system toolbox » voip

koje dodatne telefone nabaviti ?

(16:06:20) hernad: ma nabaviti 4 x bazna, 4 x slušalica 
(16:06:31) hernad: to se mora riješiti
(16:06:44) vsasa: pa treba jer se ovo ne moze pustiti u rad bez tih telefona
(16:06:46) vsasa: fali
(16:06:48) hernad: koliko sam ja shvatio to se na asterisku već sada može konfigurisati
(16:07:28) hernad: a koji je firmware na rmlh telefonima ?
(16:07:34) vsasa: ne znam
(16:07:41) hernad: da li se može manuelno firmware osvježiti
(16:07:52) hernad: ne sa njihovih servera nego da ti daš fajl 
(16:08:07) vsasa: treba pogledati, pretpostavljam da moze
(16:08:33) hernad: pretpostavljam da može da se radi backup firmware-a i restore, što bi, ako već sve funkcioniše tamo moralo biti takođe workaround za taj problem
(16:08:44) hernad: pretpostavljam da je je hardware identičan
(16:09:42) vsasa: da ali rmlh ni ne koristi to grupno, pa tako da pojma nemam da li se i kod njih to desava
(16:09:52) vsasa: ja sam testirao prilikom postavljanja kada je radilo
(16:10:00) vsasa: e sada.... poslije, ne znam
(16:10:10) hernad: aha
(16:10:16) vsasa: ha ja
(16:10:19) hernad: njima telefnoni zvone samo na sekretaricu
(16:10:23) vsasa: ma da
(16:10:30) vsasa: i kod direktora na drugi broj
(16:10:51) hernad: i sada trenutno
(16:11:03) hernad: taj bug možeš reproducirati u fbze ?
(16:11:14) vsasa: mogu
(16:12:21) vsasa: evo sad mi zvoni samo 11
(16:12:39) vsasa: i sada kada upalim ugasim
(16:12:43) vsasa: sad cu vidjeti
(16:12:56) vsasa: a to se sve vidi iz astriska fino
(16:13:06) vsasa: m1*CLI> 
    -- Executing [10@fbze:1] Dial("SIP/32-ac054770", "SIP/11&SIP/12&SIP/13&SIP/14|400|tT") in new stack
    -- Called 11
    -- Called 12
    -- Called 13
[Jul 30 16:08:55] WARNING[617]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
    -- SIP/11-00779860 is ringing

(16:13:17) vsasa: vidi se da 11-ka samo zvoni ali poziva sve
(16:13:23) vsasa: e sad da vidim nakon restarta
(16:13:51) hernad: ovo što si ti našao odnosi se na 450IP
(16:13:56) vsasa: evo sada zvone 11-ka i 12-ka
(16:13:59) hernad: da li je isti firmware 470 ip
(16:13:59) vsasa:     -- Called 11
    -- Called 12
    -- Called 13
[Jul 30 16:10:39] WARNING[16794]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
    -- SIP/12-00761bd0 is ringing
    -- SIP/11-00779860 is ringing

(16:14:14) vsasa: e odosmo ovi zatvaraju...

(16:14:17) hernad: a koji je firmware

(16:15:00) hernad: ok
(16:15:55) vsasa: Firmware version:     021230000000 / 043.00    EEPROM version:               121
(16:16:08) vsasa nije više u sobi.

_ALERT_INFO replacement in 1.4?

Does SIPAddHeader(Alert-Info:) not do it?

On 3/12/07, Nikhil Jogia <nikhil at nikhiljogia.com> wrote:
> Hi All
>
> I have just upgraded from Asterisk 1.2 to 1.4 and am having trouble with
> with one of my ATAs not ringing.   <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
>
> Basically, when I execute the Dial command, an error occurs: "Got SIP
> response 400 "In alert-info header: Empty value expected" 
>
> Now in 1.2, I just issued the following command to overcome this
> problem: Set(_ALERT_INFO=).
>
> Now in 1.4, _ALERT_INFO is deprecated, so I have to use SIPAddHeader,
> but I don't know how, or if there is a way to remove the alert-info header.
>
> Here is my dialplan snippet:
>
> exten => s,9,Playback(my-greeting)
> exten => s,10,Wait(1)
> exten => s,11,SIPAddHeader(Alert-Info: info=bellcore-r4)
> exten => s,12,Dial(SIP/600&SIP/602&SIP/603,60,tm)
> exten => s,13,Set(_ALERT_INFO=)
> exten => s,14,Dial(SIP/604,60,tm)
> exten => s,15,Voicemail(su600)
> exten => s,16,Hangup
> exten => s,115,Voicemail(sb600)
> exten => s,116,Hangup
>
> As you can see, #13 is deprecated, so extension 604 does not ring.
> Extension 600, 602 and 603 are all hooked up to Sipura ATAs and need the
> bellcore-r4 ringtone to differentiate from other incoming lines.
>
> Any ideas?

znači

na početak bi trebalo staviti:

SIPAddHeader(Alert-Info: info=bellcore-r4)

na kraj

SIPAddHeader(Alert-Info:)

siemens alert info problem

Problem with Alert_Info messages crashing the phone (the phone does not ring when it receive an Alert Info Header)

http://en.wikipedia.org/wiki/Distinctive_ring

https://lists.cs.columbia.edu/pipermail/sip-implementors/2006-June/013357.html

[Sip-implementors] Alert-INFO - no ringing at destination
Paul Kyzivat pkyzivat at cisco.com
Wed Jun 21 19:17:23 EDT 2006

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Stumer, Peggy (Com US) wrote:

Is there any way to communicate this in SIP in a standardized way? If
not, should there be, or, crap shoot?

