[Asterisk-Dev] Request for a help..............
Scott Stingel
scott at evtmedia.com
Sat Jul 3 06:53:52 MST 2004
Could you please re-post this message on the asterisk-users forum, not the
developer's forum.
Thank you
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California & London England
www.evtmedia.com
-----Original Message-----
From: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com] On Behalf Of sthiti
Sent: Saturday, July 03, 2004 6:51 AM
To: asterisk-dev at lists.digium.com
Subject: [Asterisk-Dev] Request for a help..............
Hi,
I want to setup a PBX, I am having the asterisk software... I am having
1 X100 and 1 TDM400 cards.
The phone line is connected to the X100 card and extensions to TDM card.
I am facing two problems..
1. I want to activate the voicemail in each extension? So that I can hear
the voicemails, how I can do this , I want the details configuration for
this. And which all configuration files need to be change.
2. my 2nd problems is I want to make outgoing call from extensions, for this
what I need to do. How to configure the extension.conf file and other files
too.? I mean I want to call to my home from office through any extension,
for this what will be my configuration.
Take this as my request and urgent. I am stucked up in the middle of my
work.. Please help me..
Sthiti Ranjan Sarangi
Adyasystems & Software Pvt. Ltd.
New Delhi
India
-----Original Message-----
From: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com] On Behalf Of
asterisk-dev-request at lists.digium.com
Sent: Monday, June 28, 2004 10:56 AM
To: asterisk-dev at lists.digium.com
Subject: Asterisk-Dev digest, Vol 1 #730 - 11 msgs
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"Re: Contents of Asterisk-Dev digest..."
Today's Topics:
1. Re: Manager Command Reference (Rooster)
2. sms sending and receiving (Andreas Bayer)
3. Re: HylaFAX and spandsp (Bruce Ferrell)
4. Re: sms sending and receiving (Hans Mueller)
5. voip online status (Andreas Bayer)
6. Re: voip online status (Rooster)
7. Re: DTMF digits relay with Quintum to Oh323 version
0.5.10 (Kelvin Chua)
8. Re: E option in meetme.conf (Tilghman Lesher)
9. Re: Getting URL to IAX Client Agent (Navnit Chachan)
10. Re: Getting URL to IAX Client Agent (Navnit Chachan)
--__--__--
Message: 1
From: "Rooster" <rooster at cableaz.com>
To: <asterisk-dev at lists.digium.com>
Subject: Re: [Asterisk-Dev] Manager Command Reference
Date: Sun, 27 Jun 2004 13:01:50 -0700
Reply-To: asterisk-dev at lists.digium.com
any clues as to which source files to look at?
----- Original Message -----
From: "Brancaleoni Matteo" <mbrancaleoni at espia.it>
To: <asterisk-dev at lists.digium.com>
Sent: Sunday, June 27, 2004 12:41 AM
Subject: Re: [Asterisk-Dev] Manager Command Reference
> Hi
>
> Il dom, 2004-06-27 alle 08:14, Rooster ha scritto:
> > I'm looking for a command reference for the Asterisk manager
> > (something that shows the usage for each command). Any one know
> > where I can find one?
>
> from asterisk CLI
> "show manager commands"
> will show a list of the available commands.
> regarding the syntax, the source is your friend.
>
> Matteo.
>
> --
> Brancaleoni Matteo <mbrancaleoni at espia.it>
> Espia Srl
>
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
--__--__--
Message: 2
From: Andreas Bayer <angel_azrael at gmx.de>
To: asterisk-dev at lists.digium.com
Date: Sun, 27 Jun 2004 22:28:08 +0200
Subject: [Asterisk-Dev] sms sending and receiving
Reply-To: asterisk-dev at lists.digium.com
Hi,
i heard that Asterisk supports short messages.
What channels support sms?
I heard, that sms over isdn (compatible with protocol used by "Deutsche
Telekom" ) work.
