Re: [asterisk-users] Anyone doing speech to text?
Lefteris, Thanks a lot for your detailed answer and for the valuable work you've been doing on this topic for quite some time now. Cheers, Philippe Sultan 2015-08-28 12:26 GMT+02:00 Lefteris Zafiris zaf@gmail.com: On Fri, 28 Aug 2015 12:11:14 +0300 Amelye Chatila amec...@gmail.com wrote: I have a similar situation here, I want to include TTS in my asterisk IVR system. Could someone give suggestion(s) please, I prefer open-source thanks in advance! Hello, what follows is a mostly incomplete list of Text To Speech (TTS) and Speech To Text (STT) solutions available for asterisk. -Regarding the TTS free and open source available options: Asterisk comes with festival (http://www.cstr.ed.ac.uk/projects/festival/) support (app_festival) already build in. Decent quality, supports mainly English. There is support for flite (http://www.festvox.org/flite/) available as a 3rd party plugin : http://zaf.github.io/Asterisk-Flite/ Quality at par with festival, much easier to setup and use, supports only English. Also support for espeak (http://espeak.sourceforge.net/): http://zaf.github.io/Asterisk-eSpeak/ Average quality, supports a wide range of languages. -Free plugins/scripts that provide TTS from a remote not-so-free service: GoogleTTS : http://zaf.github.io/asterisk-googletts/ Great quality, lots of languages, free of charge but NOT suitable for any serious/commercial use. It is not a service Google officially provides but just a hack that gets synthesized speech data from their translate page. It's more suitable for testing/developing and home use. MsTTS: http://zaf.github.io/asterisk-mstts/ using Microsoft's Translator voice synthesis engine. iSpeech: http://zaf.github.io/asterisk-ispeech/ using iSpeech API ( http://www.ispeech.org) -Other non free solutions: Cepstral: http://www.cepstral.com/en/telephony/asterisk Speech Technology Group: http://www.asteriskexchange.com/listings/1001 -Regarding the STT options: Google Speech: http://zaf.github.io/asterisk-speech-recog/ the API is limited at the moment in something like 50 requests/day and considered a technology preview. iSpeech: http://zaf.github.io/asterisk-ispeech/ Lumevox: http://www.lumenvox.com/partners/digium/asterisk.aspx Sphinx: http://cmusphinx.sourceforge.net/wiki/asteriskdetails Vestec: http://www.asteriskexchange.com/listings/113 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 -- _ -- 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] libpri : Q.931 Called Party Number interpreted as empty
Hi, I'm running an Asterisk server connected to a carrier over 2 E1 cards. From time to time, the Called Number Party presented by the carrier changes a bit (for some reason I don't know) and is prefixed with a byte string (e.g. : 00 34 34 39 ), which furtherly prevents libpri from getting the Called Number properly. I've managed to catch the log from libpri when that happened today. In this excerpt, 951693203 is calling 90020361589425. It shows that the Called Party Number IE actually contains the number, prefixed with a NULL byte, then 449. PRI Span: 1 PRI Span: 1 Protocol Discriminator: Q.931 (8) len=71 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 66/0x42) (Sent from originator) PRI Span: 1 Message Type: SETUP (5) PRI Span: 1 [a1] PRI Span: 1 Sending Complete (len= 1) PRI Span: 1 [04 03 90 90 a3] PRI Span: 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) PRI Span: 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) PRI Span: 1 User information layer 1: A-Law (35) PRI Span: 1 [18 03 a9 83 8c] PRI Span: 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0 PRI Span: 1ChanSel: As indicated in following octets PRI Span: 1Ext: 1 Coding: 0 Number Specified Channel Type: 3 PRI Span: 1Ext: 1 Channel: 12 Type: CPE] PRI Span: 1 [1e 02 84 83] PRI Span: 1 Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) PRI Span: 1Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] PRI Span: 1 [28 0f 53 75 6c 74 61 6e 20 50 68 69 6c 69 70 70 65] PRI Span: 1 Display (len=15) [ Sultan Philippe ] PRI Span: 1 [6c 0b 21 83 39 35 31 36 39 33 32 30 33] PRI Span: 1 Calling Number (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) PRI Span: 1Presentation: Presentation allowed of network provided number (3) '951693203' ] PRI Span: 1 [70 13 a1 00 34 34 39 39 30 30 32 30 33 36 31 35 38 39 34 32 35] PRI Span: 1 Called Number (len=21) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '' ] PRI Span: 1 -- Making new call for cref 66 PRI Span: 1 Received message for call 0xb755f2f0 on 0x9bc3c10 TEI/SAPI 0/0, call-pri is 0x9bc3c10 TEI/SAPI 0/0 PRI Span: 1 -- Processing Q.931 Call Setup PRI Span: 1 -- Processing IE 161 (cs0, Sending Complete) PRI Span: 1 -- Processing IE 4 (cs0, Bearer Capability) PRI Span: 1 -- Processing IE 24 (cs0, Channel Identification) PRI Span: 1 -- Processing IE 30 (cs0, Progress Indicator) PRI Span: 1 -- Processing IE 40 (cs0, Display) PRI Span: 1 -- Processing IE 108 (cs0, Calling Party Number) PRI Span: 1 -- Processing IE 112 (cs0, Called Party Number) PRI Span: 1 q931.c:6871 post_handle_q931_message: Call 66 enters state 6 (Call Present). Hold state: Idle Span: 1 Processing event: PRI_EVENT_RING -- Going to extension s|1 because of Complete received -- Span 1: Extension s@from-pstn does not exist. Rejecting call from '951693203'. I believe the NULL byte at the beginning of the Called Party Number IE sent by the carrier causes the problem. I'd like to know if someone had a hack to solve that issue. More information regarding my setup : 2 E1 cards : Digium Wildcard TE205P (5th Gen) Asterisk version : Asterisk 1.8.6.0 built by root @ asterisk-t2 on a i686 running Linux on 2011-09-15 13:26:08 UTC libpri version: 1.4.11.4 DAHDI Version: 2.3.0.1 Echo Canceller: OSLEC Thanks for your help! Philippe -- _ -- 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] libpri : Q.931 Called Party Number interpreted as empty
Giovanni, Thanks a lot for clearing this up. The 's' extension would match any number, and I would not be able to retrieve the actually dialed number from within the dialplan, unless I'm missing something. I'll file a ticket to solve that issue. Thanks again, Philippe On Wed, Nov 2, 2011 at 5:29 PM, giovanni.v i...@keybits.org wrote: Il 02/11/2011 15.06, Philippe Sultan ha scritto: PRI Span: 1 [70 13 a1 00 34 34 39 39 30 30 32 30 33 36 31 35 38 39 34 32 35] Yes, like you guessed the third bit (wich is part of the called number i.e.) is a NUL... but Q.931 allows any IA5 (ISO 646) character so it's a bug in libpri not in your telco side. 70|0111 I-Element: Called party number 13|00010011 Length = 19 a1|0--- Extension Bit = with extension |-001 Type of number: international number |0001 Numbering Plan: ISDN/telephony 00|0--- Spare |-000 Number digits: NUL 34|0--- Spare |-0110100 Number digits: 4 34|0--- Spare |-0110100 Number digits: 4 39|0--- Spare |-0111001 Number digits: 9 39|0--- Spare |-0111001 Number digits: 9 30|0--- Spare |-011 Number digits: 0 [...] a hack to solve that issue. Why not an 's' in your incoming context to workaround the issue? -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Philippe Sultan -- _ -- 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] libpri : Q.931 Called Party Number interpreted as empty
Issue filed : https://issues.asterisk.org/jira/browse/PRI-128 Philippe On Wed, Nov 2, 2011 at 7:00 PM, giovanni.v i...@keybits.org wrote: On 02/11/2011 17.52, Philippe Sultan wrote: The 's' extension would match any number, and I would not be able to retrieve the actually dialed number from within the dialplan, unless I'm missing something. No, obviously you missed none... but I prefer to route an incoming call to a generic destination instead of missing it. ;-) I'll file a ticket to solve that issue. Thanks, please post a pointer to the issue after that done. -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Philippe Sultan -- _ -- 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] ${HASH(SIP_CAUSE, ...)} and peer name
The destination channel dies right after your Dial statement exits,
but you can retrieve the info in the channel that's still alive :
exten = _XXX,n,Dial(SIP/${EXTEN})
exten = _XXX,n,NoOp(SIP return code : ${HASH(SIP_CAUSE,${CDR(dstchannel)})})
Works fine on the Asterisk server I'm running (1.8.3.3).