Since Alert-Info is usually carried in the first INVITE message and
expected to be acted on immediately by the recipient, there is no
obvious and simple way to negotiate this.

A straightforward way to handle would be to define a new option tag and
include it in a Require header. That would make the call fail if it
isn't supported at the other end. But that isn't always a good option.

Or you can use if the semantics you intend are in fact optional, so that
reasonable things happen if the callee ignores this.

And of course you can use this in controlled environments where the one
inserting the header knows the recipient will understand it. For
instance this makes sense when a device connects to the network through
an edge proxy that supports and protects it. That edge proxy might
insert an Alert-Info with various values depending on who is calling.

http://www.faqs.org/rfcs/rfc3261.html

20.4 Alert-Info

   When present in an INVITE request, the Alert-Info header field
   specifies an alternative ring tone to the UAS.  When present in a 180
   (Ringing) response, the Alert-Info header field specifies an
   alternative ringback tone to the UAC.  A typical usage is for a proxy
   to insert this header field to provide a distinctive ring feature.

   The Alert-Info header field can introduce security risks.  These
   risks and the ways to handle them are discussed in Section 20.9,
   which discusses the Call-Info header field since the risks are
   identical.

   In addition, a user SHOULD be able to disable this feature
   selectively.

      This helps prevent disruptions that could result from the use of
      this header field by untrusted elements.

   Example:

      Alert-Info: <http://www.example.com/sounds/moo.wav>

http://www.voip-info.org/wiki/view/Gigaset+DECT+with+activation+of+Direct+Dial+In

http://www.voip-info.org/wiki/view/Siemens+Phones

http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels

ovo mi je bila inspiracija:

http://www.voip-info.org/wiki-Asterisk+cmd+Dial#Example3Dialmultiplechannelspartiallydel

extensions.ael

GR_TEHSLUZBA="LOCAL/891011@fbze&LOCAL/891012@fbze&LOCAL/891013@fbze";

...
         _9009XXXX => {
             Wait(${EXTEN:4:2});
             Dial(SIP/${EXTEN:6});
         };

         // header iscisti
          _8910XX => {
             SIPAddHeader(Alert-Info:);
             Dial(SIP/${EXTEN:4});
         };

*CLI>    
    -- Accepting voice call from '033269291' to '200270' on channel 0/1, span 1
    -- Executing [200270@fbze:1] Goto("Zap/1-1", "fbze|10|1") in new stack
    -- Goto (fbze,10,1)
    -- Executing [10@fbze:1] Dial("Zap/1-1", "LOCAL/891011@fbze&LOCAL/891012@fbze&LOCAL/891013@fbze|400|tT") in new stack  <<<<<<<<<<<<<<<<<
    -- Called 891011@fbze
    -- Called 891012@fbze
    -- Called 891013@fbze

    -- Executing [891011@fbze:1] SIPAddHeader("Local/891011@fbze-a383,2", "Alert-Info:") in new stack
    -- Executing [891011@fbze:2] Dial("Local/891011@fbze-a383,2", "SIP/11") in new stack
    -- Called 11

    -- Executing [891012@fbze:1] SIPAddHeader("Local/891012@fbze-d3f1,2", "Alert-Info:") in new stack
    -- Executing [891012@fbze:2] Dial("Local/891012@fbze-d3f1,2", "SIP/12") in new stack
    -- Called 12

    -- Executing [891013@fbze:1] SIPAddHeader("Local/891013@fbze-8e4e,2", "Alert-Info:") in new stack
    -- Executing [891013@fbze:2] Dial("Local/891013@fbze-8e4e,2", "SIP/13") in new stack
    -- Called 13

[Jul 30 20:54:28] WARNING[29671]: chan_sip.c:12462 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '6af22651013d747f0bd440c3385d7258@192.168.66.1'. Giving up.
    -- SIP/13-007c33d0 is circuit-busy     <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< ??? 