And a heard of a channel for sms over serial connected gsm-phones. Which
phones (all phones supported by gnokii, at-modems or something else) are
supported?
bye
--__--__--
Message: 3
Date: Sun, 27 Jun 2004 13:47:57 -0700
From: Bruce Ferrell <bferrell at baywinds.org>
To: asterisk-dev at lists.digium.com
Subject: Re: [Asterisk-Dev] HylaFAX and spandsp
Reply-To: asterisk-dev at lists.digium.com
t38modem can fallback to G711, but only H323.
Florian Overkamp wrote:
> Hi,
>
>
>>-----Original Message-----
>>FAX should *really* be handled at the PSTN interface, and only T.37 or
>>T.38 used across any IP links.
>
>
> I totally agree, but since T37/T38 is not available in asterisk at
> this time and faxing over g711 does work in a reasonable number of
> situations I think we don't always have that option :-P
>
> Best regards,
> Florian
>
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
--__--__--
Message: 4
Date: Sun, 27 Jun 2004 23:10:56 +0200
From: Hans Mueller <74521ffee0851ff135013b95382277b9 at digitalcity.ch>
To: asterisk-dev at lists.digium.com
Subject: [Asterisk-Dev] Re: sms sending and receiving
Reply-To: asterisk-dev at lists.digium.com
Hi Andeas,
Andreas Bayer wrote:
>i heard that Asterisk supports short messages.
>What channels support sms?
>
>
>I heard, that sms over isdn (compatible with protocol used by "Deutsche
>Telekom" ) work.
>
IMHO there are two completely different SMS Protocoll: The first works
on any analog line. This is a known ETSI standard and is supported by
app_sms.
The second one, which Swisscom Fixnet uses (i guess this is the same
stuff as Deutsche Telekom), works ONLY on BRI. The Protocoll itself is
very
simple,
you get a short call from the SMSC (with the callerid of the SMSC), the
phonenumber of the sending cellphone and the messagetext itself is in
the UUS1
field of the call. You can see this if you do a "capi debug", it looks
like
Jun 18 11:46:29 NOTICE[-1158329424]: CONNECT_IND ID=001 #0x2a1e LEN=0073
Controller/PLCI/NCCI = 0x101
CIPValue = 0x2
CalledPartyNumber = <c1>5551234
CallingPartyNumber = <01 81>0812866130
CalledPartySubaddress = default
CallingPartySubaddress = default
BC = <88 90>
LLC = default
HLC = default
AdditionalInfo
BChannelinformation = <00 00>
Keypadfacility = default
Useruserdata = <00 01 0a>0795554444testmessage<2e>
Facilitydataarray = default
So the cellphone with the number 0795554444 sent the message
"testmessage" to 5551234, and 0812866130 is the number of the SMSC in
Swisscom fixnet.
The current chan_capi does *not* support this to be used from asterisk,
but i guess it woudn't be
very hard to put the Useruserdata into an asterisk variable and send the
call to agi to process
an incoming sms. If someone would add this, it would be cool anyway,
some big pbxes add the
caller's name to the UUS1-Field...
>And a heard of a channel for sms over serial connected gsm-phones.
>Which
>phones (all phones supported by gnokii, at-modems or something else)
are
>supported?
>
Never heard of something like that...
bye
Hans
--__--__--
Message: 5
From: Andreas Bayer <angel_azrael at gmx.de>
To: asterisk-dev at lists.digium.com
Date: Mon, 28 Jun 2004 00:52:35 +0200
Subject: [Asterisk-Dev] voip online status
Reply-To: asterisk-dev at lists.digium.com
Hi,
is it possible to check if somebodies voip-client is online ?? I think
of a feature like the icq-buddy-list. Which protocols support such a
feature?
Is it possible to use the isdn feature ccnr ("rueckruf bei nichtmelden")
in
such way?
bye
--__--__--
Message: 6
From: "Rooster" <rooster at cableaz.com>
To: <asterisk-dev at lists.digium.com>
Subject: Re: [Asterisk-Dev] voip online status
Date: Sun, 27 Jun 2004 19:01:01 -0700
Reply-To: asterisk-dev at lists.digium.com
it's on it's way... www.quadrasoftware.com
----- Original Message -----
From: "Andreas Bayer" <angel_azrael at gmx.de>
To: <asterisk-dev at lists.digium.com>
Sent: Sunday, June 27, 2004 3:52 PM
Subject: [Asterisk-Dev] voip online status
> Hi,
>
> is it possible to check if somebodies voip-client is online ?? I think
> of a feature like the icq-buddy-list. Which protocols support such a
> feature?
