Re: [asterisk-users] unable to hear voice with asterisk 1.4.15
On Monday 28 January 2008 14:00:27 Rahul Yadav wrote: > Hi all > > i am getting a serious problem.I am using asterisk 1.4.15 and dialing > outbound through sip. > The problem is that whenever i dial a number the other person can hear my > voice but i dont hear anything. Have you tried: http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions Boyko ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set CDR userfield in a realtime dialplan
On Wednesday 09 January 2008 09:54:59 Yves Räber wrote:
> Hello,
>
> I'm using Asterisk with Realtime extensions and ODBC CDR. And I'm have
> some trouble with the CDR userfield that is not changed when using the
> SET command in the realtime dialplan.
> In my dialplan (extensions.conf, the file) I'm setting the userfield
> like this :
>
> exten => s,n,Set(CDR(userfield)="X")
>
> Later, my dialplan switches to the realtime part and this is an extract
> for what is inside :
> ===
> id | context | exten | priority | app | appdata
> ===
> 12 | script | s | n| SET | CDR(userfield)="Y"
> ===
>
> I can show that the command is executed :
> -- Executing Set("SIP/siemens1-081ca290", "CDR(userfield) = Y")
>
> But in my CDR, the old value is saved (X in this case).
Into a database the line exten => s,n,Set(CDR(userfield)="X")
should be enterd as:
context | exten | priority | app | appdata
means in your case:
your_context| s|n|Set(CDR(userfield)|X
Boyko
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Re: [asterisk-users] FWD and IPCall
> > I have a problem. I have tried everything that is in the book "The > > Future of Telephony" as well as on the FWD (freeworlddialup) website, > > and there is still a problem. My asterisk box is not able to associate > > with the FWD server. I get: > > Registration Rejected by [insert IP], and I can't use my IPCall number > > to reach my Asterisk box. > > Any suggestions? If you try dnsmgr.conf enable=yes; what happens? Boyko ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] b2bua
On Saturday 05 January 2008 00:45:00 ameel wrote: > Is there a way to disable the b2bua feature in asterisk. > I would like asterisk to work as a sip server and not be involved in the > RTP path between phones. You can do it by setting BOTH peers canreinvite=yes http://www.voip-info.org/wiki/view/Asterisk+SIP+media+path Boyko ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to forward call on SIP channel after SIP response 302 Moved Temporarily
On Friday 04 January 2008 11:58:25 Tomasz Zieleniewski wrote: > Hi, > > I have the following problem that when asterisk receives SIP response 302 > it cannot forward the call > I get such debug: > [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel > type registered for 'Local' > [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to > create local channel for call forward to 'Local/[EMAIL PROTECTED]' (cause > = 66) Maybe this: "Local channel Description: Local Proxy Channel Driver Syntax: Local/[EMAIL PROTECTED]/n Configuration file: none chan_local is a pseudo-channel. Use of this channel simply loops calls back into the dialplan in a different context. Useful for recursive routing. Notes: Adding "/n" at the end of the string will make the Local channel not do a native transfer (the "n" stands for "n"o release) upon the remote end answering the line. This is an esoteric, but important feature if you expect the Local channel to handle calls _exactly_ like a normal channel. If you do not have the "no release" feature set, then as soon as the destination (inside of the Local channel0 answers the line, the variables and dial plan will revert back to that of the original call, and the Local channel will become a zombie and be removed from the active channels list. This is desirable in some circumstances, but can result in unexpected dialplan behavior if you are doing fancy things with variables in your call handling. " Boyko ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway behind SIP ATA?
On Thursday 03 January 2008 16:38:35 Remco Barendse wrote: > On Thu, 3 Jan 2008, Benchev wrote: > > Basically Grandstream HT286 is a single port FXS ATA. > > In order to interconnect GSM gateway one would need FXO. > > Are you sure it gives you "new" dialing tone or this is the * itself > > you hear? > > Yes, i am positive that i get a new dialtone from the GSM Gateway. > > If i dial DTMF codes from a SIP phone connected to Asterisk, i can see the > digits appear in the display of the GSM Gateway. But it is a bit > incovenient to call an internal extension, wait for the dialtone and then > punch in all the numbers of the cell phone i need to call. > > I would prefer Asterisk to decide where / how to route the call and send > the DTMF inband to the ATA device. Yep. I've found a gsm gateway that does "...calls from VoIP to GSM and GSM to VoIP and uses the SIP protocols so is ideal for use as a SIP Trunk on many SIP based VoIP PBX Phone Systems..." Sorry, didn't know such a thing exists. I don't think it matters dialing DTMF or not a simple dialplan trick should do. >From home (Europe) I do: [gsm-out] exten => _0N.,1,Dial(SIP/gsm_gateway) exten => _0N.,2,Hangup Means all calls starting with zero and have digits from 2-9 afterwards go here. The mobile numbers start with 088 or 089. Otherwise I dial 01 for US and 011 for International. These are just ideas. You could figure out something else that fits your needs. Boyko ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to retrieve my voice mail ... (password incorrect)
On Thursday 03 January 2008 22:15:07 William Herrera wrote: > I installed one last week (downloaded and installed the latest) and > everything went beautiful and every thing works fine, however, my client > has voice mail and no matter what phone I use, or what password I enter, or > in which way I try I always get the same answer from the server: "Password > incorrect". I even deleted the extension and recreated it with a different > number and get the same results. I looked in the voicemail.conf to verify > that everything reflected correct in there and it is. The voicemail.conf > shows ext. 200 with password 200 (which is what I enter). Have you tried to change the dtmfmode of the respective peer to inband or rfc2833 or visa versa? Boyko ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway behind SIP ATA?
On Thursday 03 January 2008 15:28:15 Remco Barendse wrote: > I have an analog GSM Gateway that is connected to a normal SIP ATA device. > > Basically what it does is this : when you call the extension nr. of the > SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) > dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia > a Grandstream HT286. > > I would like to use the GSM Gateway to route my outbound cellular calls, > how do i do this in Asterisk? Basically Asterisk should dial the extension > number and then send required number as DTMF tones to the Gateway through > the ATA. Basically Grandstream HT286 is a single port FXS ATA. In order to interconnect GSM gateway one would need FXO. Are you sure it gives you "new" dialing tone or this is the * itself you hear? Boyko ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cmd mysql
> I've been using mysql databases more and more. I've run
> across a couple
> of instances where I've either made a mistake on the ip
> address of the
> mysql database or for whatever reason, mysql wasn't
> running. In those
> instances, I've noted that the mysql command will hang
> indefinitely
> (I've counted to 40 before killing it).
>
> The offending line is:
>
> exten => s,1,MYSQL(Connect connid 192.168.103.15
> mysqladmin 'x' did)
>
> Is there a way to specify a timeout for the mysql
> command?
>
> I'm not finding anything on Google, voip-info or the
> documents in the
> add-ons directory. Any help would be appreciated.
>
By experience, after every query sequence you should use
MYSQL(Clear ${resultid}) otherwise they'll become 400.
As it is said into app_addon_sql_mysql.c
"Frees memory and datastructures associated with result
set."
Hope that helps,
Benchev
-
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за 2 години.
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За повече информация:www.btc.bg
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Re: [asterisk-users] MySQL Update from exten
> Barton Fisher wrote:
> > I've tried every combination I could find on the net
> and so far there is
> > no joy
> > The thing is I can do this update from the command
> line: Maybe some new
> > eyes might find the answer?
> >
> > exten => update,1,MYSQL(Connect connid localhost root
> password dax)
> > exten => update,n,MYSQL(QUERY resultid ${connid}
> UPDATE\ caller\ SET\
> >
>
lastcall=${LASTCALL}\,totalcalls=totalcalls+1\,currentcalls=currentcalls+1\
> > WHERE\ dnis=\'${IVR-Exten}\'\ AND\
> ani=\'${CALLERID(number)}\')
> > exten => update,n,MYSQL(Clear ${resultid})
> > exten => update,n,MYSQL(Disconnect ${connid})
> >
> > Asterisk logs says:
> > Apr 19 15:50:05 VERBOSE[19740] logger.c: -- Executing
> > MYSQL("SIP/5400-b7bbfaf0", "QUERY resultid 201 UPDATE
> caller SET
> > lastcall= 04/18/07 11:12:55, totalcalls= totalcalls+1,
> currentcalls=
> > currentcalls+1 WHERE dnis= '7690' AND ani= '5400'") in
> new stack
> > Apr 19 15:50:05 WARNING[18333] app_addon_sql_mysql.c:
> Identifier 200,
> > identifier_type 2 not found in identifier list
> > Apr 19 15:50:05 WARNING[18333] app_addon_sql_mysql.c:
> Invalid result
> > identifier 200 passed in aMYSQL_clear
> >
Probably
exten => update,n,MYSQL(QUERY resultid ${connid} UPDATE\
caller\ SET\
lastcall=${LASTCALL}\,totalcalls=${TOTALCALLS}+1\,currentcalls={CURRENTCALLS}+1\
WHERE\ dnis=\'${IVR-Exten}\'\ AND\
ani=\'${CALLERID(number)}\')
Benchev
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Re: [asterisk-users] Stuck on MySQL UPDATE
> What I'm retrying to do is update mysql field with the
> new message ID
> that was just recorded. Ideally, I'd like to specify
> the field to update using a variable ${BINID} and use
> ${NEWPHRASENAME}
> for the value - I'm not sure asterisk will allow
> using a variable for the field name and if not, I'll
> attempt to create
> an exten for each bin to update.
