Re: [asterisk-users] How to diable fax header (OR: what module could add that header?)
On 14.10.2015 21:59, Eric Cooper wrote: > On Tue, Oct 13, 2015 at 06:54:51PM +0200, Recursive wrote: >> Some of the software components involved is inserting an unwanted >> header line into the fax. The header is always formatted the same >> way; an example (using a two page fax which I have just sent): >> "13.10.2015 18:22:05" (at the left border of the page) and "001/002" >> (at the right border of the page). > > Are you using ReceiveFax()? If so, are you sure the unwanted > information is already in the TIFF file? Or could it be produced by > whatever program is being used to convert and/or view the TIFF file? > > -- > Eric Cooper e c c @ c m u . e d u > We have tested sending to third party endpoints as well as using ReceiveFAX(). In every case, the header is there, meaning it's in the TIFF file which ReceiveFAX produces. But in the meantime, I have found out where it comes from. We don't use hardware faxes, but a network fax software which is based on CAPI on the server side. So we are using a T38 CAPI emulator on the server, and it has turned that this piece of software is inserting the unwanted header. Nor Asterisk nor our fax software are responsible, but the T38 "driver" - the last place where I had thought of. I had a conversation with the developer of the emulator; he promised that he would change the software and the configuration dialog in a way that users could disable the unwanted header. Regards, Recursive -- _ -- 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] PJSIP Contact User in Dial INVITE
Hi all, This is a followup from my "Getting around semi-colons" question. I've specified: contact_user=01234567\;tgrp=01234567\;trunkcontext=telecom.co.nz under my registration section for the trunk. This is all fine and I successfully got Asterisk 13 + pjsip registered to our BroadWorks-based provider. However, I see the contact_user field only gets used upon registration, and not during an INVITE. During Registration: Contact: During INVITE: Contact: The only questions I have is: 1. Is this expected/known behaviour? eg. pjsip won't use the contact user for anything else? 2. If not, where should I be specifying the contact_user to make pjsip use it in INVITEs? Is this where the AoRs come in? 3. Are those Contact headers used for some internal reference/record keeping for the calls? Or... 4. Can I mess with the Contact Header with the PJSIP_HEADER function? Thanks again for the help, just wrapping my head around this new channel driver :) Juan -- _ -- 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] How to diable fax header (OR: what module could add that header?)
On Tue, Oct 13, 2015 at 06:54:51PM +0200, Recursive wrote: > Some of the software components involved is inserting an unwanted > header line into the fax. The header is always formatted the same > way; an example (using a two page fax which I have just sent): > "13.10.2015 18:22:05" (at the left border of the page) and "001/002" > (at the right border of the page). Are you using ReceiveFax()? If so, are you sure the unwanted information is already in the TIFF file? Or could it be produced by whatever program is being used to convert and/or view the TIFF file? -- Eric Cooper e c c @ c m u . e d u -- _ -- 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] same sip username with realms and chan_sip
Just as a reminder: absolutely anytime that you succeed in crashing Asterisk (no matter the validity of your input), please make sure that either an issue covering the situation already exists, or please take the time to create a new one. When creating an issue (or if one is not already attached), please follow these [1] instructions for obtaining a backtrace and attach the file to the issue. Very often a backtrace on an issue is sufficient for us to identify and eliminate the bug that caused it. And if you can, please replicate using a currently supported version (11, 13, master) of Asterisk compiled from the latest git head -- this helps us to be confident that it's not something already fixed, and we can skip that step and get to fixing it faster. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace On Tue, Oct 13, 2015 at 5:22 AM, Ludovic Gasc wrote: > pjsip crashes only with my realm experiments. > I'll test with the latest Asterisk 13 stable version to verify. > > However, even if I've found a solution for realm, I've the feeling that > realm in Asterisk isn't well tested/supported. > > For now, since September, I use a simpler solution in production: > integrate the account name as a prefix in the username: enough mainstream > to be sure is supported ;-) > > Ludovic Gasc (GMLudo) > http://www.gmludo.eu/ > On 11 Oct 2015 22:22, "Joshua Colp" wrote: > >> Ludovic Gasc wrote: >> >>> Hello, >>> >>> same sip username with realms is possible with Asterisk ? >>> I've tried to have this feature with Asterisk 13.3.2 and chan_pjsip, and >>> now, Asterisk crashes. >>> >> >> Did PJSIP crash in general (it's usually a build problem if that happens) >> or was it when you were experimenting with different realms and such? >> >> -- >> Joshua Colp >> Digium, Inc. | Senior Software Developer >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US >> Check us out at: www.digium.com & www.asterisk.org >> >> -- >> _ >> -- 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 > -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org -- _ -- 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] PJSIP Dialout error
Hi Guys I keep getting this "Warning" when I dial out via pjsip and the calls fail But if I do a pjsip reload it works for 1 minute WARNING[6707]: res_pjsip_outbound_authenticator_digest.c:135 digest_create_request_with_auth_from_old: Unable to create request with auth.No auth credentials for any realms in challenge. Any ideas? -- _ -- 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] Call diversion
Hi all,
i'm trying to setup a function like "secretary/director": when an user
call director number (eg. 5000), the call were firstly diverted to
secretary (5001). At this point, when secretary answer, can decide to
transfer back the call to director (5000).
Because i'm using OpenSIPS as SIP router, when this function is needed i
did a special "extension" like DIVERT_[from]_[to]:
; On DIVERT
exten => _DIVERT_.,1,Noop("from-voip: DIVERT ${CALLERID(num)} -
${EXTEN}") ; Example: DIVERT_5000_5001
exten => _DIVERT_.,n,Set(DIVFrom=${CUT(EXTEN,_,2)})
exten => _DIVERT_.,n,Set(DIVTo=${CUT(EXTEN,_,3)})
exten => _DIVERT_.,n,Noop("Divert ${CALLERID(num)} from ${DIVFrom} to
${DIVTo}")
exten => _DIVERT_.,n,Macro(services-divert) // Divert
exten => _DIVERT_.,n,Hangup()
and a macro like:
[macro-services-divert] ; Servizio direttore-segretaria
exten => s,1,Noop("Divert from ${CALLERID(num)}")
same => n,Answer
same => n,Playback(msg/msg_attendereufficiodesiderato)
same => n,Dial(SIP/voip-trunk/${DIVTo},30)
It works *BUT* when from the secretary phone the call were forwarded
back to director (and, at this point, director phone must ring !), the
call stall with an error on OpenSIPS like:
In-Dialog NOTIFY from [my asterisk box ip]
(callid=707902a659eb005b141a9491195bd3cf@voip) is not valid according to
dialog
Any hits ?
Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - central...@unisi.it
Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di
Ateneo, http://www.faq.unisi.it
signature.asc
Description: OpenPGP digital signature
--
_
-- 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
