Apr 20 17:37:27 NOTICE[20970]: app_meetme.c:1276 conf_run: Audio bytes: 640 Buffer size: 320 Apr 20 17:37:27 NOTICE[20970]: app_meetme.c:1276 conf_run: Audio bytes: 320 Buffer size: 640 Apr 20 17:37:27 NOTICE[20970]: app_meetme.c:1276 conf_run: Audio bytes: 640 Buffer size: 320 Apr 20 17:37:28 NOTICE[20970]: app_meetme.c:1276 conf_run: Audio bytes: 320 Buffer size: 640 Apr 20 17:37:28 NOTICE[20970]: app_meetme.c:1276 conf_run: Audio bytes: 640 Buffer size: 320 This always happens, even if there's only one participant in the conference. All phones are Supura/Linksys SPA-941 phones. Everything is working fine (users can talk to each other, voicemail is working, etc), exept for meetme. In meetme.conf I've got audiobuffers=32, which doesn't help. Any clue? -- Erik Hensema ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] attended transfer issue
Olivier Krief wrote: For the sake of behaviour consistency, maybe : - this unified transfer method (let's say U for unified) should be introduced in features.conf independently of previous t or T methods and it's up developpers to reuse, factorize or rewrite existing transfer code and as long as those 3 methods as supported, - and previous t or T methods should be droped sometimes later on to simplify code support. the t or T is used to determine whether someone can transfer a call, be it blink or attended, not HOW that transfer occurs. So the 'tT' discussion is completely separate from the how discussion. Don Pobanz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback(something,noanswer) on Zap?
Dmitry Ivanov wrote: Apparently, Playback(invalid,noanswer) does not work with Zap/PRI. Is this bug? Yes it does work. However, if your telco will not allow you to send 'early audio', then you can't do it. A better solution is to set the PRI hangup cause before dropping the incoming call; if you set the hangup cause to 'number not assigned' then your telco's switch will play its normal intercept message to the caller. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel and zapata configuration
Mongi LASSOUED wrote: I am trying to use asterisk with an Aculab card using ss7 protocol. i have a problem when configuring zaptel and zapata files. could you give me the right configuration of this files to get asterisk functionning with ss7 protocol? I hope that you could help me! Aculab cards do not use Zaptel. You will need to contact Aculab for help configuring their cards to work with Asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What's the best way to combine multiple VOIP lines into a single number?
Guys, thank you so much for the answers! So, if you don't mind, what are the service providers that you use? Mine does not allow multiple concurrent calls to the same number... and I don't think it offers the 'rollover' feature, either... Thanks once again, Leo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MeetMe: lots of buffer overruns/underruns when connecting over IAX
In article [EMAIL PROTECTED], Erik Hensema [EMAIL PROTECTED] wrote: Hello, Situation: I've got two asterisk 1.2.4 servers, connected to each other over the internet with IAX2 with about 20msec delay. One of the servers is hosting MeetMe. It's working fine as long as only SIP phones connected to the meetme server participate in the conference. As soon as a participant using IAX2 is connecting, lots and lots of buffer overruns and underruns are generated: Apr 20 17:37:27 NOTICE[20970]: app_meetme.c:1276 conf_run: Audio bytes: 640 Buffer size: 320 Apr 20 17:37:27 NOTICE[20970]: app_meetme.c:1276 conf_run: Audio bytes: 320 Buffer size: 640 Apr 20 17:37:27 NOTICE[20970]: app_meetme.c:1276 conf_run: Audio bytes: 640 Buffer size: 320 Apr 20 17:37:28 NOTICE[20970]: app_meetme.c:1276 conf_run: Audio bytes: 320 Buffer size: 640 Apr 20 17:37:28 NOTICE[20970]: app_meetme.c:1276 conf_run: Audio bytes: 640 Buffer size: 320 This always happens, even if there's only one participant in the conference. Those messages are not a part of the standard MeetMe. It looks like you are running a version that includes Dan Austin's dynamic buffer patch from Mantis bug #5697. That dynamic buffer patch is not really required. It was an attempt to get the pseudo-device to accept a whole frame in one write(), but this has been obviated by the non-blocking flag to careful_write(). Try updating to an unpatched 1.2.7 instead. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk FAX
Hi, How can we change the FROM address when Asterisk sends mail. For example it is sending [EMAIL PROTECTED] in FROM , I need to change to [EMAIL PROTECTED] Any help? Thanks Wazb ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk FAX
Open voicemail.conf Find serveremail=asterisk If it's commented, uncomment it Change it to the email address you want it to be. Aaron On Thu, 20 Apr 2006, Wasif wrote: Hi, How can we change the FROM address when Asterisk sends mail. For example it is sending [EMAIL PROTECTED] in FROM , I need to change to [EMAIL PROTECTED] Any help? Thanks Wazb ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM2400P
We just bougth a tdm2400p with all the modules for FXO, but we are having some troubles with the card, cause it aparently is stripping some digits from the dialed number, we tested the same server with a tdm400 and everything worked as expected. We´ve already added w before the dialed number with no results, is there any way to solve, is it a bug thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] still some moh troubles
Bart van Daal wrote: Hi, After following the suggestions on the mailing lists and the wiki I'm still experiencing choppy moh. The song plays but with frequent noise parts. - I'm using asterisk 1.2.4 on our production server and 1.2.7 on the test server. - native moh with .gsm and .pcm formats (according to Doug Lytle wrote: Actually, you'll want to use ulaw for Native MOH. CUT #!/bin/sh for filename in *mp3 do eval filename=`echo $filename | cut -f1 -d.` echo Converting $filename sox -V $filename.mp3 -t au -r 8000 -U -b -c 1 $filename.ulaw resample -ql done CUT Doug, The required formats for native MOH are entirely dependent on the codecs being used for calls. As such, there is no single format that is the right one. Looking at the original post, I'd assume Bart is using both the GSM and ULAW codecs on his system. If that is the case, he should transcode all of his MOH files to both GSM and PCM formats. Asterisk will take care of matching the formats to the codecs being used on each individual call. In addition to this, the two methods mentioned in this thread for transcoding files are almost identical. They both produce a file in the ULAW format. To demonstrate this, I took the same WAV file and ran the following two commands against it: sox test.wav -t au -r 8000 -U -b -c 1 test.ulaw sox test.wav -t ul -r 8000 -b -c 1 test.pcm The only difference between test.ulaw and test.pcm is that the former has a 41 byte header and the latter doesn't. Discounting the unnecessary header, the files are identical. As for the OP, I'm not sure what the source of his problem is. There is a lot of information out there about MOH, some of it good, some of it bad. Hopefully, someone that has experienced a problem similar to his can help him out. It would be helpful if we were sure of the codecs he is using. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Announcement System for a Charity