>
> Here the method I'd like to use:
> exten => sav,n,MYSQL(Connect connid localhost root
> password dax)
> exten => sav,n,MYSQL(QUERY resultid ${connid}UPDATE\
> dnislookup\ SET\
> ${BINID}\ =\ ${NEWPHRASENAME}\ WHERE\ dnis\ =\
> ${IVR-Exten})
>
> But I've tried this too:
> exten => sav,n,MYSQL(Connect connid localhost root
> password dax)
> exten => sav,n,MYSQL(QUERY resultid ${connid}UPDATE\
> dnislookup\ SET\
> bin2\ =\ ${NEWPHRASENAME}\ WHERE\ dnis\ =\ ${IVR-Exten})
>
> However, neither one of these saves to new value into the
> bin2 (or
> ${BINID}) field.
Just to reassure you that the method works, below is what
works with me:
exten => _1NXXNXX,n(continue),MYSQL(Query resultid
${connid} update\ cards\ set\ used\ =used+${FREEZEFUNDS}\
where\ number\ ="${CDR(accountcode)}")
First there is no "fetch" i.e.
exten => _1NXXNXX,n,MYSQL(Fetch fetchid ${resultid}
var1 var2 var3 var4)
Second there is no space "${connid}UPDATE\" should be
"${connid} UPDATE\".
And check how bin2 "remembers" the variables you assign to
it: probably
should use _bin2 or __bin2 for instance:
exten =>
_1NXXNXX,n,Set(__bin2=${MATH(${VAR3}-${VAR4},int)})
Hope that helps,
Benchev
-
Най-добрият начин да научаваш новините,
които те интересуват.Бързо лесно и безплатно!
новини с филтър
http://www.radar.bg/
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Re: [asterisk-users] Re: Arrays ???
> Im aware of ARRAY and would have used it but it only sets variables - you
> cant reference the array by name or its elements as is common in other
> programming languages.
>
> Would it be possible to use ASTDB and the "while" application? For
> instance the user inputs 3 numbers to be used for Follow me and the dial
> plan iterates thru them checking for CF CW etc etc changing the numbers if
> needs be then dialling.
>
Basically, for automation, the is usually doomed to be the callerid.
A pair of may have only one . I.e there's no need
to delete an entry with a pair just to change the value, you just
"put" THE pair with the "new" and it's overwritten.
Probably, if you use different ...
However you might do :
database put CF/123456 23456,34567,45678
And while:
exten => s,1,Set(temp=${DB(CF/${ARG1})})
exten => s,n,Set(temp1=${CUT(temp,\,,1)}); if macro
exten => s,n,Set(temp2=${CUT(temp,\,,2)})
exten => s,n,Set(temp3=${CUT(temp,\,,3)})
exten =>
s,n,Dial(LOCAL/[EMAIL PROTECTED]&LOCAL/[EMAIL PROTECTED]&LOCAL/[EMAIL
PROTECTED],15,r)
...
Another shortcut is with only one and timeout 15 secs:
database put CF/123456 23456|15
First it rings for 15 secs to 23456and in case they'd forgot to
disable CF, starts ringing 123456.
Hope the above will give you even more ideas.
Benchev
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Re: [asterisk-users] Arrays ???
On Wednesday 02 August 2006 18:09, Pele Zico wrote: > Is their a way of implementing arrays in asterisk?? What im trying to do > is allow the user to input numbers (internal and external) from his line > thats gets saved via astdb - (follow me purposes) - i would like to look t > each number individually and check for CF, CFB CFU etc etc . can this be > done or would i have to use AGI. i prefer not to use agi however *CLI> show version Asterisk SVN-trunk-r37291 *CLI> show function ARRAY -= Info about function 'ARRAY' =- [Syntax] ARRAY(var1[|var2[...][|varN]]) [Synopsis] Allows setting multiple variables at once [Description] The comma-separated list passed as a value to which the function is set will be interpreted as a set of values to which the comma-separated list of variable names in the argument should be set. Hence, Set(ARRAY(var1|var2)=1\,2) will set var1 to 1 and var2 to 2 Note: remember to either backslash your commas in extensions.conf or quote the entire argument, since Set can take multiple arguments itself. So probably with 1.4 will come. 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] sip realtime
On Thursday 27 July 2006 15:07, Andrea Spadaccini wrote: > Ciao Benchev, > > > Also register=> can be done only from a .conf file. > > Well, I'm experimenting right now with this, and I can tell you that > register => works even with static realtime. Not "even", it *must* work because if one uses realtime static, the equivalent file in /etc/asterisk i.e. sip.conf, should be deleted. Ciao, 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] sip realtime
On Wednesday 26 July 2006 00:03, marek cervenka wrote: > i'm reading a lot docs about asterisk realtime > but i cannot understand how works sip realtime static > > i need NAT/qualify for SIP. this is not possible with dynamic realtime > i want > - save data to sql > - asterisk -rx "reload" to read config (sip.conf with sip users) from sql > > it is possible? For sip realtime static you have probably read: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Static However, NAT/qualify for SIP(users) is perfectly possible with "dynamic" realtime: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip And `sip_buddies` table gives extensive opportunities(including nat and qualify). And there is the advantage of using it(realtime), you do not need to reload when a new user comes. (This is valid for the needed extensions and voicemail attributes, as well) Sorry for twisting a bit your question, but basically "realtime static" means to store a .conf file into a database(in which case you must delete its equivalent from /etc/asterisk); "realtime" is when you store users, peers and friends into the database, keeping the skeletons of sip&iax2.conf files in /etc/astrisk. In that case the "users, peers and friends" sip or iax2 info is being read "on the fly". The appropriate extensions though, must be addressed with "switch => Realtime " statement from extensions.conf. Since all .conf files exist they have precedence. Also register=> can be done only from a .conf file. Hope it 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] Asterisk Realtime Macros
On Monday 24 July 2006 19:47, Jon Scottorn wrote: > If I setup the extension like such: > >exten => 101,1,Dial(SIP/101) >exten => 102,1,Dial)SIP/102) > > I can dial back and forth between the two phones. > > When I switch it to use the stdexten macro and change the extension like > such > >exten => 101,1,Macro(stdexten,101,sip/101) >exten => 102,1,Macro(stdexten,102,sip/102) > > I can not dial each extension and this is what reports on asterisk cli: > > My question is what has to be in the mysql extenstions_table to get the > macro to work? > > Here is what is in my extensions_table: > > mysql> select * from extensions_table; > ++-+---+--+---+--+ > > | id | context | exten | priority | app | appdata | > > ++-+---+--+---+--+ > > | 1 | default | 101 |1 | Macro | stdexten,101,sip/101 | > | 2 | default | 102 |1 | Macro | stdexten,102,sip/102 | > You should substitute commas with a "pipe"; i.e. your "select" should look: | 1 | default | 101 |1 | Macro | stdexten|101|sip/101 | | 2 | default | 102 |1 | Macro | stdexten|102|sip/102 | On the wiki is complete mess with the commas ... 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] Asterisk + fax
> If there any way to pass on that problem, like i know the source should > cancel the echo on the line. > > In addition i am trying to connect regular fax through ata to asterisk > with no success. > > Regular Fax machine -> ata -> Asterisk. > > ata is registering to the asterisk as regular extension. Instead of phone > after the ata i have a fax machine. i am trying to send a fax to my ATA FAX > MACHINE with no success it's falling after the dialing i don't see the > connection stage. > > I think i am missing something. > Any help will be appreciated. The ATA must be T.38 enabled. For instance: PAP2 is not. 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] Now that Nufone is dead...
On Wednesday 05 July 2006 20:00, John Kington wrote: > At 09:29 AM 7/5/2006 +0300, you wrote: > >I have tollfree numbers with Nufone working OK. > >But what I like most is the regular numbers with charge/month > >but no charge/min on incoming calls... > > Did your tollfree number(s) with Nufone get cut-off in April? > Did you keep the same number or did you signup for another number? > I requested Nufone transfer my tollfree number in May and it is still > not working (code is 77-4). I am wondering if this has happened to > everyone or if my number fell through the cracks. > The one cut-off is steel dead(the same code), but I am keeping it ...it 's not only you. I hope somewhere down the road to have it back. ___ --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] Need help with config-files
> > You [internal] is rather a macro than [internal]
> >
>
> Is a macro reqally required? If wildcards suffice, why use a macro?
Thanks for getting involved, since you might be in better help.
I his case maybe yes, since his [internal] is a macro-like, is it not?
> > [internal]
> > exten => _ZX[0-8]X,1,DBget(temp=CFIM/${EXTEN})
> > exten => _ZX[0-8]X,2,Dial(SIP/${temp})
> > exten => _ZX[0-8]X,3,Dial(SIP/${EXTEN},20)
> > exten => _ZX[0-8]X,102,Goto(${EXTEN},3)
> > exten => _ZX[0-8]X,4,VoiceMail(u${EXTEN})
> > exten => _ZX[0-8]X,104,VoiceMail(b${EXTEN})
> > exten => _ZX[0-8]X,5,Hangup
> >
> > I would suggest
> > [internal]
> > exten => _1001,1,Macro(stdexten,1001,sip/1001)
> > exten => _1002,1,Macro(stdexten,1002,sip/1002)
>
> Any special reason to use a number three times in the call to the macro?
> If it is inherently the same, use one parameter.
The only reason is to show him where the extension goes and where the device
(in this case his extension and peer name are equal to each other)
> e.g:
>
> exten => _1001,1,Macro(stdexten,${EXTEN},sip/${EXTEN})
>
> And maybe eliminate the second parameter.