>
> Is it possible to use the isdn feature ccnr ("rueckruf bei
> nichtmelden")
in
> such way?
>
> bye
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
--__--__--
Message: 7
Subject: Re: [Asterisk-Dev] DTMF digits relay with Quintum to Oh323
version
0.5.10
From: Kelvin Chua <kchua at up.edu.ph>
To: asterisk-dev at lists.digium.com
Date: Mon, 28 Jun 2004 10:42:55 +0800
Reply-To: asterisk-dev at lists.digium.com
dtmf works from quintum to asterisk using inband dtmf in g711 alaw. i
have not tested dtmf from * to quintum. i think it might work.
btw, the "dtmf timeout" problem in call transfers exist in all of our
h323 clients. might be a chan_h323 thingie...
On Wed, 2004-06-23 at 15:12, Dmitry Mishchenko wrote:
> On Wednesday 23 June 2004 08:38, Kelvin Chua wrote:
> > what quintum model are you using? we are using tenor AX with
> > chan_h323 and dtmf is passed fine but the timeout is really short
> > therefore only 2 out of 4 digits are collected during transfers.
> >
> Kelvin, could you clarify: are you saying DTMFs working when you are
> sending
> DTMFs from Quintun to astersisk of from asterisk to Quintum? or both
> directions?
> What codec are you using for this setup?
>
> Luis, may be there is a sense to try the latest version of oh323.
>
> Dmitry
>
> > On Tue, 2004-06-15 at 07:43, Luis Mata wrote:
> > > Hello:
> > >
> > > Has any one being able to received dtmf digits from
> > > Quintum gateways to asterisk coming from Oh323 channel driver. For
> > > some reason no matter how I setup the Oh323 channel drivers
> > > whether is inband or Outband (STRING or TONE ) . Am unable to get
> > > the tone meessages I have included the trace from oh323.trc in
> > > here to see if someone has any ideas.. When ever a user hits a key
> > > this is what the system gets...
> > >
> > >
> > > 2:34.893 ThreadID=0x00009011 RTP Found existing
session 1
> > > 2:34.903 LogChanTx:819f6b0 H323RTP Transmit start of
talk
> > > burst: 480
> > > 2:35.182 RTP Jitter:818a498 RTP First data: ver=2
> > > pt=G729 psz=20 m=0 x=0 seq=38 ts=6080 src=2864434397 ccnt=0
> > > 2:36.484 LogChanRx:8189fe8 H323RTP Receiver written
> > > timestamp 16160
> > > 2:36.844 LogChanTx:819f6b0 H323RTP Transmitter sent
> > > timestamp 16080
> > > 2:36.904 LogChanTx:819f6b0 RTP Transmit
statistics:
> > > packets=101 octets=2020 avgTime=20 maxTime=21 minTime=20
> > > 2:37.024 LogChanTx:819f6b0 H323RTP Transmit end of
talk
> > > burst: 17440
> > > 2:37.044 LogChanTx:819f6b0 H323RTP Transmit start of
talk
> > > burst: 17600
> > > 2:37.184 RTP Jitter:818a498 RTP Receive
statistics:
> > > packets=101 octets=2020 lost=0 tooLate=0 order=0 avgTime=20
> > > maxTime=79 minTime=0 jitter=7 maxJitter=11
> > > 2:37.823 LogChanTx:819f6b0 H323RTP Transmit end of
talk
> > > burst: 23840
> > > 2:38.503 LogChanRx:8189fe8 H323RTP Receiver written
> > > timestamp 32320
> > > 2:38.863 LogChanTx:819f6b0 H323RTP Transmitter sent