I'm putting together an Asterisk server for a local charity to use as an announcement system. I've been thinking about how to write the dialplan to allow different options for different groups' announcements, as well as mailboxes for the various groups and the charity's administrators. Of course, this would also need to include an option for the heads of the different groups to modify their announcements. Before I write it, I was wondering if anyone had an extensive dialplan or an AGI script that already did something like this. I know it'll only take a couple of hours to write and test this, but I thought if someone has something already written, I could just borrow it from you. Thanks, Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] enablling Te110p with PRI
Hi gurus... I have connected an asterisk with a te110p/pri to a GSM ericsson switch, all apears to be write. But when i try to make an outbond call from asterisk to the te110p group, the folowing error is logged: -- Executing Dial(SIP/201-5923, ZAP/1-1/0971200152|20|r) in new stack == Everyone is busy/congested at this time (1:0/0/1) Question: Is there a how to connect the Asterisk to an ericsson sw? What other test can i do against the switch?. Thanks in advance... this is the te110p configuration... asterisk1*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 asterisk1*CLI zap show channel 1 Channel: 1CLI File Descriptor: 19 Span: 1k1*CLI Extension: LI Dialing: noLI Context: default Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 1 Signalling Type: PRI Signalling Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF PRI Flags: PRI Logical Span: Implicit Hookstate (FXS only): Onhook asterisk1*CLI [EMAIL PROTECTED] asterisk]# cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS/CRC4 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear (In use) 3 WCT1/0/3 Clear (In use) 4 WCT1/0/4 Clear (In use) 5 WCT1/0/5 Clear (In use) 6 WCT1/0/6 Clear (In use) 7 WCT1/0/7 Clear (In use) 8 WCT1/0/8 Clear (In use) 9 WCT1/0/9 Clear (In use) 10 WCT1/0/10 Clear (In use) 11 WCT1/0/11 Clear (In use) 12 WCT1/0/12 Clear (In use) 13 WCT1/0/13 Clear (In use) 14 WCT1/0/14 Clear (In use) 15 WCT1/0/15 Clear (In use) 16 WCT1/0/16 HDLCFCS (In use) 17 WCT1/0/17 Clear (In use) 18 WCT1/0/18 Clear (In use) 19 WCT1/0/19 Clear (In use) 20 WCT1/0/20 Clear (In use) 21 WCT1/0/21 Clear (In use) 22 WCT1/0/22 Clear (In use) 23 WCT1/0/23 Clear (In use) 24 WCT1/0/24 Clear (In use) 25 WCT1/0/25 Clear (In use) 26 WCT1/0/26 Clear (In use) 27 WCT1/0/27 Clear (In use) 28 WCT1/0/28 Clear (In use) 29 WCT1/0/29 Clear (In use) 30 WCT1/0/30 Clear (In use) 31 WCT1/0/31 Clear (In use) AVISO LEGAL: Esta información es privada y confidencial y está dirigida únicamente a su destinatario. Si usted no es el destinatario original de este mensaje y por este medio pudo acceder a dicha información por favor elimine el mensaje. La distribución o copia de este mensaje está estrictamente prohibida. Esta comunicación es sólo para propósitos de información y no debe ser considerada como propuesta, aceptación ni como una declaración de voluntad oficial de NUCLEO S.A. La transmisión de e-mails no garantiza que el correo electrónico sea seguro o libre de error. Por consiguiente, no manifestamos que esta información sea completa o precisa. Toda información está sujeta a alterarse sin previo aviso. . This information is private and confidential and intended for the recipient only. If you are not the intended recipient of this message you are hereby notified that any review, dissemination, distribution or copying of this message is strictly prohibited. This communication is for information purposes only and shall not be regarded neither as a proposal, acceptance nor as a statement of will or official statement from NUCLEO S.A. . Email transmission cannot be guaranteed to be secure or error-free. Therefore, we do not represent that this information is complete or accurate and it should not be relied upon as such. All information is subject to change without notice. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Suggestion Request: Coloc Provider in Chicago, IL area
Hello all! I always prefer to get referrals from fellow professionals, and this is such a request. I'm looking for the following: 1. Colocation providers in the chicago area to store a small server for the purpose of setting up a VOIP service (including pstn connection via Digium cards) for between 100-10,000 users. Obviously value is a big part, but reliability and network speed are also factors, as well as whether or not they'd let the client have T1/T3 lines for his PSTN connects. 2. PSTN-SIP OR PSTN-IAX providers that client can optionally use instead of Digium T# interface cards. (Network proximity to Chicago area a MUST, obviously) I thank all of you in advance, your experience and feelings towards people you've dealt with in the past are far more insightful than reading company websites. Cheers all! Sherwood McGowan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Announcement System for a Charity