>
> > etc...
> > and then use a macro i.e.:
> > [macro-stdexten] ; lastcaller, BlackListing, CFIM
> > ;; Standard extension macro:
> > ;; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
> > ;; ${ARG2} - Device(s) to ring
> > ;;
> > exten => s,1,LookupBlacklist ; If CID blacklisted, goto 102
>
> Isn't "jump to priority" deprecated?
Yep, and also DBget and DBput but we don't know which version he uses :)
> Anyway, you have too many numbers below. Why not use "n" and labels
> instead?
Absolutely!
> exten => s,1,Do,Something
> ; maybe replace the following two by a very long GotoIf with an obscure
> ; functions syntax ;-)
> exten => s,n,A(test)
> exten => s,n,GotoIf($[ "${testresult}" = "bad" ],bad)
> exten => s,n,GoOn(AsUsual)
> exten => s,n,Hangup
> exten => s,n(bad),Handle(bad)
>
> This makes it possible to add/remove lines without editing everything in
> the neighbourhood.
>
Looking at his extensions and sip.conf I thought ,the best
for Thomas is to simplify the things down to the point everything gets
working. Probably to use regexten in the sip peers?
I think [internal] is the place to declare the internal extensions
not in [incoming-from-sip] and once a call comes to enter
a macro loop.
Thanks,
Benchev
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Re: [Asterisk-Users] Need help with config-files
On Tuesday 04 July 2006 17:55, Thomas Jacobsen wrote:
> Hello,
>
> I decided to resend the files, because i made alot of typos in them. -
> Please use these files instead.
You [internal] is rather a macro than [internal]
[internal]
exten => _ZX[0-8]X,1,DBget(temp=CFIM/${EXTEN})
exten => _ZX[0-8]X,2,Dial(SIP/${temp})
exten => _ZX[0-8]X,3,Dial(SIP/${EXTEN},20)
exten => _ZX[0-8]X,102,Goto(${EXTEN},3)
exten => _ZX[0-8]X,4,VoiceMail(u${EXTEN})
exten => _ZX[0-8]X,104,VoiceMail(b${EXTEN})
exten => _ZX[0-8]X,5,Hangup
I would suggest
[internal]
exten => _1001,1,Macro(stdexten,1001,sip/1001)
exten => _1002,1,Macro(stdexten,1002,sip/1002)
etc...
and then use a macro i.e.:
[macro-stdexten] ; lastcaller, BlackListing, CFIM
;; Standard extension macro:
;; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
;; ${ARG2} - Device(s) to ring
;;
exten => s,1,LookupBlacklist ; If CID blacklisted, goto 102
exten => s,2,Set(temp=${DB(CFIM/${ARG1})}); Get CFIM (Call Forward IM) key
exten => s,3,GotoIf($["${temp}" = ""]?,6); go to s,6 if no CFIM
exten => s,4,GotoIf($[${temp:0:6} = ${temp}]?s|5:s|9) ; internal # of 6 digits
or int more than 6
exten => s,5,Dial(Local/[EMAIL PROTECTED]/n) ; Unconditional forward
exten => s,6,Set(DB(lastcaller/${ARG1})=${CALLERIDNUM}) ; Note the last caller
exten => s,7,Dial(${ARG2},20) ; Call the device with a 20 sec.If no j NOT
jumping
exten => s,8,Goto(s-${DIALSTATUS},1)
exten => s,9,Set(temp=${DB(CFIM/${ARG1})}) ; Get again CFIM
exten => s,10,Set(CDR(accountcode)=${ARG1}) ; set ARG1 = accountcode thus
changing whom to bill
exten => s,11,Playback(pls-hold-while-try) ; advise him
exten => s,12,Dial(Local/[EMAIL PROTECTED]/n)
exten => s,102,Goto(blacklisted,s,1) ; Blacklisted CallerID
exten => s,103,Goto(s,7) ; No CFIM key
exten => s-NOANSWER,1,Voicemail(u${ARG1})
exten => s-NOANSWER,2,Goto(s,1) ; If they press #, return to start
exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail -> busy
exten => s-BUSY,2,Goto(s,1) ; If they press #, return to start over
exten => _s-.,1,Goto(s-NOANSWER,1); Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If they press * -> VoicemailMain
You should change it or use the sample from the original
extensions.conf to answer your needs.
The macro does not need to be included in a context.
And as you see priorities jump n+101.
Hope it helps.
Benchev
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Re: [Asterisk-Users] Now that Nufone is dead...
On Tuesday 04 July 2006 17:32, Martin Joseph wrote: > Who says nufone is dead? > > I use them, but my did is through sellvoip.net > I have tollfree numbers with Nufone working OK. But what I like most is the regular numbers with charge/month but no charge/min on incoming calls... ___ --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-biz] Selling Bulgarian (+3592) DIDs at 1.5 USD
Edno utochnenie... Ako ne sym bulgarska firma i kupja DID ot vas shte polucha li Invoice za BG DIDs origination? > nikakaw! nie imame nyakolko stotin klienta izwan BG. Mojete da poglednete > http://bgnumber.info za podrobna informaciya. > > - Original Message ----- > From: "Benchev" <[EMAIL PROTECTED]> > To: "Enky" <[EMAIL PROTECTED]> > Sent: Thursday, 25 May, 2006 21:36 > Subject: Re: [asterisk-biz] Selling Bulgarian (+3592) DIDs at 1.5 USD > > > Zdarweite, > > Znachi ako preprodam vash DID na > > client v USA njama problem? > > > > Benchev > > > >> Zdraweyte, > >> > >> principno ne e nezakonno. Licenzionniya rejim nalaga iziskwaniya za > >> parametrite na wrazkata do tochkata na terminaciya, no tazi tochka e w > >> Bulgaria, kadeto signalizaciyata se preobrazuwa i taka iziskwaniyata na > >> zakona sa spazeni. > >> > >> - Original Message - > >> From: "Benchev" <[EMAIL PROTECTED]> > >> To: > >> Cc: "Enky" <[EMAIL PROTECTED]> > >> Sent: Wednesday, 24 May, 2006 23:01 > >> Subject: Re: [asterisk-biz] Selling Bulgarian (+3592) DIDs at 1.5 USD > >> > >> > On Monday 22 May 2006 21:55, Enky wrote: > >> >> Selling Bulgarian (+3592) DIDs at 1.5 USD. Minimum 100 pcs. > >> > > >> > Is it legal to sell Bulgarian DIDs to a foreign entities? > >> > > >> > 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]Asterisk <-IP-> Siemens HiPath 3750
Hi Nguyen , I haven't got the opportunity to make my project real due to business obstacles, but I still think that it should work. All that follows is a theory, but there are guys on the list that might help you with more practical advises. > I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the > manual of Hipath 3500 yet (have to buy from local vendor), so I was not > sure are these thing possible > > Scenario: Asterisk|TE110P->TMS2|Hipath 3750 ->(16 CO lines) PSTN I had the same idea because I wanted to save on the card side(single span), and use the Hipath as a "channel bank" :-) > - Is this possible for Asterisk Users call out using CO lines? Some of > Siemens guys told me that I need an DISA card for this? Is this true? Most of the time the Siemens guys don't know what is Asterisk. Basically TE110P *is* a DISA since it gives Direct Inward System Access (if this is what they mean by DISA) Below is a threat I found with exactly the same scenario like yours: http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.html And this proves that the idea must work. > - When the call arrived from PSTN through CO line, can it be forwarded to > Asterisk? Again, they says that we require the DISA card. As far as anything gets into Asterisk then you are free to do whatever you want. I don't know what DISA they are talking about? Do they mean S2M or similar thing(but TMS2 is S2M)? Anyone? Sorry for not being able to help, but hope somebody else would 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] just softphone
On Thursday 18 May 2006 16:32, Ralph Liebessohn wrote: > On 5/18/06, Benchev <[EMAIL PROTECTED]> wrote: > > > I'm trying to start with Asterisk, but I could not put 2 softphones to > > > talk. The asterisk server rejects the connections always when I dial. > > > > > > May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from > > > > Try: > > extensions.conf > > [default] > > include => internal > > [internal] > > exten => 311000,1,Dial(SIP/teste1) > > exten => 311000,2,Hangup ; Hangup is good > > > > exten => 312000,1,Dial(SIP/teste2) > > exten => 312000,2,Hangup > > > > exten => 313000,1,Dial(SIP/teste3) > > exten => 313000,2,Hangup > > > > Put context=internal or default in all your sip "friends". > extensions.conf ( I've changed internal by from-sip) > > [default] > include => from-sip > include => demo > > [from-sip] > exten => 9222,1,Dial(SIP/9222,25) > exten => 9222,2,Hangup > > exten => 9223,1,Dial(SIP/9223,25) > exten => 9223,2,Hangup > > exten => 31200,1,Dial(SIP/312000,25) > exten => 31200,2,Hangup > > > sip.conf > > [general] > context=default > port=5060 > bindaddr=0.0.0.0 > ;srvlookup=yes > > > [9222] > type=friend > callerid = "Nome - 9222" <9222> > username=9222 > secret=9222 > host= dynamic > context=from-sip > dtmfmode=rfc2833 > nat=yes > canreinvite=nocontext=internal Ralph, > canreinvite=nocontext=internal Is the above a copy/paste mistake? Try context=from-sip or default as in you case that's declared in extensions.conf. Delete context=internal. On the other hand, this thing X-light, does it require Use Auth ID or something similar? Try to turn it on/off. 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] just softphone
