search for: fax_detected

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2004 May 01
1
Fax Detect problem (have consulted archives, wiki & irc)
Hi All, I'm using an X100P to connect to PSTN ( context=from-pstn ). I'm trying to get fax detection to work. Using the simplest dialplan, I cannot get * to detect fax tones: [from-pstn] exten => s,1,Answer exten => fax,1,Goto(ext-fax,999,1) The fax is never detected (ie: Goto never executed) All that I see on the * -vvvvc console is: -- Executing Answer("Zap/1-1",
2023 Apr 28
1
Asterisk translates 200 OK + SDP into 488 not acceptable here after both side agreed on codec.
Hi List Asterisk 16.28.0 in use. PJSIP in use Two endpoints Both using IPv6 One Endpoint on UDP, the other via TLS. Both with: t38_udptl=yes ;fax_detect=yes ;fax_detect_timeout=30 rtp_ipv6=yes Both sides are T.38 capable and detect fax tone so no need for fax detection on asterisk. Voice calls between the two work fine. But on a Fax call, I see this situation: A <=> Asterisk
2017 Jun 05
3
asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'
On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: >>> Just a guess (without knowing about your network), but are the two ends >>> points on public networks and visible to one another? If not the reinvite >>> may be passing an internal (nat'ed)
2004 Jul 05
2
fax detection and X100P
Hi everybody I am having problem detecting fax with my X100P. I have RedHat 8 as OS and an X100P and a TDM400P. The X100P being plugged into PSTN. I have successfully installed tiff-v3.5.7 and spandsp-0.0.1 and also patched Asterisk wthout problem. Here is my zapata.conf file context=cda signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5
2006 May 16
1
tdm2400p: fax detection not working
...e: _zapata.conf_: context = inbound faxdetect = incoming language = it musiconhold = default signalling = fxs_ks callerid = "call_or_fax" channel => 1 extensions.conf: [inbound] exten => s,1,Answer exten => s,2,Playback(something) exten => s,3,HangUp exten => fax,1,NoOp(fax_detected) exten => fax,2,HangUp but asterisk always play the 'something' sound ignoring incoming faxes. Any ideas? TIA Giorgio Incantalupo
2016 Jul 01
2
CALLERID on pjsip doesn't work?
Asterisk 13.8 Is CALLERID(all) supposed to wok for pjsip? When I do this: exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) same => n,Dial(PJSIP/phone123, 30) I expect the callerid to be as set, but is always seems to be "phone123", the name of the endpoint. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL:
2015 Jul 29
2
PJSIP T.38 issues
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Thanks for your reply Larry. Le 27/07/2015 01:22, Larry Moore a ?crit : > I think the "488 Not acceptable here" is occurring because the channel > connecting through is not T.38 capable, that will be the IAX channel > from iaxmomdem. This is what T38gateway is supposed to do. And I'm very happy to report that after one more
2019 Apr 22
2
Incoming SIP call, outgoing SIP registration. PJSIP.
Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple
2016 Jul 04
2
CALLERID on pjsip doesn't work?
On 1 July 2016 at 17:41, Joshua Colp <jcolp at digium.com> wrote: > > >> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) >> same => n,Dial(PJSIP/phone123, 30) >> > > Your exten line has no priority, is that how it is in your dialplan? > Actually no, I stole that line from an earlier email to this list. Mine has a priority.
2006 Jun 09
3
FXO registration and VegaStream
I am trying to configure a VegaStream 50 FXO to work with asterisk. The problem that I am having is that the VegaStream does not support incoming registration from asterisk. VegaStream only allows outbound registration. My question is does asterisk allow incoming registration from an FXO? If yes how? Or better yet, has anybody been able to make the VegaStream FXO work with asterisk? According
2006 Mar 30
3
asterisk doesn't wait for whole extension
Hi, maybe a dumb question, but it seems that some calls are directed to our central dial in number despite the extensions the callers say they dialled. E.g. they dial 1234-567, asterisk recognizes 12345, it says this is an unknown extension, where it is right, and redirects the call to the central dial in extension 1234-0. This only seems to happen when the numbers are dialled manually. When