Why not use [EMAIL PROTECTED] It's got the AMP/FreePBX already installed, so it'd be easy for them to maintain, and should do what you want.. -Original Message- From: Nabeel Jafferali [mailto:[EMAIL PROTECTED] Sent: Thursday, April 20, 2006 2:40 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Announcement System for a Charity I'm putting together an Asterisk server for a local charity to use as an announcement system. I've been thinking about how to write the dialplan to allow different options for different groups' announcements, as well as mailboxes for the various groups and the charity's administrators. Of course, this would also need to include an option for the heads of the different groups to modify their announcements. Before I write it, I was wondering if anyone had an extensive dialplan or an AGI script that already did something like this. I know it'll only take a couple of hours to write and test this, but I thought if someone has something already written, I could just borrow it from you. Thanks, Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] still some moh troubles
Use madplay. I tried everything and madplay works the best. -Original Message- From: Bart van Daal [mailto:[EMAIL PROTECTED] Sent: Thu 4/20/2006 6:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: Subject: [Asterisk-Users] still some moh troubles Hi, After following the suggestions on the mailing lists and the wiki I'm still experiencing choppy moh. The song plays but with frequent noise parts. - I'm using asterisk 1.2.4 on our production server and 1.2.7 on the test server. - native moh with .gsm and .pcm formats (according to http://astrecipes.net/?n=152) - compiled ztdummy as a timing source any pointers on how to dig deeper into the problem or remedy it are very much appreciated regards, Bart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Announcement System for a Charity
Why not use [EMAIL PROTECTED] It's got the AMP/FreePBX already installed, so it'd be easy for them to maintain, and should do what you want.. I considered using [EMAIL PROTECTED], but the installation did not detect my network card and I kind of gave up. However, regardless, does [EMAIL PROTECTED] have built-in functionality to have a hidden menu for a user to modify a recorded file (i.e. a file played as an option on the IVR). And I don't mean from the [EMAIL PROTECTED] web interface, I mean an option that can be added to the IVR menu. Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM2400P
blackgecko wrote: We just bougth a tdm2400p with all the modules for FXO, but we are having some troubles with the card, cause it aparently is stripping some digits from the dialed number, we tested the same server with a tdm400 and everything worked as expected. We´ve already added w before the dialed number with no results, is there any way to solve, is it a bug Asterisk does NOT listen for dialtone before dialing. Many consider that a bug, or a design defect. Unfortunately no one who has the skills in coding to fix that sees it as an issue. Multiple w's may fix it if that is really the problem. Remember that w to wait before dialing ONLY works in DTMF. If you are forced or want to use pulse dialing, too bad. w doesn't work in that case. You may want to monitor a line while dialing out and see if that is really the problem, though I would think it should be there with the TDM400 as well. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Announcement System for a Charity
Does AMP also let you split up each charity so that each only has access to manage their own content? That seems to me to be a pretty big limitation of all the Asterisk management software out there. It's designed to be used by one company to manage their own config, not to be used by many 'organisations' to manage their own data. Kind of like Asterisk being used as a carrier solution rather than a hosted PBX solution. Doug. -Original Message- From: Nabeel Jafferali [mailto:[EMAIL PROTECTED] Sent: Thursday, April 20, 2006 1:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Announcement System for a Charity Why not use [EMAIL PROTECTED] It's got the AMP/FreePBX already installed, so it'd be easy for them to maintain, and should do what you want.. I considered using [EMAIL PROTECTED], but the installation did not detect my network card and I kind of gave up. However, regardless, does [EMAIL PROTECTED] have built-in functionality to have a hidden menu for a user to modify a recorded file (i.e. a file played as an option on the IVR). And I don't mean from the [EMAIL PROTECTED] web interface, I mean an option that can be added to the IVR menu. Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] attended transfer issue
Olivier Krief wrote: snip So it should be implemented in Asterisk and it's up to Polycom, Snom and others to design phones that at least do not prevent people to use # sign based unified transfer method if they wish to. I would HOPE that either the transfer key could be reprogrammed or the transfer function in Asterisk could be changed to match one another, and it also work with other than SIP phones. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Announcement System for a Charity
Does AMP also let you split up each charity so that each only has access to manage their own content? That seems to me to be a pretty big limitation of all the Asterisk management software out there. It's designed to be used by one company to manage their own config, not to be used by many 'organisations' to manage their own data. Kind of like Asterisk being used as a carrier solution rather than a hosted PBX solution. In my situation, that is not an issue because the only modifiable part of this installation needs to be IVR-accessible and is only to record announcements for different groups by the respective groups. However, I see your point. They need a sort of tenant capability. Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enablling Te110p with PRI
Rafael Visser wrote: Hi gurus... I have connected an asterisk with a te110p/pri to a GSM ericsson switch, all apears to be write. But when i try to make an outbond call from asterisk to the te110p group, the folowing error is logged: -- Executing Dial(SIP/201-5923, ZAP/1-1/0971200152|20|r) in new stack == Everyone is busy/congested at this time (1:0/0/1) Question: Is there a how to connect the Asterisk to an ericsson sw? What other test can i do against the switch?. Thanks in advance... What does the exact Dial line look like in your extensions.conf? Is 0971200152 the number that the other end is expecting? For instance, our Shoretel requires the country code be added, for instance 1503XXX. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MeetMe: lots of buffer overruns/underruns when connecting over IAX