> I'm trying to start with Asterisk, but I could not put 2 softphones to > talk. The asterisk server rejects the connections always when I dial. > > May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 192.168.0.106 > > What is necessary to put it to work? > There is no need to configure external lines. > > extensions.conf > > [internal1] > exten => 311000,1,Dial(SIP/teste1) > > [internal2] > exten => 312000,1,Dial(SIP/teste2) > > [internal3] > exten => 313000,1,Dial(SIP/teste3) > [teste1] > > > sip.conf > > [teste1] > type=friend > username=teste1 > secret=123 > > qualify=yes > nat=no > host=dynamic > canreinvite=no > context=internal > > [teste2] > type=friend > username=teste2 > secret=123 > > qualify=yes > nat=no > host=dynamic > canreinvite=no > context=internal2 > > [teste3] > type=friend > username=teste3 > secret=123 > > qualify=yes > nat=no > host=dynamic > canreinvite=no > context=internal3 Debug/verbose is too short, but probably your peers cannot "meet" in a mutual context. Try: extensions.conf [default] include => internal [internal] exten => 311000,1,Dial(SIP/teste1) exten => 311000,2,Hangup ; Hangup is good exten => 312000,1,Dial(SIP/teste2) exten => 312000,2,Hangup exten => 313000,1,Dial(SIP/teste3) exten => 313000,2,Hangup Put context=internal or default in all your sip "friends". Hope that would 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
Re: [Asterisk-Users] Please help me...Urgent
> Thank you for your quick response. I have successfully implemented > Intercom (Dialling within my office LAN) using Asterisk. To implement this, > I am using X-Lite Softphone. > > Now, I want to make calls to US using VoIP Asterisk. I think that there is > no need of any external hardware to implement pure VoIP solution. Am I > right? > > I have registered with Vebtel (VoIP IP Telephony Service provider). They > had given me one VoIP Modem called "Voice Finder AP 200" and the below > values: > > Inbound Number: 123456789 > Public IP Number: 55.23.789.145 > Password: xyz > > (These values are dummy values) > > Currently we are making US calls using VoIP provided by "Vebtel". Now, I > want to make US calls using this Vebtel service from Asterisk. How can I do > this? > > I am unable to understand where to give above mentioned values? What > configuration files I should use to implement this using the Vebtel SIP > provider? Do I need to provide any more values to implement this using > Asterisk from Vebtel? > > Waiting for your quick response. Thank you. Hi, You have sent this 10 times and received at least 20 answers but there is no development in you query! Some people have email filters for Urgent. The only "Urgent" in your case is that you urgently need to go to the wiki and have a long reading. Please, show some efforts, benevolence or something... Benchev P.S. http://www.voip-info.org/wiki/view/Asterisk+Configurations+for+connecting+with+VOIP+providers ___ --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] Anyone using the GSMgateway from CyberTelecom ?
Hi Adibar, It took me some time to answer because I was waiting for a positive confirmation from that client of mine. I have no confirmation, but hey, I concider that as good news. It appears that Sam and Cyber-telecom did a good job providing the right support. That model of GSM-gateway works fine. As I did mention before, me personally, I have one from the older models ( 5 months now) and it is rock solid. Thanks very much for your concern. Cheers, Benchev On Tuesday 09 May 2006 13:58, adibar wrote: > Hi Benchev > > Mine is working now. It was set to a courious "billing mode". > After inserting the following command via an analog phone: > > *00#000# hangup > *00#100#0# hangup > > I could finaly use it. I guess you can try it too. Keep > me informed of your success, I wonder if these settings > do help you too. > > Greets > Adibar > > On Wed, Apr 26, 2006 at 10:17:51PM +0300, Benchev wrote: > > Thanks Adibar, (sorry List:-) > > > > You have at least an offer. The only thing I've got so far > > was a promiss to have a decision "tomorrow" .For a month > > now, "tomorrow" is every day after the day before "tomorrow". > > > > However, I hope, that in order to keep it's good reputation CyberTelecom > > will consider FREE OF CHARGE EXCHANGE of the useless GSM boxes > > or do immediate refund or something. > > > > Let's see... > > Cheers, > > Benchev > > > > On Wednesday 26 April 2006 01:17, adibar wrote: > > > Hi Benchev > > > > > > News from the front. Sam is kinda offering me an exchange > > > of my box. But I should return it to him at my cost ;-) > > > > > > Last word is not spoken yet on that, cause I'm really > > > not amused on this ;-) > > > > > > Keeping you in loop. > > > > > > Greets > > > Adibar > > > > > > On Mon, Mar 27, 2006 at 10:05:01AM +0300, Benchev wrote: > > > > Actually I've got five, but the first one I have received > > > > around Xmas and I don't have these problems with it. > > > > I use spa3000 as FXO and the gsm gateway works > > > > seamlessly inbound, outbound, DISA, no annoying sounds, > > > > no DTMF problems. There is one problem however, the gateway does > > > > not transfer correctly the CID to the FXO(at least in my case) > > > > but this could be a sipura problem as well. > > > > > > > > Now, the other 4 seam to be a different model or something > > > > and one should be very careful ordering that thing since you never > > > > know which model you are going to receive. > > > > They are used with "no-brand-name" FXO/FXS ATAs > > > > but I don't think that the ATAs is the problem. > > > > > > > > Everything goes wrong when the gateway is tested > > > > as a "dock-n-talk" (dialing through it connected to > > > > one of the RJ11 with an ordinary phone set). First there > > > > is no DTMF recognition whatsoever, and second tha gateway > > > > does not sense the hangup and start making the "noises". > > > > > > > > Hope Sam could solve the problem with the factory or > > > > exchange the goods with working ones. > > > > Benchev > > > > > > > > > Outch... Four of them and not working... That hurts. > > > > > How do you connect them to * ? As I'm using only one > > > > > for me an X100P-FXO is sufficiant and seems to work as > > > > > good as attaching a real anlog phone. > > > > > > > > > > Btw. I saw that www.voipsolutions.be is selling them > > > > > also, but for 165.- euro > > > > > > > > > > On Sun, Mar 26, 2006 at 03:28:07PM +0200, Benchev wrote: > > > > > > Hi Adibar, > > > > > > Thanks very much for the answer. > > > > > > We are also struggling (with 4 of them :( ) > > > > > > and will let you know how the things develop, too, > > > > > > in case of success. > > > > > > > > > > > > Have a nice Sunday, > > > > > > Benchev > > > > > > > > > > > > > Hi Benchev > > > > > > > > > > > > > > I'm still in contact with Sam, but currently no changes. > > > > > > > The device is still in an unusable state for me, as it > >
Re: [Asterisk-Users] plainvoip - IAX2 call rejected
> Is anybody using "plainvoip" provider with IAX2? They seem to support
> IAX but it rejects my calls.
>-- Executing Dial("SIP/11-cb98", "IAX2/[EMAIL PROTECTED]") in new
> stack
> -- Called [EMAIL PROTECTED]
> May 14 00:33:32 WARNING[26580]: chan_iax2.c:5553 socket_read: Call
> rejected by 66.199.240.2: No authority found
>
> My registration goes through OK.
> My dial plan:
>
> exten => _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]
Try
exten => _1NXXNXX,2,Dial,IAX2/plainvoip/${EXTEN}
or
exten => _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
Benchev
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Re: [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?
