Re: [asterisk-users] Dialplan not reading MySQL table
Цитат на писмо от Doug Shubert [EMAIL PROTECTED]: Hello, I'm trying to use MySQL for Dialplans and have followed the Asterisk RealTime Extensions setup. The MySQL table is called extensions and I have entered two records.. ext 1000 and 2000. I also added switch = Realtime/[EMAIL PROTECTED] in extensions.conf and extensions = mysql,asterisk,extensions in extconfig.conf I do a *CLI dialplan reload but when I show the dialplan it has 0 extensions There is a spelling mistake in the switch statement extentions But try to use different names in extconfig.conf for the switch and table i.e.: exten_sions = mysql,asterisk,extensions_table or something, and then switch = Realtime/[EMAIL PROTECTED] to avoid problems. Benchev - SCENA - Единственото БЕЗПЛАТНО списание за мобилни комуникации и технологии. http://www.bgscena.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
Re: [Asterisk-Users] TDM04B/TDM2401E Card
On Monday 13 February 2006 20:43, housi mueller wrote: Hi there, I plan to use Aterisk in our small office. Until now we used a Panasonic D-1232 Super Hybrid System. The figure is representing the future configuration I where thinking about to have in the office. Question 1: We need only 4 lines and I thought to buy a TDM04B or a TDM2401E card. There is quite a price difference. Which card would you recommend me to buy. Question 2: Is such a configuration as shown on the figure with a TDM04B/TDM2401E card at all realizable? I am not familiar with that model in particular, but I've done some reasurch about D500. I think all of them have BRI interface so you may consider a BRI http://www.junghanns.net/asterisk/page17.html or http://www.avm.de/en/Produkte/Server-Produkte/C4/index.js.html or http://www.eicon.com/worldwide/products/MediaGateways/diva-server-v4bri.htm to interconnect Panasonic and your Asterisk. By my opinion Digium cards are more end user/provider oriented which is not your case. Like they say card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Let me know how did you do it. benchev ___ --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] Billing inbound calls per minute
On Monday 06 February 2006 09:25, JP Carballo wrote:
snip
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
Would you share with me how'd you do billing on a DID
(if you do), and through what Technology?
Anything that goes Local here is ANSWEREDTIME zero.
Thanks,
benchev
That probably explains it.
IIRC, from when I was still testing ASTCC, when calling a Local channel,
the AGI suffers from short term memory loss and forgets the values of
channel variables even if /n is used in the dial string.
I checked my test server logs and while I can verify that ASTCC's CDR
does have blank duration and billsec fields for the Local calls, *'s CDR
records them.
Similar here, and I read the patch from Darren May, 2005
where Local/$phone/$res-{path}|30|HL/n was changed to
Local/[EMAIL PROTECTED]{path}|30|HL/n
snip
I do billing based on account number so clients are free to call from
any phone. I don't check callerid.
Since each account is based on the phone number registered by the
client, I can just chop off the 2 digit prefix and set their callerid
with the result.
Yes, I do that also with another instance of astcc, I call astcc-disa.agi
to allow clients from outside to enter * and do things.
[makecall]
exten = s,1,Set(CALLERID(num)=${CARDNO:2})
exten = s,n,DeadAGI(astcc.agi,${CARDNO})
exten = s,n,Goto(nf2xsubmenu,s,1)
All my calls are routed to IAX2 or SIP or Zap.
And this is my problem because my target is to use Local, but
please follow my answer, within that thread, to Darren.
Thanks very much for your help.
benchev
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Re: [Asterisk-Users] Billing inbound calls per minute
On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
I've been playing with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the Connect fee(if I put one)
and keeps it that way no matter how long
the call is ...( if no Connect fee -stays empty).
i.e.
[inbound]
exten = 1122334455,1,Set(CALLERID(number)=${EXTEN})
exten = 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten = 1122334455,3,Hangup
Michiel van Baak wrote:
DeadAGI is for hungup channels, not for active channels.
That might be a problem.
Try this:
exten = h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
On Monday 06 February 2006 09:25, JP Carballo wrote:
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
Would you share with me how'd you do billing on a DID
(if you do), and through what Technology?
Anything that goes Local here is ANSWEREDTIME zero.
On Saturday 11 February 2006 06:32, Darren Wiebe wrote:
Are you running a relatively recent version of ASTCC? Say within the
last 6 months. The answeredtime = 0 bug was supposed to have been fixed
by http://bugs.digium.com/view.php?id=4300 Unless something has changed
in Asterisk that affects this
Thanks Daren,
Yes, my version of astcc is the most recent one.
Asterisk-1.2.4
I have found you patch 0004300 from 16 May 2005.
Probably it's time to reverse it back since something has changed
in Asterisk that affects this... as you said.
My observation is:
If I keep:
$dialstr = Local/[EMAIL PROTECTED]{path}|30|HL/n( . ($maxtime * 60 * 1000) .
:6:3);
Either the billseconds is empty(when dial out through Local), either there is
aZOMBIE when dialing in.
I put back the dialstring to:
Local/$phone/$res-{path}|30|HL/n( . ($maxtime * 60 * 1000) .
:6:3);
The only difference that it looks only for is a default context.
extensions.conf
[inbound]
; 10 digits DID = _XX = cardnumber
;
exten = _XX ,1,Answer()
exten = _XX ,n,Set(DB(RCID/${CALLERIDNUM})=${CALLERIDNUM})
exten = _XX ,n,Set(realcid=${DB(RCID/${CALLERIDNUM})})
exten = _XX ,n,Noop(${REALCID})
;exten = _XX ,n,Set(TIMEOUT(digit)=4)
exten = _XX ,n,Set(CALLERID(number)=${EXTEN})
exten = _XX ,n,Set(CALLERID(name)= ${REALCID})
;exten = t,3,Goto(h|1)
;exten = _XX 2,Goto(s|1)
;exten = s,1,Wait,1 ; is this preventing HUP?
exten = h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${CALLERIDNUM},4)
; must be h,1 as per Michiel van Baak note(above).
exten = h,2,Hangup
[internal]
; i.e. 360 1234567 = DID = card
exten = 3601234567,1,Macro(stdexten,3601234567,sip/did_owner)
[default]
include = internal
[personal]
exten = t,1,Hangup
include = inbound
Result:
- ANSWEREDTIME is OK
- inbound call billed on the callee
- there is CALLERID(name) for callerid in the cdrs(kind of)
There is still a small but looong problem - Timeout about 10
secs long while the IAX2/incoming Hangup in personal,t,1.
But CDR is updated after that and the call is billed as expected.
Sorry for the long explanation.
What do you think? Is there something suspicious in
that solution?
Thanks,
benchev
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Re: [Asterisk-Users] Billing inbound calls per minute
On Monday 06 February 2006 09:25, JP Carballo wrote:
Michiel van Baak wrote:
On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
Hi,
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?
I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the Connect fee(if I put one)
and keeps it that way no matter how long
the call is ...( if no Connect fee -stays empty).
i.e.
[inbound]
exten = 1122334455,1,Set(CALLERID(number)=${EXTEN})
exten = 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten = 1122334455,3,Hangup
DeadAGI is for hungup channels, not for active channels.
That might be a problem.
Try this:
exten = h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
Would you share with me how'd you do billing on a DID
(if you do), and through what Technology?
Anything that goes Local here is ANSWEREDTIME zero.
Thanks,
benchev
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Re: [Asterisk-Users] Billing inbound calls per minute
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?
