Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
On 24/10/2014 12:47 AM, Tim Nelson wrote: - Original Message - On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? The originating endpoint is an IAXmodem controlled by Hylafax. Actual call flow is IAXmodem --G.711u via localhost-- Asterisk (old version with no T.38 support) --G.711u-- Asterisk 11.x --G.711u/T.38-- ITSP The problem lies on the Asterisk 11.x system not being able to reinvite to T.38 on the call leg with the ITSP, and given the ITSP does not do this either, the call is stuck in G.711u with varying performance. :/ --Tim IAXmodem (other host on network) - Asterisk 1.2 (IAX) - Asterisk 1.8 with Fax Gateway Patch - SIP provider - PSTN Fax destination I have successfully sent a fax using a full page image via an Asterisk 1.2 system which forwards the request to my Asterisk 1.8 over an IAX channel, Asterisk 1.8 has the T.38 Fax Gateway patch installed. The outbound call triggered the T.38 gateway and the fax was received without error. I have ECM disabled in my IAX modem configuration in Hylafax. I don't have Asterisk 11 running to test with at this time however I confirmed the T.38 Gateway functions in Asterisk 11 when testing it. -- Accepting AUTHENTICATED call from 192.168.54.18: requested format = ulaw, requested prefs = (ulaw|alaw|slin), actual format = alaw, host prefs = (alaw|ulaw), priority = mine -- Executing [PSTN Number@FAX-T30:1] Dial(IAX2/faxgw-iax-1210, SIP/PSTN Number@itsp-fax,55) in new stack == Using SIP RTP TOS bits 184 -- Called SIP/PSTN Number@itsp-fax -- SIP/itsp-fax-000b is making progress passing it to IAX2/faxgw-iax-1210 -- SIP/itsp-fax-000b is making progress passing it to IAX2/faxgw-iax-1210 == Using SIP RTP TOS bits 184 -- SIP/itsp-fax-000b answered IAX2/faxgw-iax-1210 [Oct 25 23:24:11] NOTICE[27896]: channel.c:4220 __ast_read: Dropping incompatible voice frame on IAX2/faxgw-iax-1210 of format slin since our native format has changed to 0x8 (alaw) -- Got Fax Tone CED Chan SIP/itsp-fax-000b [1] Sending T.38 Params Peer Is IAX2/faxgw-iax-1210 [0] -- Request on IAX2/faxgw-iax-1210 [0] Storing I: SIP/itsp-fax-000b [1] == Using UDPTL TOS bits 184 -- Negotiated on SIP/itsp-fax-000b [4] Ignoring I: IAX2/faxgw-iax-1210 [0] -- T.38 Gateway starting for chan SIP/itsp-fax-000b and peer IAX2/faxgw-iax-1210 pbx*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format FirstMsgLastMsg IAX2/faxgw-iax-1210 192.168.54.18faxgw-iax 01210/4 00010/5 0ms -0001ms ms alawRx:NEW Tx:ACK 1 active IAX channel pbx*CLI fax show sessions Current FAX Sessions: Channel Tech FAXID Type Operation State File(s) SIP/itsp-fax-000 Spandsp1 T.38 receiveActive (null) 1 FAX sessions -- Executing [h@FAX-T30:1] GotoIf(IAX2/faxgw-iax-1210, 0?2:3) in new stack -- Goto (FAX-T30,h,3) -- Executing [h@FAX-T30:3] NoOp(IAX2/faxgw-iax-1210, Finish if_FAX-T30_37) in new stack -- Executing [h@FAX-T30:4] NoOp(IAX2/faxgw-iax-1210, Call/Fax Ended 2014-10-25 23:27:38 +0800) in new stack -- Connection Statistics Bit Rate :14400 ECM : No Pages : 1 == Spawn extension (FAX-T30, PSTN Number, 1) exited non-zero on 'IAX2/faxgw-iax-1210' -- Hungup 'IAX2/faxgw-iax-1210' -- _ -- 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] Questions on musiconhold.conf custom mode
Am 25.10.2014 00:09, schrieb Olivier: Hello, I need to play some musiconhold content starting at a random duration from the start. Thanks to mode=custom option and either madplay or mpg123 programs, I could successfully get what I was after on a Debian Wheezy system. Now I realized sox version on my target system (Debian Squeeze) cannot convert to MP3 format. So I'm looking after workarounds. 0. I've read many mpg123 or madplay examples. All of them are clutered with option converting MP3 input file into an appropriate format that Asterisk requires for music on hold. What is the name of this appropriate format ? sln ? wav ? 1. Is there a player like mpg123, that can repeat content in appropriate format (see above) to stdout but can read from anything different from MP3 ? 2. Is there an option on Squeeze to convert audio files to MP3 (reverse coversion works OK). 3. Which options could I have for such custom MOH, if I was building on system without g729 transaltion capabilites ans with g729-only SIP trunks or phones ? Is the gsm-format an option for you? So you may convert your moh-File to gsm: sox YouWavFile.wav -r 8000 -c1 MohFile.gsm If you really need mp3 you have to compile sox with mp3-support by yourself OR maybe this is a solution on Debian: http://www.howtoinstall.co/en/debian/wheezy/main/libsox-fmt-mp3/ -Thorsten- -- _ -- 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] Voicemail ODBC Storage
Hi, Is there any reason why ODBC voicemail storage requires varchar for most fields? For example, is there anything stopping me using a BIGINT or similar for origtime or INT for duration? Kind regards, Dan -- _ -- 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] make asterisk do something when an outgoing call is picked up
Hi,
how can I make asterisk do something when an outgoing call is picked up?