Hi Adibar, Thank you. I have also received some "recipes" from Sam. I have sent the codes to the client and waiting for him to confirm the good news. It is amazing sometimes how slow some things happen. I'll let you know. Regards, Benchev > Hi Benchev > > Mine is working now. It was set to a courious "billing mode". > After inserting the following command via an analog phone: > > *00#000# hangup > *00#100#0# hangup > > I could finaly use it. I guess you can try it too. Keep > me informed of your success, I wonder if these settings > do help you too. > > Greets > Adibar > > On Wed, Apr 26, 2006 at 10:17:51PM +0300, Benchev wrote: > > Thanks Adibar, (sorry List:-) > > > > You have at least an offer. The only thing I've got so far > > was a promiss to have a decision "tomorrow" .For a month > > now, "tomorrow" is every day after the day before "tomorrow". > > > > However, I hope, that in order to keep it's good reputation CyberTelecom > > will consider FREE OF CHARGE EXCHANGE of the useless GSM boxes > > or do immediate refund or something. > > > > Let's see... > > Cheers, > > Benchev > > > > On Wednesday 26 April 2006 01:17, adibar wrote: > > > Hi Benchev > > > > > > News from the front. Sam is kinda offering me an exchange > > > of my box. But I should return it to him at my cost ;-) > > > > > > Last word is not spoken yet on that, cause I'm really > > > not amused on this ;-) > > > > > > Keeping you in loop. > > > > > > Greets > > > Adibar > > > > > > On Mon, Mar 27, 2006 at 10:05:01AM +0300, Benchev wrote: > > > > Actually I've got five, but the first one I have received > > > > around Xmas and I don't have these problems with it. > > > > I use spa3000 as FXO and the gsm gateway works > > > > seamlessly inbound, outbound, DISA, no annoying sounds, > > > > no DTMF problems. There is one problem however, the gateway does > > > > not transfer correctly the CID to the FXO(at least in my case) > > > > but this could be a sipura problem as well. > > > > > > > > Now, the other 4 seam to be a different model or something > > > > and one should be very careful ordering that thing since you never > > > > know which model you are going to receive. > > > > They are used with "no-brand-name" FXO/FXS ATAs > > > > but I don't think that the ATAs is the problem. > > > > > > > > Everything goes wrong when the gateway is tested > > > > as a "dock-n-talk" (dialing through it connected to > > > > one of the RJ11 with an ordinary phone set). First there > > > > is no DTMF recognition whatsoever, and second tha gateway > > > > does not sense the hangup and start making the "noises". > > > > > > > > Hope Sam could solve the problem with the factory or > > > > exchange the goods with working ones. > > > > Benchev > > > > > > > > > Outch... Four of them and not working... That hurts. > > > > > How do you connect them to * ? As I'm using only one > > > > > for me an X100P-FXO is sufficiant and seems to work as > > > > > good as attaching a real anlog phone. > > > > > > > > > > Btw. I saw that www.voipsolutions.be is selling them > > > > > also, but for 165.- euro > > > > > > > > > > On Sun, Mar 26, 2006 at 03:28:07PM +0200, Benchev wrote: > > > > > > Hi Adibar, > > > > > > Thanks very much for the answer. > > > > > > We are also struggling (with 4 of them :( ) > > > > > > and will let you know how the things develop, too, > > > > > > in case of success. > > > > > > > > > > > > Have a nice Sunday, > > > > > > Benchev > > > > > > > > > > > > > Hi Benchev > > > > > > > > > > > > > > I'm still in contact with Sam, but currently no changes. > > > > > > > The device is still in an unusable state for me, as it > > > > > > > only allows one call, which results in wild-beeping on > > > > > > > terminating the call. > > > > > > > But I still hope, that Sam finds anywhere a tech-person > > > > > > > who
Re: [Asterisk-Users] astcc: need partial pin code
> > [paygo-forward]
> > exten => _1NXXNXX,1,Set(CALLERID(all)=${CALLING})
> > exten => _1NXXNXX,2,Playback(pls-hold-while-try)
> > exten => _1NXXNXX,3,DeadAGI(astcc.agi,${CARDNO},${TEMP},4)
> > exten => _1NXXNXX,4,Hangup
> > ${CALLING},${CARDNO},${TEMP} are coming from an inbound context
>
> In my case they don't always. If it is an call from my system, local or
> remote phone, I have the caller-id, but if the user call into the system
> via PSTN, than I do not get the caller-id!
You can control then from the dialplan:
[from-pstn] ; full interrogation
exten => 1231231234,1,DeadAGI(astcc-disa.agi)
exten => 1231231234,2,Hangup
[from-internal-outbound]
exten => _1NXXNXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4); or 5
exten => _1NXXNXX,2,Hangup
> My idea was not to check if PIN=YES, but the name of the agi.
> if the astcc.agi is used, than treat it as PIN=NO
> if the astcc-disa is used, than treat it as PIN=YES
See above
> I have added in my astcc.agi different rates according to the card number.
> I would need to do this twice! My solution does not need me to make it
> twice.
In the astcc.agi I would use
my $sth = $dbh->prepare("SELECT * FROM routes WHERE " .
$dbh->quote($number) . " RLIKE pattern ORDER BY LENGTH(pattern) DESC");
I'd create a table routes_special with the same scheme as routes
and put my special prices there.
Then in astcc-disa.agi
my $sth = $dbh->prepare("SELECT * FROM routes_special WHERE " .
$dbh->quote($number) . " RLIKE pattern ORDER BY LENGTH(pattern) DESC");
Cheers,
Benchev
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Re: [Asterisk-Users] astcc: need partial pin code
> >> Just to give you an idea
> >> I would suggest you to make two .agi files:
> >> astcc.agi and astcc-disa.agi
> >> In astcc.agi you'd leave everithing as it is, and enable
> >> PIN =YES through the astcc-admin.cgi.
> >> Thus you could dial without interogation:
> >> exten => _1NXXNXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
> >>
> >> astcc-disa.agi is a copy of astcc.agi so
> >> # cp astcc.agi astcc-disa.agi.
> >> # pico astcc-disa.agi.
> >
> > pico astcc.agi.
> >
> >> Find the line:
> >> # At this point we have a valid card number.
> >> and coment out everything until:
> >> # At this point we have a valid card and pin number.
> >> You can dial from outside:
> >> exten => 1234567894,1,DeadAGI(astcc-disa.agi)
> >> and will de asked for cardnumber and pin.
> >>
> >> Some mobile phones support "w" inside of a dialstring i.e.
> >> 1234567894w123456789012#w159753# .Fist part is the
> >> DID you dial to enter * .
> >> * asks for a cardnumber and the mobile waits for you on "w"
> >> to push"Enter",
> >> * asks for a pin and phone waits for you on "w" to push "Enter"
> >> for the last string.
> >> After all that you would here:Please enter the number you wish to
> >> dial...
>
> How about this solution:
>
> 1. we make a sym link instead of a copy. That avoids later update
> problems!!!
> ln -s astcc.agi astcc-disa.agi
>
> 2. instead of
> if ($config{'pinstatus'} eq "YES") {
>
> (please help me to write it, but what I want is:)
> if ("my called name is 'astcc-disa' ") {
Hi,
astcc is such a neat and stable piece that I would hardly dare to
to mess with it.
My idea was 1) you need a PIN=YES because otherwise pins are
not generated; 2)you need a PIN because most of the guys
use CALLERIDNUM as cardnumber, and that's open to see. I think it was JP Carballo suggested earlier to authenticate against another variable i.e.
as I do in one system of ours:
[paygo-forward]
exten => _1NXXNXX,1,Set(CALLERID(all)=${CALLING})
exten => _1NXXNXX,2,Playback(pls-hold-while-try)
exten => _1NXXNXX,3,DeadAGI(astcc.agi,${CARDNO},${TEMP},4)
exten => _1NXXNXX,4,Hangup
${CALLING},${CARDNO},${TEMP} are coming from an inbound context
and are set to show the person the call is forwarded to , who actually
is calling and NOT his own cardnumber, as well as to charge the appropriate CARDNO etc. But, briefly, that way you could skip usage of pins.
Although, PINS are good thing.
Excuse me, I am not playing clever, I just don't quite understand what you
wanted to do, so I am giving max information I can(maybe redundant)
hoping for an accidental hit.
3)I think even if you use quiet=5 i.e.(astcc.agi,${CARDNO},${TEMP},5)
a PIN will be expected(if PIN=YES). That's way a soft link won't work
because you need to comment out the pin validation snippet in your
inner .agi and leave it for you "disa" .agi
Out of curiosity to see what DBI is doing behind the scene:
I started: asterisk -gc
Found sub connect_db() in astcc.agi and change it to:
sub connect_db() {
my $dsn = "DBI:mysql:database=$config{'dbname'};host=$config{'dbhost'}";
$dbh->disconnect if $dbh;
$dbh = DBI->connect($dsn, $config{'dbuser'}, $config{'dbpass'});
$dbh -> trace(3, 'dbitrace.log' ); # this is the only change
}
Then I did cat dbitrace.log (the same directory I started
asterisk) and saw very interesting things.
Hope I'm not waisting your time with already know things, but I love to
discuss about asterisk :-)
Benchev
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Re: [Asterisk-Users] astcc: need partial pin code
Hi Ronald,
Small mistake, see bellow:
Benchev
> Just to give you an idea
> I would suggest you to make two .agi files:
> astcc.agi and astcc-disa.agi
> In astcc.agi you'd leave everithing as it is, and enable
> PIN =YES through the astcc-admin.cgi.
> Thus you could dial without interogation:
> exten => _1NXXNXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
>
> astcc-disa.agi is a copy of astcc.agi so
> # cp astcc.agi astcc-disa.agi.
> # pico astcc-disa.agi.
pico astcc.agi.
> Find the line:
> # At this point we have a valid card number.
> and coment out everything until:
> # At this point we have a valid card and pin number.
> You can dial from outside:
> exten => 1234567894,1,DeadAGI(astcc-disa.agi)
> and will de asked for cardnumber and pin.
>
> Some mobile phones support "w" inside of a dialstring i.e.
> 1234567894w123456789012#w159753# .Fist part is the
> DID you dial to enter * .
> * asks for a cardnumber and the mobile waits for you on "w"
> to push"Enter",
> * asks for a pin and phone waits for you on "w" to push "Enter"
> for the last string.
> After all that you would here:Please enter the number you wish to dial...
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Re: [Asterisk-Users] astcc: need partial pin code
On Thursday 27 April 2006 11:08, Ronald Wiplinger wrote:
> Ronald Wiplinger wrote:
> > I have not used astcc with pin codes so far, since I set-up the phone
> > number as card number.
> >
> > Some of my users want now to dial in to the system and than use their
> > card, which is their phone number.
> > For that I would need a way of authentication, like a pin.
> >
> > I want to use something like:
> > What is your card number:
> > Enter your pin:
> > Enter your destination phone number: > phone number>
> >
> > Is there a code snip available for that?
> >
> > Keyin needs always more time, we need to allow longer spaces between
> > the digits, therefore we need to allow the # to finish the dialstring
> > faster. I wonder if we can use one dialstring for all:
> > cardnumber*pin*destination-number
> >
> > How can a user end the call and dial a new number, without hanging up?
> >
> > The user has usually a desk phone (=card number), and this dialin
> > should work parallel, but of course it assumes still that only one
> > card is in use.
> >
> >
> > bye
> >
> > Ronald Wiplinger
>
> I tried now the examples in the wiki, but they do not fit!!!
> If I use in configure Require Pins Yes then everyone needs a pin code!
