[asterisk-users] Problems with sendinf to list. Was: Re: Paging in Asterisk
On Mo, 15 Okt 2007, [EMAIL PROTECTED] wrote: > I've been trying to send a message to the list for the past 3 days, but > I neither get bounces nor the message appearing in the list, so someone > on IRC sugested I reply to an existing message. Same with me here! -- Volker Sauer * Poststrasse 1/601 * 64293 Darmstadt * Germany E-Mail/Jabber: volker(at)volker-sauer.de * http://www.volker-sauer.de PGPKey-Fingerprint: DB26 11C7 B12E 0B27 3999 2E4F 7E35 4E4D 5DD5 D0E0 http://wwwkeys.de.pgp.net/pks/lookup?op=get&search=0x7E354E4D5DD5D0E0 signature.asc Description: Digital signature ___ --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
[asterisk-users] Problem: features (from features.conf) not available if call was originated by manager API or call file
Hello asterisk-users,
I setup my asterisk to support several features like
automon,blindxfer,atxfer,parkcall etc. by using features.conf and the
global variable
DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in
extension.conf. Every Dial() command in my diaplan has the appropriate
parameters out of {tTkWwW}.
For calls from my SIP phones everything works fine. Pressing #1 will
transfer, pressing ## will automon and so on.
But if I originate the call from either a call file (which is used by a
callback application on my setup) or the manager api (which is used by a
webadressbook, which automatically dials and connects to the phone on
your desk) these features are not available even the manager api jumps
to the same context as a normal call from a phone and uses the same
Dial() command with {tTwWkK}. This means, that alle the nice feature
keys which "normally" work do not if I originate the call over the
webfrontend.
Therefore, either the global variable is not known to the call which
came from the call file or the manager api or something else is going
wrong.
Here's my call file:
Channel: SIP/tol/06151154260
Account: t-online
CallerID: 03222XXX <03222XXX>
MaxRetries: 4
RetryTime: 15
WaitTime: 60
Context: doCallBackVolkerStage2
Extension: s
Priority: 1
(the Context doCallBackVolkerStage2 will run DISA).
and that's the PHP code orgiginating the calls:
$socket = fsockopen("127.0.0.1","5038", $errno, $errstr, $timeout);
fputs($socket, "Action: Login\r\n");
fputs($socket, "UserName: webdial\r\n");
fputs($socket, "Secret: XX\r\n\r\n");
fputs($socket, "Action: Originate\r\n");
fputs($socket, "Channel: $channel\r\n");
fputs($socket, "Exten: $tonumber\r\n");
fputs($socket, "CallerID: $callerid\r\n");
fputs($socket, "Context: doLocalCalls\r\n");
fputs($socket, "MaxRetries: 1\r\n");
fputs($socket, "RetryTime: 15\r\n");
fputs($socket, "WaitTime: 60\r\n");
fputs($socket, "Priority: 1\r\n\r\n");
fputs($socket, "SetLanguage: de\r\n\r\n");
fputs($socket, "Action: Logoff\r\n\r\n");
(the Context doLocalCalls is the same as on any other SIP phone. It will run
Dial().
Any hint on what I'm doing wrong or where to check else?
Regards
Volker
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Re: [asterisk-users] Problem: features (from features.conf) not available if call was originated by manager API or call file
TECTED],1", "SIP/tol/XXX||ktwKTW") in new stack -- Called tol/06151154260 == Spawn extension (macro-intDial, s, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'intDial' == Spawn extension (macro-intDial, s, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' -- SIP/tol-08240f78 is making progress passing it to SIP/cisco2-08247f88 -- SIP/tol-08240f78 is ringing -- SIP/tol-08240f78 is making progress passing it to SIP/cisco2-08247f88 -- SIP/tol-08240f78 answered SIP/cisco2-08247f88 Now, the phone and the remote party are called with Dial with ktwKTW. This should do it. But: nothing! I declare this a bug, now! Regards Volker > > Regards, > Atis > > > -- > Atis Lezdins > VoIP Developer, > IQ Labs Inc. > [EMAIL PROTECTED] > Skype: atis.lezdins > Cell Phone: +371 28806004 > Work phone: +1 800 7502835 > -- Volker Sauer * Poststrasse 1/601 * 64293 Darmstadt * Germany E-Mail/Jabber: volker(at)volker-sauer.de * http://www.volker-sauer.de PGPKey-Fingerprint: DB26 11C7 B12E 0B27 3999 2E4F 7E35 4E4D 5DD5 D0E0 http://wwwkeys.de.pgp.net/pks/lookup?op=get&search=0x7E354E4D5DD5D0E0 signature.asc Description: Digital signature ___ --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
[asterisk-users] CLI not showing DTMF
Hi, after I upgraded from 1.2 to 1.4.13 the CLI does not show DTMF anymore, even at high debug level. Do I need to activate that? Regards Volker -- Volker Sauer * Poststrasse 1/601 * 64293 Darmstadt * Germany E-Mail/Jabber: volker(at)volker-sauer.de * http://www.volker-sauer.de PGPKey-Fingerprint: DB26 11C7 B12E 0B27 3999 2E4F 7E35 4E4D 5DD5 D0E0 http://wwwkeys.de.pgp.net/pks/lookup?op=get&search=0x7E354E4D5DD5D0E0 signature.asc Description: Digital signature ___ --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] Jitterbuffer issues
Hi Tony,
please do not send HTML-only messages to mailing lists. There are a lot
of people using mail programs that do not display html. You're html
messages are annoying and that's the reason why you get only a few
answers.
