[asterisk-users] Asterisk versions supporting Path header?
The bug tracker includes several issues relating to Path (RFC 3327) support. It is not clear which version actually included the patch and which versions are working. Could anybody update these issues in Jira with a brief comment about the supported versions? https://issues.asterisk.org/jira/browse/ASTERISK-16884 original patch against chan_sip / Asterisk 1.8 Status is "Fixed", but not version is recorded, which version was this merged in? https://issues.asterisk.org/jira/browse/ASTERISK-21084 chan_pjsip Path support Satus is Fixed for v12.1.0 - is that only for chan_pjsip, or is Path also supported in chan_sip in any versions up to 12.1.0? https://issues.asterisk.org/jira/browse/ASTERISK-25666 Path header ignored (looks like a regression?) reported for 13.6.0 - which is the last version where it did work? https://jira.digium.com/browse/SWP-2484 "add Path header support to chan_sip" Internal Jira link - does this issue contain any further details about the versions supported? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [CFP] reminder! FOSDEM RTC dev-room talks: deadline Friday
Reminder: speaker's deadline this Friday, 27 November at 23:59 UTC We have already received several really exciting talk proposals but there is still time for people to propose talks or encourage friends or colleagues to speak. Many other dev-rooms also have a deadline in the next few days and if your topic is applicable to more than one dev-room, you are welcome to make more than one submission. Please contact us or put a note in the memo field at the top of the talk proposal if you do that. All projects are encouraged to consider making a lightning talk too, it is an excellent opportunity to get exposure for your project: even though you only have 15 minutes, it can be a much larger and more diverse audience than in some dev-rooms. For full details, please see the call for papers: http://danielpocock.com/fosdem-2016-free-rtc-dev-room-and-lounge We invite all potential speakers and participants to discuss the selection process and other aspects of FOSDEM on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc -- _ -- 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] [CFP] FOSDEM 2016, RTC devroom, speakers, volunteers neeeded
XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on 28 and 29 January 2016 http://wiki.xmpp.org/web/Summit_19 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit We are also considering a more general RTC or telephony summit, potentially on 29 January. Please join the Free-RTC mailing list and send an email if you would be interested in participating, sponsoring or hosting such an event. Social events and dinners = The traditional FOSDEM beer night occurs on Friday, 29 January On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc Spread the word and discuss === If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: http://planet.jabber.orgral...@ik.nu http://planet.sip5060.net dan...@pocock.pro http://planet.opentelecoms.org dan...@pocock.pro Please also link to the Planet sites from your own blog or web site. Contact === For discussion and queries, please join the free-rtc mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Daniel Pocock Ralph Meijer Saúl Ibarra Corretgé Iain R. Learmonth -- _ -- 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] Asterisk in the RTC Quick Start Guide
Asterisk is mentioned in quite a few places in the RTC Quick Start Guide[1] I've put up a blog today about my work on this book and some questions[2] for discussion. I'd be particularly interested in any feedback from the Asterisk community about just how Asterisk fits into the federated SIP and RTC environment and whether this book makes it easier for people. Regards, Daniel 1. http://rtcquickstart.org 2. http://danielpocock.com/rtc-quick-start-becoming-a-book-now-in-beta -- _ -- 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] WebRTC meeting Norfolk, 15 October 2014
