Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is 1234567@domain.net. I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------------------------------------------------ A mind is like a parachute... it only works when it's open.
Use SetCallerID(1234567). Tan telappliant.com ----- Original Message ----- From: "Marian Danisek" <majo@sunteq.sk> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 09, 2003 3:23 PM Subject: [Asterisk-Users] caller id Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is 1234567@domain.net. I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------------------------------------------------ A mind is like a parachute... it only works when it's open. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, How do I optionally hide the caller id on outgoing calls on chan_zap? Ie. calling h323 -> asterisk -> chan_zap -> isdn provider. Using setcallerid() to clear the callerid won't work since my provider requires a callerid. AFAIK one has to send something along with the callerid (set a flag in the call setup or similar) to indicate that the callerid is private. Any pointers? - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/Xfec2TEAILET3McRAoI6AJ472yR6aRUjC25WhZdToaXj4NrY6gCgkt0H S7G2GHgp8hCYDiYoEScRrQY=XCMO -----END PGP SIGNATURE-----
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Tuesday 09 September 2003 17:54, Tais M. Hansen wrote:> How do I optionally hide the caller id on outgoing calls on chan_zap? Ie. > calling h323 -> asterisk -> chan_zap -> isdn provider.Problem solved. I made app_dial.c take an option to change hidecallerid/restrictcid flag. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/XkvX2TEAILET3McRAmIQAJ9Lk6A5r6CPBDlIYgcdb+XmI1UHXQCfdANP WlQBNDX6qIIhzu14U3eAUy8=kmtq -----END PGP SIGNATURE-----
> At 23:53 9-9-2003 +0200, you wrote: > >On Tuesday 09 September 2003 17:54, Tais M. Hansen wrote: > > > How do I optionally hide the caller id on outgoing calls on chan_zap? Ie. > > > calling h323 -> asterisk -> chan_zap -> isdn provider. > > > >Problem solved. I made app_dial.c take an option to change > >hidecallerid/restrictcid flag. > > That would be something I'd care to see in my own systems too. Is the patch > public ? > > > Best regards, > Florian Overkamp >Most phone providers have a number that you can use to dissable the sending of the caller ID.. For BT in the UK its 141 so for example if you are dialing 5551234 it will send callerID so to stop it you dial 1415551234 and it blocks the callerID.. Can be easily controlled in the dial plan.. Later.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze
I've come late into this thread, so I risk saying things that you all will just shake your heads at and say, "Duh!" Historically, though, from WAY back in the days of electromechanical switches, reverse battery mainly provided answer supervision. Its usefulness pretty much went away with the advent of SS7, except for those cases where end users resold their POTS services (such as hotels and motels, which usually paid extra for the service). The battery would then reverse BACK to normal again after the call was terminated. During this reversal (obviously), the voltage would transition past zero, and it would also suffice for disconnect supervision. Aside from the hotel/motel scenario, telcos have recently been providing disconnect supervision solely by means of removal of battery from the circuit. This feature continues to be of value in situations where analog CPE would continue to keep the line seized were it not for the removal of the battery -- key systems having lines on hold, answering machines, etc. Personally, I would very much like to see the reverse battery feature built in to the FXS cards that work on asterisk. I say this because I am starting to go back to my roots in the industry by looking for old step-by-step line finders, selectors, and connectors. Answer supervision via some electromechanical means would be preferable than trying to cobble an ISDN D channel over to that old stuff. Just my 2 cents.> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Kevin Walsh > Sent: Sunday, May 02, 2004 1:05 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] RE: Caller ID > > Steve Underwood [steveu@coppice.org] wrote: > > Wake up. > > > Sorry, I must have drifted off for a while. Thanks for the > alarm call. > > > > > The reversal detection is a complete waste of time. Totally > unnecessary. > > Pointless. A line break detector would have much more use, > as it would > > give a reliable disconnect detection on many lines. (Actually, > > reversal detection would have years ago, but its not much > use any more). > > > Perhaps both would be good then: A polarity reversal > detector for determining the start of a Caller*ID sequence > and a line break detector for, err, detecting line breaks. > Actually, my X101P seems to detect hangups just fine, so I've > not had cause to check whether the detection is done in the > hardware or in the driver. If you say that it's not done in > hardware then I'll take your word for it for the moment. > > > > > All you need for these CLI requirements is to monitor for > some energy > > on the line. Since these FXOs are not being used in banks > of hundreds, > > you will never notice this MIPs this uses. > > > I'd still prefer to see this done in hardware, rather than in > some sort of idle loop in the driver or the application. > Call me old fashioned, but I prefer it when unnecessary > overheads are not measured in MIPs. :-) > > Perhaps the new FXO module for the TDMxxB has, or will have, > hardware support for the above. If it has, and I'm sure I > heard somewhere that it does, then that's great. An X102P, > with similar support, would no doubt be welcome too. > > -- > _/ _/ _/_/_/_/ _/ _/ _/_/_/ _/ _/ > _/_/_/ _/_/ _/ _/ _/ _/_/ _/ K e v i n > W a l s h > _/ _/ _/ _/ _/ _/ _/ _/_/ kevin@cursor.biz > _/ _/ _/_/_/_/ _/ _/_/_/ _/ _/ > > _______________________________________________ > 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 > > >
Hello ! I'm really new to asterisk, in fact I have been only using it as a simple PBX, now a customer mine needs to show a different "call back number" or a different Caller ID, because they have the customers service in another country ....and they use my asterisk server for outgoing calls only through an IAX provider ... There is a way to do that ? Any lights ?, any links ? ... can somebody point me to the right direction ? Thanks in advance ! Pepe
Good day all MY caller Id does not.I have asked for caller Id on the line and in my vpb.conf I have callerid=on This is what I have in extensions.conf exten => s,1,Wait,2 ; Wait a second, just for fun exten => s,2,Answer ; Answer the line exten => s,3,DigitTimeout,3 ; Set Digit Timeout to 5 seconds exten => s,4,ResponseTimeout,4 ; Set Response Timeout to 10 seconds exten => s,5,BackGround(bi) Am I missing something,this is a voicetronix openline4 card Please Help
How can I change that when there's no Caller ID instead of displaying asterisk it display something like Unknown. Because everyone is confuse when they see a call coming from asterisk. Thanks Martin
Stefan Gofferje wrote:> ...what probably would be a good idea, because a call from "asterisk" > really looks strange... I have been searching for the position in > source but haven't found it yet. Didn't spend too much effort anyway... > But if one of the maintainers would do that, it would be nice...I assuming this is when using SIP. I was annoyed by this and make an adjustment which works nicely. channels/chan_sip.c around line 132 look for #define DEFAULT_CALLERID "asterisk" swap that to Unknown and you're in good shape.
I have a question: Why is't possible to see Caller ID on the analog phones? If I'm wrong pls tell me how to do to see Caller ID on analog phones. Thank you. mihaid __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Hi, I'm using TDM04B and Asterisk 1.0.5. How can I setup the Asterisk so that I get caller ID? I do not get caller ID currently. Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050614/56722d2f/attachment.htm
Hello,
I am implementing asterisk in my office in India. I am
facing a problem of caller ID. Any kind of help is appreciated.
As a call comes to asterisk console it shows
ERROR[27863]: callerid.c:260 callerid_feed: fsk_serie made mylen < 0 (-7)
WARNING[27863]: chan_zap.c:5434 ss_thread: CallerID feed failed: Success
Aug 18 11:33:45 WARNING[27863]: chan_zap.c:5476 ss_thread: CallerID
returned with error on channel 'Zap/4-1'
Thankyou
Gurminder
If this question has an obvious answer forgive me, I'm a noob. I'm planning to make a system configured as below: POTS <---> FXO (400P) <--> Asterisk <---> FXS (400P) <----> Analog phone The question I have is, if an incoming call from the POTS line has caller ID information, does/is/can that information be passed onto the analog phone so it's caller id display will show the info? If so, is there anything I need to do to make this happen or does it *just work*? Thanks. -- Michael J. Lynch What if the hokey pokey IS what it's all about -- author unknown
Hi to all,
I am having problems with the caller id using IAX. The
caller id feature does not function for an incoming
IAX2 call when the incoming caller hides the caller
id.
The caller id is presented as blank on my phone
instead of the number i set it to be. It works fine
otherwise and also works fine when using a sip trunk
(even for anonymous calls).