On Thursday 20 April 2006 19:07, Tony Mountifield wrote: In article [EMAIL PROTECTED], Erik Hensema [EMAIL PROTECTED] wrote: One of the servers is hosting MeetMe. It's working fine as long as only SIP phones connected to the meetme server participate in the conference. As soon as a participant using IAX2 is connecting, lots and lots of buffer overruns and underruns are generated: [...] Those messages are not a part of the standard MeetMe. It looks like you are running a version that includes Dan Austin's dynamic buffer patch from Mantis bug #5697. That dynamic buffer patch is not really required. It was an attempt to get the pseudo-device to accept a whole frame in one write(), but this has been obviated by the non-blocking flag to careful_write(). Try updating to an unpatched 1.2.7 instead. Thanks, that solved it. I was running a bristuffed 1.2.4 which seemingly also includes that patch. However, zaphfc seems broken beyond repair so I'm ditching my hfc card and I'm going to use another isdn card. I'm now running an unpatched asterisk again. -- Erik Hensema ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Announcement System for a Charity
NerdVittles.com has a dialout announcement system article. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Thursday, April 20, 2006 11:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Announcement System for a Charity I'm putting together an Asterisk server for a local charity to use as an announcement system. I've been thinking about how to write the dialplan to allow different options for different groups' announcements, as well as mailboxes for the various groups and the charity's administrators. Of course, this would also need to include an option for the heads of the different groups to modify their announcements. Before I write it, I was wondering if anyone had an extensive dialplan or an AGI script that already did something like this. I know it'll only take a couple of hours to write and test this, but I thought if someone has something already written, I could just borrow it from you. Thanks, Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Won't start after SVN Trunk Update
Hi: I deleted old modules in /usr/lib/asterisk/modules before make install. I built zaptel and libpri before asterisk. Modprobe zaptel and modprobe -v wctdm executed witiout complaint. Starting asterisk produced the output below with several warnings and a failure. Can someone help, please. I double-spaced the warnings in the text below. The first warning is about music on hold because it gets depricated because I turned off chan modem loading. When it loads it fails to find a common object as well. It seems like I'm missing one or more module that may have remained from an older build, but I'm not sure what I haven't deleted from the earlier builds. It is probably obvious to a few of you. Thanks very much in advance for your help. _ ScottSuSE:/usr/src/asterisk # asterisk -vgc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk SVN-trunk-r21707, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk Dynamic Loader loading preload modules: == Parsing '/etc/asterisk/modules.conf': Found == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxStatus == Manager registered action MailboxCount == Manager registered action ListCommands == Manager registered action WaitEvent == Parsing '/etc/asterisk/manager.conf': Found Apr 20 08:27:58 NOTICE[13559]: cdr.c:1116 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 == UDPTL allocating from port range 4500 - 4999 Asterisk PBX Core Initializing Registering builtin applications: [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [ExecIfTime] == Registered application 'ExecIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [Set] == Registered application 'Set' [ImportVar] == Registered application 'ImportVar' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [res_musiconhold.so]Apr 20 08:27:58 WARNING[13559]: loader.c:726 __load_resource: new style res_musiconhold.so (0x30) loaded RTLD_LOCAL = (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 Apr 20 08:27:58 WARNING[13559]: res_musiconhold.c:1007 load_moh_classes: The old musiconhold.conf syntax has been deprecated! Please refer to the sample configuration for information on the new syntax. [res_indications.so]Apr 20 08:27:58 WARNING[13559]: loader.c:726 __load_resource: new style res_indications.so (0x10) loaded RTLD_LOCAL = (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' -- Registered indication country 'de'
[Asterisk-Users] Asterisk FAx-to-Email
Hi, I get error when my DID hit to asterisk box which I am using for FAX to Email Service. Sometimes Fax goes through but mostly I get communication error on Fax Machine and on Asterisk I get Comfort noise support incomplete in Asterisk (RFC 3389) error. I am using SIP with G711. My Did provider cannot turn off VAD and Echo from his side, so is there any option or setting I can do at my side to make FAX service more reliable Thanks Wazb ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Won't start after SVN Trunk Update
Fred Noris wrote: Hi: I deleted old modules in /usr/lib/asterisk/modules before make install. I built zaptel and libpri before asterisk. Modprobe zaptel and modprobe -v wctdm executed witiout complaint. Starting asterisk produced the output below with several warnings and a failure. Can someone help, please. I double-spaced the warnings in the text below. The first warning is about music on hold because it gets depricated because I turned off chan modem loading. When it loads it fails to find a common object as well. It seems like I'm missing one or more module that may have remained from an older build, but I'm not sure what I haven't deleted from the earlier builds. It is probably obvious to a few of you. Thanks very much in advance for your help. _ ScottSuSE:/usr/src/asterisk # asterisk -vgc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk SVN-trunk-r21707, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk Dynamic Loader loading preload modules: == Parsing '/etc/asterisk/modules.conf': Found == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxStatus == Manager registered action MailboxCount == Manager registered action ListCommands == Manager registered action WaitEvent == Parsing '/etc/asterisk/manager.conf': Found Apr 20 08:27:58 NOTICE[13559]: cdr.c:1116 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 == UDPTL allocating from port range 4500 - 4999 Asterisk PBX Core Initializing Registering builtin applications: [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [ExecIfTime] == Registered application 'ExecIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [Set] == Registered application 'Set' [ImportVar] == Registered application 'ImportVar' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [res_musiconhold.so]Apr 20 08:27:58 WARNING[13559]: loader.c:726 __load_resource: new style res_musiconhold.so (0x30) loaded RTLD_LOCAL = (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 Apr 20 08:27:58 WARNING[13559]: res_musiconhold.c:1007 load_moh_classes: The old musiconhold.conf syntax has been deprecated! Please refer to the sample configuration for information on the new syntax. [res_indications.so]Apr 20 08:27:58 WARNING[13559]: loader.c:726 __load_resource: new style res_indications.so (0x10) loaded RTLD_LOCAL = (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' -- Registered