Philippe
On Mon, Jul 11, 2011 at 11:01 PM, ik ido...@gmail.com wrote:
Hello,
I'm trying to figure out what was the return code of SIP for a call.
The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to
retrieve the peer name using ${CHANNEL(peername)}, I have an error message
that CHANNEL does not have peername or it is not available to be used.
I tried to print it with NOOP on a live channel, and also after hangup, both
with the same error message.
So how can I get SIP_CAUSE, or how can I get the peer name ?
Thanks,
Ido
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Re: [asterisk-users] conferencing without DAHDI
Hi Klaus, The module is app_confbridge, and the application is ConfBridge. I had been using it for a while because it's really easy to use : you don't need any configuration file, and you get cool announcements upon conference events from a playback channel. The options work pretty much like meetme, although I would have liked to have a 'x' option to close the conference when the last marked user leaves. Moreover, I couldn't have the playback channel speak French, from what I've read in the source code, I think that feature would require a configuration file because the playback channel is not a per user option. Philippe On Mon, Feb 8, 2010 at 12:56 PM, Olle E. Johansson o...@edvina.net wrote: 8 feb 2010 kl. 12.29 skrev Klaus Darilion: Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. In Asterisk trunk there's a new conference bridge module you can test. There are also some third-party modules out there, like app_conference. /O -- _ -- 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 -- Philippe Sultan -- _ -- 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] conferencing without DAHDI
And by the way, app_confbridge is included in the 1.6.2 series (at least). On Mon, Feb 8, 2010 at 1:49 PM, Philippe Sultan philippe.sul...@gmail.com wrote: Hi Klaus, The module is app_confbridge, and the application is ConfBridge. I had been using it for a while because it's really easy to use : you don't need any configuration file, and you get cool announcements upon conference events from a playback channel. The options work pretty much like meetme, although I would have liked to have a 'x' option to close the conference when the last marked user leaves. Moreover, I couldn't have the playback channel speak French, from what I've read in the source code, I think that feature would require a configuration file because the playback channel is not a per user option. Philippe On Mon, Feb 8, 2010 at 12:56 PM, Olle E. Johansson o...@edvina.net wrote: 8 feb 2010 kl. 12.29 skrev Klaus Darilion: Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. In Asterisk trunk there's a new conference bridge module you can test. There are also some third-party modules out there, like app_conference. /O -- _ -- 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 -- Philippe Sultan -- Philippe Sultan -- _ -- 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] conferencing without DAHDI
Philippe, what exactly is a playback channel? Is it a pseudo participant playing back the announcements? Yes. Announcements are played to the conference members by creating a channel of type 'Bridge' which streams the sound files. thanks klaus Further, is there somewhere a documentation Well, there is no sample configuration in the tarball because ConfBridge does require any configuration file. 'core show application ConfBridge' in the CLI will give you the options list. You'd probably also want to take a look at the app_confbridge.c file. Very short and readable for such a powerful app. Philippe -- _ -- 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] Asterisk and XMPP Jingle : testers needed
Dear community members,
I'm happy to announce that we now have code that allows you to use
your XMPP (Jabber) client like a softphone to place SIP or PSTN (or
whatever channel Asterisk supports) calls.
The XMPP clients that support Jingle that I and others have tested are :
- Pidgin (Linux, Ubuntu 9.10), version 2.6.2 : OK
- Empathy (Linux, Ubuntu 9.10), version 2.28.1.1 : OK
- Psi (Windows XP), version 0.13 : Call establishes, but no sound
(seems to be a problem with Speex)
For the moment, one can only place calls from the XMPP client to
Asterisk, but soon, you'll be able to receive calls on your XMPP
client too.
Please test the following branch :
http://svn.digium.com/svn/asterisk/team/phsultan/jingle-support
Or visit this ticket : https://issues.asterisk.org/view.php?id=15634
The doc/jabber.txt in the code contains code snippets and
configuration examples. Hereafter is an example of how to place a call
to an Asterisk server through the Jingle channel. The user places a
Jingle call to Asterisk from his XMPP client's UI, which triggers a
chat message being sent back to him, asking him to enter a number to
call. And that's it, Asterisk just relays the call to the configured
destination (here, a registered SIP phone).