> If I use in configure Require Pins NO then calling in people will just
> need to know a valid card number!!!
>
> How can I overcome this?
>
> How can I re-write:
> exten => _77.,1,Answer
> exten => _77.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:2},3)
> exten => _77.,3,Hangup
>
> sothat the dialstring:
> 77*123456789012*159753*011886939775516 would be splitted into:
> ${CARDNUM}=123456789012
> ${PIN}=159753
> ${DESTINATION}=0118869397755516
>
> with a mysql lookup of the cardnum in astcc get the pin and compare to
> the given pin. If all is ok, than use the dial command
>
Hi Ronald,
Just to give you an idea
I would suggest you to make two .agi files:
astcc.agi and astcc-disa.agi
In astcc.agi you'd leave everithing as it is, and enable
PIN =YES through the astcc-admin.cgi.
Thus you could dial without interogation:
exten => _1NXXNXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
astcc-disa.agi is a copy of astcc.agi so
# cp astcc.agi astcc-disa.agi.
# pico astcc-disa.agi.
Find the line:
# At this point we have a valid card number.
and coment out everything until:
# At this point we have a valid card and pin number.
You can dial from outside:
exten => 1234567894,1,DeadAGI(astcc-disa.agi)
and will de asked for cardnumber and pin.
Some mobile phones support "w" inside of a dialstring i.e.
1234567894w123456789012#w159753# .Fist part is the
DID you dial to enter * .
* asks for a cardnumber and the mobile waits for you on "w"
to push"Enter",
* asks for a pin and phone waits for you on "w" to push "Enter"
for the last string.
After all that you would here:Please enter the number you wish to dial...
Hope, this helps.
Benchev
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Re: [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?
Thanks Adibar, (sorry List:-) You have at least an offer. The only thing I've got so far was a promiss to have a decision "tomorrow" .For a month now, "tomorrow" is every day after the day before "tomorrow". However, I hope, that in order to keep it's good reputation CyberTelecom will consider FREE OF CHARGE EXCHANGE of the useless GSM boxes or do immediate refund or something. Let's see... Cheers, Benchev On Wednesday 26 April 2006 01:17, adibar wrote: > Hi Benchev > > News from the front. Sam is kinda offering me an exchange > of my box. But I should return it to him at my cost ;-) > > Last word is not spoken yet on that, cause I'm really > not amused on this ;-) > > Keeping you in loop. > > Greets > Adibar > > On Mon, Mar 27, 2006 at 10:05:01AM +0300, Benchev wrote: > > Actually I've got five, but the first one I have received > > around Xmas and I don't have these problems with it. > > I use spa3000 as FXO and the gsm gateway works > > seamlessly inbound, outbound, DISA, no annoying sounds, > > no DTMF problems. There is one problem however, the gateway does > > not transfer correctly the CID to the FXO(at least in my case) > > but this could be a sipura problem as well. > > > > Now, the other 4 seam to be a different model or something > > and one should be very careful ordering that thing since you never > > know which model you are going to receive. > > They are used with "no-brand-name" FXO/FXS ATAs > > but I don't think that the ATAs is the problem. > > > > Everything goes wrong when the gateway is tested > > as a "dock-n-talk" (dialing through it connected to > > one of the RJ11 with an ordinary phone set). First there > > is no DTMF recognition whatsoever, and second tha gateway > > does not sense the hangup and start making the "noises". > > > > Hope Sam could solve the problem with the factory or > > exchange the goods with working ones. > > Benchev > > > > > Outch... Four of them and not working... That hurts. > > > How do you connect them to * ? As I'm using only one > > > for me an X100P-FXO is sufficiant and seems to work as > > > good as attaching a real anlog phone. > > > > > > Btw. I saw that www.voipsolutions.be is selling them > > > also, but for 165.- euro > > > > > > On Sun, Mar 26, 2006 at 03:28:07PM +0200, Benchev wrote: > > > > Hi Adibar, > > > > Thanks very much for the answer. > > > > We are also struggling (with 4 of them :( ) > > > > and will let you know how the things develop, too, > > > > in case of success. > > > > > > > > Have a nice Sunday, > > > > Benchev > > > > > > > > > Hi Benchev > > > > > > > > > > I'm still in contact with Sam, but currently no changes. > > > > > The device is still in an unusable state for me, as it > > > > > only allows one call, which results in wild-beeping on > > > > > terminating the call. > > > > > But I still hope, that Sam finds anywhere a tech-person > > > > > who can hand me out the correct setup-information. > > > > > > > > > > As soon as I get it in a working state, I will let you > > > > > know it ;-) > > > > > > > > > > Adibar > > > > > > > > > > On Sat, Mar 25, 2006 at 09:55:56PM +0200, Benchev wrote: > > > > > > Hi Adibar, > > > > > > Any success with the gsm gateway? > > > > > > I have exactly the same problem with units received this month. > > > > > > The codes given by Sam are not working... > > > > > > Please, let me know if you have discovered something. > > > > > > Thanks in advance, > > > > > > Benchev > > > > > > > > > > > > > But these are the wrong instructions again. Same as those > > > > > > > ones you sent me allready. I've got the small box for £60 > > > > > > > The only reaction I get is if I press just "*". Then the > > > > > > > display changes to "SET___". After that there is silence > > > > > > > for about 15 seconds. Pressing any keys is only allowd up > > > > > > > to four digits. So also the given password is to long for > > > > > > > entering. After the 15 seconds or the four digits I get a
Re: [Asterisk-Users] iax2 show netstats
> i've been using iax2 show netstats and i wonder if someone could explain > what all these means, just in case i have them wrong. Because i am looking > for something that tells me that there is delay , and/or packet loss. > > LOCAL - > REMOTE Channel RTT Jit Del Lost % > Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts IAX2/iaxBBG-16384 > 1000 -10-1 -1 0 -1 000 0 0 00 >0 IAX2/iaxBBG-16386 16 -10-1 -1 0 -1 10 > 40 0 0 00 0 The new Jitterbuffer in Asterisk Steve Kann .. 5) Testing and monitoring: -- You can test the effectiveness of PLC and the new jitterbuffer's detection of loss by using the new CLI command "iax2 test losspct ". This will simulate n percent packet loss coming _in_ to chan_iax2. You should find that with PLC and the new JB, 10 percent packet loss should lead to just a tiny amount of distortion, while without PLC, it would lead to silent gaps in your audio. "iax2 show netstats" shows you statistics for each iax2 call you have up. The columns are "RTT" which is the round-trip time for the last PING, and then a bunch of s tats for both the local side (what you're receiving), and the remote side (what the other end is telling us they are seeing). The remote stats may not be complete if the remote end isn't using the new jitterbuffer. The stats shown are: * Jit: The jitter we have measured (milliseconds) * Del: The maximum delay imposed by the jitterbuffer (milliseconds) * Lost: The number of packets we've detected as lost. * %: The percentage of packets we've detected as lost recently. * Drop: The number of packets we've purposely dropped (to lower latency). * OOO: The number of packets we've received out-of-order * Kpkts: The number of packets we've received / 1000. ... 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] iax limit question
> I found a solution... I just has to enter an Answer
> line and now it behaves as I wanted. Here is the
> working code:
>
> [inbound]
> exten => 1234567,1,Set(GROUP()=limit)
> exten => 1234567,2,GotoIf($[${GROUP_COUNT()}>2]?103)
> exten => 1234567,3,Dial(Zap/5&Zap/6,25,tT)
> exten => 1234567,4,Voicemail,u110
> exten => 1234567,5,hangup
> exten => 1234567,103,Answer
> exten => 1234567,104,Playtones(busy)
> exten => 1234567,105,Wait(5)
> exten => 1234567,106,Hangup
Check for OUTBOUND_GROUP variable in
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup
It provides interesting capability "to set the amount of calls on the called
channel but also on the calling channel".
In this case you should not need Answer.
Benchev
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Re: [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?