The background is that I would like to record incoming and outgoing
phone calls. In order to do this, I need to play an announcement
telling the person calling or being called that the call will be
recorded.
Here's what I'm trying to do:
call comes in:
if(I pick up) {
play announcement to caller;
start recording;
let me talk to the caller;
end recording when call ends;
send recording to my email account;
} else {
record voice mail;
}
call goes out:
if(call is picked up) {
play announcement to callee;
if(callee hangs up) {
end call;
} else {
start recording;
let me talk to callee;
end recording when call ends;
send recording to my email account;
}
} else {
call ends;
offer me to automatically call again later;
}
Please keep in mind that I'm new to asterisk and just got it to work.
Searching for having asterisk do something when an outgoing call is
picked up has been unsuccessful other than that I found out that you can
have it make outgoing calls automatically to play pre-recorded messages:
So asterisk does have a way to detect when a call is picked up and a way
of doing something when that happens.
What I have working so far is incoming and outgoing calls and voicemail
for one phone/user, which is a basic set up I'm trying extend and
improve now.
--
Again we must be afraid of speaking of daemons for fear that daemons
might swallow us. Finally, this fear has become reasonable.
--
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[asterisk-users] Asterisk 13.0.0 Now Available!
The Asterisk Development Team is pleased to announce the release of Asterisk 13.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases Asterisk 13 is the next major release series of Asterisk. It is a Long Term Support (LTS) release, similar to Asterisk 11. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions For important information regarding upgrading to Asterisk 13, please see the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13 A short list of new features includes: * Asterisk security events are now provided via AMI, allowing end users to monitor their Asterisk system in real time for security related issues. * Both AMI and ARI now allow external systems to control the state of a mailbox. Using AMI actions or ARI resources, external systems can programmatically trigger Message Waiting Indicators (MWI) on subscribed phones. This is of particular use to those who want to build their own VoiceMail application using ARI. * ARI now supports the reception/transmission of out of call text messages using any supported channel driver/protocol stack through ARI. Users receive out of call text messages as JSON events over the ARI websocket connection, and can send out of call text messages using HTTP requests. * The PJSIP stack now supports RFC 4662 Resource Lists, allowing Asterisk to act as a Resource List Server. This includes defining lists of presence state, mailbox state, or lists of presence state/mailbox state; managing subscriptions to lists; and batched delivery of NOTIFY requests to subscribers. * The PJSIP stack can now be used as a means of distributing device state or mailbox state via PUBLISH requests to other Asterisk instances. This is analogous to Asterisk's clustering support using XMPP or Corosync; unlike existing clustering mechanisms, using the PJSIP stack to perform the distribution of state does not rely on another daemon or server to perform the work. And much more! More information about the new features can be found on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Documentation A full list of all new features can also be found in the CHANGES file: http://svnview.digium.com/svn/asterisk/branches/13/CHANGES For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0 Thank you for your continued support of Asterisk! -- _ -- 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] Voicemail ODBC Storage
On Sat, Oct 25, 2014 at 4:09 PM, Dan Journo d...@keshercommunications.com wrote: Hi, Is there any reason why ODBC voicemail storage requires varchar for most fields? For example, is there anything stopping me using a BIGINT or similar for origtime or INT for duration? Yes. app_voicemail uses a message envelope file to hold the metadata regarding the voice mail. When the ODBC retrieve function pulls the database records, it writes that data out to a temporary message envelope file for playback/manipulation by other functions. This process does not examine the column types, but instead simply looks at the column names and writes the data values out to the file using the types that it expects each column name to have. So, changing those types will not work out well for you. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