context jingle-in {
s = {
Answer();
SendText(Please enter the number you wish to call);
Set(NEWEXTEN=${JABBER_RECEIVE(asterisk-xmpp,${CALLERID(name)})});
SendText(Calling ${NEWEXTEN} ...);
Dial(SIP/${NEWEXTEN);
Hangup();
}
}
Thanks,
Philippe
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Re: [asterisk-users] Asterisk Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error
Or you can disable the digest-md5 authentication mechanism on OpenFire. I remember an old related bug : https://issues.asterisk.org/view.php?id=11644 On Mon, Jul 6, 2009 at 8:55 PM, Julian Lyndon-Smithaster...@dotr.com wrote: usetls=no Julian jonas kellens wrote: On Mon, 2009-07-06 at 16:18 +0100, Julian Lyndon-Smith wrote: I can assure you that it works, and that it works well. We use it ;) My jabber.conf : [general] debug=yes ;;Turn on debugging by default. autoprune=no ;;Auto remove users from buddy list. autoregister=yes ;;Auto register users from buddy list. [asterisk] ;;label type=client ;;Client or Component connection serverhost=192.168.2.5 ;;Route to server for example talk.google.com username=aster...@192.168.2.5 ;;Username with optional roster. secret=XX ;;Password port=5222 ;;Port to use defaults to 5222 usetls=yes ;;Use tls or not usesasl=yes ;;Use sasl or not statusmessage=I am Asterisk ;;Have custom status message for Asterisk. ;timeout=100 ;;Timeout on the message stack. Then I get the following : [Jul 6 20:07:57] JABBER: asterisk INCOMING: ?xml version='1.0' encoding='UTF-8'?stream:stream xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:client from=openfire.jocan.local id=56ff9859 xml:lang=en version=1.0stream:featuresmechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismDIGEST-MD5/mechanismmechanismPLAIN/mechanismmechanismANONYMOUS/mechanismmechanismCRAM-MD5/mechanism/mechanismscompression xmlns=http://jabber.org/features/compress;methodzlib/method/compressionauth xmlns=http://jabber.org/features/iq-auth/register xmlns=http://jabber.org/features/iq-register//stream:features [Jul 6 20:07:57] JABBER: asterisk OUTGOING: auth xmlns='urn:ietf:params:xml:ns:xmpp-sasl' mechanism='DIGEST-MD5'/ [Jul 6 20:07:57] JABBER: asterisk INCOMING: challenge xmlns=urn:ietf:params:xml:ns:xmpp-saslcmVhbG09Im9wZW5maXJlLmpvY2FuLmxvY2FsIixub25jZT0iSngyRVZCRmlDNlI4K1hlMU5rbm9PUUNWT1VEN1pGMEpXcnRydUxjdiIscW9wPSJhdXRoIixjaGFyc2V0PXV0Zi04LGFsZ29yaXRobT1tZDUtc2Vzcw==/challenge [Jul 6 20:07:57] JABBER: asterisk OUTGOING: response xmlns='urn:ietf:params:xml:ns:xmpp-sasl'dXNlcm5hbWU9ImFzdGVyaXNrIixyZWFsbT0ib3BlbmZpcmUuam9jYW4ubG9jYWwiLG5vbmNlPSJKeDJFVkJGaUM2UjgrWGUxTmtub09RQ1ZPVUQ3WkYwSldydHJ1TGN2Iixjbm9uY2U9IjQzZTVmYjFkNjZiMTU2OGI1MDFjNzk0ZDQ0MzMyYzFiIixuYz0wMDAwMDAwMSxxb3A9YXV0aCxkaWdlc3QtdXJpPSJ4bXBwLzE5Mi4xNjguMi41IixyZXNwb25zZT1kNGUxYzQ0ZDM0OGNjNWJkN2E2MzJiNzdmZjRjZTQ0OCxjaGFyc2V0PXV0Zi04/response [Jul 6 20:07:57] JABBER: asterisk INCOMING: failure xmlns=urn:ietf:params:xml:ns:xmpp-saslnot-authorized//failure [Jul 6 20:07:57] ERROR[24565]: res_jabber.c:606 aji_act_hook: JABBER: encryption failure. possible bad password. I am 100% sure I have the correct password ! I even took a very simple password without any special characters... Can you advise ?? Jonas. ___ -- 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 -- Philippe Sultan ___ -- 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] Qualify sip users behind remote registrar
Hi everybody, From an Asterisk console, I'd like to retrieve information from SIP users (eg. their contact address) that are registered on a Kamailio (OpenSER) server. Kamailio is defined as a peer in my sip.conf file, and it looks like the 'sip qualify peer' command can help me get the information I need. However, this command applies only to SIP peers explicitly defined in sip.conf, but my Asterisk server is not aware of the registrations in Kamailio. Basically, I'd like to have 'sip qualify peer usern...@peer', where only 'peer' is defined in sip.conf and where 'username' is unknown by Asterisk. Is there any trick, or another command I could use to achieve this? Philippe ___ -- 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] What's the difference between the Jabber Client Mode And Component Mode?
Hi Tony, There are quite a few differences between the two modes. 1) connections Component connections use a different TCP port than the regular ports (5222 or 5223) use for client connections. For example, the default port on jabberd2 for component connections is 5347. Component connections cannot be encrypted with TLS, and rely on a specific authentication mechanism. 2) XMPP packets handling Client connections allow you to receive XMPP packets (presence/iq/message) from buddies you explicitly allowed to, and send XMPP packets to buddies that explicitly allowed you to do so. Buddies are identified with a Jabber ID (JID). Basically, the authorization process is ruled by a mutual subscription mechanism. Component connections on the other hand allow you to do more. As an example, suppose you connect your Asterisk server as a component identified by 'asterisk'. Then, any XMPP packet sent to JIDs like 'u...@asterisk' will be routed to your Asterisk server for further processing. Also, your Asterisk server can send XMPP packets from any JID like 'u...@asterisk'. Connecting as a component is basically the same as connecting a new domain to an XMPP server. At the institute I work for, we use Asterisk as a component for groupchat/meetme connections. Asterisk informs the users connected to a given groupchat that someone (identified with a phone extension like 1...@asterisk) has entered/left a meetme conference. Coupled with a Web interface, you get a conferencing service that brings chat and audio functions together. Philippe On Tue, Feb 3, 2009 at 10:59 AM, tony luo tony.luo0...@gmail.com wrote: Hi All, I am doing some research on the intergration of Jabber and Asterisk. I have tried Jabber Client Mode. It's cool and works fine. But there's few information on the Component Mode. What's the difference between these two mode? I finished the configuration on jabber.conf and I am using openfire. What shall I do in the openfire to make them intergrated? Looks forward to your suggestions. Regards Tony ___ -- 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 -- Philippe Sultan ___ -- 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] NAT router for Linux
If you set the 'bindaddr' to your private IP address, the Gtalk connection from your Asterisk server to my Gtalk client (running on Windows) works fine. That's at least what we've tested together Julien, right? If the STUN packets are properly exchanged between Asterisk and the Gtalk client you're trying to communicate with, there should not be any problem. The thing is that you and I did not test the Asterisk - NAT box - Internet - NAT -box - Gtalk client since my Gtalk client had a public IP. I don't advise you to purchase a NAT router to test this scenario though. Philippe On Sat, Jan 24, 2009 at 6:32 PM, Julien Claassen jul...@c-lab.de wrote: Hello everyone! This is my problem: I try to do gtalk, but my asterisk server uses the local IP 127.0.0.1 or perhaps the 192.168.*.*. Now I've heard, that a NAT router can help there. I was told it's the way the windows-world does the trick, when they sit behind a router/phonebox/modem. Does anyone know a good software that will do the trick on Linux? I'm running Debian Lenny and one important thing: I can't use a GUI to configure anything. Any help is highly apreciated! Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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 -- Philippe Sultan ___ -- 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] gtalk/jingle full report
Hi Julien, The Gtalk call to your buddy fails because of a mismatch in the UDP ports for RTP. Try to disable the 'strictrtp' option in your rtp.conf file. Question : did you scramble the IP addresses? The Jingle call fails because of Google's XMPP network refusing to relay jingle packets wrapped in iq stanzas. There's no chance to have Jingle working on their network, you'll have to test another server like 'jabber.org'. From what I can read, the buddy you're trying to place calls to has a Telepathy client with Gtalk support, so we should be able to call him soon :) Also please file your bug report on the bug tracker : http://bugs.digium.com Thanks! Philippe On Tue, Oct 28, 2008 at 12:41 AM, Julien Claassen [EMAIL PROTECTED] wrote: Hello everyone! Philippe, you told me to make a bugreport. Well, here it comes, I'm still not sure, if tis is a bug or a miss-configuration. So I've put up a collection of configurations/output/debug files from a simple asterisk session testing the gtalk call. You can download it here: http://juliencoder.de/ap.txt Or I can mail it, just tell me where and I'll attach it to a mail. Please someone try to take a look, tell me what you'd need in addition and I'll happily provide it. Kindest regards and thanks Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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 ___ -- 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] jingle/gtalk still very troubling