> > > timestamp 32160
> > > 2:39.043 LogChanRx:8189fe8 RTP Jitter buffer size
> > > decreased to 26765 (3345ms)
> > > 2:39.183 RTP Jitter:818a498 RTP Receive
statistics:
> > > packets=201 octets=4020 lost=0 tooLate=0 order=0 avgTime=19
> > > maxTime=47 minTime=6 jitter=4 maxJitter=11
> > > 2:40.873 LogChanTx:819f6b0 H323RTP Transmitter sent
> > > timestamp 48240
> > > 2:41.183 RTP Jitter:818a498 RTP Receive
statistics:
> > > packets=301 octets=6020 lost=0 tooLate=0 order=0 avgTime=20
> > > maxTime=203 minTime=0 jitter=9 maxJitter=27
> > > 2:42.873 LogChanTx:819f6b0 H323RTP Transmitter sent
> > > timestamp 64320
> > > 2:43.185 RTP Jitter:818a498 RTP Receive
statistics:
> > > packets=401 octets=8020 lost=0 tooLate=0 order=0 avgTime=20
> > > maxTime=50 minTime=5 jitter=3 maxJitter=27
> > > 2:43.773 LogChanRx:8189fe8 RTP Jitter buffer size
> > > decreased to 26757 (3344ms)
> > > 2:43.833 LogChanRx:8189fe8 H323RTP Receiver written
> > > timestamp 48480
> > > 2:44.893 LogChanTx:819f6b0 H323RTP Transmitter sent
> > > timestamp 80400
> > > 2:45.184 RTP Jitter:818a498 RTP Receive
statistics:
> > > packets=501 octets=10020 lost=0 tooLate=0 order=0 avgTime=19
> > > maxTime=47 minTime=7 jitter=3 maxJitter=27
> > > 2:45.793 LogChanRx:8189fe8 RTP Jitter buffer size
> > > decreased to 26749 (3343ms)
> > > 2:45.853 LogChanRx:8189fe8 H323RTP Receiver written
> > > timestamp 64640
> > > 2:46.766 RTP Jitter:818a498 RTP SentSenderReport:
> > > ssrc=1213391464 ntp=3296245119.793380910 rtp=23680 psent=145
osent=2900
> > > 2:46.766 RTP Jitter:818a498 RTP
SentReceiverReport:
> > > ssrc=2864434397 fraction=0 lost=0 last_seq=0 jitter=13 lsr=0
dlsr=0
> > > 2:46.766 RTP Jitter:818a498 RTP Sending SDES:
> > > 2:46.893 LogChanTx:819f6b0 H323RTP Transmitter sent
> > > timestamp 96480
> > > 2:47.184 RTP Jitter:818a498 RTP Receive
statistics:
> > > packets=601 octets=12020 lost=0 tooLate=0 order=0 avgTime=20
> > > maxTime=48 minTime=6 jitter=1 maxJitter=27
> > > 2:47.823 LogChanRx:8189fe8 RTP Jitter buffer size
> > > decreased to 26741 (3342ms)
> > > 2:47.863 LogChanRx:8189fe8 H323RTP Receiver written
> > > timestamp 80800
> > > 2:48.903 LogChanTx:819f6b0 H323RTP Transmitter sent
> > > timestamp 112560
> > > 2:49.184 RTP Jitter:818a498 RTP Receive
statistics:
> > > packets=701 octets=14020 lost=0 tooLate=0 order=0 avgTime=20
> > > maxTime=48 minTime=8 jitter=3 maxJitter=27
> > > 2:49.863 LogChanRx:8189fe8 RTP Jitter buffer size
> > > decreased to 26733 (3341ms)
> > > 2:49.883 LogChanRx:8189fe8 H323RTP Receiver written
> > > timestamp 96960
> > >
> > >
> > >
> > > Thanks...
> > >
> > > ; Valid values for this option are:
> > > ; Q931 - Q.931 Keypad Information Element
> > > ; STRING - H.245 string
> > > ; TONE - H.245 tone
> > > ; RFC2833 - RFC2833
> > > ;
> > >
> > > userInputMode=STRING I have try either STRING or AUDIO with our
> > > luck....