Re: [Asterisk-Users] Asterisk Won't start after SVN Trunk Update
Fred Noris wrote: [res_snmp.so]Apr 20 08:27:58 WARNING[13559]: loader.c:726 __load_resource: new style res_snmp.so (0x0) loaded RTLD_LOCAL Apr 20 08:27:58 WARNING[13559]: loader.c:744 __load_resource: Key routine returned NULL in module /usr/lib/asterisk/modules/res_snmp.so Apr 20 08:27:58 WARNING[13559]: loader.c:753 __load_resource: 5 errors loading module /usr/lib/asterisk/modules/res_snmp.so, aborted Apr 20 08:27:58 WARNING[13559]: loader.c:850 print_and_load: Loading module res_snmp.so failed! These messages mean that the module in question hasn't been updated yet to use the new module loader. So you can't use that module with SVN-trunk until somebody does so. Otherwise, the system is usable; I am running it even though a module or two hasn't been upgraded and I see those messages on startup. B. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Won't start after SVN Trunk Update
Fred Noris wrote: Hi: I deleted old modules in /usr/lib/asterisk/modules before make install. I built zaptel and libpri before asterisk. Modprobe zaptel and modprobe -v wctdm executed witiout complaint. Starting asterisk produced the output below with several warnings and a failure. Can someone help, please. I double-spaced the warnings in the text below. The first warning is about music on hold because it gets depricated because I turned off chan modem loading. When it loads it fails to find a common object as well. It seems like I'm missing one or more module that may have remained from an older build, but I'm not sure what I haven't deleted from the earlier builds. It is probably obvious to a few of you. Thanks very much in advance for your help. _ ScottSuSE:/usr/src/asterisk # asterisk -vgc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk SVN-trunk-r21707, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk Dynamic Loader loading preload modules: == Parsing '/etc/asterisk/modules.conf': Found == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxStatus == Manager registered action MailboxCount == Manager registered action ListCommands == Manager registered action WaitEvent == Parsing '/etc/asterisk/manager.conf': Found Apr 20 08:27:58 NOTICE[13559]: cdr.c:1116 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 == UDPTL allocating from port range 4500 - 4999 Asterisk PBX Core Initializing Registering builtin applications: [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [ExecIfTime] == Registered application 'ExecIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [Set] == Registered application 'Set' [ImportVar] == Registered application 'ImportVar' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [res_musiconhold.so]Apr 20 08:27:58 WARNING[13559]: loader.c:726 __load_resource: new style res_musiconhold.so (0x30) loaded RTLD_LOCAL = (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 Apr 20 08:27:58 WARNING[13559]: res_musiconhold.c:1007 load_moh_classes: The old musiconhold.conf syntax has been deprecated! Please refer to the sample configuration for information on the new syntax. [res_indications.so]Apr 20 08:27:58 WARNING[13559]: loader.c:726 __load_resource: new style res_indications.so (0x10) loaded RTLD_LOCAL = (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' -- Registered
[Asterisk-Users] channels change names
I'm writing a php script to dial numbers and connect them to a conference. This is fairly straightforward: Action: originate Channel: Local/[EMAIL PROTECTED] Context: default Exten: $extension Priority: 1 This is pretty straightforward. However, the script then loads the list of members in the conference (using the meetme list ... command). For local extensions this works fine - the list of members shows the right channels, etc. The problem I'm having is that if the extension is external, the conference list shows a Local/$extension channel at the start, and then once the call is completed, it changes the channel to whatever was dialed. I'm probably not explaining it properly, but what I'd like to have happen is that I get one consistent channel name from the start of the connection - it doesn't matter what it is, as long as it doesn't change. As things stand, the conference list isn't accurate, unless I wait about 5 seconds after adding someone before updating the list. Thanks for any suggestions you might have here! -- Jon-o Addleman - http://redowl.dyndns.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enablling Te110p with PRI
Sorry, at the switch side, the spam is ABL (automatic blocked), is like to be Not Aligned. Im not sure about the required parameters to configure the ericsson with isdn-pri. So, lets just wait if someone help me with the isdn config first.. Thanks. Steven Ringwald [EMAIL PROTECTED]@lists.digium.com con fecha 20/04/2006 03:47:06 p.m. Por favor, responda a [EMAIL PROTECTED]; Por favor, responda a Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Enviado por: [EMAIL PROTECTED] Destinatarios:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com CC: Asunto: Re: [Asterisk-Users] enablling Te110p with PRI Rafael Visser wrote: Hi gurus... I have connected an asterisk with a te110p/pri to a GSM ericsson switch, all apears to be write. But when i try to make an outbond call from asterisk to the te110p group, the folowing error is logged: -- Executing Dial(SIP/201-5923, ZAP/1-1/0971200152|20|r) in new stack == Everyone is busy/congested at this time (1:0/0/1) Question: Is there a how to connect the Asterisk to an ericsson sw? What other test can i do against the switch?. Thanks in advance... What does the exact Dial line look like in your extensions.conf? Is 0971200152 the number that the other end is expecting? For instance, our Shoretel requires the country code be added, for instance 1503XXX. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Visite nuestro Sitio http://www.personal.com.py AVISO LEGAL: Esta información es privada y confidencial y está dirigida únicamente a su destinatario. Si usted no es el destinatario original de este mensaje y por este medio pudo acceder a dicha información por favor elimine el mensaje. La distribución o copia de este mensaje está estrictamente prohibida. Esta comunicación es sólo para propósitos de información y no debe ser considerada como propuesta, aceptación ni como una declaración de voluntad oficial de NUCLEO S.A. La transmisión de e-mails no garantiza que el correo electrónico sea seguro o libre de error. Por consiguiente, no manifestamos que esta información sea completa o precisa. Toda información está sujeta a alterarse sin previo aviso. . This information is private and confidential and intended for the recipient only. If you are not the intended recipient of this message you are hereby notified that any review, dissemination, distribution or copying of this message is strictly prohibited. This communication is for information purposes only and shall not be regarded neither as a proposal, acceptance nor as a statement of will or official statement from NUCLEO S.A. . Email transmission cannot be guaranteed to be secure or error-free. Therefore, we do not represent that this information is complete or accurate and it should not be relied upon as such. All information is subject to change without notice. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with TE110P configuration
Hello list! I have a TE110P card installed with asterisk at home 2.0 When I try to make a call, I have the following error in the /var/log/asterisk/full Apr 19 17:53:34 WARNING[14304] chan_unicall.c: Unicall/1 event Protocol failure Apr 19 17:53:34 VERBOSE[14304] logger.c: -- Unicall/1 protocol error. Cause 32771 Apr 19 17:53:34 WARNING[14304] chan_unicall.c: MFC/R2 UniCall/1 Channel echo cancel Apr 19 17:53:34 DEBUG[14304] chan_unicall.c: disabled echo cancellation on channel 1 Apr 19 17:53:34 WARNING[14304] chan_unicall.c: MFC/R2 UniCall/1 - 1001 [1/ 1/Idle /Idle ] Apr 19 17:53:34 WARNING[14304] chan_unicall.c: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Apr 19 17:53:34 WARNING[14304] chan_unicall.c: MFC/R2 UniCall/2 - 1001 [2/ 2/Seize ack /Seize ack ] Apr 19 17:53:34 WARNING[14304] chan_unicall.c: MFC/R2 UniCall/2 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 My unicall.conf is [EMAIL PROTECTED] asterisk]# cat unicall.conf [channels] usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=co-land,30,1,16 protocolend=cpe group = 1 loglevel = 255 context= e1-incoming channel = 1-15 channel = 17-31 ;skip time slot 16 and my zaptel.conf is # MFC/R2 normalmente no usa CRC4 span=1,1,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101 loadzone=us defaultzone=us Somebody could help me? thanks to all Max ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queues and the '*' key
[EMAIL PROTECTED] asterisk]# asterisk -V Asterisk SVN-branch-1.2-r8632M I was wondering if there was some documentation I was missing on the '*' key and queues. I have my features setup to use *x, where x is a #, but these features don't work for calls coming in from a queue. As soon as the '*' button is hit, the call is disconnected. I have a vague memory of reading about this somewhere, but searched @ the wiki AND through google aren't turning up anything useful. Sean begin:vcard fn:Sean Kennedy n:Kennedy;Sean org:Rickey Wong DDS Inc adr;dom:A115;;2937 Veneman Ave;Modesto;CA;95356 email;internet:[EMAIL PROTECTED] title:Chief Information Officer tel;work:209-577-0777 x44 tel;fax:209-529-3209 tel;cell:209-485-2834 x-mozilla-html:TRUE url:http://www.qualitydentists.com version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agents and Realtime
Is there a way to get the agents.conf file from a realtime database or at least use the realtime static format? -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Won't start after SVN Trunk Update
Hi Brian: Thanks for that. So, I assume I just need to edit modules.conf and put noload statements for the offending modules? --to get * to run, that is, as it is not. At 01:31 PM 4/20/2006, you wrote: Fred Noris wrote: [res_snmp.so]Apr 20 08:27:58 WARNING[13559]: loader.c:726 __load_resource: new style res_snmp.so (0x0) loaded RTLD_LOCAL Apr 20 08:27:58 WARNING[13559]: loader.c:744 __load_resource: Key routine returned NULL in module /usr/lib/asterisk/modules/res_snmp.so Apr 20 08:27:58 WARNING[13559]: loader.c:753 __load_resource: 5 errors loading module /usr/lib/asterisk/modules/res_snmp.so, aborted Apr 20 08:27:58 WARNING[13559]: loader.c:850 print_and_load: Loading module res_snmp.so failed! These messages mean that the module in question hasn't been updated yet to use the new module loader. So you can't use that module with SVN-trunk until somebody does so. Otherwise, the system is usable; I am running it even though a module or two hasn't been upgraded and I see those messages on startup. B. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Background() and Read()
I'm having some issues with Background() and Read() commands.
See the example below. This is when I wait for Background to finish playing the
sound file, before entering '12345#'.
All works fine.
hestia*CLI
-- Executing Answer(SIP/2944093-3366, ) in new stack
-- Executing Wait(SIP/2944093-3366, 1) in new stack
-- Executing BackGround(SIP/2944093-3366, if-u-know-ext-dial) in new
stack
-- Playing 'if-u-know-ext-dial' (language 'en')
-- Executing Read(SIP/2944093-3366, number||) in new stack
-- User entered '12345'
-- Executing NoOp(SIP/2944093-3366, 12345) in new stack
== Auto fallthrough, channel 'SIP/2944093-3366' status is 'UNKNOWN'
However, if I start to enter digits before Background() is finished, background
stops playing the file, and nothing happens after this point. I keep hitting #
and still no reply. It didn't even execute the Read().
hestia*CLI
-- Executing Answer(SIP/2944093-6437, ) in new stack
-- Executing Wait(SIP/2944093-6437, 1) in new stack
-- Executing BackGround(SIP/2944093-6437, if-u-know-ext-dial) in new
stack
-- Playing 'if-u-know-ext-dial' (language 'en')
Here's extensions.conf:
exten = 1000,1,Answer
exten = 1000,2,Wait,1
exten = 1000,3,Background(if-u-know-ext-dial)
exten = 1000,4,Read(number||)
exten = 1000,5,NoOp(${number})
Anyone got any ideas?