Hi Julien, I just placed a call from my GoogleTalk account to your Asterisk server, reached your voicemail (or at least I guess cause the welcome message is in German), and left a message. Cheers, Philippe On Tue, Oct 28, 2008 at 10:24 AM, Julien Claassen [EMAIL PROTECTED] wrote: Hello Philippe! Would you by any chance have asterisk running with gtalk? I saw your mail there. If so perhaps we could test. Because all others I have found either don't have gtalk, so we tried jingle, which was still a bit problematic or if they had pure gtalk, they weren't really upto the VOIP-part. If someone else has gtalk/asterisk I'd be glad to test. My acount is: [EMAIL PROTECTED] Thanks in advance! Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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 -- Philippe Sultan ___ -- 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] jingle/gtalk still very troubling
Hi Julien, Please file a ticket on the bug tracker. We'll have a deeper look at your configuration there and figure out what's happening. Thanks! Philippe On Sun, Oct 26, 2008 at 8:15 PM, Julien Claassen [EMAIL PROTECTED] wrote: Evening Philippe! Here's what jabber show connected says: Jabber Users and their status: User: [EMAIL PROTECTED]/Talk - Connected Number of users: 1 I'll have to ask my friends, what their clients say. Although I suppose as my friend already send me a text message he saw me. And the state of me having no resource still appears after hours of running asterisk, with no change in configuration. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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 -- Philippe Sultan ___ -- 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] jingle/gtalk still very troubling
Hi Julien, Gtalk channels work with GoogleTalk clients. Empathy (based on the Telepathy framework) has a Gtalk implementation that is reported to work with Asterisk, too. Jingle channels should work with other Jingle implementations, but there are only a few of them around. One reason is that the Jingle specifications are not yet standardized. We try to keep Asterisk's Jingle implementation as close to the spec as possible though. Work is being done by the Telepathy guys on this area too. I've set up a publicly accessible Jingle Asterisk server, reachable at [EMAIL PROTECTED] Subscribe to this JID's presence status and you'll get automatically registered, you can then place Jingle calls to an echo server. Cheers, Philippe On Sun, Oct 26, 2008 at 1:25 PM, Julien Claassen [EMAIL PROTECTED] wrote: Hi! I just tried to call a friend using jingle, but I got refused. Errorcode was 502, he tried to call me, heard it ringing once and then it stopped. I used: originate jingle/gtalk_account/[EMAIL PROTECTED] [application] I'm registered to googletalk, but this should mean no harm, or should it. Once I was able to receive a text-message from him, but couldn't respond, I don't know how to. Remember I use asterisk only, no soft- or hardphone. Does anyone have suggestions, where to look, what to try? Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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 -- Philippe Sultan ___ -- 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] jingle/gtalk still very troubling
The originate command should work. Make sure that the user you're placing the Gtalk/Jingle call is in the buddy list and has Jingle capabilities. The 'jabber show buddies' command will give you that info. Cheers! Philippe On Sun, Oct 26, 2008 at 3:57 PM, Julien Claassen [EMAIL PROTECTED] wrote: Hello Philippe! Do I need a googletalk client? Or can I just use asterisk's originate CLI command? I was under the illusion I could. Otherwise it's a bit problematic. I canonly use text-based applications and they better support JACK audio Connection Kit, for my soundcard is not simple standard. I had problems with that before. Do I need to especially configure my firewall, besides opening all outbound ports? I'm in a small local network, so do I also have to configure port-forwarding. As I said: we succeeded in sending me a text-message, but audio won't work. Signalling is fine, but then establishing the connection always failed. Kindest regards and thanks so far Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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 -- Philippe Sultan ___ -- 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] jingle/gtalk still very troubling
Hi Julien, On Sun, Oct 26, 2008 at 4:51 PM, Julien Claassen [EMAIL PROTECTED] wrote: Hi! There's something strange. I have entered a couple of buddies. On has Jingle capability and two have resources (Home and Telepathy), but my own account does have no resource, I put myself in the buddies list. Is tat supposed to be? The account Asterisk connects with (in jabber.conf) appears in the buddy list, with a default resource named 'asterisk', and has Jingle capabilities. Usually, when you see a buddy without any resource, it means that this buddy is in your roster, but is not currently connected. And again about those ports: Accept the 5222 port, do all the other necessary ports have to be opened from the outside (or requested from there) or are they opened from my end? And if they need to be opened from the outside: whichports do I have to open in the firewall (taken from the rtp.conf or is there a range simply given by some standard? Gtalk and Jingle channels use Asterisk's RTP stack. The UDP port is negociated and can take any value, in the range specified in rtp.conf for Asterisk, unknown for the remote peer. Philippe ___ -- 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] jingle/gtalk still very troubling
Strange, are you both connected to Google's XMPP server? Sometimes it takes a little time before retrieving your roster on Gtalk. Does Asterisk appear as connected on your friend's buddy list? Also, what does the 'jabber show connected' say? Cheers, Philippe On Sun, Oct 26, 2008 at 5:53 PM, Julien Claassen [EMAIL PROTECTED] wrote: Well, so asterisk seems to think, that I'm not connected, for I don't see a resource Asterisk or Talk with my name. That shouldn't really be. :-( Any ideas on fixing this? Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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 -- Philippe Sultan ___ -- 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] gtalk dialstring?
Hi Julien, bach [Oct 25 21:18:11] ERROR[28847]: chan_gtalk.c:908 gtalk_alloc: no gtalk capable clients to talk to. [Oct 25 21:18:11] NOTICE[28847]: channel.c:3243 __ast_request_and_dial: Unable to request channel gtalk/gtalk_account/[EMAIL PROTECTED] The syntax is correct. Make sure that you have the [gtalk_account] section inside your jabber.conf file, you can also check the connection to the GoogleTalk XMPP server by issuing these commands : jabber show connected jabber show buddies (in Asterisk 1.6) Cheers, Philippe ___ -- 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] AGI and prepaid billing + Radius
Hi Bilal, On Tue, Sep 23, 2008 at 11:11 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Dear Philippe; Thanks a lot for ur kindly answer. How can I use the Radius with CDR (Accounting)? Here is the documentation : http://svn.digium.com/view/asterisk/branches/1.4/doc/radius.txt?view=markup About PortaOne's billing systems: Do u mean I can use the PortaOne's billing systems Radius client (to be fixed at Asterisk side), and customize this client to be used with any RADIUS based billing system? Yep. This client is written in PERL, and uses the Authen::Radius API. You can integrate it with Asterisk (see the doc in the link I sent), and adapt it to make it work with any RADIUS server. Regards, Philippe ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI and prepaid billing + Radius
Hi Bilal, Asterisk's RADIUS support is limited to CDRs, that is, the last A in AAA (Accounting). As for Authentication and Authorization, Asterisk integrates very well with PortaOne's billing systems (PortaBilling + PortaSIP), if you use their PERL RADIUS client : http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth I guess if you tweak that RADIUS client a bit, you can make it work with any RADIUS based billing system. Cheers, Philippe On Tue, Sep 23, 2008 at 10:35 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Yes it answer and big thanks. I have another question (which might be not related alot to AGI) if u can help me: If Asterisk support Radius, so we can build Prepaid Billing with Radius to communicate via Radius as standard communication method? Regards Bilal --- On Tue, 9/23/08, Benjamin Jacob [EMAIL PROTECTED] wrote: From: Benjamin Jacob [EMAIL PROTECTED] Subject: Re: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com, [EMAIL PROTECTED] Date: Tuesday, September 23, 2008, 6:39 AM Hi Bilal, Yes it is definitely possible. And I've done it myself for a couple of our clients. Does that answer your two questions? cheers - Ben. --- On Tue, 9/23/08, bilal ghayyad [EMAIL PROTECTED] wrote: From: bilal ghayyad [EMAIL PROTECTED] Subject: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com Date: Tuesday, September 23, 2008, 9:52 AM Hi All; Did anyone do an prepaid billing application via AGI? I would like to know if that is possible. Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Philippe Sultan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to encrypt passwords stored in the realtime database?