I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the Connect fee(if I put one)
and keeps it that way no matter how long
the call is ...( if no Connect fee -stays empty).
i.e.
[inbound]
exten = 1122334455,1,Set(CALLERID(number)=${EXTEN})
exten = 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten = 1122334455,3,Hangup
DeadAGI is for hungup channels, not for active channels.
That might be a problem.
Try this:
exten = h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
Thanks, tried that several ways but no help since ${EXTEN}=h.
Probably will try with CHANISAVAIL or ${CHANNEL} or something...
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
Sure ASTCC works, but I am talking about inbound calls
where 1122334455 is a DID as well as a card number being
charged for the incoming calls. Thus ${EXTEN}=DID=card i.e.
exten = 1122334455,1,Set(CALLERID(number)=${EXTEN})
Mayby I should not assosiate DID from card(user) and create a separate
peer for the DID on a different port.
Any other ideas? Thanks.
benchev
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[Asterisk-Users] Billing inbound calls per minute
Hi,
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?
I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the Connect fee(if I put one)
and keeps it that way no matter how long
the call is ...( if no Connect fee -stays empty).
i.e.
[inbound]
exten = 1122334455,1,Set(CALLERID(number)=${EXTEN})
exten = 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten = 1122334455,3,Hangup
Thanks in advance,
benchev
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Re: [Asterisk-Users] callback script?
On Thursday 02 February 2006 11:40, Arne Morten Johansen wrote: How do I setup a Callback script? This script does what I want to do. But how do I set it up? http://www.junghanns.net/en/callback.html I see it uses PHP for scriptlanguage. So where do I place it (the .agi)? /var/lib/asterisk/agi-bin and should be 755 benchev ___ --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] Dundi key Problem
On Wednesday 01 February 2006 19:48, Jonathan k. Creasy wrote: I am getting the following message when trying to lookup up a number via Dundi: Feb 1 13:39:24 NOTICE[20146]: pbx_dundi.c:1309 update_key: No such key 'office.pbx.bluegrass.net.pub' for creating RSA encrypted shared key for '00:a0:c9:55:91:89'! I have created keys on each box with astgenkey -n office.pbx.bluegrass.net using the host name for each box of course. I then copied the .pub files to the /var/lib/asterisk/keys folder from each box to the other box. What am I missing? inkey=office.pbx.bluegrass.net benchev ___ --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] Spa3k and ISDN
On Friday 27 January 2006 13:29, Manuel Dominguez wrote: Hello all, I have an ISDN termination box (TR1) that converts ISDN(Bri) to 2 normal analogue lines. The same number is assigned to these lines. These lines are connected to 2 spa3k registered to my asterisk box. When calls arrive, TR1 try to pass call to the first spa. If spa not takes the call immediately then try to pass to the other spa. The only configuration I found works is to put the parameter 'PSTN Answer Delay' to 0 in each spa. The problem is Call CID. I suppose the problem is that Asterisk not sees the CID because the spa takes several seconds to know. In the Spa status page appears the CID but never in the asterisk box or extensions. Connect a phone with a display through the FXS port and let the PSTN line to ring through to the VOIP. That way you can check do you receive callerid at all. You could play with the delay secs untill you are sure you see a callerid(probably only number). And most probably the callerid method should be ETSI. Any experience with as how a GSM CLIP is read by spa3000? Anyone? benchev ___ --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 set caller id?
On Monday 23 January 2006 06:03, Ronald Wiplinger wrote:
C F wrote:
The one you demonstrate should have *never* worked.
well, it did,
Pre 1.2 you do:
exten = s,1,SetCIDNUM(12345789)
Post 1.2 you do:
exten = s,1,Set(CALLERID(num)=123456789)
I need to get the callers phone number there!
How can I do it now?
exten = _91NXXNXX,3,NoOp(SetCallerID(${username}))
exten =
_91NXXNXX,4,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:${TRUNKMSD}},${TAR
IFF})
Hi, I had similar problem and Tony Mountifield gave me the idea
If you want your script to be compatible with both 1.0 and 1.2, try
something like this:
$calleridname = $input{calleridname} || (($input{callerid} =~ /(.*)/) ? $1
: unknown);
$callerid = ($input{callerid} =~ /(.*)/) ? $1 : $input{callerid};
So I did put the first line in astcc.agi as:
my $callerid = $input{calleridname} || (($input{callerid} =~ /(.*)/) ? $1 :
unknown);
and then:
exten = _1NXXNXX,1,Set(CALLERID(all)=${CALLERIDNAME} - ${CALLERIDNUM})
exten = _1NXXNXX,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
...
Worked for me, hope it gives you a clue.
benchev
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Re: [Asterisk-Users] Caller ID and Sipura Router
On Saturday 21 January 2006 20:30, Conrad Beckert wrote: Could anyone please help me with that: I have an analog telephone connected to Asterisk using a Sipura 2002 ATA. When calling the extension, the caller ID presented is always the number of that extension rather than the number of the calling one. While I learned that this is the new standard behaviour (?) of Asterisk, I want to show the original caller ID. I tried the options o and f in the dial command - e.g. Don't know about f but o is Operator extension, used for operator exit by pressing zero in voicemail exten = 1002,4,dial(sip/2999,20,o) no avail. The phone rings and shows 2999 instead of the calling party! The SIPURA seems to be ok: when I connect to Sipgate/Nikotel etc. directly, everything is ok What's wrong? My Asterisk Version is 1.2.1 sip.conf [2999] type=friend secret=x callerid=Analog Phone 1002 regexten=1002 etc... exten = 1002,1,Dial(sip/2999,20) exten = 1002,2,Hangup Does this give you a clue? benchev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial() Jumping behaviour and Vesrsion 1.2
see inline
The version 1.2 Dial() command does not use the n+101 jumping
behaviour by default. I know about the j option and setting
priorityjumping=yes as described here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial
But if I use the default behaviour does that mean I have to check the
DIALSTATUS to determine whether or not to go to voicemail?
No but is good idea
For example I used to do this:
exten = s,1,Dial(SIP/[EMAIL PROTECTED],20,t)
exten = s,2,Voicemail(u${EXTEN})
exten = s,3,Goto(s,200)
;
exten = s,102,Voicemail(b${EXTEN})
exten = s,103,Goto(s,200)
;
exten = s,200,Playback(CallAgainRealSoon)
exten = s,201,Hangup
;
exten = h,1,Hangup
So in 1.2 would I do the following or am I missing something?