Extensions.conf entry:
exten => xxxxxxxxxxx,1,SetCallerID(xxxxx)
exten => xxxxxxxxxxx,1,NoOp(${CALLERID(number)})
exten => xxxxxxxxxxx,2,Dial(SIP/xxxx)
On the CLI:
Executing SetCallerID("IAX2/xxx.xx.xx.xxx:4569-2",
"xxxxx") in new stack
Executing Dial("IAX2/xxx.xx.xx.xxx:4569-2",
"SIP/xxxxx") in new stack
-- Called xxxxx
The SIP TRACE:
SIP read from 192.168.254.52:50254:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.254.78:5060;branch=z9hG4bK113ae6bd;rport
From: "Unknown"
<sip:Unknown@192.168.254.78>;tag=as29a6b6c1
To:
<sip:xxxx@192.168.254.52:50254;user=phone>;tag=f147d1518b13372d
Call-ID:
087abe3f7f737c8c316525c71f25cbe4@192.168.254.78
CSeq: 102 INVITE
User-Agent: Grandstream BT110 1.0.7.11
Contact: <sip:xxxx@192.168.254.52:50254;user=phone>
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Supported: replaces
Content-Length: 216
SIP read from 192.168.254.52:50254:
BYE sip:Unknown@192.168.254.78 SIP/2.0
Via: SIP/2.0/UDP
192.168.254.52:50254;branch=z9hG4bK0bb47f6f5a9cd5a0
From:
<sip:xxxx@192.168.254.52:50254;user=phone>;tag=f147d1518b13372d
To: "Unknown"
<sip:Unknown@192.168.254.78>;tag=as29a6b6c1
Supported: replaces
Call-ID:
087abe3f7f737c8c316525c71f25cbe4@192.168.254.78
CSeq: 22862 BYE
User-Agent: Grandstream BT110 1.0.7.11
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
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Hello all,
All my sip users are identified by their name.lastname (mine would be
dov.bigio).
But I have to associate them to extension numbers too, so I did the following on
my extensions.conf.
The problem is that when a call is logged on CDR and also the caller ids that
appear for end users is without the "." (dot).
So if I call someone, this person would see a call coming from
"dovbigio". And he won't be able to call back to me, since
"dovbigio" is not a valid user.
Is this some kind of but, or I am doing something wrong here?
Thank you
Dov
--
[default]
exten => 435,1,Goto(01.ramais_nomes,dov.bigio,1)
[ramais_nomes]
exten => dov.bigio,1,Macro(ramais,dov.bigio,435)
[macro-ramais]
exten => s,1,SetCallerID(${CALLERID}|a)
exten => s,2,SetCDRUserField(INTERNA)
exten => s,3,Dial(SIP/${ARG1},15,r)
exten => s,4,VoiceMail(u${ARG2})
exten => s,5,Hangup
exten => s,104,VoiceMail(b${ARG2})
Exten => s,105,Hangup
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Hello
I am using X100P card and set up asterisk box in india. can any one tell
me the procedure for caller ID detection.
here is zapata.conf file
usecallerid=yes
cidsignalling=dtmf
cidstart = polarity
callerid=asreceived
immediate=yes
The CLI output I get is
Starting simple switch on 'Zap/1-1'
-- Executing Answer("Zap/1-1", "") in new stack
-- Executing NoOp("Zap/1-1", "") in new stack
-- Executing Playback("Zap/1-1", "auth-thankyou") in new
stack
-- Playing 'auth-thankyou' (language 'en')
-- Executing Festival("Zap/1-1", "Press 1 for sheeju")
in new stack
== Parsing '/etc/asterisk/festival.conf': Found
== CDR updated on Zap/1-1
-- Executing Dial("Zap/1-1", "Zap/4") in new stack
-- Called 4
-- Zap/4-1 is ringing
-- Zap/4-1 is ringing
Jan 14 11:05:28 WARNING[1641]: chan_zap.c:3928 zt_handle_event: Didn't
finish Caller-ID spill. Cancelling.
-- Zap/4-1 is ringing
-- Zap/4-1 is ringing
-- Zap/4-1 is ringing
-- Zap/4-1 is ringing
-- Zap/4-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/4-1
== Spawn extension (incoming, 1, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1
Thanks in advance
sheeju
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I have a quick Caller*ID question.
I have an inbound call to my PBX which I am attempting to bridge with
a PSTN number (specifically my cell phone, so when someone dials my
extension the cell phone rings).
In my extentions.conf I have:
; Daniel -- 1102
exten => 1102,1,Answer()
exten => 1102,2,Set(DIALEDNUM=1102)
exten => 1102,3,Wait(2)
exten => 1102,4,Playback(pls-wait-connect-call)
exten => 1102,5,Wait(1)
exten =>
1102,6,Dial(SIP/2102&SIP/3102&SIP/4102&SIP/1987654321@porta,
33,mj)
exten => 1102,7,Voicemail(su{$EXTEN})
exten => 1102,8,Hangup()
exten => 1102,106,Voicemail(sb{$EXTEN})
exten => 1102,107,Hangup()
where "porta" is my SIP account with the company that provides my
PSTN connection. I know for a fact that I can set any caller ID I
want (because I've done it with ATAs) and my carrier will pass it;
however, my question is, how do I get my asterisk box to pass the
original Call*ID instead of the number assigned to me by my provider?
this is the entry in sip.conf
[porta]
type=peer
secret=corbe9845
username=portasip
host=68.145.125.95
;fromuser=17862065496
fromdomain=66.165.175.35
insecure=very
;nat=yes
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