Well, if someone steals the md5secret (HA1) for a given username and realm, he can use it to authenticate to the SIP proxy or B2BUA that serves the target user. On both sides (SIP client and proxy or B2BUA), the values to be compared are the computed results of MD5(HA1:nonce:HA2), where : HA1 = MD5(username:realm:password) and HA2=MD5(Method:Request-URI) The nonce string is generated by the SIP server, as well as the realm value. So, even without knowing the user's password, you can still get access to his SIP account. On Wed, Aug 20, 2008 at 10:17 PM, BJ Weschke [EMAIL PROTECTED] wrote: Igor Hernandez wrote: I was thinking the same thing I believe Tzafrir just alluded to. If the passwords are encrypted in the DB with a public key then...asterisk needs to have the private key stored somewhere to be able to decrypt the values to authenticate the user. In this way there is nothing preventing whoever intrudes your boxes from getting that key and decrypting the values himself. I might be missing something though and if thats the case chime in, I'm interested in this issue. Regards, You are. md5secret simply stores the crypt hash. When it receives the password attempt, it too, is crypted using MD5 algorithm and then the two hashes are compared. Using MD5 crypt hash, there is no way to decrypt the hash. It's a brute force methodology to get the password back if you've lost it. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Philippe Sultan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Radius
Ciao Salvo, That is not directly possible. But, you can integrate a GPL PERL RADIUS client with Asterisk : http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth It works good, I use it to make Asterisk work as an IVR with a billing system, that acts as a RADIUS server. Cheers, -- Philippe Sultan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Gtalk setup
Works ok on my side. Any debug messages from your console you could post here? Thanks, Philippe On Mon, Aug 11, 2008 at 9:16 AM, vivek rastogi [EMAIL PROTECTED] wrote: Hi, I've just followed http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talkinstructions from wiki, And i always get my jabber (GoogleTalk account for asterisk server) not registred: asterisk1*CLI jabber show connected Jabber Users and their status: User: myasteriskaccount - Disconnected asterisk1*CLI jabber test ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Philippe Sultan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a component in Jabber network
Hi Eric, no that's not possible, Asterisk being connected either as a regular client or component. In the configuration you describe, Asterisk needs to be allowed by the XMPP user (gmail.com) to send messages to his/her address. This is achieved under the mutual subscription mechanism. However, the opposite call configuration works ok if you use the JabberReceive application, which gives a flexible way to access telephony resources from Gtalk/Jingle through Asterisk. That is, back to your example, a GoogleTalk user can pass any SIP URI to an Asterisk gateway. JabberReceive is still at a testing stage, if you want to give a try, please check : http://bugs.digium.com/view.php?id=12569 Cheers, -- Philippe Sultan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a component in Jabber network
Hi Antonio, here is the corresponding section of my jabber.conf file, that allows our Asterisk server to connect to our local XMPP server (jabberd2), as a component. [asterisk-component] type=component serverhost=jabber.inria.fr username=asterisk secret=*** port=5347 Depending on your XMPP server, the port number may be different. Can you tell us more about what you want to achieve? Cheers, Philippe On Fri, Jun 27, 2008 at 4:54 PM, Antonio Anderson M. de Souza [EMAIL PROTECTED] wrote: Hi Everybody, Does anybody have some tutorial how to configure Asterisk in the component mode in a Jabber service, i already configured, and tested it in the client mode, and it worked fine, but i think the component is the best solution for what i need to implement. Thank you very much, -- Antonio Anderson M. Souza Project Manager Voice Technology Rua: Libero Badaró, 293 Cj 30D - 30o. andar CEP: 01009-907 [EMAIL PROTECTED] phone: +55 11 3588-0188 mobile: +55 11 8863-0693 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Philippe Sultan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and XMPP (Jabber) : testing new application JabberReceive
Hi Julian, How difficult would it be to have a JabberReceive Event *initiate* a channel ? I think that could be done. And you could also place Originate commands over AMI, as you mentioned it. You might be interested in BJ's work, as it covers that topic : http://www.asterisk.org/node/48440 Cheers, Philippe ___ -- 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] Asterisk and XMPP (Jabber) : testing new application JabberReceive
Hi Julian,
[...]
What can you do with it? Well, a direct usage of this application is
to make an easy to use GoogleTalk voice gateway out of Asterisk. Here
is an example (assuming the asterisk-xmpp account is configured) :
context gtalk-in {
s = {
NoOp(Caller id : ${CALLERID(all)});
Answer();
JabberSend(asterisk-xmpp,${CALLERID(name),Please enter the
number you wish to call);
JabberReceive(${CALLERID(name)},NEWEXTEN);
How can you assume that the message you are waiting for is the right one
? Let's say that you have 10 channels each doing a JabberReceive at the
same time - how does the channel know how to get the right message, let
alone the right data ?
(2 channels may be waiting for a NewExten message, others for a
GetSomeDataFromSomeOtherPlace message )
Well, in the example, as long as you have 10 simultaneous GoogleTalk
calls from 10 different buddies, that won't be a problem. The first
argument of JabberReceive is used by the channel to identify the
Jabber ID it expects to read data from. Therefore, a message coming
from a specified buddy (identified by his JID) will be passed by
res_jabber to the channel that is waiting for data from this buddy.
In the case when several channels are waiting for data from the same
JID, res_jabber passes the message to every channel that matches.
Although this is less likely to happen, I tried to address this issue
by using the thread tag to track chat conversations
(http://www.xmpp.org/extensions/xep-0201.html). Unfortunately, very
few XMPP clients implement this conversation tracking mechanism (and
GoogleTalk does not).
Philippe
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[asterisk-users] Asterisk and XMPP (Jabber) : testing new application JabberReceive
Friends,
a new dialplan application is now available for testing :
http://svn.digium.com/view/asterisk/team/phsultan/jabberreceive/
The corresponding feature request is located here :
http://bugs.digium.com/view.php?id=12569
What can you do with it? Well, a direct usage of this application is
to make an easy to use GoogleTalk voice gateway out of Asterisk. Here
is an example (assuming the asterisk-xmpp account is configured) :
context gtalk-in {
s = {
NoOp(Caller id : ${CALLERID(all)});
Answer();
JabberSend(asterisk-xmpp,${CALLERID(name),Please enter the
number you wish to call);
JabberReceive(${CALLERID(name)},NEWEXTEN);
JabberSend(asterisk-xmpp,$(CALLERID(name),(Calling ${NEWEXTEN} ...);
Dial(SIP/${NEWEXTEN);
Hangup();
}
}
In this example, when Asterisk receives a GoogleTalk voice call
request from a GoogleTalk buddy, it answers the call, and asks the
buddy to enter a number over an XMPP (Jabber) chat session. Then,
Asterisk dials the extension (accessible over SIP), which results in a
GoogleTalk to SIP call.