Actually I've got five, but the first one I have received around Xmas and I don't have these problems with it. I use spa3000 as FXO and the gsm gateway works seamlessly inbound, outbound, DISA, no annoying sounds, no DTMF problems. There is one problem however, the gateway does not transfer correctly the CID to the FXO(at least in my case) but this could be a sipura problem as well. Now, the other 4 seam to be a different model or something and one should be very careful ordering that thing since you never know which model you are going to receive. They are used with "no-brand-name" FXO/FXS ATAs but I don't think that the ATAs is the problem. Everything goes wrong when the gateway is tested as a "dock-n-talk" (dialing through it connected to one of the RJ11 with an ordinary phone set). First there is no DTMF recognition whatsoever, and second tha gateway does not sense the hangup and start making the "noises". Hope Sam could solve the problem with the factory or exchange the goods with working ones. Benchev > Outch... Four of them and not working... That hurts. > How do you connect them to * ? As I'm using only one > for me an X100P-FXO is sufficiant and seems to work as > good as attaching a real anlog phone. > > Btw. I saw that www.voipsolutions.be is selling them > also, but for 165.- euro > > On Sun, Mar 26, 2006 at 03:28:07PM +0200, Benchev wrote: > > Hi Adibar, > > Thanks very much for the answer. > > We are also struggling (with 4 of them :( ) > > and will let you know how the things develop, too, > > in case of success. > > > > Have a nice Sunday, > > Benchev > > > > > Hi Benchev > > > > > > I'm still in contact with Sam, but currently no changes. > > > The device is still in an unusable state for me, as it > > > only allows one call, which results in wild-beeping on > > > terminating the call. > > > But I still hope, that Sam finds anywhere a tech-person > > > who can hand me out the correct setup-information. > > > > > > As soon as I get it in a working state, I will let you > > > know it ;-) > > > > > > Adibar > > > > > > On Sat, Mar 25, 2006 at 09:55:56PM +0200, Benchev wrote: > > > > Hi Adibar, > > > > Any success with the gsm gateway? > > > > I have exactly the same problem with units received this month. > > > > The codes given by Sam are not working... > > > > Please, let me know if you have discovered something. > > > > Thanks in advance, > > > > Benchev > > > > > > > > > But these are the wrong instructions again. Same as those > > > > > ones you sent me allready. I've got the small box for £60 > > > > > The only reaction I get is if I press just "*". Then the > > > > > display changes to "SET___". After that there is silence > > > > > for about 15 seconds. Pressing any keys is only allowd up > > > > > to four digits. So also the given password is to long for > > > > > entering. After the 15 seconds or the four digits I get a > > > > > busy-signal. No password-prompt, no "LINE CON". Nada... > > > > > > > > > > Adibar > > > > > > > > > > On Sat, Mar 11, 2006 at 05:12:40AM +0800, Sam Tam wrote: > > > > > > Hello > > > > > > > > > > > > > > > > > > To solved the beeping problem you need to first enter the > > > > > > configuration mode. > > > > > > > > > > > > > > > > > > > > > > > > I .Entry into SETTING STATUS > > > > > > 1) Pick up the phone, > > > > > > press the button of "0 ** #"; > > > > > > > > > > > > 2) Screen display: "?SETUP?"; > > > > > > Input pass: input pass word: "332808" > > > > > > Then will display "IMPUT CON" > > > > > > > > > > > > > > > > > > you can change the box working mode . > > > > > > > > > > > > use the command > > > > > > > > > > > > *000100#0#for set defaut ,billing mode. > > > > > > *000100#1#for one long tone mode > > > > > > *000100#2#for long tone mode > > > > > > > > > > > > > > > > > > Sam > > > > > > --
Re: [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?
Hi Adibar, Any success with the gsm gateway? I have exactly the same problem with units received this month. The codes given by Sam are not working... Please, let me know if you have discovered something. Thanks in advance, Benchev > But these are the wrong instructions again. Same as those > ones you sent me allready. I've got the small box for £60 > The only reaction I get is if I press just "*". Then the > display changes to "SET___". After that there is silence > for about 15 seconds. Pressing any keys is only allowd up > to four digits. So also the given password is to long for > entering. After the 15 seconds or the four digits I get a > busy-signal. No password-prompt, no "LINE CON". Nada... > > Adibar > > On Sat, Mar 11, 2006 at 05:12:40AM +0800, Sam Tam wrote: > > Hello > > > > > > To solved the beeping problem you need to first enter the configuration > > mode. > > > > > > > > I .Entry into SETTING STATUS > > 1) Pick up the phone, > > press the button of "0 ** #"; > > > > 2) Screen display: "?SETUP?"; > > Input pass: input pass word: "332808" > > Then will display "IMPUT CON" > > > > > > you can change the box working mode . > > > > use the command > > > > *000100#0#for set defaut ,billing mode. > > *000100#1#for one long tone mode > > *000100#2#for long tone mode > > > > > > Sam > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of adibar > > Sent: Saturday, March 11, 2006 4:55 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Anyone using the GSMgateway from > > CyberTelecom ? > > > > Hi Dan > > > > To connect the gsm-gateway I'm using a X100P which is sufficiant > > and which works. But I'm exeriencing problems with the device > > itself. The gateway acts like a POTS for me, but according a > > sales-representive of CT they do use it like a phone ?!? > > Dialing in works fine (beside the currently missing CID). Calls > > are taken from ZAP and are routed internaly. I can pickup, talk > > and... hangup. But after hanging up the gateway starts beeping > > like hell and requires a reset. Outgoing calls are even worse. > > I can dial, opposite takes the call, gateway hangs up and beeps > > again. Therefore... reset again. > > To be shure that it's neither * nor the X100P I tried the same > > with an old analog phone. Same result. > > So, you see. as I'm currently not ready to use it, I even could > > not think about tweaking, fine-tuning and codec-testing ;-) > > As soon as the tech-departement from CT comes (hopefully) > > back to me with a solution I'm the going to treat that box ;-) > > > > adibar > > > > On Fri, Mar 10, 2006 at 10:15:44PM +0300, [EMAIL PROTECTED] wrote: > > > Hi adiar, > > > > > > Just have a few questions. Hope u can give me the answers. > > > > > > Well im currently having analog lines to my asterisk which is used to > > > bridge calls from PSTN to Voip (DISA). Im facing issues with hang up > > > supervision etc. Im thinking of bridging calls using the GSM gateway > > > in future. Just want to know will it work well with codecs and > > > disconnect supervision. I need g729 to work well i have low bandwdth > > > too. Im facing issues with TDM11B + analog lines. So will the GSM > > > gateway + TDM11B a better choice? > > > > > > Please some one advice.. > > > > > > Dan > > > > > > On 10/03/06, adibar <[EMAIL PROTECTED]> wrote: > > > > Hi > > > > > > > > I would like to say that I'm a happy user of it. But > > > > currently I'm still waiting for support from CT. > > > > My gateway is here, but it's doing nasty things, > > > > so currently I'm not a good source for information. > > > > > > > > Sorry > > > > Adibar > > > > > > > > On Fri, Mar 10, 2006 at 07:03:25PM +0300, [EMAIL PROTECTED] wrote: > > > > > Hi, > > > > > > > > > > I wrote this thread to find someone who uses this. Please write if > > > > > u r using this as i would like to clarify regarding disconnect > > > > > supervision. > > > > > > > > > > Thanks > > > > > > > >
Re: [Asterisk-Users] Asterisk RealTime Question, Please help
> >> I need many contexts because I have around 1000 DID's each with 5-10
> >> Extensions.
> >> These DID numbers are changed or added very frequently and whenever
> >> there is a change I have to change Extensions.conf manually. So please
> >> tell me how can I do this dynamically without changing Extensions.conf
> >> and help me configure Asterisk.
> >
> > I presume you have about 1000 DID numbers and each of this numbers
> > may ring to
> > 5-10 users of yours, right?
> >
> > If so, make a context in you extensions.conf and include in it a switch
> > like that:
> > [ever_changing_dids]
> > switch => Realtime/[EMAIL PROTECTED]
> >
> > Now you can insert in your extensions_table imaginary DID 9876543210:
> >
> > INSERT INTO `extensions_table` VALUES ('', 'ever_changing_dids',
> > '9876543210',
> > 1, 'Dial', 'SIP/user1:SIP/user2:SIP/user3:SIP/user4:SIP/user8:SIP/user12|
> > 20');
> > You can do that for many thousands of DIDs without changing
> > extensions.conf.
> >
> My current setup is exactly similar to which you have suggested. My DID
> numbers
> are added or changed very frequently and all the time I have to change some
> config file manually and should reload Asterisk or atleast call Extensions
> reload. I do want these things to be manual, Can't I have the Asterisk to
> directly get the contexts from Mysql DB without giving them in config
> files? If
> this is possible then we can have a realtime dynamic Asterisk.
>
> The other approcah can be to match the context itself with some regular
> expression. But I do not know how do this or whether this is possible? I
> will have a context something like this
>
> [XX]
> switch => Realtime/@extensions
>
> So all contexts will be directed to Mysql DB matching regex but
> [XX] is
> not acting as regex as expected it just matches context XX.
I am afraid I'm loosing you.
However, try to change in extconfig.conf
extentions => mysql,astcc,extensions_table
to:
exten_sions => mysql,astcc,extensions_table
and then switch => Realtime/@exten_sions
Then in extensions.conf:
[inbound]
switch => Realtime/@exten_sions
In mysql do:
INSERT INTO `extensions_table` VALUES ('', 'inbound',
'9876543210', 1, 'Dial', 'SIP/user1|20');
*CLI> show dialplan inbound
should show: Alt. Switch => 'Realtime/@exten_sions'
not what you have used to see with a static extensions.conf but you can do:
server*CLI> realtime load exten_sions context inbound
Column Name Column Value
id 1
context inbound
exten 9876543210
priority 1
app Dial
appdata SIP/user1|20
and see how realtime took that.
On the other hand, no you can not create contexts on the fly and
out of nothing. Actually the context scheme is the backbone of your system
and should be thought over and set up beforehand.
The connection between particular context and realtime is the switch.
When you insert into the extensions_table a set, which context corresponds
to where the switchis, and this is read in realtime without the need of
reloading.
Pretty much that's all I can do to help. Sorry.
Benchev
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Re: [Asterisk-Users] Asterisk RealTime Question, Please help
> I need many contexts because I have around 1000 DID's each with 5-10
> Extensions.
> These DID numbers are changed or added very frequently and whenever there
> is a change I have to change Extensions.conf manually. So please tell me
> how can I do this dynamically without changing Extensions.conf and help me
> configure Asterisk.
I presume you have about 1000 DID numbers and each of this numbers may ring to
5-10 users of yours, right?
If so, make a context in you extensions.conf and include in it a switch
like that:
[ever_changing_dids]
switch => Realtime/[EMAIL PROTECTED]
Now you can insert in your extensions_table imaginary DID 9876543210:
INSERT INTO `extensions_table` VALUES ('', 'ever_changing_dids', '9876543210',
1, 'Dial', 'SIP/user1:SIP/user2:SIP/user3:SIP/user4:SIP/user8:SIP/user12|
20');
You can do that for many thousands of DIDs without changing extensions.conf.