  == Everyone is busy/congested at this time (1:0/1/0)
                                                              <<<<<<<<<<<<<<<<<<<<< kad je utvrdio da je SIP/13 busy onda je prešao na ZAP/g1/891013
    -- Executing [891013@fbze:3] Dial("Local/891013@fbze-8e4e,2", "ZAP/g1/891013") in new stack            <<<<<<<<<<<<<<<<<<<<<<<<<<<<<
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called g1/891013
[Jul 30 20:54:28] WARNING[29671]: chan_sip.c:12997 handle_response: Remote host can't match request BYE to call '6af22651013d747f0bd440c3385d7258@192.168.66.1'. Giving up.

    -- Zap/2-1 is proceeding passing it to Local/891013@fbze-8e4e,2
    -- Local/891013@fbze-8e4e,1 is proceeding passing it to Zap/1-1
    -- Zap/2-1 is ringing                                                 ???????????????? šta sad ovo, zap je na kraju nazvao 891013 broj - a to je neki vanjski broj ?!!?

    -- Local/891013@fbze-8e4e,1 is ringing
    -- SIP/12-007bebe0 is ringing                               <<<<<<<<<<<<<<< SIP/12
    -- Local/891012@fbze-d3f1,1 is ringing
    -- SIP/11-007b92d0 is ringing                               <<<<<<<<<<<<<<< SIP/11
    -- Local/891011@fbze-a383,1 is ringing
    -- Zap/2-1 answered Local/891013@fbze-8e4e,2
    -- Local/891013@fbze-8e4e,1 stopped sounds
    -- Local/891013@fbze-8e4e,1 answered Zap/1-1                <<<<<<<<<<<<<<< ženski glas mi se javio :) i prekinuo ... heh fakat je fbze zaptel nazvao 891013

izgleda da gigaset bazna stanica uopšte i ne može da handlira više od dva poziva simultano

kasnije pričao sa sašom i pitao ga, ok misli takođe da mogu samo dva poziva simultano da se obave

e sad ostaje da se ova gore varijanta podesi da u slučaju kad je neka ekstenzija stvarno zauzeta da se ne desi da se zove preko zaptela neki 89xxxx broj kao što se gore desilo

(09:51:06) hernad: vidi sa fbde da li je kod njih zvonjava nakon reseta ok
(09:51:55) hernad: trebali bi uloviti grešku kada prestanu obje slušalice zvoniti
(09:52:05) vsasa: pa jeste, rekao sam ti jucer, kada je faruk resetovao bilo je ok
(09:52:11) hernad: pa bez reseta promjeniti grupni poziv kako sam ja uradio
(09:52:19) hernad: znam da kada je resetovao da je bilo ok
(09:52:26) hernad: pitanje je da li je OSTALO ok
(09:52:32) vsasa: a ha
(09:52:35) hernad: ako je bug ponovo će prestati
(09:52:37) vsasa: dobro, provjerit cu
(09:52:39) hernad: a navodno je bug
(09:52:46) hernad: i kad on prestane zvoniti
(09:54:03) hernad: u ovoj shemi koju sam ja stavio
(09:54:35) hernad: vidi da li umjesto 8910 radi A910
(09:55:03) hernad: tako da se ne bi desilo da ako je zauzeta ekstenzija da se poziva neki broj u žepču :)
(09:55:40) hernad: znači vidi da li asterisk u dial planu dozvoljava alfanumerike
(09:56:39) hernad: ja sam juče testirajući uočio da kada stavim bilo šta u 
                    SIPAddHeader(Alert-Info: bilo_sta) da tada nijedna ekstenzija ne zvoni

pored ovoga treba prebaciti u fbze odmah slušalice tako da na svakoj baznoj stanici imamo max dvije slušalice

postavio u fbde šemu kao i u fbze što se tiče ovog clearing-a

i evo sada:

m1*CLI> 
  == ISDN1#02: Incoming call '01725288203' -> '4569801'
    -- Executing [4569801@demo:1] Dial("CAPI/ISDN1#02/4569801-33", "LOCAL/A91051@demo&LOCAL/A91052@demo&SIP/53
    DEFTIMEOUT=60|400|tT") in new stack
    -- Called A91051@demo
    -- Executing [A91051@demo:1] SIPAddHeader("Local/A91051@demo-f0f4,2", "Alert-Info:") in new stack
    -- Executing [A91051@demo:2] Dial("Local/A91051@demo-f0f4,2", "SIP/51") in new stack
    -- Called 51
    -- Called A91052@demo
    -- Executing [A91052@demo:1] SIPAddHeader("Local/A91052@demo-7eff,2", "Alert-Info:") in new stack
    -- Executing [A91052@demo:2] Dial("Local/A91052@demo-7eff,2", "SIP/52") in new stack
    -- Called 52
[Jul 31 09:18:54] WARNING[13022]: chan_sip.c:2841 create_addr: No such host: 53
    DEFTIMEOUT=60
[Jul 31 09:18:54] WARNING[13022]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
    -- SIP/51-00832b40 is ringing
    -- Local/A91051@demo-f0f4,1 is ringing
    -- SIP/52-00841d30 is ringing
    -- Local/A91052@demo-7eff,1 is ringing
       > ISDN1#02: CAPI INFO 0x3490: Normal call clearing

ali ne zvoni 53 koja je na drugom gigaset-u ?!????