> > >
> > >
> > > Configuration of OpenH323 channel driver
> > >
> > > ----------------------------------------
> > > Version: 0.5.10
> > > Listening on address: XXX.XXX.XXX.XXX
> > > Gatekeeper used: <No Gatekeeper>
> > > FastStart/H245Tunnelling/H245inSetup: ON/ON/ON
> > > Supported format(s): G729A<0>
> > > Jitter buffer limits (min/max): 20-10000 ms
> > > TCP port range: 10000 - 20000
> > > UDP (RAS) port range: 40001 - 59999
> > > UDP (RTP) port range: 10000 - 40000
> > > IP Type-of-Service value: 0
> > > User input mode: 0
> > > Max number of inbound H.323 calls: 120
> > > Max number of outbound H.323 calls: 120
> > > Max number of simultaneous H.323 calls: 120
> > >
> > >
> > >
> > > _______________________________________________
> > > Asterisk-Dev mailing list
> > > Asterisk-Dev at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-dev
> > > To UNSUBSCRIBE or update options visit:
> > > http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
> > _______________________________________________
> > Asterisk-Dev mailing list
> > Asterisk-Dev at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
>
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
--__--__--
Message: 8
From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
To: asterisk-dev at lists.digium.com
Subject: Re: [Asterisk-Dev] E option in meetme.conf
Date: Sun, 27 Jun 2004 23:01:13 -0500
Reply-To: asterisk-dev at lists.digium.com
On Friday 11 June 2004 20:49, Ryan Courtnage wrote:
> Hi all,
>
> I'm toying with the 'E' option for MeetMe:
>
> 'E' =97 select an empty pinless conference (added after v0.7.2)
>
>
> [ext-meetme]
> exten =3D> 80,1,MeetMe(|E)
> exten =3D> _8[1-6],1,MeetMe(${EXTEN}) ; 81-86 defined in meetme.conf
>
>
> Works great when a single extension dials into 80.
> However, if a second extension dials 80 (while the 1st extension is
> still in a conference), * will pin the CPU and require a restart.
What kind of restart? Asterisk soft restart, kill -9 restart, warm
reboot, or cold reboot?
Also, what processor are you using, and what options did you turn on in
zconfig.h? The contents of zaptel.conf and zapata.conf might be useful,
too, depending upon the answers above.
=2D-=20
Tilghman
--__--__--
Message: 9
From: "Navnit Chachan" <navnit at tentrams.com>
To: <asterisk-dev at lists.digium.com>
Subject: Re: [Asterisk-Dev] Getting URL to IAX Client Agent
Date: Mon, 28 Jun 2004 10:39:41 +0530
Reply-To: asterisk-dev at lists.digium.com
Hi,
This is exactly what the doctor ordered.
As an issue, have you tried using Queue+URL. Dial+URL works for me but
not
Queue+URL.
Looking into the app_queue code, just when the url is to be sent, (Line
1167), ast_channel_supports_html(peer) returns a zero even though I am
using an iax client. The same client can recieve the URL when using
Dial+URL from *.
Am I missing something here?
Thanx
Navnit
----- Original Message -----
From: "Jean-Denis Girard" <jd-girard at esoft.pf>
To: <asterisk-dev at lists.digium.com>
Sent: Saturday, June 26, 2004 2:36 AM
Subject: Re: [Asterisk-Dev] Getting URL to IAX Client Agent
> Steven Sokol wrote:
> > I'm working on adding URL support to IAX Phone. It will simply pop
> > the default browser with the URL. The bigger question is: how do
> > you want
to go
> > about _sending_ the URLs? Copy/paste/send?
> >
> > Click a button that pops up a dialog asking you to enter the URL?
> > The dialog box could contain a set of "Speed Dial URLs" for commonly
> > transmitted info. It could also contain a drop-down list for
> > recently
sent
> > URLs. Would that work?