Another approach, also no changing the extension.conf:
[ever_changing_dids]
#include includes/ever_changing_dids.conf
ever_changing_dids.conf
exten =>
9876543210,1,Dial(SIP/user1&SIP/user2&SIP/user3&SIP/user4&SIP/user8&SIP/user12|
20)
etc...
However this requires *CLI> reload
Hope I've guessed right.
Benchev
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Re: [Asterisk-Users] Asterisk RealTime Question, Please help
> > I was able to install Asterisk and Asterisk-addons and use them > > successfully. But I have a problem now, I have many contexts and it looks > > like Asterisk is unable to find the context given directly in Mysql DB > > unless I specify it in Extensions.conf to switch it to RealTime. If I add > > a new context in Mysql then I have to add it in Extensions.conf and > > reload extensions whenever I need a new context. Please tell me if there > > is a way to avoid all this and make Asterisk take contexts directly from > > Mysql without mentioning that context in Extensions.conf. If this is > > possible then I can make my Asterisk RealTime actually and modify > > contexts directly in Mysql. Idea from the wiki: ; If regcontext is specified, Asterisk will dynamically create and destroy a ; NoOp priority 1 extension for a given peer who registers or unregisters with ; us. The actual extension is the 'regexten' parameter of the registering ; peer or its name if 'regexten' is not provided. More than one regexten may ; be supplied if they are separated by '&'. Patterns may be used in regexten. ; ;regcontext=sipregistrations That means that you should creat a "mother" context in extensions.conf: [sipregistrations] But first I would try to add a field "regcontext" along with "regexten"(which already there) in sip_users table since for the trick to work ... read http://www.voip-info.org/wiki-Asterisk+sip+regcontext Hope this will 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] CDR question
> > i'm trying without success to change the dst (destination) entry of the
> > cdr. I'm using the following:
> > exten=>_2006234500254.,2,SetVar(CDR(dst)=${EXTEN:10})
> > exten=>_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)
> > I want to record into the cdr only the called number, but in the cdr
> > appears the prefix 2006234500254.
But this should do it:
exten=>_2006234500254.,2,Set(CDR(dst) = ${EXTEN:10})
exten=>_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)
Benchev
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Re: [Asterisk-Users] CDR question
On Tuesday 14 March 2006 17:15, Benchev wrote:
> > i'm trying without success to change the dst (destination) entry of the
> > cdr. I'm using the following:
> > exten=>_2006234500254.,2,SetVar(CDR(dst)=${EXTEN:10})
> > exten=>_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)
> > I want to record into the cdr only the called number, but in the cdr
> > appears the prefix 2006234500254.
>
> Would you try:
> exten=>_2006234500254.,2,Set(destination = ${EXTEN:10})
> exten=>_2006234500254.,3,Dial(OH323/[EMAIL PROTECTED],60,tr)
Yep fast end stupid. Sorry, it wont work that way.
Benchev
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Re: [Asterisk-Users] CDR question
> i'm trying without success to change the dst (destination) entry of the
> cdr. I'm using the following:
> exten=>_2006234500254.,2,SetVar(CDR(dst)=${EXTEN:10})
> exten=>_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)
> I want to record into the cdr only the called number, but in the cdr
> appears the prefix 2006234500254.
Would you try:
exten=>_2006234500254.,2,Set(destination = ${EXTEN:10})
exten=>_2006234500254.,3,Dial(OH323/[EMAIL PROTECTED],60,tr)
Benchev
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Re: [Asterisk-Users] Asterisk RealTime Question, Please help
> > > > I was able to install Asterisk and Asterisk-addons and use them > > > > successfully. But I have a problem now, I have many contexts and it > > > > looks like Asterisk is unable to find the context given directly in > > > > Mysql DB unless I specify it in Extensions.conf to switch it to > > > > RealTime. If I add a new context in Mysql then I have to add it in > > > > Extensions.conf and reload extensions whenever I need a new context. > > > > Please tell me if there is a way to avoid all this and make Asterisk > > > > take contexts directly from Mysql without mentioning that context in > > > > Extensions.conf. If this is possible then I can make my Asterisk > > > > RealTime actually and modify contexts directly in Mysql. > > > > Idea from the wiki: > > ; If regcontext is specified, Asterisk will dynamically create and > > destroy a ; NoOp priority 1 extension for a given peer who registers or > > unregisters with ; us. The actual extension is the 'regexten' parameter > > of the registering ; peer or its name if 'regexten' is not provided. > > More than one regexten may ; be supplied if they are separated by '&'. > > Patterns may be used in regexten. ; > > ;regcontext=sipregistrations > > That means that you should creat a "mother" context in extensions.conf: > > [sipregistrations] > > > > But first I would try to add a field "regcontext" along with > > "regexten"(which already there) in sip_users table since for the trick to > > work ... read http://www.voip-info.org/wiki-Asterisk+sip+regcontext > > > OK, that will enable the auto generation of a context but as the new > context won't have a switch statement it doesn't help with this > problem... I may try writing a "default switch if no matching context > found" type patch. Well, it wont generate a context, it would rather "register" the extension of the new user under [sipregistrations] And, maybe now is the time to warn that regexten was created to facilitate a sip-user extensions' "propagation" within an * network; there is a discussion "Clustering" going on the list, see for details. As for the switch, since "context is optional: (switch => Realtime/@realtime_ext) and if left off, RealTime will use the current context, in this case "sipregistrations"." Means: [sipregistrations] switch => Realtime/@realtime_ext ;realtime_ext or whatever the table name is Ok i'am guessing "sans voir" here since I don't understand why so many contexts are needed? Hope it 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] Action after _caller_ has hungup(cmd Dial 'g'-option)
> Hmm, both of you recommend a solution with the dial cmd in an > agi-script, i would prefer a direct solution but i guess there is none. There is - "H" - Allow the calling party to hang up by hitting the '*' DTMF digit. I though that your main concern was how to cachup the hangup and deal with the result of a call(see my previous email ), which is bigger pain than H. Sorry misunderstanding you. 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] Action after _caller_ has hungup(cmd Dial 'g'-option)
> There's the "g"-option for the Dial-cmd that allows to execute the next
> extensions in the current context when the callee hangs up.
>
> I would need the same for a call where the caller hangs up, concretely
> i have to inform a agi-application of the end of a call. Does someone
> know a way to do this from the dialplan?
You could add after your agi-application a hangup
section starting with "h":
exten => _X.,n,AGI,agi-application.agi
exten => h,1,Set(result=${DIALSTATUS})
exten => h,2,GotoIf($["${RESULT}" = "CANCEL" | "${RESULT}" = "NOANSWER" |
"${RESULT}" = "BUSY" | "${RESULT}" = "CHANUNAVAIL" | "${RESULT}" =
"CONGESTION" | "${RESULT}" = ""]?500:3)
exten => h,3,Do_things
exten => h,500,Congestion
benchev
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Re: [Asterisk-Users] Limiting Sip Calls ?
> Is there any way not using group count, to limit calls received by every > endpoint SIP?.. > > Outgointlimit and Incominglimit seems to be deprecated on 1.2.x branch. > > Is there another command to do that? > http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup ___ --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] Re: Anyone using the GSMgateway from CyberTelecom ?
> >Do you have any success receiving the caller id with your TDM400 FXO? > >I have the same problem when I connect the GSM gateway to a SPA3000 FXO > > line and thought this a Sipura's problem. On a phone connected to the GSM > > gateway I can see the callerid, but not on the Sipura's PSTN line ... > this is no more and no less the same problem as I do have. > > It appears it's then not really the TDM400 FXO module. I have another > option to test: I do have a similar ATA like the Sipura, but made by > Grandstream. > > It's here at home; I will take it to the office tomorrow and see if it > can read the caller id from the GSM gateway. > > Even my gsm unit does indeed pass the callerID when I connect it to a > cheap, dead simple analog phone! > > BTW: Do you have a manual for the gateway? Thanks Aldo, No I do not have a manual and I don't believe such a thing exist. Actually, that GSM gateway is a "Dock-N-Talk" kind of thing with the exception that the handset is "imbedded", so pretty much no need of a manual. Is your Grandstream a HT-488? If so you might be able to simulate the spa3000 case. Please, let me know what happened. Best regards, 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] need help
> need help, iam installing areskiCC and have a problem > after that create extension for calling card and after dial > > exten => 17000,3,DeadAgi,a2billing.php > > i see messages : a2billing.php no such file in directory, i tired copy > that file that file aready copy in agi-bin. Try chmod 755 /var/lib/asterisk/agi-bin/a2billing.php 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] Re: Anyone using the GSMgateway from CyberTelecom ?
> Basically you just plug it into an analog interface after installing the > GSM chip. > > The voice quality is good even in my office; a sort of radio waves-black > hole. Normally most cellphones just disappear when they are there.. > > The only problem I have so far is that the TDM400 FXO module does not > seem to read the caller id. > > A regular phone shows it, if I switch connections. > > It might be a problem of configuration of the TDM card; I have looked in > the wiki and googled around, but I do not know how I can change the way > a zaptel card reads the callerid. > > I will try to upgrade to 1.2.x asap to see if this helps. Hi, Do you have any success receiving the caller id with your TDM400 FXO? I have the same problem when I connect the GSM gateway to a SPA3000 FXO line and thought this a Sipura's problem. On a phone connected to the GSM gateway I can see the callerid, but not on the Sipura's PSTN line ... Thanks, 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