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Re: [Asterisk-Users] enablling Te110p with PRI
Rafael Visser wrote: Hi gurus... I have connected an asterisk with a te110p/pri to a GSM ericsson switch, all apears to be write. But when i try to make an outbond call from asterisk to the te110p group, the folowing error is logged: -- Executing Dial(SIP/201-5923, ZAP/1-1/0971200152|20|r) in new stack == Everyone is busy/congested at this time (1:0/0/1) One thing that is obvious here is you are using 1-1 instead of 1. A better thing would be to use groups. Check /etc/asterisk/zapata.conf and /etc/asterisk/extensions.conf Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom MWI
I have tried everything from voip-info and I still cant get the Polycom 501/601 to display the MWI indicator light. Everything else works just fine. I am using FreePBX set to users and devices mode. Here is the MWI section of the phonexxx.cfg file: mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="*97" msg.mwi.2.subscribe="" msg.mwi.2.callBackMode="disabled" msg.mwi.2.callBack="" msg.mwi.3.subscribe="" msg.mwi.3.callBackMode="disabled" msg.mwi.3.callBack="" msg.mwi.4.subscribe="" msg.mwi.4.callBackMode="disabled" msg.mwi.4.callBack="" msg.mwi.5.subscribe="" msg.mwi.5.callBackMode="disabled" msg.mwi.5.callBack="" msg.mwi.6.subscribe="" msg.mwi.6.callBackMode="disabled" msg.mwi.6.callBack=""/ /msg i have also tried msg.mwi.1.callBackMode="register" Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call recording
Hi all, Is there a way to record a call conversation starting in the middle of the call? I know I can recording whole conversation with mixmonitor, but I prefer only recording certain part of the conversation. Thnx. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk (RFC 3389)
Hi, I am getting this message when my DID hit to asterisk box which I am using for FAX to Email Service. Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible Any cure for that. Thanks Wazb ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound calls are failing
Thanks
Inserting a w did resolve the problem. I saw another post from
today where somebody else is having the same problem with a
TDM2400P. Hopefully someday Asterisk will be coded to wait for a dial tone.
nb
On 4/19/06, Time Bandit [EMAIL PROTECTED] wrote:
When dialing an outbound number, sometimes all the digits are not dialed properly on the outside line. In the dial plan I added a SayDigits to the outbound rule and it properly reads back the entire number that was entered
on the phone before dialing.Is asterisk dialing too quickly, is there anyway to insert a pause or wait for a dial tone on the external line?* is probably starting to dial too fast. Try to add a w in your dial
string to make it wait.Like : Dial(ZAP/g0,w${EXTEN})w adds half a second pause. You can put more w to make it wait longer.hth___--Bandwidth and Colocation provided by
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RE: [Asterisk-Users] Polycom MWI
Put your voicemailbox number (usually extension) in the 1.subscribe field. Bill From: [EMAIL PROTECTED] on behalf of Kerry Garrison Sent: Thu 4/20/2006 7:32 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Polycom MWI I have tried everything from voip-info and I still cant get the Polycom 501/601 to display the MWI indicator light. Everything else works just fine. I am using FreePBX set to users and devices mode. Here is the MWI section of the phonexxx.cfg file: mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=*97 msg.mwi.2.subscribe= msg.mwi.2.callBackMode=disabled msg.mwi.2.callBack= msg.mwi.3.subscribe= msg.mwi.3.callBackMode=disabled msg.mwi.3.callBack= msg.mwi.4.subscribe= msg.mwi.4.callBackMode=disabled msg.mwi.4.callBack= msg.mwi.5.subscribe= msg.mwi.5.callBackMode=disabled msg.mwi.5.callBack= msg.mwi.6.subscribe= msg.mwi.6.callBackMode=disabled msg.mwi.6.callBack=/ /msg i have also tried msg.mwi.1.callBackMode=register Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.techdatapros.com http://www.techdatapros.com/ winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call recording
On Thu, Apr 20, 2006 at 07:41:48PM -0400, Wai Wu spake thusly: Hi all, Is there a way to record a call conversation starting in the middle of the call? I know I can recording whole conversation with mixmonitor, but I prefer only recording certain part of the conversation. Thnx. From http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial: # w: Allow the called user to start recording after pressing *1 or what # defined in features.conf (Asterisk v1.2.x); requires # Set(DYNAMIC_FEATURES=automon) # W: Allow the calling user to start recording after pressing *1 or what # defined in features.conf (Asterisk v1.2.x); requires # Set(DYNAMIC_FEATURES=automon) See the rest of that page for more about it. I haven't used it myself, but it looks like what you need! -- Jon-o Addleman - http://redowl.dyndns.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium cards, so disappointing !
Our production asterisk server has TE411P and we route close to 50-70K of calls per month through its ports. We have NEVER EVER had any issues with faxing (close to 3k/month) with faxes connected on one of the spans of the card. Moreover, we have had quite a success receiving the faxes with iaxmodem+hylafax thanks to Lee Howard that we're now gradually switching the fax machines to iaxmodem+hylafax combo. Faxes are sensitive to timing and configuration settings of your asterisk. Once your system is tuned to perfection you should have no problems faxing at all despite the official stance from Digium. issues). Then we switched to a TE411P for the hardware echo cancellation. Now we want to receive fax ( 20/day) on it and guess what ? Since April 2006 (again a few months after we bought our brand new card), officially, fax communications is not supported with Digium cards ( http://www.voip-info.org/wiki-Asterisk+fax ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Background() and Read()
You should use the 'filename' parameter of Read to play the audio so
that it captures the input. Currently what's happening is that digits
entered whilst background is running are passed into the dialplan
context, since there's no match in the dialplan and you don't have an
'i' extension it will time out and hang up on you if you wait. To test
this, add an exten = i,1,Playback(invalid) and you'll see that this is
the case.