postavio ovako

GR_OFFICE="LOCAL/A91051@demo&LOCAL/A91052@demo&SIP/53" 

dakle, pošto je sip 53 na drugom gigaset-u praktično bi trebalo da ovako radi

i kaže

[Jul 31 09:18:54] WARNING[13022]: chan_sip.c:2841 create_addr: No such host: 53

isto je kao i kada stavim

LOCAL/A91053@demo

postavio ovako

GR_OFFICE="LOCAL/A91051@demo&LOCAL/A91052@demo" 

a u liniju dodao ovako

        4569801 => {
                Dial(${GR_OFFICE}&SIP/53,400,tT);
        };

sada zvone 51 i 53 ?????

  == ISDN1#02: Incoming call '004530758695' -> '4569801'
    -- Executing [4569801@demo:1] Dial("CAPI/ISDN1#02/4569801-36", "LOCAL/A91051@demo&LOCAL/A91052@demo
    DEFTIMEOUT=60&SIP/53|400|tT") in new stack
    -- Called A91051@demo
[Jul 31 09:25:49] NOTICE[16694]: chan_local.c:566 local_alloc: No such extension/context A91052@demo
    DEFTIMEOUT=60 creating local channel
[Jul 31 09:25:49] WARNING[16694]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'LOCAL' (cause 0 - Unknown)
    -- Executing [A91051@demo:1] SIPAddHeader("Local/A91051@demo-037e,2", "Alert-Info:") in new stack
    -- Executing [A91051@demo:2] Dial("Local/A91051@demo-037e,2", "SIP/51") in new stack
    -- Called 51
    -- Called 53
    -- SIP/53-007c7aa0 is ringing
    -- SIP/53-007c7aa0 answered CAPI/ISDN1#02/4569801-36

vratio opet na staru postavku, sada zvoni samo 51, a svi su dostupni

  == ISDN1#02: Incoming call '01725288203' -> '4569801'
    -- Executing [4569801@demo:1] Dial("CAPI/ISDN1#02/4569801-3a", "LOCAL/A91051@demo&LOCAL/A91052@demo&LOCAL/A91053@demo
    DEFTIMEOUT=60|400|tT") in new stack
    -- Called A91051@demo
    -- Executing [A91051@demo:1] SIPAddHeader("Local/A91051@demo-ad47,2", "Alert-Info:") in new stack
    -- Executing [A91051@demo:2] Dial("Local/A91051@demo-ad47,2", "SIP/51") in new stack
    -- Called 51
    -- Called A91052@demo
[Jul 31 09:33:14] NOTICE[20739]: chan_local.c:566 local_alloc: No such extension/context A91053@demo
    DEFTIMEOUT=60 creating local channel
[Jul 31 09:33:14] WARNING[20739]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'LOCAL' (cause 0 - Unknown)
    -- Executing [A91052@demo:1] SIPAddHeader("Local/A91052@demo-88bc,2", "Alert-Info:") in new stack
    -- Executing [A91052@demo:2] Dial("Local/A91052@demo-88bc,2", "SIP/52") in new stack
    -- Called 52
    -- SIP/51-008365a0 is ringing
    -- Local/A91051@demo-ad47,1 is ringing
       > ISDN1#02: CAPI INFO 0x3490: Normal call clearing

(10:41:11) hernad: čekaj malo saša
(10:41:26) hernad: ovo nisi podesio izgleda kako treba
(10:41:30) hernad: kaže ovo:
(10:41:45) hernad: No such extension/context A91052@demo 
(10:41:51) hernad: ni ne može zvonit
(10:42:03) hernad: gdje je onaj dio
(10:42:23) hernad: ono definisanje _A910XX ...
(10:42:27) hernad: stavi sve što si mjenjao
(10:44:09) hernad: postoji li 52 u extension.ael ?
(10:44:14) hernad: kao da ne postoji
(10:44:25) hernad: garant ne postoji

u dijelu globals

 GR_OFFICE="LOCAL/A91051@demo&LOCAL/A91052@demo&LOCAL/A91053@demo" 

u dijelu demo dolazni poziv koristi

        4569801 => {
                Dial(${GR_OFFICE},400,tT);
        };

zatim u dijelu demo

        // harun
        50 => {
                Dial(SIP/50,${DEFTIMEOUT},tT);
                Voicemail(50@default,u);
        };

        // faruk
        51 => {
                SIPAddHeader(Alert-Info:);
                Dial(SIP/51,${DEFTIMEOUT},tT);
                Voicemail(51@default,u);
        };

        // amir
        52 => {
                SIPAddHeader(Alert-Info:);
                Dial(SIP/52,${DEFTIMEOUT},tT);
                Voicemail(52@default,u);
        };

        // harun
        53 => {
                SIPAddHeader(Alert-Info:);
                Dial(SIP/53,${DEFTIMEOUT},tT);
                Voicemail(53@default,u);
        };

        .....