No but is good idea
or you can use n instead of s
or
exten = 1234,1,Dial(SIP/[EMAIL PROTECTED],20,t)
exten = 1234,n,Voicemail(u${EXTEN)
exten = 1234,dial+101,Voicemail(b${EXTEN})
or even better use a macro (since your way is very close to that)
(see below)
exten = s,1,Dial(SIP/[EMAIL PROTECTED],20,t)
exten = s,2,GotoIf($[${DIALSTATUS } = BUSY]?10)
exten = s,3,GotoIf($[${DIALSTATUS } = NOANSWER]?20)
;
exten = s,10,Voicemail(b${EXTEN})
exten = s,11,Goto(s,100)
;
exten = s,20,Voicemail(u${EXTEN})
exten = s,21,Goto(s,100)
;
exten = s,100,Playback(CallAgainRealSoon)
exten = s,101,Hangup
;
exten = h,1,Hangup
[your-internal-context]
exten =1234,1,Macro(stdexten,${EXTEN},SIP/${EXTEN},20,t)
[macro-stdexten]; straight from the extensionsconf sample
;; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
;; ${ARG2} - Device(s) to ring
;exten = s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
;exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on
status(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
;
;exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail
w/ unavail announce
;exten = s-NOANSWER,2,Goto(s,1); If they press #, return to
start
;exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy
announce
;exten = s-BUSY,2,Goto(s,1); If they press #, return to start
;
;exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no
answer
;exten = a,1,VoicemailMain(${ARG1})
benchev
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Re: [Asterisk-Users] Dropping incompatible voice frame
On Thursday 19 January 2006 13:48, Joseph Rothstein wrote: I am now getting these messages on a second box running a different version of Asterisk. If anyone has any idea what is causing these, or how to avoid them I would be very grateful. 157 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw I have had a similar issue but was saying :of format slin since our native format has changed to ulaw whatever. The problem was: wrong configuration of FXO port dialplan(spa3000). Kind of - simultaneous use of PSTN dialplan and Call Forward Settings on User tab... This is just a guess since your info is not enough. benchev ___ --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] sipura ata 3000 UK (BT) CAllerid
On Wednesday 18 January 2006 23:00, Conrad Wood wrote: Hi I wonder whether anyone got the Sipura ata 3000 to decode British Telecoms callerid and pass it to asterisk? The userguide seems to suggest that this is not possible, is that right? Conrad Check this http://www.ainslie.org.uk/callerid/nopcsoft.htm#UNIX benchev ___ --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] realtime voicemail
On Monday 16 January 2006 09:02, [EMAIL PROTECTED] wrote: i tried to setup realtime voicemail recently with 1.2.1 but couldn't get it to work. no matter what i do. it still looks for config in the voicemail.conf file. (BTW realtime sip extensions works fine) here's the voicemail line in extconfig.conf: voicemail = mysql,asterisk,voicemail here's the mysql schema: CREATE TABLE voicemail ( uniqueid int(11) NOT NULL auto_increment, customer_id bigint NOT NULL default '0', context varchar(50) NOT NULL default '', mailbox bigint NOT NULL default '0', password varchar(10) NOT NULL default '0', fullname varchar(50) NOT NULL default '', email varchar(50) NOT NULL default '', pager varchar(50) NOT NULL default '', stamp timestamp NOT NULL default CURRENT_TIMESTAMP on update CURRENT_TIMESTAMP, attach varchar(3) NOT NULL default 'yes', saycid varchar(3) NOT NULL default 'yes', hidefromdir varchar(3) NOT NULL default 'no', PRIMARY KEY (uniqueid), KEY mailbox_context (mailbox,context) ) TYPE=MyISAM; Put in voicemail.conf searchcontexts=yes and do not forget to stop and start *. Reload may not do. benchev ___ --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] codec_gsm.c:194 gsmtolin_framein: Invalid GSM data
Hi guys, Anyone seen something like below(see below the line)? Machine P2 w/512MB RAM Debian (testing) ; kernel 2.6.12-1-386 asterisk 1.2.1-n-all incl. astcc For many months now I went through * 1.07, 1.09 and never saw something like that. Even with 1.2.0, a month now, at the beginning everything was fine, and suddenly codec_gsm.c:194 gsmtolin_framein: Invalid GSM data thing started. It happens with astcc (but not immediatelly) as you can see after being interogated about the card (after astcc-accountnum2). When there is no card check (silent level 4) sometimes there is no problem, but some times it happens when the callee party answers. The warning lines repeat hundreds of times and it is not possible to hear anything. I do suspect, however that it happens when an answering machine answers the call.(not sure) At the same time I can triger MOH, put it on hold, dial my mobile, open all tree lines and listen stereo with no problem. I think it is something to do with the machine since on another one: Machine AMD2.8G w/1G RAM Debian (testing) ; kernel 2.6.12-1-386 asterisk 1.2.1-n-all incl. astcc everything seams to work fine? Any experience of a kind? benchev -- Executing Answer(SIP/asterisk-d349, ) in new stack -- Executing NoOp(SIP/asterisk-d349, Call from spa3000) in new stack -- Executing DeadAGI(SIP/asterisk-d349, astcc.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi -- Playing 'digits/6' (language 'en') -- Playing 'astcc-accountnum2' (language 'en') Jan 14 16:48:19 WARNING[15662]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data Jan 14 16:48:19 WARNING[15662]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data ... Jan 14 16:48:20 WARNING[15662]: format_gsm.c:155 gsm_read: Short read (1) (No such file or directory)! -- Playing 'digits/14' (language 'en') Jan 14 16:48:22 WARNING[15662]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data Jan 14 16:48:22 WARNING[15662]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data ... Jan 14 16:48:22 WARNING[15662]: format_gsm.c:155 gsm_read: Short read (1) (No such file or directory)! Jan 14 16:48:22 WARNING[15662]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data ... Jan 14 16:48:23 WARNING[15662]: format_gsm.c:155 gsm_read: Short read (6) (No such file or directory)! -- Playing 'digits/10' (language 'en') Jan 14 16:48:23 WARNING[15662]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data Jan 14 16:48:23 WARNING[15662]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data ... Jan 14 16:48:24 WARNING[15662]: format_gsm.c:155 gsm_read: Short read (2) (No such file or directory)! -- Playing 'astcc-phonenum' (language 'en') Jan 14 16:48:25 WARNING[15662]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data ... -- AGI Script astcc.agi completed, returning 0 -- ___ --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] GSM Gateway / Terminal for sale
On Thursday 05 January 2006 17:09, Chris Bagnall wrote: Single port GSM Gateway support 900 / 1800 GSM mode with external antenna. Brand new unit and all of them will be tested before dispatch. Extremely easy to setup and can be used out of the box without any configuration. So should be good alternatively of phonecell or nokia pbx etc.. Units are located in UK and £60 GBP per unit excluding shipping. Has anyone bought one of these and able to offer some feedback? I'm seriously considering a GSM gateway to take advantage of the spare SIM cards lying around still inside their 12-month contracts. Looking at the website in question, delivery is £17.37 for a 6-day delivery, or £10 for a 30+ day delivery, both of which seem a bit high for an item apparently located in the UK. I have got one and and is working fine. It's exactly for cards lying around still inside their 12-month contracts.. Actually it's full extend PnP-SIM-card-GSM-gateway(I forgot to unlock the pin:) so keep this in mind). There are 2 fxs ports, but I use just one; points to a SPA3000. The other could go to a phone set, too(I did test it) And that's it... pretty much . Anything else you want to do is * job and dial plans. When one calls from outside, first is getting authenticated against CallerId and could then dial internal or any other destination. It's a week I have it and works no problem. It is a little big, but much cheaper than other solutions, I have checked around. The one I have came from HK during Xmas and took a little longer, but the freight was fine with me because I like things that just work. Hope that helps, benchev ___ --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] GSM Gateway / Terminal for sale