Douglas Garstang wrote:
I'm having some issues with Background() and Read() commands.
See the example below. This is when I wait for Background to finish playing the sound file, before entering '12345#'.
All works fine.
hestia*CLI
-- Executing Answer(SIP/2944093-3366, ) in new stack
-- Executing Wait(SIP/2944093-3366, 1) in new stack
-- Executing BackGround(SIP/2944093-3366, if-u-know-ext-dial) in new
stack
-- Playing 'if-u-know-ext-dial' (language 'en')
-- Executing Read(SIP/2944093-3366, number||) in new stack
-- User entered '12345'
-- Executing NoOp(SIP/2944093-3366, 12345) in new stack
== Auto fallthrough, channel 'SIP/2944093-3366' status is 'UNKNOWN'
However, if I start to enter digits before Background() is finished, background
stops playing the file, and nothing happens after this point. I keep hitting #
and still no reply. It didn't even execute the Read().
hestia*CLI
-- Executing Answer(SIP/2944093-6437, ) in new stack
-- Executing Wait(SIP/2944093-6437, 1) in new stack
-- Executing BackGround(SIP/2944093-6437, if-u-know-ext-dial) in new
stack
-- Playing 'if-u-know-ext-dial' (language 'en')
Here's extensions.conf:
exten = 1000,1,Answer
exten = 1000,2,Wait,1
exten = 1000,3,Background(if-u-know-ext-dial)
exten = 1000,4,Read(number||)
exten = 1000,5,NoOp(${number})
Anyone got any ideas?
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Re: [Asterisk-Users] channels change names
Probably because the Local proxy channel drops out once the two sides have been bridged. If you want the Local chan to stay up, use the /n parameter and the local channel won't perform the native transfer. This does have it's own problems, but should do what you want. eg: Channel: Local/[EMAIL PROTECTED]/n Jon-o Addleman wrote: I'm writing a php script to dial numbers and connect them to a conference. This is fairly straightforward: Action: originate Channel: Local/[EMAIL PROTECTED] Context: default Exten: $extension Priority: 1 This is pretty straightforward. However, the script then loads the list of members in the conference (using the meetme list ... command). For local extensions this works fine - the list of members shows the right channels, etc. The problem I'm having is that if the extension is external, the conference list shows a Local/$extension channel at the start, and then once the call is completed, it changes the channel to whatever was dialed. I'm probably not explaining it properly, but what I'd like to have happen is that I get one consistent channel name from the start of the connection - it doesn't matter what it is, as long as it doesn't change. As things stand, the conference list isn't accurate, unless I wait about 5 seconds after adding someone before updating the list. Thanks for any suggestions you might have here! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queues and the '*' key
From: http://www.voip-info.org/wiki/view/Asterisk+cmd+AgentCallbackLogin More info Unlike with AgentLogin the agent is not permanently off-hook (on-line). Instead the agent will be called at the designated extension when a new queue caller has been assigned to him. The agent goes off-hook and if ackcall is set to yes, must confirm with # that she is ready to take the call (it might be smart to include this instruction in the optional queue announcement). Press * to hang-up on the caller. And: http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue Description Queue(queuename|options|optionalurl|announceoverride|timeout) The option string may contain zero or more of the following characters: * 'H' — allow caller to hang up by hitting *. Sean Kennedy wrote: [EMAIL PROTECTED] asterisk]# asterisk -V Asterisk SVN-branch-1.2-r8632M I was wondering if there was some documentation I was missing on the '*' key and queues. I have my features setup to use *x, where x is a #, but these features don't work for calls coming in from a queue. As soon as the '*' button is hit, the call is disconnected. I have a vague memory of reading about this somewhere, but searched @ the wiki AND through google aren't turning up anything useful. Sean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] channels change names
Another option would be to set/pass a variable and use that instead of the current channel variable value. Keeping the Local chan up just to maintain a constant variable value may be alot of overkill (longterm) compared to rewriting your code to set and use your own variable (shortterm). On 4/20/06, Peter Fern [EMAIL PROTECTED] wrote: Probably because the Local proxy channel drops out once the two sides have been bridged. If you want the Local chan to stay up, use the /n parameter and the local channel won't perform the native transfer. This does have it's own problems, but should do what you want. eg: Channel: Local/[EMAIL PROTECTED]/n Jon-o Addleman wrote: I'm writing a php script to dial numbers and connect them to a conference. This is fairly straightforward: Action: originate Channel: Local/[EMAIL PROTECTED] Context: default Exten: $extension Priority: 1 This is pretty straightforward. However, the script then loads the list of members in the conference (using the meetme list ... command). For local extensions this works fine - the list of members shows the right channels, etc. The problem I'm having is that if the extension is external, the conference list shows a Local/$extension channel at the start, and then once the call is completed, it changes the channel to whatever was dialed. I'm probably not explaining it properly, but what I'd like to have happen is that I get one consistent channel name from the start of the connection - it doesn't matter what it is, as long as it doesn't change. As things stand, the conference list isn't accurate, unless I wait about 5 seconds after adding someone before updating the list. Thanks for any suggestions you might have here! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