Volker
On Fr, 02 Nov 2007, Tony Plack <[EMAIL PROTECTED]> wrote:
>
> <!--
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> -->
> style="font-family:'Tahoma';font-size:10pt;">When initiating a call from a
> SIP phone to another SIP phone through Asterisk 1.4 (latest SVN), I get the
> following:
>
> [Nov 2
> 10:08:55] WARNING[7292] abstract_jb.c: Failed to put first frame in the
> jitterbuffer on channel 'SIP/5001-08266108'
> [Nov 2
> 10:08:55] WARNING[7292] abstract_jb.c: Failed to put first frame in the
> jitterbuffer on channel 'SIP/5000-08299448'
> [Nov 2
> 10:08:55] WARNING[7292] chan_iax2.c: Resyncing the jb. last_delay 0, this
> delay -309029745, threshold 1000, new offset 309029745
> [Nov 2
> 10:08:55] WARNING[7292] chan_iax2.c: Resyncing the jb. last_delay 0, this
> delay -374875680, threshold 1000, new offset 374875680
>
> First is a
> question about why, when the channel is initially created, does the
> jitterbuffer fail to put in the first frame. Here is the output from
> the console that shows more detail:
>
> --
> Called 5000
> --
> SIP/5000-08299448 is ringing
> Extension
> Changed 5000 new state InUse for Notify User 5002
> --
> SIP/5000-08299448 answered SIP/5001-08266108
> [Nov 2
> 10:08:55] WARNING[7292]: abstract_jb.c:469 create_jb: Failed to put first
> frame in the jitterbuffer on channel 'SIP/5001-08266108'
> --
> adaptive jitterbuffer created on channel SIP/5001-08266108
> [Nov 2
> 10:08:55] WARNING[7292]: abstract_jb.c:469 create_jb: Failed to put first
> frame in the jitterbuffer on channel 'SIP/5000-08299448'
> --
> adaptive jitterbuffer created on channel SIP/5000-08299448
> [Nov 2
> 10:08:55] WARNING[7292]: chan_iax2.c:794 jb_warning_output: Resyncing the jb.
> last_delay 0, this delay -309029745, threshold 1000, new offset
> 309029745
> [Nov 2
> 10:08:55] WARNING[7292]: chan_iax2.c:794 jb_warning_output: Resyncing the jb.
> last_delay 0, this delay -374875680, threshold 1000, new offset
> 374875680
> Extension
> Changed 5001 new state Idle for Notify User 5004
> Extension
> Changed 5001 new state Idle for Notify User 5002
> Extension
> Changed 5001 new state Idle for Notify User 5000
> --
> adaptive jitterbuffer destroyed on channel SIP/5000-08299448
> == Spawn
> extension (macro-stdexten, s, 8) exited non-zero on 'SIP/5001-08266108' in
> macro 'stdexten'
> == Spawn
> extension (macro-stdexten, s, 8) exited non-zero on
> 'SIP/5001-08266108'
> --
> adaptive jitterbuffer destroyed on channel SIP/5001-08266108
>
>
> As can be seen,
> the jitterbuffer is created after the first failure.
>
> Second question,
> why is chan_iax2 involved in a SIP to SIP connection? Is chan_iax2
> required for the new jitterbuffer?
>
>
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Volker Sauer * Poststrasse 1/601 * 64293 Darmstadt * Germany
E-Mail/Jabber: volker(at)volker-sauer.de * http://www.volker-sauer.de
PGPKey-Fingerprint: DB26 11C7 B12E 0B27 3999 2E4F 7E35 4E4D 5DD5 D0E0
http://wwwkeys.de.pgp.net/pks/lookup?op=get&search=0x7E354E4D5DD5D0E0
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