But this application is not restricted to GoogleTalk voice calls, and
it can be used within any call context. Code snippets are available in
the corresponding feature request under the bugtracker as well as in
doc/jabber.txt.
The codebase is Asterisk's SVN trunk, which is merged to the
jabberreceive branch on a regular basis. To install it, follow these
steps :
#svn co http://svn.digium.com/view/asterisk/team/phsultan/jabberreceive/
jabberreceive
#cd jabberreceive
#./configure
#make
#make install
Note for Linux users : the Gnome IM+ToIP client Empathy (starting from
version 0.23.1) is now compatible with Asterisk, which allows users to
place voice calls over a GoogleTalk channel from their Empathy client
to Asterisk.
Please give your feedback!
Thanks i advance,
Philippe
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Re: [asterisk-users] handling jabber status
Thanks for the snippet, I re-wrote it (badly) for regular extensions.conf usage, and verified it's also working here on 1.6, though I do get a warning about JabberStatus being depreciated. Yes, JabberStatus is being moved from an dialplan application to a function (JABBER_STATUS), because it's just retrieving a variable. Philippe ___ -- 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] handling jabber status
Hi Benoit, Anyone already did that (changing jabber status/ status message of many accounts) or know if it's even remotly possible ?? We discussed that during the last XSF devcon in Brussels. Actually Asterisk (or any other XMPP client) cannot change the Jabber status on behalf of another Jabber user, even if you connect it as a component to your XMPP server. This behaviour is forbidden by the XMPP specs. To be able to do this, you can use OpenFire along with its Asterisk plugin, or patch your own XMPP server. I had written a patch for Jabberd2 some time ago, but I'm not aware of anything that would be applicable to Ejabberd (written in Erlang). I'm in the process of extending Asterisk's Hints dialplan to XMPP notifications for authorized users. Those notifications would be carried over XMPP as message stanza of type 'headline'. That will be a first step toward implementing PEP (Personal Eventing via Pubsub). Although I understand that this won't answer your specific need, I thought you might be interested in knowing this. I can build a private branch to make this code available if you're interested in testing it. Philippe ___ -- 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] handling jabber status
Hi Matt,
On Wed, Jun 4, 2008 at 1:05 AM, Matthew Gibson [EMAIL PROTECTED] wrote:
I'd be interested to know more about the status abilities as well, we've
tried to test jabberstatus application, but it doesn't seem to function as
we expect, it should be returning 0,1,2,3,4,5 based on users current status,
but switching to away doesn't seem to change it from 0 to 2 .. .
this could be an interesting thread :)
JabberStatus is supposed to retrieve the XMPP status of a buddy, and
store it in a diaplan variable. I just tested it on my Asterisk (1.6)
server.
Here is an example of how to use it :
1234 = {
JabberStatus(asterisk-gmail,[EMAIL PROTECTED],STATUS);
if (${STATUS}=1) {
NoOp(User is online and active, ring his Gtalk client.);
Dial(Gtalk/asterisk-gmail/[EMAIL PROTECTED]);
} else {
NoOp(Prefer the SIP phone);
Dial(SIP/1234);
}
}
Matt, if you're experiencing some problems with this application, for
example on a 1.4 system, do not hesitate to file a bug report.
Cheers,
Philippe
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Re: [asterisk-users] using gtalk received instant messages in the dialplan
Hi Erik,
There's ongoing work on the topic, you might be want to have a look at
BJ Weschke's note and branch : http://www.asterisk.org/node/48440 .
The final purpose of this code is to allow users to send/receive AMI
commands over an XMPP connection to Asterisk. Placing an Originate
action in this situation would then answer your need.
You might also be interested in what's being worked here :
http://bugs.digium.com/view.php?id=12569 . Basically, we want to
implement a JabberReceive dialplan application, which along with
JabberSend (or SendText) allows users (as long as they're authorized
to as buddies) to control calls going through the dialplan.
E.g., you can use Asterisk as a Gtalk gateway, by configuring this
sample GoogleTalk context :
[gtalk-in]
exten = s,1,answer()
exten = s,n,SendText(What extension to call?)
exten = s,n,JabberReceive(${CALLERID(name)},RCV_EXT)
exten = s,n,Dial(SIP/[EMAIL PROTECTED])
exten = s,n,hangup()
Cheers!
Philippe
On Wed, May 21, 2008 at 11:57 PM, Erik de Wild: Tripple-o
[EMAIL PROTECTED] wrote:
I have been doing some reading about gtalk and asterisk. Most of it is
pointed to enable using gtalk for making phonecalls. Would it be
possible to use gtalk instant messaging input (just some text send to
the gtalk account configured on an asterisk box) into the dialplan.
This way you could use gtalk im to trigger all kind of events like
sending an sms, adding sip entries to the system, start conferences
etc. etc. The basic question is: is it possible to store the received
Gtalk message into a variable that can be used to trigger events in a
dialplan (which isn't actually a dial plan anymore) or doesn't
anything like that exists at this moment. Is this just a crazy thought
or does the idea of triggering events in the dialplan via Gtalk im
input make sense.
I was thinking about a call with the im message as a variable in it
starting a local channel that goes into the relevant part of the
dialplan.
examples:
sms: Could you please call Mark 0612345678 for sending an sms
sip_entry: 500 cdwtg_34$ ALAW snom320 for adding a sip
entry to sip.conf with number 500 and password cdwtg_34$ for a snom320
conference: 0591234567 0201234567 0612345678: for setting up a
conference between this numbers
All the logic has to be in the dialplan or scripts but it all should
start with receiving a message send by a gtalk client. My personal
opinion is that it would make a great and easy to explain user
interface that can be used from every pc and every pda or smartphone.
