[asterisk-users] Disable fax detect on specific incoming DID
Hello, our gateway receive incoming calls from an outside gateway for multiple DIDs. For some of them we want fax detection, for other no. To do so, faxdetect is set to yes, but how to disable the fax detection for a specific incoming DID? For those DIDs, we just want to forward the call to a real fax machine DID which will do the job. Thanks for any hint Regards -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] agitator - FastAGI reverse proxy
Hello, FastAGI doesn't get (or deserve) much love these days but a lot of people are still widely using it. Here is a small project of mine, started trying to scratch my own itch, that some might find useful. https://github.com/zaf/agitator It is a reverse proxy for the FastAGI protocol with some interesting features. Most noticeably request based routing, HA/failover, load balancing and TLS encryption for FastAGI sessions (something asterisk unfortunately is still lacking). It is written in Go and it is quite fast and light on resources. First public release, feedback and patches are more than welcome ;) Regards, Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disable fax detect on specific incoming DID
The easiest way is to just run the Dial() command to forward the call to the hard fax without ever Answer()-ing the call. Without an Answer() on the call, Asterisk can't listen for fax detection (because the call hasn't been set up and there is no audio leg yet). Thank you, Noah Engelberth -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Friday, January 16, 2015 5:59 AM To: Asterisk Users Subject: [asterisk-users] Disable fax detect on specific incoming DID Hello, our gateway receive incoming calls from an outside gateway for multiple DIDs. For some of them we want fax detection, for other no. To do so, faxdetect is set to yes, but how to disable the fax detection for a specific incoming DID? For those DIDs, we just want to forward the call to a real fax machine DID which will do the job. Thanks for any hint Regards -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_genconf fails with Empty configuration - no spans
On Thu, Jan 15, 2015 at 12:58:26PM -0600, Russ Meyerriecks wrote: On Thu, Jan 15, 2015 at 2:05 AM, Bertrand LUPART - Linkeo.com bertrand.lup...@linkeo.com wrote: However, dahdi_genconf keeps finding no span: What am i missing? It looks like your driver is loaded correctly. My guess would be maybe the dahdi-tools is packaged as an older version that doesn't know about the newer te435 card. You could hand craft the config file using the info in the card's manual http://www.digium.com/sites/digium/files/quad-span-digital-card-user-manual.pdf The error is no spans. What is the output of: dahdi_span_assignment list -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Asterisk pjsip auto dtmf mode
Hello Asterisk Users,
I have been looking for similar auto dtmf mode implementation on pjsip, but
didn't see it coming, so I decided to give it a try.
My basic plan was to do it as simple as possible with minimum changes
because I am not familiar with all Asterisk code. My idea is to use rfc4733
settings, but when going over the codecs check if telephone-event appear
and if not set the dtmf mode to inband on rtp instance.
I would appreciate if someone would look at what I did and see if I didn't
do stupid things. If you think this is something you would like to add to
one of the next releases I am willing to help - add the additional dtmf
mode ...
I based my development on 13.1.0. The following are my changes:
In res/res_pjsip_sdp_rtp.c (I added session_media to get_codecs and used it
in order to update dtmf settings on rtp instance when telephone-event is
not included in the sdp):
150:
static void get_codecs(struct ast_sip_session *session, const struct
pjmedia_sdp_media *stream, struct ast_rtp_codecs *codecs, struct
ast_sip_session_media *session_media)
159:
char fmt_param[256];
int tel_event = 0;
177:
ast_copy_pj_str(name, rtpmap-enc_name, sizeof(name));
if (strcmp(name,telephone-event) == 0) {
tel_event++;
}
202:
}
if (tel_event==0) {
ast_rtp_instance_dtmf_mode_set(session_media-rtp,
AST_RTP_DTMF_MODE_INBAND);
}
/* Get the packetization, if it exists */
241:
get_codecs(session, stream, codecs, session_media);
In res/res_pjsip_session.c (Just activated DSP also on RFC dtmf mode - I
didn't find a way to test the rtp instance dtmf settiings because
session_media pointer is not there. Any advice for doing so would be
appreciated):
1062:
if (endpoint-dtmf == AST_SIP_DTMF_INBAND || endpoint-dtmf ==
AST_SIP_DTMF_RFC_4733) {
dsp_features |= DSP_FEATURE_DIGIT_DETECT;
}
In channels/chan_pjsip.c (1 change similar to the above, and 2 more changes
to send inband dtmf when rtp instance dtmf settings is inband)
543:
if (session-endpoint-dtmf == AST_SIP_DTMF_INBAND ||
session-endpoint-dtmf == AST_SIP_DTMF_RFC_4733) {
ast_dsp_set_features(session-dsp,
DSP_FEATURE_DIGIT_DETECT);
1420:
if (!media || !media-rtp ||
(ast_rtp_instance_dtmf_mode_get(media-rtp) == AST_RTP_DTMF_MODE_INBAND)) {
return -1;
1523:
if (!media || !media-rtp ||
(ast_rtp_instance_dtmf_mode_get(media-rtp) == AST_RTP_DTMF_MODE_INBAND)) {
return -1;
That's it!!! It works fine for me. Any remarks / advice would be
appreciated.
Yaron.
--
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