        // problem grupnih adresa
        _A910XX => {
                SipAddHeader(Alert-Info:);
                Dial(SIP/${EXTEN:4});
        };

ispravio ovo

A910XX => {
                SipAddHeader(Alert-Info:);
                Dial(SIP/${EXTEN:4});

=>

A910XX => {
                SIPAddHeader(Alert-Info:);
                Dial(SIP/${EXTEN:4});

mada opet isto...

evo nakon reseta oba gigaset-a

  == ISDN1#02: Incoming call '01725288203' -> '4569801'
    -- Executing [4569801@demo:1] Dial("CAPI/ISDN1#02/4569801-3d", "LOCAL/A91051@demo&LOCAL/A91052@demo&LOCAL/A91053@demo
    DEFTIMEOUT=60|400|tT") in new stack
    -- Called A91051@demo
    -- Executing [A91051@demo:1] SIPAddHeader("Local/A91051@demo-0e9e,2", "Alert-Info:") in new stack
    -- Executing [A91051@demo:2] Dial("Local/A91051@demo-0e9e,2", "SIP/51") in new stack
    -- Called 51
    -- Called A91052@demo
[Jul 31 09:58:05] NOTICE[560]: chan_local.c:566 local_alloc: No such extension/context A91053@demo
    DEFTIMEOUT=60 creating local channel
[Jul 31 09:58:05] WARNING[560]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'LOCAL' (cause 0 - Unknown)
    -- Executing [A91052@demo:1] SIPAddHeader("Local/A91052@demo-6b7d,2", "Alert-Info:") in new stack
    -- Executing [A91052@demo:2] Dial("Local/A91052@demo-6b7d,2", "SIP/52") in new stack
    -- Called 52
    -- SIP/52-007c7aa0 is ringing
    -- Local/A91052@demo-6b7d,1 is ringing
    -- SIP/51-00832b40 is ringing
    -- Local/A91051@demo-0e9e,1 is ringing
       > ISDN1#02: CAPI INFO 0x3490: Normal call clearing

zvone 51 i 52, ali ne i 53

i ovdje se vidi da poziva

• 51
• pa 53
• pa 52

zašto ????

postavio sam 2 gigaset-a sa po 2 telefona

• Gigaset C470IP
• Gigaset C455IP

i naštimao da se u dialplanu zovu extenzije 11, 12, 13 i 14 ako se pozove 10.

m1*CLI> 
    -- Executing [10@fbze:1] Dial("SIP/32-007ca360", "LOCAL/A91011@fbze&LOCAL/A91012@fbze&LOCAL/A91013@fbze&LOCAL/A91014@fbze|400|tT") in new stack
    -- Called A91011@fbze
    -- Executing [A91011@fbze:1] SIPAddHeader("Local/A91011@fbze-fe94,2", "Alert-Info:") in new stack
    -- Executing [A91012@fbze:1] SIPAddHeader("Local/A91012@fbze-3019,2", "Alert-Info:") in new stack
    -- Executing [A91012@fbze:2] Dial("Local/A91012@fbze-3019,2", "SIP/12") in new stack
    -- Executing [A91011@fbze:2] Dial("Local/A91011@fbze-fe94,2", "SIP/11") in new stack
    -- Called 12
    -- Called 11
    -- Called A91012@fbze
    -- Called A91013@fbze
    -- Called A91014@fbze
    -- Executing [A91014@fbze:1] SIPAddHeader("Local/A91014@fbze-de56,2", "Alert-Info:") in new stack
    -- Executing [A91014@fbze:2] Dial("Local/A91014@fbze-de56,2", "SIP/14") in new stack
    -- Executing [A91013@fbze:1] SIPAddHeader("Local/A91013@fbze-51fa,2", "Alert-Info:") in new stack
    -- Executing [A91013@fbze:2] Dial("Local/A91013@fbze-51fa,2", "SIP/13") in new stack
    -- Called 14
    -- Called 13
    -- SIP/14-007feee0 is ringing
    -- Local/A91014@fbze-de56,1 is ringing
    -- SIP/12-007fab80 is ringing
    -- Local/A91012@fbze-3019,1 is ringing
    -- SIP/13-00803240 is ringing
    -- Local/A91013@fbze-51fa,1 is ringing
    -- SIP/11-b005e130 is ringing
    -- Local/A91011@fbze-fe94,1 is ringing
  == Spawn extension (fbze, 10, 1) exited non-zero on 'SIP/32-007ca360'
  == Spawn extension (fbze, A91013, 2) exited non-zero on 'Local/A91013@fbze-51fa,2'
  == Spawn extension (fbze, A91011, 2) exited non-zero on 'Local/A91011@fbze-fe94,2'
  == Spawn extension (fbze, A91014, 2) exited non-zero on 'Local/A91014@fbze-de56,2'
  == Spawn extension (fbze, A91012, 2) exited non-zero on 'Local/A91012@fbze-3019,2'

i evo sada, zvone sve na testu.