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Re: [asterisk-users] UPDATED Asterisk Jingle Extensions.conf
Hi Ali, I have sent a previous email with a problem that I solved by using component mode. In this mode the asterisk server acts as a subdomain. So I can call [EMAIL PROTECTED], [EMAIL PROTECTED] That's a nice way of using Asterisk's component capability. Which XMPP/Jingle client are you using? However I want it to call the number in dialed initially I.e 1000 or 1001 etc etc etc. Any way to do this parsing using Asterisk ? If your XMPP/Jingle client can send DTMF, you can use Asterisk's DISA application that will collect the entered digits and place a new call : http://www.voip-info.org/wiki-Asterisk+cmd+DISA If your XMPP/Jingle client cannot send DTMF, then please open a feature request on the bug tracker : http://bugs.digium.com/, along with an Asterisk's debug output and detailed description of the feature. Cheers, Philippe ___ -- 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] jingle with Asterisk + PSTN
Hi Ali, On Fri, Mar 28, 2008 at 5:31 PM, Ali Jawad [EMAIL PROTECTED] wrote: Hi All I am developing a client that uses libjingle to do xmpp stuff with ejabberd. I can also make audio calls between those clients. What I am trying to archive now is to send calls to pstn using jingle. I was told in the jingle-dev community that asterisk can do that. Asterisk speaks Jingle indeed. If you're using a libjingle based client, you'll have to set up a GoogleTalk connection to Asterisk through the chan_gtalk channel driver. Those good pointers will help : http://www.voip-info.org/wiki/view/Asterisk+Google+Talk http://taug.ca/node/43 Note : the chan_jingle channel driver implements the Jingle (not GoogleTalk related), so it won't work with libjingle even though the names sound close. Is there any way to send jingle audio calls to asterisk and will it understand them ? If yes..can I forward those calls to PSTN ? PSTN, SIP, H323, SCCP, ... or any protocol implemented in Asterisk! Philippe ___ -- 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] jingle with Asterisk + PSTN
On Mon, Mar 31, 2008 at 4:51 PM, Ali Jawad [EMAIL PROTECTED] wrote: So should I register directly on the asterisk server or should I send the voice calls through ejabberd to asterisk ? You can't register an XMPP client on Asterisk, because it's not an XMPP server. The required steps to establish a Jingle voice call to Asterisk are : - register Asterisk on an XMPP server (ex. talk.google.com, but it can be any XMPP server) ; - register your XMPP (Jingle capable) client on any XMPP server ; - make sure both these XMPP clients are buddies ; - place your call directly to Asterisk. In order to have your Jingle call relayed to the PSTN, you must edit your Asterisk dialplan accordingly. Philippe ___ -- 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] Gtalk with asterisk
Hi Naveen, For which i installed the iksemel but this didnt help me out. I couldnt find the res_jabber.so file any where in the asterisk source directory. Still when i run the command make menuselect the channel driver chan_gtalk shows xxx (dependencies not met). How can i register gtalk with asterisk. If you can provide me with some basic details i can carry forward. Either the iksemel library has not been properly installed on your system, or Asterisk did not detect it. The following link may help you, as it includes hints to troubleshoot iksemel + GnuTLS installations : http://www.voip-info.org/wiki/view/Asterisk+Google+Talk However, feel free to open a bug report if you've made sure you have a properly installed iksemel stack. Note that as of Asterisk 1.6, GnuTLS was replaced by OpenSSL which is now used by Asterisk as the encrypting protocol for iksemel. -- Philippe Sultan ___ -- 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] OT : OpenSER Summit Pavilion - 17th to 19th of March, 2008 , San Jose, US
I'm taking the liberty to announce this event on the Asterisk mailing list, as Asterisk and OpenSER form a valuable combination in SIP architectures. The second edition of OpenSER Summit will take place in San Jose, USA ,on the 17th of March, 2008, during VonX Spring 2008 pre-conference events. This is the first US edition of the OpenSER Summit - to learn more about the agenda and layout of the event, see http://www.openser.org/mos/view/OpenSER-Summit-2008. All participants to register via OpenSER site before Friday, March 7th, will get free access to the OpenSER event. This OpenSER Summit edition is sponsored by a parallel OpenSER related event, the OpenSER Pavilion . The pavilion is a common exhibiting area - booth 1027, inside VoN expo, gathering, under the OpenSER name, six different companies working or using the project. You can find more about the OpenSER Pavilion and its participants here - http://www.openser.org/mos/view/OpenSER-Pavilion-2008. The dual event, OpenSER Summit Pavilion is new concept of a more complex event, aiming to create a larger diversity and to give more power to the understanding of the OpenSER project. ___ -- 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] FW: jabber
Hi Clive, Hi all, Do some one experiencing running jabber applications (jabberstatus...) in asterisk? I do experinced Asterisk 1.4.18 and wish to start it, however I got such result. IBM*CLI help jabber No such command 'jabber'. IBM*CLI help jabberstatus No such command 'jabberstatus'. Any one can help me on this, or may be I miss out somethings that cause jabber applications did'nt install. It looks like res_jabber is not installed on your system. If res_jabber is loaded, the output of the 'module show like jabber' command should match the following : *CLI module show like jabber Module Description Use Count res_jabber.so AJI - Asterisk Jabber Interface 0 1 modules loaded The jabber/XMPP related modules depend on the iksemel library, which depends on GnuTLS. Note that starting from the 1.6 series, GnuTLS is not used anymore by res_jabber, as we moved to OpenSSL. You should check your module installation configuration via 'make menuselect', and make sure your system supports the libraries required by res_jabber. Philippe ___ -- 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] gtalk and dtmf
Hi Adam, I've been googling for half an hour, i found some sort of jingle protocol which i'm not sure what to use for but it might be the solution? It seems to me that my problem is sending the dtmf tones, not receiving them, so this is really gtalk related. You've spotted the problem, you cannot send DTMF tones with your GoogleTalk client, even though Asterisk is capable of receiving them. I know the people of the Jabbin project were working on this topic, maybe you can try their gtalk compliant client : http://www.jabbin.com Philippe ___ -- 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] Asterisk as XMPP component. How to use it ?
Hi Olivier, At the opposite, I think it could be useful for an Asterisk server to act as XMPP User Activity provider (ie update XEP-0108 field with on-the-phone value). Do you agree ? This is indeed a direction we should consider in order to relay call and device state information to XMPP users. As those kind of messages are exchanged via PEP (Personal Eventing via Pubsub), we'll have to implement that too. But still, I'm not sure we can provide this kind of presence information on behalf of XMPP users if we connect Asterisk as a component. On the other hand, by connecting Asterisk as a XMPP client, we could easily pass call and device state information to that client's buddy list. The main drawback of that method is that Asterisk needs to be provided with the user password. Is there any XMPP client supporting User Activity ? I know Psi supports PEP, not sure about User Activity though. Is Asterisk capable of getting or sending such User Activity messages ? No. Philippe ___ -- 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] Jabber and Asterisk?
Indeed, through the dialplan configuration, Asterisk allows you to place calls from whatever channel type (SIP, H323, ISDN, ..) to GoogleTalk clients. The revert call configuration is trickier though, as the GoogleTalk client user interface does not allow you to dial numbers over DTMF for example. A possible workaround is to use the powerful 'Originate' command available in Asterisk : http://www.voip-info.org/tiki-index.php?page=Asterisk+Manager+API+Action+Originate Also, BJ Weschke is currently working on linking XMPP (Jabber) with AMI : http://www.asterisk.org/node/48440 Finally, the chan_jingle channel is available starting from Asterisk 1.6). Jingle is the set of XSF (XMPP Standards Foudation) specifications that will provide Voice + Video over IP to XMPP. Jingle is its final stage at XSF, see : http://blog.xmpp.org/?p=32. There are not so many Jingle clients available around, so test reports against Asterisk's Jingle interface are very welcome! Philippe On Jan 27, 2008 10:55 AM, Michiel van Baak [EMAIL PROTECTED] wrote: On 04:41, Sun 27 Jan 08, Vincent wrote: Hello I'm curious about what can be done when using Jabber with Asterisk. What are good examples of this combination? What I do is make asterisk send a message to one of my jabber accounts when a call comes in. Also, because asterisk is in my buddy list I can see if it's online or not :) This is just a simple usecase for it. with chan_gtalk it should be possible to do voicecalls using google talk, but I never tried that. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ -- 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 -- Philippe Sultan ___ -- 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] Discover Asterisk 1.4 :: Google Talk, XMPP and Jabber Integration!