On Friday 06 January 2006 00:19, Jean-Michel Hiver wrote: I have got one and and is working fine. It's exactly for cards lying around still inside their 12-month contracts.. Actually it's full extend PnP-SIM-card-GSM-gateway(I forgot to unlock the pin:) so keep this in mind). There are 2 fxs ports, but I use just one; points to a SPA3000. The other could go to a phone set, too(I did test it) And that's it... pretty much . Anything else you want to do is * job and dial plans. When one calls from outside, first is getting authenticated against CallerId and could then dial internal or any other destination. It's a week I have it and works no problem. It is a little big, but much cheaper than other solutions, I have checked around. Sounds pretty cool! Is the antenna detachable? Can you replace it with a longer antenna which can be stuck somewhere with decent GSM reception? For Remco, no I don't know who the producer is, but as far as I can tell the box is Chinese or something close. The antenna is 30cm tall, on magnetic stand connected to a cable about 1.5m long, which could become longer I guess. One could substitute the body with a longer on, unscrewing it from the stand I'm keeping it sticked upon my metal desk light, hanging from the ceiling upside down, but looking through the window for a gsm cell :) Hope you'll like it. benchev ___ --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] GSM Gateway / Terminal for sale
On Thursday 05 January 2006 21:31, stotaro wrote: I have got one and and is working fine. It's exactly for cards lying around still inside their 12-month contracts.. Actually it's full extend PnP-SIM-card-GSM-gateway(I forgot to unlock the pin:) so keep this in mind). There are 2 fxs ports, but I use just one; points to a SPA3000. The other could go to a phone set, too(I did test it) And that's it... pretty much . Anything else you want to do is * job and dial plans. When one calls from outside, first is getting authenticated against CallerId and could then dial internal or any other destination. It's a week I have it and works no problem. It is a little big, but much cheaper than other solutions, I have checked around. Sounds pretty cool! Is the antenna detachable? Can you replace it with a longer antenna which can be stuck somewhere with decent GSM reception? Cheers, Jean-Michel. Can it be used to send SMS via asterisk? Not by it self (It is rather cellsocket kind of thing), but with an appropriate sms application, why not? i.e. see http://tuxmobil.org/phones_linux_sms.html for hints. benchev ___ --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] Cell phone dock/switch as Asterisk FXO source
On Tuesday 03 January 2006 05:48, Paul Dugas wrote: On Mon, 2006-01-02 at 16:06 +, Jonathan Attwood wrote: I use a Dock-n-Talk in conjuction with a Sipura SPA3000 Asterisk. Does this unit require any funky dialing when placing outbound calls from * through the phone? Do the docs indicate operation is any different between CDMA, TDMS, AMPS, or GSM phones? I'd guess not or, if so, it was simple to handle it in the dialplan but I'm curious anyway. I've been considering this as a way to have work calls that come to my cell appear different to the server. At the moment, I have my GSM phone forward calls to the house when it's off so I can't really tell between them. I have good experience with a GSM-box I've bought from cybertelecom and SPA3000. GSM-box acts as a Dock-n-Talk because is it allows in and out dialing. The advantage is that one doesn't need even a mobile phone, but only a SIM card. The whole thing is like porting a number. There are 2 FXS ports. One could go to an ordinary phone, the other to SPA3000. The disadvantage is that you have one more number for your friends to remember. Otherwise is stable, and as-easy- as-PnP instalation, if you don't forget to disable the pin lock as I did :-) benchev ___ --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] Cell phone dock/switch as Asterisk FXO source
On Tuesday 03 January 2006 15:37, Noah Swint wrote: Do you have a url for the device? http://cyber-telecom.net/store/index.php?cPath=1 From: [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source Date: Tue, 3 Jan 2006 11:02:30 +0200 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc3-f17.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Tue, 3 Jan 2006 01:03:53 -0800 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 43C5041E7;Tue, 3 Jan 2006 02:02:15 -0700 (MST) Received: from psmtp.com (exprod5mx19.postini.com [64.18.0.159])by lists.digium.com (Postfix) with SMTP id 7F12641D8for asterisk-users@lists.digium.com;Tue, 3 Jan 2006 02:02:08 -0700 (MST) Received: from source ([69.93.167.194]) (using TLSv1) byexprod5mx19.postini.com ([64.18.4.10]) with SMTP; Tue, 03 Jan 2006 01:02:11 PST Received: from [84.22.2.1] (helo=chick)by server.mgmhost.net with esmtps (SSLv3:RC4-MD5:128) (Exim 4.52)id 1Eti3X-0006T8-0sfor asterisk-users@lists.digium.com; Tue, 03 Jan 2006 04:02:11 -0500 X-Message-Info: TiNwL5K19MGed4lSuBRh1tXSI4SUwww6i3LK9Bv9faQ= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com User-Agent: KMail/1.8.2 References: [EMAIL PROTECTED]fa074e5c06010 [EMAIL PROTECTED][EMAIL PROTECTED] calhost.localdomain X-AntiAbuse: This header was added to track abuse,please include it with any abuse report X-AntiAbuse: Primary Hostname - server.mgmhost.net X-AntiAbuse: Original Domain - lists.digium.com X-AntiAbuse: Originator/Caller UID/GID - [47 12] / [47 12] X-AntiAbuse: Sender Address Domain - mail.bg X-Source: X-Source-Args: X-Source-Dir: X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [90/4] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:asterisk [EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:asterisk [EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 03 Jan 2006 09:03:53.0160 (UTC) FILETIME=[9E9DCC80:01C61044] On Tuesday 03 January 2006 05:48, Paul Dugas wrote: On Mon, 2006-01-02 at 16:06 +, Jonathan Attwood wrote: I use a Dock-n-Talk in conjuction with a Sipura SPA3000 Asterisk. Does this unit require any funky dialing when placing outbound calls from * through the phone? Do the docs indicate operation is any different between CDMA, TDMS, AMPS, or GSM phones? I'd guess not or, if so, it was simple to handle it in the dialplan but I'm curious anyway. I've been considering this as a way to have work calls that come to my cell appear different to the server. At the moment, I have my GSM phone forward calls to the house when it's off so I can't really tell between them. I have good experience with a GSM-box I've bought from cybertelecom and SPA3000. GSM-box acts as a Dock-n-Talk because is it allows in and out dialing. The advantage is that one doesn't need even a mobile phone, but only a SIM card. The whole thing is like porting a number. There are 2 FXS ports. One could go to an ordinary phone, the other to SPA3000. The disadvantage is that you have one more number for your friends to remember. Otherwise is stable, and as-easy- as-PnP instalation, if you don't forget to disable the pin lock as I did :-) benchev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GSM-gateway setup
Merry Christmas List, Any body with experience on the GSM-gatewas that Cyber-telecom.net sell? The thing keeps on asking for a PASS and ... pretty much that's all. Help anyone? benchev ___ --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] Calls not incoming to any extension from remoteproxy server
On Thursday 22 December 2005 22:13, abhishek wrote:
Thanks a lot for the reply. But i am sucessfully getting registered on the
remote proxy, so that i am getting right outputs as u said. I think that is
why only i am able to route calls outside to remote proxy,
The problem is when i am writing
register = user:[EMAIL PROTECTED]/1234
, the outside calls are not coming to 1234 extension , which is a Xlite
client.
The files configuration are as
sip.conf
register = user:[EMAIL PROTECTED]/1234
[1234]
type=friend
host=dynamic
context=test_in
user=phone
regexten=1234
extensions.conf
[test_in]
exten= 1236,1,Dial(SIP/sandhu)
exten= 1235,1,Dial(SIP/1235)
exten= _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten= 1234,1,Dial(SIP/1234)
I would try to separate incoming and outgoing extensions
to different contexts, for instance:
[test_in]
exten= 1234,1,Dial(SIP/1234)
[test_out]
exten= _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
and make include = both to you [default] context and put
context = default in your [1234] definition
I think this was important in order to follow the correct dialing priority.