I'll be in Norfolk, VA for xTupleCon in October On 15 October, there will be two events for WebRTC: 14:15 a talk about the xTuple WebRTC extension at xTupleCon - must register for xTupleCon to attend this 17:30 a technical / developer workshop at xTuple's offices - free, anybody welcome, even if not attending xTupleCon, RSVP through Eventbrite[1] Please see my blog[2] for more comments about all of this and feel free to email me in advance if you have questions about it or if you may like to meet up there. 1. http://www.eventbrite.com/e/browser-based-webrtc-telephony-for-web-apps-workshop-tickets-13002257101 2. http://danielpocock.com/xtuplecon-webrtc-talk-schedule-change -- _ -- 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] WebRTC / Rejecting secure audio stream errors
I've seen the following appear in some tests with Asterisk 11.11: WARNING[3938][C-0003]: chan_sip.c:10535 process_sdp: Rejecting secure audio stream without encryption details: audio 54908 UDP/TLS/RTP/SAVPF 109 0 8 101 Specifically, it always happens from a Firefox 24 host but it works without this error from another host running Firefox 26 I did a diff on the SDP and couldn't see anything obviously different except one thing: Firefox 24 only has host candidates for ICE (TURN support was only added in Firefox 25). Is there any way that could cause this error though? It appears the encryption details are sufficient and do not otherwise differ between Firefox 24 and 26: --- ff-24.txt 2014-08-25 15:02:20.452383599 +0200 +++ ff-26.txt 2014-08-25 15:01:42.472346613 +0200 @@ -1,12 +1,12 @@ v=0 -o=Mozilla-SIPUA-24.7.0 14737 0 IN IP4 0.0.0.0 +o=Mozilla-SIPUA-26.0 18111 0 IN IP4 0.0.0.0 s=SIP Call t=0 0 -a=ice-ufrag:301212e4 -a=ice-pwd:d7430f468514f1f2d326d3c944691fbf -a=fingerprint:sha-256 E2:53:6A:FA:6D:E2:3F:7E:24:82:0F:E3:27:34:D1:CC:50:31:42:82:5F:DF:34:9A:4F:42:D1:6D:B7:DB:5C:43 -m=audio 54908 UDP/TLS/RTP/SAVPF 109 0 8 101 -c=IN IP4 10.10.1.144 +a=ice-ufrag:2ff98ac6 +a=ice-pwd:dc22648d73c4b421274f31c1953828d4 +a=fingerprint:sha-256 F7:52:A3:46:A4:C3:99:36:83:05:7A:8F:B6:CC:A9:17:0A:45:04:79:3D:D7:F5:39:BE:1D:F3:FF:DA:81:DB:7C +m=audio 51390 UDP/TLS/RTP/SAVPF 109 0 8 101 +c=IN IP4 195.8.117.59 a=rtpmap:109 opus/48000/2 a=ptime:20 a=rtpmap:0 PCMU/8000 @@ -14,17 +14,21 @@ a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv -a=candidate:0 1 UDP 2113667327 10.10.1.144 54908 typ host -a=candidate:1 1 UDP 2113667327 192.168.1.161 52081 typ host -a=candidate:2 1 UDP 2113667327 195.8.117.161 54978 typ host -a=candidate:0 2 UDP 2113667326 10.10.1.144 58499 typ host -a=candidate:1 2 UDP 2113667326 192.168.1.161 33161 typ host -a=candidate:2 2 UDP 2113667326 195.8.117.161 36491 typ host +a=setup:actpass +a=candidate:0 1 UDP 2122252543 10.10.1.90 60221 typ host +a=candidate:1 1 UDP 1686110207 195.8.117.200 60221 typ srflx raddr 10.10.1.90 rport 60221 +a=candidate:2 1 UDP 8388607 195.8.117.59 51390 typ relay raddr 195.8.117.59 rport 51390 +a=candidate:3 1 UDP 2122187007 192.168.150.1 38505 typ host +a=candidate:0 2 UDP 2122252542 10.10.1.90 55368 typ host +a=candidate:1 2 UDP 1686110206 195.8.117.200 55368 typ srflx raddr 10.10.1.90 rport 55368 +a=candidate:2 2 UDP 8388606 195.8.117.59 51391 typ relay raddr 195.8.117.59 rport 51391 +a=candidate:3 2 UDP 2122187006 192.168.150.1 46478 typ host +a=rtcp-mux <-> (22 headers 22 lines) --- +--- (22 headers 26 lines) --- Sending to 195.8.117.60:5060 (no NAT) Sending to 195.8.117.60:5060 (no NAT) -Using INVITE request as basis request - kbr110264479udsqistu +Using INVITE request as basis request - hqs8q0vi6pgckcu59a8r Found peer 'example.org' for 'anonymous' from 195.8.117.60:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 109 @@ -35,5 +39,53 @@ Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 -[Aug 25 14:59:29] WARNING[3938][C-0003]: chan_sip.c:10535 process_sdp: Rejecting secure audio stream without encryption details: audio 54908 UDP/TLS/RTP/SAVPF 109 0 8 101 +Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) +Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) +Peer audio RTP is at port 195.8.117.59:51390 -- _ -- 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] force_avp ignored?