> >
> > Thanks,
> >
> > Steve
> >
> >
>
> Well, here is my current patch to libiaxclient. As you will see, it
> does not support sending an URL from the client, just receiving URL
> which may be sent by * via Dial+URL (or Queue+URL), usefull in call
> center environment. I don't really know how to best handle sending an
> URL from the client, but copy/paste/send seems ok to me.
>
>
> Thanks,
>
> --
> Jean-Denis Girard
>
> ====================================================
> Essential Software - Ingénierie Informatique
> Solutions Linux & Open Source en Polynésie française
> ----------------------------------------------------
> http://www.esoft.pf/
> Tél: (689) 54 12 95
> ====================================================
>
------------------------------------------------------------------------
----
----
> diff -Naur iaxclient-20040617/lib/iaxclient.h
iaxclient-20040617-JDG/lib/iaxclient.h
> --- iaxclient-20040617/lib/iaxclient.h 2004-06-10 14:01:23.000000000
> -1000
> +++ iaxclient-20040617-JDG/lib/iaxclient.h 2004-06-17
14:40:30.135164832 -1000
> @@ -47,6 +47,7 @@
> #define IAXC_EVENT_TEXT 1
> #define IAXC_EVENT_LEVELS 2
> #define IAXC_EVENT_STATE 3
> +#define IAXC_EVENT_URL 4 /* URL push via IAX(2) */
>
> #define IAXC_CALL_STATE_FREE 0
> #define IAXC_CALL_STATE_ACTIVE (1<<1)
> @@ -62,7 +63,11 @@
> #define IAXC_TEXT_TYPE_FATALERROR 4
> #define IAXC_TEXT_TYPE_IAX 5
>
> -
> +#define IAXC_URL_URL 1 /* URL received */
> +#define IAXC_URL_LDCOMPLETE 2 /* URL loading complete */ #define
> +IAXC_URL_LINKURL 3 /* URL link request */ #define IAXC_URL_LINKREJECT
> +4 /* URL link reject */ #define IAXC_URL_UNLINK 5 /* URL unlink */
>
> #define IAXC_EVENT_BUFSIZ 256
> struct iaxc_ev_levels {
> @@ -85,12 +90,19 @@
> char local_context[IAXC_EVENT_BUFSIZ];
> };
>
> +struct iaxc_ev_url {
> + int callNo;
> + int type;
> + char url[IAXC_EVENT_BUFSIZ];
> +};
> +
> typedef struct iaxc_event_struct {
> int type;
> union {
> struct iaxc_ev_levels levels;
> struct iaxc_ev_text text;
> struct iaxc_ev_call_state call;
> + struct iaxc_ev_url url;
> } ev;
> } iaxc_event;
>
> diff -Naur iaxclient-20040617/lib/iaxclient_lib.c
iaxclient-20040617-JDG/lib/iaxclient_lib.c
> --- iaxclient-20040617/lib/iaxclient_lib.c 2004-06-10
14:01:23.000000000 -1000
> +++ iaxclient-20040617-JDG/lib/iaxclient_lib.c 2004-06-17
14:40:30.136164680 -1000
> @@ -509,6 +509,52 @@
> iaxc_post_event(ev);
> }
>
> +/* handle IAX URL events */
> +void handle_url_event( struct iax_event *e, int callNo ) {
> +iaxc_event ev;
> +
> + if(callNo < 0) return;
> +
> + ev.ev.url.callNo = callNo;
> + ev.type = IAXC_EVENT_URL;
> + strcpy( ev.ev.url.url, "" );
> +
> + switch( e->subclass ) {
> + case AST_HTML_URL:
> + ev.ev.url.type = IAXC_URL_URL;
> + if( e->datalen ) {
> + if( e->datalen > IAXC_EVENT_BUFSIZ ) {
> + fprintf( stderr, "ERROR: URL too long %d > %d\n",
> + e->datalen, IAXC_EVENT_BUFSIZ );
> + } else {
> + strncpy( ev.ev.url.url, e->data, e->datalen );
> + }
> + }
> + fprintf( stderr, "URL:%s\n", ev.ev.url.url );
> + break;
> + case AST_HTML_LINKURL:
> + ev.ev.url.type = IAXC_URL_LINKURL;
> + fprintf( stderr, "LINKURL event\n" );
> + break;
> + case AST_HTML_LDCOMPLETE:
> + ev.ev.url.type = IAXC_URL_LDCOMPLETE;
> + fprintf( stderr, "LDCOMPLETE event\n" );
> + break;
> + case AST_HTML_UNLINK:
> + ev.ev.url.type = IAXC_URL_UNLINK;
> + fprintf( stderr, "UNLINK event\n" );
> + break;
> + case AST_HTML_LINKREJECT:
> + ev.ev.url.type = IAXC_URL_LINKREJECT;
> + fprintf( stderr, "LINKREJECT event\n" );
> + break;
> + default:
> + fprintf( stderr, "Unknown URL event %d\n", e->subclass ); break;
> + }
> + iaxc_post_event( ev );
> +}
> +
> void handle_audio_event(struct iax_event *e, int callNo) {
> int total_consumed = 0;
> int cur;
> @@ -599,6 +645,9 @@
> case IAX_EVENT_PONG: /* we got a pong */
> //fprintf(stderr, "**********GOT A PONG!\n");
> break;
> + case IAX_EVENT_URL:
> + handle_url_event(e, callNo);
> + break;
> default:
> iaxc_usermsg(IAXC_STATUS, "Unknown event: %d for call %d", e->etype,
callNo);
> break;
> diff -Naur iaxclient-20040617/lib/libiax2/src/iax.c
iaxclient-20040617-JDG/lib/libiax2/src/iax.c
> --- iaxclient-20040617/lib/libiax2/src/iax.c 2004-06-17
14:37:02.586716992 -1000
> +++ iaxclient-20040617-JDG/lib/libiax2/src/iax.c 2004-06-17
14:40:30.137164528 -1000
> @@ -1944,6 +1944,8 @@
> case AST_HTML_URL:
> if (!e->etype)
> e->etype = IAX_EVENT_URL;
> + e->subclass = fh->csub;
> + e->datalen = datalen;
> if (datalen) {
> memcpy(e->data, fh->iedata, datalen);
> }
>
--__--__--
Message: 10
From: "Navnit Chachan" <navnit at tentrams.com>
To: <asterisk-dev at lists.digium.com>
Subject: Re: [Asterisk-Dev] Getting URL to IAX Client Agent
Date: Mon, 28 Jun 2004 10:42:02 +0530
Reply-To: asterisk-dev at lists.digium.com
This is implemented would be good.
----- Original Message -----
From: "Steven Sokol" <ssokol at sokol-associates.com>
To: <asterisk-dev at lists.digium.com>
Sent: Saturday, June 26, 2004 12:20 AM
Subject: RE: [Asterisk-Dev] Getting URL to IAX Client Agent
> Navnit Chachan wrote:
> > I am using IAX Phone from skol associates
I'm working on adding URL support to IAX Phone. It will simply pop the
default browser with the URL. The bigger question is: how do you want
to go about _sending_ the URLs? Copy/paste/send?
Click a button that pops up a dialog asking you to enter the URL? The
dialog box could contain a set of "Speed Dial URLs" for commonly
transmitted info. It could also contain a drop-down list for recently
sent URLs. Would that work?
Thanks,
Steve
>
> I may be wrong, but as far as I know:
>
> The only IAX client that supports URL is gnophone, which is basically
> dead since it's only IAX1... There is no support for URL in the
> libiaxclient on which IAXPhone is based.
>
> I'm currently working on yet another iax2 client with support for URL,
> but it's too early in the development to show anything.
>
> --
> Jean-Denis Girard
>
> ====================================================
> Essential Software - Ingénierie Informatique
> Solutions Linux & Open Source en Polynésie française
> ----------------------------------------------------
> http://www.esoft.pf/
> Tél: (689) 54 12 95
> ====================================================
>
> _______________________________________________
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