To see the difference you could type now: show dialplan test_in
and after : show dialplan default
Also when forming a dial string keep in mind that
X = any digit from 0-9, Z = any digit from 1-9,
N = any digit from 2-9
means to use:
exten= _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
when dialing US/Canada and:
exten= _9011N.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
when dialing other desitinations...
Another thing I can see now is that there isn't a peer
(or you don't show it?) for the remote proxy i.e.:
[remote_proxy]
type=peer (or friend)
host=proxy-ip
context=whatever_they_say
etc
Your [1234] is for the Xlite
and [remote_proxy] for your provider.
Hope that helps,
benchev
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Re: [Asterisk-Users] IAX No Authority found
On Thursday 22 December 2005 13:40, Leandro Martini - ISAT DGL wrote:
Guys,
I,m facing a little tricky issue here, is there anybody that faced the same
issue or knows how to solve this?
I have 2 *, trunked with IAX
From ServerA I can call ServerB without any problems
If I call from ServerB to ServerA i get the following message :
ServerB
Dec 21 11:12:23 WARNING[2849]: chan_iax2.c:6967 socket_read: Call rejected
by 10.0.100.125: No authority found
-- Hungup 'IAX2/campinas-16384'
== No one is available to answer at this time (1:0/0/0)
== Auto fallthrough, channel 'IAX2/8512-7' status is 'NOANSWER'
-- Hungup 'IAX2/8512-7'
ServerA
Dec 21 13:08:12 NOTICE[2420]: chan_iax2.c:6772 socket_read: Rejected
connect attempt from 10.20.0.20, who was trying to reach '[EMAIL PROTECTED]'
The iasx.conf is as follows:
serverB
[campinas]
qualify=yes
type=friend
auth=rsa
If you use auth=rsa
then must show where is it:
inkeys=campinasOrwhatever
benchev
;username=campinas
;secret=campinasvoip
host=10.0.100.125
trunk=yes
notransfer=yes
disallow=all
allow=speex
serverA
[saopaulo]
qualify=yes
type=friend
auth=rsa
;username=saopaulo
;secret=saopaulovoip
host=10.20.0.20
trunk=yes
notransfer=yes
The Dial string is this one:
On serverA
exten = _85XX,1,Dial(IAX2/saopaulo/${EXTEN},60,t)
On serverB
exten = _74XX,1,Dial(IAX2/campinas/${EXTEN},60,t)
Is there anything missing ???
Happy holidays to you all !!!
Leandro Martini
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[Asterisk-Users] Fwd: Legacy PBX - * - Voip Calls problems
If have installed a TE110P and have connected it to my Mitel 200SX. I can dial to the Mitel via the T1 connection but when I dial from the Mitel to try and go out the Asterisk box via Voip it fails. I can see the calls getting to the Asterisk box from the Mitel but it just loops though its Zap channels then fails. Do I have spilt incoming and out going channels on a T1? Thanks, -Scott Scott, Sorry for approaching you personally but not sure you are still subscribed to the list (copy to the list anyway) I have found your efforts about Legacy PBX - * - Voip Calls problems goggling but not a full thread. Your scheme is very intriguing since I intend to interconnect an *+TE110P -(PRI)-Siemens HiPath3750. Asterisk+TE110P will provide only VOIP dialing to the Siemens HiPath3750 members(about 400). Mitel is more more IP oriented than Siemens but any info you could point me to, or share would be of great value. Thanks and Merry Xmas everyone, benchev --- ___ --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: Fw: Legacy PBX - * - Voip Calls problems
Scott, Great, really! I'll remember that the correct T1 Crossover cable! And I was correct thinking that with *+TE110P a legacy PBX becomes simply a 400-slots-channel bank, wasn't I? Thanks again and Merry Christmas;, benchev On Thursday 22 December 2005 22:29, Scott Wolfe wrote: Hi there, I was able to get this going. I just needed to create a dialing plan in the Mitel so that calls would go through the asterisk box. The Mitel I was using is not the IP Mitel. It works and the only hang up I had was making sure I had made the correct T1 Crossover cable. -Scott - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: Asterisk-users-list asterisk-users@lists.digium.com Sent: Thursday, December 22, 2005 11:40 AM Subject: Legacy PBX - * - Voip Calls problems If have installed a TE110P and have connected it to my Mitel 200SX. I can dial to the Mitel via the T1 connection but when I dial from the Mitel to try and go out the Asterisk box via Voip it fails. I can see the calls getting to the Asterisk box from the Mitel but it just loops though its Zap channels then fails. Do I have spilt incoming and out going channels on a T1? Thanks, -Scott Scott, Sorry for approaching you personally but not sure you are still subscribed to the list (copy to the list anyway) I have found your efforts about Legacy PBX - * - Voip Calls problems goggling but not a full thread. Your scheme is very intriguing since I intend to interconnect an *+TE110P -(PRI)-Siemens HiPath3750. Asterisk+TE110P will provide only VOIP dialing to the Siemens HiPath3750 members(about 400). Mitel is more more IP oriented than Siemens but any info you could point me to, or share would be of great value. Thanks and Merry Xmas everyone, benchev ___ --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] [offtopic] Asterisk -IP- Siemens HiPath 4000
On Wednesday 21 December 2005 14:47, Dmitry Ivanov wrote: Hello! Is it possible to connect Siemens HiPath 4000 to Asterisk? What equipment required on Siemens side? I mean IP not E1. I am also to deal with HiPath, but 3750. I came to the conclusion that buying anything on the HiPath side is stupid. A Digium TE110P or Sangoma101(both PRI) would deliver E1 (31channels) to HiPath and use it as a channel bank. A card as th above would cost you about $600 but buyin analog modules could be expensive. But, please, any other opinions are highly appreciated. benchev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls not incoming to any extension from remote proxy server
On Thursday 22 December 2005 04:45, abhishek wrote:
Hi all,
I am testing my hands on asterisk , but got stuck. Let me tell you i am
only using its VOIP functionlities
I have registered the asterisk server at a remote proxy server. My
clients registered at asterisk server can make outgoing calls , but the
calls made from outside is not incoming to any extension.
I have written
user:[EMAIL PROTECTED]/1234
register = user:[EMAIL PROTECTED]/1234
is it not?
And when you do sip show registry
you see
server*CLI sip show registry
HostUsername Refresh State
proxy-ip:5060user105 Registered
Hope that gines you a clue.
benchev
in sip.conf.
and 1234are defined as
[1234]
type=friend
host=dynamic
context=test_in
user=phone
regexten=1234
in extensions.conf i am using
[test_in]
exten= 1236,1,Dial(SIP/sandhu)
exten= 1235,1,Dial(SIP/1235)
exten= _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten= 1234,1,Dial(SIP/1234)
My clients are on Xlite softphone.
Can anybody help out ?/
Abhishek
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[Asterisk-Users] ASTCC/ASTCC anything wrong with that?
List ... Darren,
In order to use a provider with unusual prefix 00
i.e. 001NXXNXX and providing failover to other providers with
the usual 1NXXNXX, decided to:
1. Change dialstr like that: IAX2/$res-{path}$phone|30|HL (excerpt from
below)
if ( $res-{tech} eq IAX2 ) {
$dialstr =
IAX2/$res-{path}/$phone|30|HL(
. ( $maxtime * 60 * 1000 )
. :6:3);
2. EVery trunk is closed lake that:
iaxprovider/
otherprovider/00
yetanother/
Q: see anything very wrong with that?