evo ga sada kada zauzmem jedan sip

    -- Executing [10@fbze:1] Dial("SIP/32-008054e0", "LOCAL/A91011@fbze&LOCAL/A91012@fbze&LOCAL/A91013@fbze&LOCAL/A91014@fbze|400|tT") in new stack
    -- Called A91011@fbze
    -- Called A91012@fbze
    -- Called A91013@fbze
    -- Called A91014@fbze
    -- Executing [A91011@fbze:1] SIPAddHeader("Local/A91011@fbze-92e5,2", "Alert-Info:") in new stack
    -- Executing [A91011@fbze:2] Dial("Local/A91011@fbze-92e5,2", "SIP/11") in new stack
    -- Executing [A91012@fbze:1] SIPAddHeader("Local/A91012@fbze-af75,2", "Alert-Info:") in new stack
    -- Executing [A91012@fbze:2] Dial("Local/A91012@fbze-af75,2", "SIP/12") in new stack
    -- Called 11
    -- Executing [A91014@fbze:1] SIPAddHeader("Local/A91014@fbze-39c4,2", "Alert-Info:") in new stack
    -- Executing [A91014@fbze:2] Dial("Local/A91014@fbze-39c4,2", "SIP/14") in new stack
    -- Called 12
    -- Executing [A91013@fbze:1] SIPAddHeader("Local/A91013@fbze-2ec5,2", "Alert-Info:") in new stack
    -- Executing [A91013@fbze:2] Dial("Local/A91013@fbze-2ec5,2", "SIP/13") in new stack
    -- Called 14
    -- Called 13
    -- SIP/11-00806dc0 is ringing
    -- Local/A91011@fbze-92e5,1 is ringing
    -- Got SIP response 486 "Busy Here" back from 192.168.66.180
    -- SIP/12-b005e110 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
  == Auto fallthrough, channel 'Local/A91012@fbze-af75,2' status is 'BUSY'
    -- Local/A91012@fbze-af75,1 is busy
    -- SIP/14-007e0840 is ringing
    -- Local/A91014@fbze-39c4,1 is ringing
    -- SIP/13-007e4ba0 is ringing
    -- Local/A91013@fbze-2ec5,1 is ringing
  == Spawn extension (fbze, 10, 1) exited non-zero on 'SIP/32-008054e0'
  == Spawn extension (fbze, A91011, 2) exited non-zero on 'Local/A91011@fbze-92e5,2'
  == Spawn extension (fbze, A91014, 2) exited non-zero on 'Local/A91014@fbze-39c4,2'
  == Spawn extension (fbze, A91013, 2) exited non-zero on 'Local/A91013@fbze-2ec5,2'
  == Spawn extension (fbze, 11, 4) exited non-zero on 'SIP/50-007ca360'

(12:03:22) vsasa: ali evo i ovdje se patim
(12:03:34) vsasa: kako izbrisati registrovanu slušalicu sa baze ??
(12:03:51) vsasa: konstatno mi ovu novu ovdje prijavljuje kao 3-ću slušalicu, a treba da bude druga
(12:04:15) hernad: i dalje registrovana na prvoj bazi
(12:04:23) hernad: ?
(12:04:47) vsasa: ne ne, nego na drugoj bazi kada sam prebacio uporno je stavlja kao 3-cu slusalicu,
           a treba kao drugu
(12:05:02) vsasa: nekad je bila neka prijavljena i ostala zapamćena pod 2
(12:05:09) vsasa: ona slušalica naša
(12:05:45) hernad: ali to i nije neki problem što je ona interno treća slušalica
(12:05:50) vsasa: hm
(12:05:52) hernad: bitno je da se druga ne logira
(12:05:52) vsasa: pa ne znam
(12:05:58) vsasa: sad ću vidjeti
(12:06:20) hernad: bitno je da je mapiranje sip - gigaset korektno  
(12:08:40) hernad: meni što se tiče fbze sve izgleda korektno
(12:08:52) hernad: barem što se tiče zvonjave i handliranja asteriska
(12:08:58) hernad: što se tiče
(12:09:01) hernad: redoslijeda poziva
(12:09:07) hernad: vsasa
(12:09:16) hernad: neka te ne buni 
(12:09:20) hernad: što u logu idu pozivi
(12:09:36) hernad: npr 12 14 13 11
(12:09:43) hernad: to je nebitni
(12:09:47) hernad: to je asinhron proces
(12:09:57) hernad: tako i za ringing
(12:09:58) hernad: isto
(12:10:34) hernad: on odradi launch tih poziva istovremeno 
                 pa koji se kad javi to se registruje u log-u

sada evo podesio sve i testirao da jasko zove ext 11, a ja 10 sa drugog telefona i evo