Hi Matt, Can an Asterisk server hold logins for multiple Japper accounts on a remote Jabber server, and carry multiple Jabber calls simultaneously the way it can carry multiple SIP (or IAX, or ZAP, etc) calls? If so, is each of those Jabber calls as lightweight as, say, each SIP call? If not, is there a way to increase the capacity of Asterisk to carry about as many Jabber calls as it can carry SIP calls? You can hold multiple logins on your Asterisk server by configuration. You can also carry simultaneous Gtalk/Jingle talks simultaneously, over a single Jabber account or several. As for the performance comparison with SIP calls, I can't really tell. It would be nice to have some feedback on that. Cheers, Philippe ___ -- 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] Gtalk callerID
Hi Abel, Is there a way to catch de gtalkID of a caller that´s calling my asterisk gtalk account? the caller id is not properly set, only its ANI part is. I just proposed a patch in order to retrieve the CALLERID(name) variable from the Dialplan, see http://bugs.digium.com/view.php?id=11549. The CALLERID(name) value will be set to the Gtalk ID of the remote peer. Thanks for pointing this out! Philippe ___ --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] Asterisk as a SIP to XMPP Jingle voice gateway
Hi Eric, I'm looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Asterisk can connect as a client or component to a XMPP server. XMPP components are typically used as gateways between XMPP and other IM services such as MSN or Yahoo. You can connect Asterisk to GoogleTalk's XMPP network as a client only, which will therefore be accessible through a presence subscription mechanism just like a usual client. On the other hand, you can connect Asterisk as a component to your locally administered XMPP server, for example. A 'service discovery' request to the server will show the Asterisk server as being available. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? Asterisk can be used to call Gtalk users from SIP phones, and vice versa. Configuration examples are given here : http://www.voip-info.org/wiki/view/Asterisk+Google+Talk The call configuration is handled in the Dialplan in that case. If you need to place a call from a XMPP client to a SIP URI, you'll also have to find a client that's able to to so. I know that GoogleTalk and Jabbin both speak XMPP + Gtalk. However, the GoogleTalk client's user interface does not allow you to place a call to anything but another XMPP client from your buddy list, without offering the ability to enter either a SIP URI or phone number. A possible workaround was available here : http://bugs.digium.com/view.php?id=8659 As for Jingle, Asterisk tries to follow the latest set of specifications (code only available from SVN trunk), which are not completed yet. Philippe ___ --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] GTALK problem
If I calling asterisk with GTALK in english everything is ok, however, some of my friends with the italian version of gtalk they cannot have the audio. Audio problems might be experienced with older Gtalk clients. Version 1.0.0.104 is reported to work. The following resources may help you : http://www.voip-info.org/wiki/view/Asterisk+Google+Talk#Bugsampknownissues http://bugs.digium.com/view.php?id=10512 Hope this will help you solve the problem, Philippe ___ --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] VoIP+IM with Asterisk+Jabber
Hi Alejandro, the Jabber module in Asterisk is available starting from the 1.4 series. Therefore, you can connect Asterisk as a client (or component) to your Jabber server after you've upgraded to 1.4. You'll get detailed information here : http://www.voip-info.org/wiki-Asterisk+Jabber http://www.voip-info.org/wiki/view/Asterisk+Google+Talk Philippe ___ --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] VoIP+IM with Asterisk+Jabber
On 8/31/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Ows, I suppose that * can only do c2s to google talk to which I did and I got audio both ways. Yet I have not seen anything so far how * could do a s2s to google talk. Indeed, the Jabber module was not designed to make Asterisk a Jabber server. Philippe ___ --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] Gtalk/Jabber connect issues in 1.4.8
Hi Bruce, [EMAIL PROTECTED] Google's server is expecting you to provide a valid gmail address here, suffixed with @gmail.com Cheers, Philippe ___ --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] gtalk - no audio
Hi Demuel, On 6/22/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Yeah, just the same as the sample configuration by mog. However, if I am using a gtalk application in asterisk to dial googletalk buddy, my voip phone is suddenly connected to the googletalk buddy though at the googletalk client software it is still waiting to be accepted or not accepted. I mean from my voip phone perspective, there is just one ring to make a call to the googletalk buddy unlike in the jingle application wherein there are successive ring before the googletalk buddy accepts the call. That's strange. I was not able to reproduce this problem, that is, when dialing an extension that points to a GoogleTalk client from a SIP phone, I *always* have to wait for the GoogleTalk client to accept the call. That's just like if you had Asterisk automatically answer GoogleTalk calls. Do you have any file streamed to the SIP phone by Asterisk? Philippe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gtalk - no audio
What is the main distinction between Jingle and Gtalk here? How should I generate the file streamed to the SIP phone by Asterisk? I really have no clue :). Maybe you can open a bug report so that we can dig into this problem. Thanks! Philippe ___ --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] gtalk - no audio
Hi Koen This works fine when I call this account from my personal gtalk. But others have some very strange problems. In most cases, I see the call coming into Asterisk and executing normally. On the callers side, the call looks like it was answered, but there's no audio. In some other cases, the call doesn't even appear to be answered, although I see a normal execution on Asterisk. Can you please open a bug report that describes your problem, and attach an Asterisk debug output for a failed call to the report? Thanks, Philippe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gtalk - no audio
Philippe, what part of the channel code handles the ringing and dialling.
From my
experience here, making a call from googletalk to a voip phone inside a
firewalled
environment does not pose any problem. But making call from voip phone to
googletalk is
kinda tricky.
Well, chan_gtalk being a channel, its PBX functions are all gathered
in a ast_channel_tech structure :
/*! \brief PBX interface structure for channel registration */
static const struct ast_channel_tech gtalk_tech = {
.type = Gtalk,
.description = Gtalk Channel Driver,
.capabilities = ((AST_FORMAT_MAX_AUDIO 1) - 1),
.requester = gtalk_request,
.send_digit_begin = gtalk_digit_begin,
.send_digit_end = gtalk_digit_end,
.bridge = ast_rtp_bridge,
.call = gtalk_call,
.hangup = gtalk_hangup,
.answer = gtalk_answer,
.read = gtalk_read,
.write = gtalk_write,
.exception = gtalk_read,
.indicate = gtalk_indicate,
.fixup = gtalk_fixup,
.send_html = gtalk_sendhtml,
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
};
demuel, do you have an extensions.conf (or ael) snippet for a VoIP
phone - Asterisk - GoogleTalk call scenario? I wonder why this does
not work in your case.
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[asterisk-users] res_jabber over OpenSSL ready for testing
Hi everybody, I'd like to have the feedback from the community regarding this patch : http://bugs.digium.com/view.php?id=9972 res_jabber currently relies on the iksemel API to handle TLS connections, which assumes GnuTLS to be installed on the system. The basic idea of the proposed modifications is to bypass iksemel's API when sending/receiving TLS secured data and use OpenSSL instead. What you'll need on your system : - OpenSSL installed (tested version 0.9.8b) ; - iksemel installed (tested version 1.2), with or without GnuTLS. I was able to have this patched res_jabber working with Google's Jabber server (TLS required), as well as with our jabberd2 server (with or without TLS) at INRIA. On reason why we should consider moving to OpenSSL is because other modules in Asterisk use it to secure connections. Also, the iksemel API does not deal with TLS connections properly, which leads res_jabber to misbehave when TLS is activated (for example, see bug #9738 : http://bugs.digium.com/view.php?id=9738). Note : the proposed patch applies to the SVN trunk branch. Thanks for your help! Philippe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