Thanks,
benchev
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Re: [Asterisk-Users] ASTCC/ASTCC anything wrong with that?
Thanks.
It works fine. I was just curious about
any collateral damages.
Thanks again,
benchev
On Monday 12 December 2005 16:42, Darren Wiebe wrote:
Try it out. It looks to me like it would work but I've been wrong
often. :-)
Darren Wiebe
[EMAIL PROTECTED] wrote:
List ... Darren,
In order to use a provider with unusual prefix 00
i.e. 001NXXNXX and providing failover to other providers with
the usual 1NXXNXX, decided to:
1. Change dialstr like that: IAX2/$res-{path}$phone|30|HL (excerpt from
below)
if ( $res-{tech} eq IAX2 ) {
$dialstr =
IAX2/$res-{path}/$phone|30|HL(
. ( $maxtime * 60 * 1000 )
. :6:3);
2. EVery trunk is closed lake that:
iaxprovider/
otherprovider/00
yetanother/
Q: see anything very wrong with that?
Thanks,
benchev
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Re: AW: [Asterisk-Users] Asterisk 1.2.0 AddOn's compile error with MySQL5.0.15
On Thursday 24 November 2005 00:01, Rainer Maier wrote: Hi Matt, I did not move the whole asterisk directory I just put a link to it. (ln -s /usr/src/asterisk-1.2.0 /usr/src/asterisk) Then I tried to compile but the error stayed. I also tried with MySQL 4.1.15 and had the same error. snip res_config_mysql.c res_config_mysql.c: In function 'realtime_mysql': res_config_mysql.c:117: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c: In function 'realtime_multi_mysql': res_config_mysql.c:224: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c: In function 'update_mysql': res_config_mysql.c:313: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c: In function 'config_mysql': res_config_mysql.c:376: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c: In function 'realtime_mysql_status': res_config_mysql.c:648: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c:650: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c:652: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c:656: warning: incompatible implicit declaration of built-in function 'snprintf' cc -shared -Xlinker -x -o res_config_mysql.so res_config_mysql.o -lmysqlclient -lz-L/usr/local/mysql/lib -L/usr/local/mysql/lib/mysql rm app_saycountpl.o sv5000:/usr/src/asterisk-addons-1.2.0# All these are warnings. Actually if you do make install it will work. Why those warnings, however, don't know. Let us know what happened. benchev ___ --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] voicemail clients
On Wednesday 23 November 2005 17:43, Joao Pereira wrote:
Hello to all
I have clients registered with names (joao, manuel, etc...) and clients
reistered with numbers (123, 120,...).
To make the number clients receive voicemail, I have this:
exten = _X,1,Answer
exten = _X,2,Wait(1)
exten = _X,3,VoiceMail(u${EXTEN})
exten = _X,4,Playback(vm-goodbye)
exten = _X,5,Hangup
but for the name clients I need these 5 lines for each...
exten = pereira,1,Answer
exten = pereira,2,Wait(1)
exten = pereira,3,VoiceMail(u${EXTEN})
exten = pereira,4,Playback(vm-goodbye)
exten = pereira,5,Hangup
Is there any way I can solve this? making all calls that reach this
point go to the voicemail?
Very ... kind of embarrassing, but interesting .
What if you assign in voicemail.conf
7373472 = 1234,pereira,[EMAIL PROTECTED]
and then exten = pereira,3,VoiceMail(u7373472)
As you understand pereira is the vanity # of 7373472
and charge pereira for having vanity # :-)
Don't forget to tell us what happened!
benchev
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Re: [Asterisk-Users] Asterisk 1.2.0 AddOn's compile error with MySQL 5.0.15
On Monday 21 November 2005 23:49, Rainer Maier wrote: Hi all, I want to compile asterisk's newest version with mysql's newest version, but I ran into a big problem. At compile time for asterisk-addons-1.2.0 I get the following errors: make -- snip -- cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include -I/usr/local/mysql/include/mysql -c -o res_config_mysql.o If you didn't do mv /usr/src/asterisk-1.2.0 /usr/src/asterisk that might be your problem. But cannot be seen because of your --snip--. Further down I don't see either CFLAGS+=-I../asterisk-1.2.0 which the other way around. Hope that helps. benchev res_config_mysql.c res_config_mysql.c: In function 'realtime_mysql': res_config_mysql.c:117: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c: In function 'realtime_multi_mysql': res_config_mysql.c:224: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c: In function 'update_mysql': res_config_mysql.c:313: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c: In function 'config_mysql': res_config_mysql.c:376: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c: In function 'realtime_mysql_status': res_config_mysql.c:648: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c:650: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c:652: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c:656: warning: incompatible implicit declaration of built-in function 'snprintf' cc -shared -Xlinker -x -o res_config_mysql.so res_config_mysql.o -lmysqlclient -lz-L/usr/local/mysql/lib -L/usr/local/mysql/lib/mysql sv5000:/usr/src/asterisk-addons-1.2.0# Now the details: I wanted to set up a plain asterisk computer without any more programms. I set up a plain debian sarge system and installed kernel 2.6.14.2. I downloaded, unpacked mysql-5.0.15 under /usr/src/mysql-5.0.15. Then I put the link /usr/src/mysql to this directory. I compiled and installed mysql successfully. I then downloaded asterisk-1.2.0.tar.gz and unpacked it to /usr/src/asterisk-1.2.0 I compiled and installed it successfully with make, make install and make-samples. I then downloaded asterisk-addons-1.2.0.tar.gz and unpacked it to /usr/src/asterisk-addons-1.2.0 I tried to compile and had the problem that the compiler did not find the mysql includes an libs. I had to modify Makefile first. First I added this directory to the MODS, CFLAGS and MLFLAGS. It would be nice to have them in the next update. Afterwards the compiler stopped with the above error's. Is there a new 'snprintf' version used ? Do you have a solution for that ? At the end are the compiler etc. versions. Makefile at /usr/src/asterisk-addons-1.2.0 --- - --- V MODS+=$(shell if [ -d /usr/local/mysql/include ] || [ -d MODS+MODS+/usr/local/mysql/include/mysql ] || [ -d /usr/include/mysql ] MODS+|| [MODS+-d /usr/local/include/mysql ] || [ -d MODS+/usr/local/mysql/include ] || [ -d /opt/mysql/include ]; then echo MODS+cdr_addon_mysql.so app_addon_sql_m ysql.so res_config_mysql.so; MODS+fi) CFLAGS+=$(shell if [ -d /usr/local/mysql/include ]; then echo -I/usr/local/mysql/include; fi) CFLAGS+=$(shell if [ -d /usr/local/mysql/include/mysql ]; then echo -I/usr/local/mysql/include/mysql; fi) --- CFLAGS+=$(shell if [ -d /usr/include/mysql ]; then echo CFLAGS+-I/usr/include/mysql; fi) =$(shell if [ -d CFLAGS+/usr/local/include/mysql ]; then echo CFLAGS+-I/usr/local/include/mysql; fi) =$(shell if [ -d CFLAGS+/opt/mysql/include/mysql ]; then echo CFLAGS+-I/opt/mysql/include/mysql; fi) MLFLAGS= MLFLAGS+=$(shell if [ -d /usr/lib/mysql ]; then echo -L/usr/lib/mysql; MLFLAGS+fi) =$(shell if [ -d /usr/lib64/mysql ]; then echo MLFLAGS+-L/usr/lib64/mysql; fi) =$(shell if [ -d /usr/local/mysql/lib ]; then echo -L/usr/local/mysql/lib; fi) MLFLAGS+=$(shell if [ -d /usr/local/mysql/lib/mysql ]; then echo -L/usr/local/mysql/lib/mysql; fi)--- MLFLAGS+=$(shell if [ -d /usr/local/lib/mysql ]; then echo MLFLAGS+-L/usr/local/lib/mysql; fi) =$(shell if [ -d MLFLAGS+/opt/mysql/lib/mysql ]; then echo -L/opt/mysql/lib/mysql; fi) Details for compiler and libs: dpkg -l | grep gcc ii gcc 4.0.2-1 The GNU C compiler ii gcc-3.3-base 3.3.6-7 The GNU Compiler Collection (base package) ii gcc-4.0 4.0.2-2 The GNU C compiler ii gcc-4.0-base 4.0.2-2 The GNU Compiler Collection (base package) ii libgcc1 4.0.2-2 GCC support library dpkg -l | grep libssl-dev ii