m1*CLI> 
    -- SIP/11-007fab80 is ringing
    -- SIP/11-007fab80 answered SIP/50-007ca360
    -- Executing [10@fbze:1] Dial("SIP/32-00806dc0", "LOCAL/A91011@fbze&LOCAL/A91012@fbze&LOCAL/A91013@fbze&LOCAL/A91014@fbze|400|tT") in new stack
    -- Called A91011@fbze
    -- Called A91012@fbze
    -- Called A91013@fbze
    -- Called A91014@fbze
    -- Executing [A91011@fbze:1] SIPAddHeader("Local/A91011@fbze-cd18,2", "Alert-Info:") in new stack
    -- Executing [A91011@fbze:2] Dial("Local/A91011@fbze-cd18,2", "SIP/11") in new stack
    -- Called 11
    -- Executing [A91012@fbze:1] SIPAddHeader("Local/A91012@fbze-20f5,2", "Alert-Info:") in new stack
    -- Executing [A91012@fbze:2] Dial("Local/A91012@fbze-20f5,2", "SIP/12") in new stack
    -- Called 12
    -- Executing [A91013@fbze:1] SIPAddHeader("Local/A91013@fbze-3e29,2", "Alert-Info:") in new stack
    -- Executing [A91013@fbze:2] Dial("Local/A91013@fbze-3e29,2", "SIP/13") in new stack
    -- Called 13
    -- Executing [A91014@fbze:1] SIPAddHeader("Local/A91014@fbze-e7fc,2", "Alert-Info:") in new stack
    -- Executing [A91014@fbze:2] Dial("Local/A91014@fbze-e7fc,2", "SIP/14") in new stack
    -- Called 14
    -- SIP/13-007e5790 is ringing
    -- Local/A91013@fbze-3e29,1 is ringing
    -- SIP/11-007d6a10 is ringing
    -- Local/A91011@fbze-cd18,1 is ringing
    -- Got SIP response 486 "Busy Here" back from 192.168.66.180
    -- SIP/12-007e0840 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
  == Auto fallthrough, channel 'Local/A91012@fbze-20f5,2' status is 'BUSY'
    -- Local/A91012@fbze-20f5,1 is busy
  == Spawn extension (fbze, 10, 1) exited non-zero on 'SIP/32-00806dc0'
  == Spawn extension (fbze, A91013, 2) exited non-zero on 'Local/A91013@fbze-3e29,2'
  == Spawn extension (fbze, A91011, 2) exited non-zero on 'Local/A91011@fbze-cd18,2'
  == Spawn extension (fbze, A91014, 2) exited non-zero on 'Local/A91014@fbze-e7fc,2'
  == Spawn extension (fbze, 11, 4) exited non-zero on 'SIP/50-007ca360'

zvoni samo 13 extenzija.

http://forums.whirlpool.net.au/forum-replies-archive.cfm/927308.html

Simlutaneous phone calls¶

I have always known the C470IP can only support two simultaneous VOIP calls plus the PSTN. I never thought this would be a limitation as I could not conceive in a home environment when I ever needed to do 3 simultaneous voice calls. If I really wanted to do 3, my plan was to connect the C470IP's PSTN line to an FXS port on my Asterisk box. That would give me the ability to do 3 simulateous SIP VOIP connections through the Astersisk box, but 2 via SIP to the C470IP handsets and 1 via analogue PSTN to the C470IP handsets.

When trying to work out how I want my home system to work I thought a good start was to ring all handsets when a call comes in. The logical thing to do in Asterisk is to setup a ring group to ring all extensions when a call comes in. To my initial surprise only two of the extensions rang. I quickly realised this is probably due to the 2 SIP call limitation, even though I have not got an active call going. All three extensions were registered with Asterisk, however when Asterisk tried to call each extension in the ringgroup, the Asterisk CLI reports only 2 of them successfully ringing.

This is not a big issue for me as I'm sure I can work out an alternative. For example I could ring a hunt group that tries each extension in turn, or I could get the C470IP to ring two or three extensions when an incoming call from one of the providers or the PSTN comes in. But it is something that you should keep in mind.

asterisk-users - Anyone used Siemens SIP/Dect phones?

kod faruka u fbde postavio jedan trik

 4569800 => {
                Dial(SIP/50,400,tT);
        };

        4569801 => {
                for (x=0; ${x} < 20; x=${x} + 1) {
                        Dial(sip/51,8,tT);
                        Dial(sip/52,8,tT);
                        Dial(sip/53,8,tT);
                };
        };