Re: [Asterisk-Users] Realtime Problems
On Monday 21 November 2005 11:18, scott wrote: Hi Thank you for your reply. I have tried various definitions in the sipusers table but none seem to be working :-( I have attached mey structure and content export below for your attention. Many thanks Scott Pinhorne -- -- Table structure for table `sip_users` -- CREATE TABLE `sip_users` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `accountcode` varchar(20) default NULL, `amaflags` varchar(7) default NULL, `callgroup` varchar(10) default NULL, `callerid` varchar(80) default NULL, `canreinvite` char(3) default 'yes', `context` varchar(80) default NULL, `defaultip` varchar(15) default NULL, `dtmfmode` varchar(7) default NULL, `fromuser` varchar(80) default NULL, `fromdomain` varchar(80) default NULL, `fullcontact` varchar(80) default NULL, `host` varchar(31) NOT NULL default '', `insecure` varchar(4) default NULL, `language` char(2) default NULL, `mailbox` varchar(50) default NULL, `md5secret` varchar(80) default NULL, `nat` varchar(5) NOT NULL default 'no', `deny` varchar(95) default NULL, `permit` varchar(95) default NULL, `mask` varchar(95) default NULL, `pickupgroup` varchar(10) default NULL, `port` varchar(5) NOT NULL default '', `qualify` char(3) default NULL, `restrictcid` char(1) default NULL, `rtptimeout` char(3) default NULL, `rtpholdtimeout` char(3) default NULL, `secret` varchar(80) default NULL, `type` varchar(6) NOT NULL default 'friend', `username` varchar(80) NOT NULL default '', `disallow` varchar(100) default 'all', `allow` varchar(100) default 'g729;ilbc;gsm;ulaw;alaw', `musiconhold` varchar(100) default NULL, `regseconds` int(11) NOT NULL default '0', `ipaddr` varchar(15) NOT NULL default '', `regexten` varchar(80) NOT NULL default '', `cancallforward` char(3) default 'yes', PRIMARY KEY (`id`), UNIQUE KEY `name` (`name`), KEY `name_2` (`name`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1 ROW_FORMAT=DYNAMIC AUTO_INCREMENT=2 ; -- -- Dumping data for table `sip_users` -- INSERT INTO `sip_users` VALUES (1, '114', NULL, NULL, NULL, '114', 'yes', 'default', '192.168.10.136', 'info', NULL, NULL, '114', '192.168.10.136', NULL, NULL, NULL, NULL, 'no', NULL, NULL, NULL, NULL, '', NULL, NULL, NULL, NULL, '114', 'friend', '114', 'all', 'g729;ilbc;gsm;ulaw;alaw', NULL, 0, '', '', 'yes'); From wiki: You do not need to insert the ipaddr, port or regseconds information. These columns will be updated periodicaly by RealTime. That's valid also for the fullcontact; it appears if: rtcachefriends=yes in sip.conf otherwise stays empty. If I am not mistaken you have filled in fullcontact with 114. benchev ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime IAX
On Saturday 19 November 2005 01:05, Carlos Chavez wrote:
I've been having a problem dialing IAX extensions since I implemented
Realtime for IAX Extensions. The problem is that I cannot seem to dial in
a simplified manner an extension like:
_9001.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:3})
snip
port: 0
Try port:4569 or whatever port you're using
because by default it try port 0 and than you have congestion
as a result.
benchev
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Re: [Asterisk-Users] Asterisk realtime extensions context inclusion
Don't know will this do, but a simple comparison
may give you a hint:
1.2-rc2 extentions.conf
[default]
switch = Realtime/[EMAIL PROTECTED]
switch = Realtime/[EMAIL PROTECTED]
;switch = Realtime/[EMAIL PROTECTED]
1.0.9 extentions.conf
[default]
include = astcc
include = internal
[astcc]
exten = _011N.,1,Set(CALLERID(name)=${CALLERIDNAME} - ${CALLERIDNUM})
exten = _011N.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten = _011N.,3,Hangup
[internal]
exten = _4XXX,1,Dial(sip/${EXTEN},20,r)
or_whatever = _way, you, use
Both do the same job. See how simple is with realtime?
You could do also:
[default]
switch = Realtime/[EMAIL PROTECTED]
switch = Realtime/[EMAIL PROTECTED]
[default_vonage]
switch = Realtime/[EMAIL PROTECTED]
[default_sipgate]
switch = Realtime/[EMAIL PROTECTED]
Hope it helps?
benchev
On Monday 14 November 2005 09:44, Daniel Clark wrote:
Thanks for the reply, it's an approach I didn't think of to simply include
the information from the other contexts into where I would be including
from. In most cases that would work, but not in my case. Each user of my
system will be able to place outgoing calls using their own sip connection
(as in one they create with sipgate or vonage etc). To ensure that each
user can dial out with their own sip connection and nobody else's they are
each getting their own context and that context is the only place in the
dialplan to dial that particular external sip connection. For a small
amount of users it's possible to include all the information in each
context, however I'm dealing with 15,000 users and would like a database
small enough to fit on the hard disk!
Would it not be possible to do something with the Goto app? In each persons
dialplan I can have an extension to catch internal numbers and then forward
to another context using exten = 1,1,Goto(context2) or something like
that?
I have to stick with the database option as there are other applications
that need quick access to the information it holds. It's not really
possible to generate the flat file for all the contexts when at some times
that would mean generating the file over 1,000 a day and reloads of the
database each time. If I can stick with the realtime database in any way, I
would much prefer to.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of snacktime
Sent: 14 November 2005 07:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk realtime extensions context
inclusion
On 11/13/05, Daniel Clark [EMAIL PROTECTED] wrote:
Hi
I'm using asterisk realtime to control all of my extensions. As part of
this I need to be able to dynamically create new contexts and extensions.
The new contexts I create will also include existing contexts. Does anybody
know the how to specify context inclusion for asterisk realtime as the
database only has colums for id, context, exten, priority, app and appdata.
You can't. Since those other contexts are in the database, why not just
select them and then insert them into the newly created context?
Or better yet dump realtime and generate extensions.conf from your own
database schema. You could even use the realtime schema with just a
couple of extra fields for things like include files, that way you dont'
have to throw away the work you have already done.
Asterisk doesn't handle database failures very well. Maybe it's been fixed
now, but for instance a dialplan reload used to wipe out your whole
dialplan if the database was down instead of just skipping the reload. I
spent quite a bit of time writing an application for ARA at one point, only
to toss it all out after seeing how it actually worked. I still think it's
a good idea, and I don't mean to disparage those who put all the work into
it, but it's implementation leaves something to be desired.
Chris
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Re: [Asterisk-Users] Setting up externip and local net for Asterisk behind NAT
I was not successful to make my Asteirsk receive calls. Please help me to set it up. snip But the problem begins when I dial to my Asterisk server from my other phone which is 514-854-7804 What is this other number? Is it a DID, ported, what? Presume it DID... On debugging, I get the following screen which shows that it is successfully receiving the call but is not directing it to my extension 201, which is X-Lite. === Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 read about s here: http://www.voip-info.org/wiki/index.php?page=Asterisk+standard+extensions or better start: http://www.voip-info.org/wiki/index.php?page=Asterisk%20howto%20dial%20plan snip === My sip.conf is === [general] snip [201] username=201 type=friend secret= qualify=no port=5060 nat=yes host=dynamic dtmfmode=rfc2833 context=internal canreinvite=no callerid=Home Phone 201 [trunk] username=8 type=peer secret=47076 host=209.167.xxx.xxx [trunk_incoming] type=user secret=47076 host=209.167.xxx.xxx === My extensions.conf has following lines to deal with incoming SIP calls === [external] exten = _X.,1,Dial(SIP/201,60,tr) snip As I said above: presumed DID add: [incoming] exten = 5148547804, 1, Dial(SIP/201,60,tr) and include it to internal: [internal] include = incoming Thanks, Zeeshan A Zakaria Is it not? let us know is it ok. benchev ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any experince with Voip Reach
Hi, Any experiences with Voip Reach. The prices seem quite tempting. They are also listed as Digium partner. Is the service decent? Why that flash(Mozilla doesn't like most most of it, Konqueror went strait to bed). Any IAX2 termination? Thanks in advance, benchev ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g.729 codec
On Monday 14 November 2005 07:20, Mark Quitoriano wrote: Hi, is there a howto to install g.729 codec on asterisk? http://www.digium.com/downloads/ftp/asterisk/g729/README ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] REaltime does not unregister sip peers on the fly
Hi, chick*CLI show version files chan_sip.c File Revision chan_sip.cRevision: 1.907 chick*CLI show version files pbx_realtime.c File Revision pbx_realtime.cRevision: 1.15 chick*CLI show version Asterisk CVS HEAD built by root @ chick on a i686 running Linux on 2005-11-09 14:28:18 UTC extconfig.conf sipusers = mysql,asterisk,sip_buddies sippeers = mysql,asterisk,sip_buddies iaxusers = mysql,asterisk,iax_buddies iaxpeers = mysql,asterisk,iax_buddies extentions = mysql,asterisk,extensions_table Registered sip friends work great through realtime. However, setting in sip.conf: rtcachefriends=yes rtupdate=yes rtautoclear=yes ignoreregexpire=no ( trying to have show peers available, but attempting to clear the cache) and deleting the secret the friend , this friend :) stays registered: Seeding..., Saved... and all. Weren't they supposed with realtime to get busted on the fly at the next registration without sip reload? Or just created (on the fly)? Anybody? benchev ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] REaltime does not unregister sip peers on the fly but not only...
In addition to the mail below, It is not the realtime! ARA is great. Moving the peers to sip.conf, and ignoring extconfig.conf for a test, discovered that, when left empty (secret=blank_space) is ignored as commented (;secret=whatever). Obviously the sip channel was actually prepared for the realtime sip_buddies table. Means, columns secretmd5secret were left empty are to be considered Not set. Thinking out load, for me, secret=blank_space meant that either the client should have literally blank password or should not be able to register, isn't it. If you don't want secrets you comment it like this (;secret=not_needed). However, do not leave secret empty if you require passwords from your users. Simply set secret=/ or something similar :). On Saturday 12 November 2005 17:45, [EMAIL PROTECTED] wrote: Hi, chick*CLI show version files chan_sip.c File Revision chan_sip.cRevision: 1.907 chick*CLI show version files pbx_realtime.c File Revision pbx_realtime.cRevision: 1.15 chick*CLI show version Asterisk CVS HEAD built by root @ chick on a i686 running Linux on 2005-11-09 14:28:18 UTC extconfig.conf sipusers = mysql,asterisk,sip_buddies sippeers = mysql,asterisk,sip_buddies iaxusers = mysql,asterisk,iax_buddies iaxpeers = mysql,asterisk,iax_buddies extentions = mysql,asterisk,extensions_table Registered sip friends work great through realtime. However, setting in sip.conf: rtcachefriends=yes rtupdate=yes rtautoclear=yes ignoreregexpire=no ( trying to have show peers available, but attempting to clear the cache) and deleting the secret the friend , this friend :) stays registered: Seeding..., Saved... and all. Weren't they supposed with realtime to get busted on the fly at the next registration without sip reload? Or just created (on the fly)? Anybody? benchev ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI environment dump callerid
Hi,
Since * 1.2-beta1 (incl CVS HEAD) there is a change in the
callerid's output to STDERR when an AGI environment
dump is requested:
Asterisk CVS HEAD built by root @ chick on a i686 running
Linux on 2005-11-06 16:35:14 UTC
AGI Environment Dump:
-- accountcode =
-- callerid = 1234689
-- calleridname = Callee Name
-- callingani2 = 0
-- callingpres = 0
-- callingtns = 0
-- callington = 0
-- channel = SIP/22-f55e
-- context = default
-- dnid = 19147858756
-- enhanced = 0.0
-- extension = 19147858756
-- language = en
-- priority = 1
-- rdnis = unknown
-- request = dump.agi
-- type = SIP
-- uniqueid = 1131381756.13
but ... Connected to Asterisk 1.0.9 currently running on dog
(pid = 28360)
AGI Environment Dump:
-- accountcode =
-- callerid = Callee Name 1234689
-- channel = SIP/22-9351
-- context = default
-- dnid = 19147858756
-- enhanced = 0.0
-- extension = 19147858756
-- language = en
-- priority = 1
-- rdnis = unknown
-- request = dump.agi
-- type = SIP
-- uniqueid = 1131381457.0
Thus my question was which is the future-to-be callerid
format?
1. -- callerid = 1234689
-- calleridname = Callee Name
OR
2. -- callerid = Callee Name 1234689
Nothing wrong with that in general since clid, as
${CDR(clid)}, is still being written correctly in 1.0.7,
1.0.9,
1.2-beta12 and CVS HEAD in the usual cdr database/table,
and in any custom table through
$dbh-quote($callerid).
However, since * 1.2-beta1 (incl CVS HEAD), when
AGI(perl) script try $callerid=$input{callerid} it results
to $dbh-quote($callerid) calleridnum(by
default it appears eq to callerid), only.
/* Obviously, because in res_agi.c $Revision: 1.53 $:
fdprintf(fd, agi_callerid: %s\n, chan-cid.cid_num ?
chan-cid.cid_num : unknown);
fdprintf(fd, agi_calleridname: %s\n, chan-cid.cid_name ?
chan-cid.cid_name : unknown); */
Changing to $callerid=$input{calleridname} is inserted as
requested.
Trying to group both callerid attributes results in an empty
string.
Playing with the dilaplan yet damages ${CDR(clid)}
record.
Any thoughts?
benchev
-
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