dakle 8 sekundi je dovoljno da 2 puta zazvoni svaki telefon, i evo CLI output-a

m1*CLI> 
  == ISDN1#02: Incoming call '01725288203' -> '4569801'
    -- Executing [4569801@demo:1] Set("CAPI/ISDN1#02/4569801-d5", "x=0") in new stack
    -- Executing [4569801@demo:2] GotoIf("CAPI/ISDN1#02/4569801-d5", "1?3:8") in new stack
    -- Goto (demo,4569801,3)
    -- Executing [4569801@demo:3] Dial("CAPI/ISDN1#02/4569801-d5", "sip/51|8|tT") in new stack
    -- Called 51
    -- SIP/51-0087ed60 is ringing
    -- Nobody picked up in 8000 ms
    -- Executing [4569801@demo:4] Dial("CAPI/ISDN1#02/4569801-d5", "sip/52|8|tT") in new stack
    -- Called 52
    -- SIP/52-00856a80 is ringing
    -- Nobody picked up in 8000 ms
    -- Executing [4569801@demo:5] Dial("CAPI/ISDN1#02/4569801-d5", "sip/53|8|tT") in new stack
    -- Called 53
    -- SIP/53-0087ed60 is ringing
    -- Nobody picked up in 8000 ms
    -- Executing [4569801@demo:6] Set("CAPI/ISDN1#02/4569801-d5", "x=1") in new stack
    -- Executing [4569801@demo:7] Goto("CAPI/ISDN1#02/4569801-d5", "2") in new stack
    -- Goto (demo,4569801,2)
    -- Executing [4569801@demo:2] GotoIf("CAPI/ISDN1#02/4569801-d5", "1?3:8") in new stack
    -- Goto (demo,4569801,3)
    -- Executing [4569801@demo:3] Dial("CAPI/ISDN1#02/4569801-d5", "sip/51|8|tT") in new stack
    -- Called 51
    -- SIP/51-00875250 is ringing
    -- SIP/53-00846900 answered CAPI/ISDN1#02/4569801-d5

to je sada ok.

proslijedio faruku info

Faruk, ja sam evo kontaktirao naše dobavljače iz sarajeva i dobio sam pozitivan odgovor što se tiče downgrade-a firmware-a.

Odmah su mi rekli da mogu da donesem kod njih telefon i oni će hardware-skim putem vratiti na default postavke i default firmware.

Trebao bi da kontaktiraš siemens.de od kojih si naručio telefone pa da vidiš da li mogu da ti urade istu stvar.

Dakle,

novi firmware (koji nevalja) je: 021230000000/043.00 EEPROM version: 121

default firmware koji dođe na telfonima je: 020970000000/043.00 EEPROM version: 114

dakle, trebaju da vrate na ovaj default.

najvjerovatnije da imaju ovlašteni servis tih telefona i oni to mogu da urade.

pozdrav 

na http://www.planetasterisk.org

Siemens Gigaset S675IP / Critical Bug on last firmware [ID: 58037]

Critical Bug on last firmware

I have a CRITICAL BUG with last S675IP firmware (02097 and 02123) :

From time to time (randomly), when a IP call is received the handset doesn't ring... And there is no call indication on the phone.

The only way to solve the problem is to restart the DECT base (power off / on)

I've also this problem with C455IP DECT base (02123 and 02097).
The previous firmware (before 02097) didn't have this problem.

Configuration summary :
3 Handsets (one C45, one S67H, one S68H)
1 Base S675IP
5 VoIP account on the same Asterisk (1.4.x)
- 61 : handset 1 (call & receive)
- 62 : handset 2 (call & receive)
- 63 : handset 3 (call & receive)
- 68 : handset 1, handset 2, handset 3 (receive)
- 69 : handset 1, handset 2, handset 3 (receive)

I've done two Ethernet capture with wireshark (ethereal) when trying to call the handset 63.

- When it works correctly
1 0.000000 ASTERISK -> S675IP SIP/SDP Request: INVITE sip:63@S675IP:5251, with session
2 0.128398 S675IP -> ASTERISK SIP Status: 100 Trying
3 0.813618 S675IP -> ASTERISK SIP Status: 180 Ringing
4 5.160596 S675IP -> ASTERISK UDP Source port: 5251 Destination port: 5060
5 5.181266 S675IP -> ASTERISK SIP/SDP Status: 200 OK, with session description
6 5.181455 ASTERISK ->SIP675IP SIP Request: ACK sip:63@S675IP:5251
7 8.713445 S675IP -> ASTERISK SIP Request: BYE sip:60@ASTERISK
8 8.713592 ASTERISK -> S675IP SIP Status: 200 OK

-When the problem occurs
1 2.046400 ASTERISK -> S675IP SIP/SDP Request: INVITE sip:63@S675IP:5251 with session
2 2.144135 S675IP -> ASTERISK SIP Status: 100 Trying
3 32.048449 ASTERISK -> S675IP SIP Request: CANCEL sip:63@S675IP:5251
4 32.147599 S675IP -> ASTERISK SIP Status: 200 OK
5 32.152493 S675IP -> ASTERISK SIP Status: 487 Request Cancelled

When the problem occurs, the S675IP DECT base doesn't send the "180 Trying".

Is there any way to solve this problem (without hard restart) ?
Is it possible to reboot the DECT base remotely (hidden url ?) ?

Thanks

Emmanuel.
emmanuel38 () at 2008-09-01 22:11 GMT