I'm using v11.11 I tried setting: force_avp=yes in a SIP peer in sip.conf and it seems to be ignored. The WebRTC client sends an INVITE with "RTP/SAVPF" and Asterisk is still sending back 183 and 200 responses with the UDP/TLS/RTP/SAVPF string Are there some limitations with this option or does it depend on any other settings? Is there any debugging I can enable to understand what is going wrong? -- _ -- 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] chan_motif / res_xmpp problems
On 22/07/14 18:20, Joshua Colp wrote: > Daniel Pocock wrote: > > > >> >> FYI, I'm using the Debian packages, latest is 11.10.2~dfsg-1~bpo70+1 >> >> Has the chan_motif / xmpp / ICE stuff changed significantly in 12.x >> releases? > > Nope. > >> Is there any way I can enable ICE debugging? > > Not within 11. In 12 there is a module as part of the PJSIP work which > forwards logging messages from the PJ core into Asterisk log messages. > Has ice-udp been tested against Jitsi already? If not, could you please comment on the clients it has been tested with so I can see if they work against my Asterisk setup? -- _ -- 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] chan_motif / res_xmpp problems
On 21/07/14 15:12, Daniel Pocock wrote: > On 21/07/14 14:33, Joshua Colp wrote: >> Daniel Pocock wrote: >>> >>> I've now replicated my setup on a host with a single IPv4 address and I >>> am still having trouble with the ICE negotiation. >>> >>> I am trying to call from Jitsi to Asterisk through a Prosody XMPP >>> server. Asterisk successfully registers with the XMPP server and >>> appears to be available in the buddy list in Jitsi. Jitsi is being run >>> with the "-4" command line option to use IPv4 only just in case Asterisk >>> doesn't like to see IPv6 ICE candidates. >>> >>> I try clicking to make an audio-only call from Jitsi. In the Asterisk >>> logging (xmpp set debug on) I see the incoming "session-initiate" XML >>> stanza but Asterisk does not send any XML back. >>> >>> I definitely have "icesupport=yes" in rtp.conf and I've tried it with >>> and without specifying a TURN server from each end. >>> >>> Is this working for anybody? >> >> What does your motif.conf configuration file contain? If it is not >> configured then it will not be associated with the account and the >> Jingle support will not be present. >> > > It is largely based on the default config: > > > [default](!) > disallow=all > allow=ulaw > allow=h264 > context=incoming-motif ; Default context that incoming sessions will land in > > ;maxicecandidates = 10 ; Maximum number of ICE candidates we will offer > ;maxpayloads = 30 ; Maximum number of payloads we will offer > > [asterisk](default) > disallow=all > allow=alaw > allow=ulaw > transport=ice-udp > connection=asterisk > context=incoming_xmpp > > > > and in xmpp.conf: > > [asterisk] > type=client > serverhost=some-host > username=asterisk@some-host > secret=-- > usetls=yes > usesasl=yes > status=available > statusmessage="I may be available" > timeout=5 > > FYI, I'm using the Debian packages, latest is 11.10.2~dfsg-1~bpo70+1 Has the chan_motif / xmpp / ICE stuff changed significantly in 12.x releases? Is there any way I can enable ICE debugging? -- _ -- 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] chan_motif / res_xmpp problems
On 21/07/14 14:33, Joshua Colp wrote: > Daniel Pocock wrote: >> >> I've now replicated my setup on a host with a single IPv4 address and I >> am still having trouble with the ICE negotiation. >> >> I am trying to call from Jitsi to Asterisk through a Prosody XMPP >> server. Asterisk successfully registers with the XMPP server and >> appears to be available in the buddy list in Jitsi. Jitsi is being run >> with the "-4" command line option to use IPv4 only just in case Asterisk >> doesn't like to see IPv6 ICE candidates. >> >> I try clicking to make an audio-only call from Jitsi. In the Asterisk >> logging (xmpp set debug on) I see the incoming "session-initiate" XML >> stanza but Asterisk does not send any XML back. >> >> I definitely have "icesupport=yes" in rtp.conf and I've tried it with >> and without specifying a TURN server from each end. >> >> Is this working for anybody? > > What does your motif.conf configuration file contain? If it is not > configured then it will not be associated with the account and the > Jingle support will not be present. > It is largely based on the default config: [default](!) disallow=all allow=ulaw allow=h264 context=incoming-motif ; Default context that incoming sessions will land in ;maxicecandidates = 10 ; Maximum number of ICE candidates we will offer ;maxpayloads = 30 ; Maximum number of payloads we will offer [asterisk](default) disallow=all allow=alaw allow=ulaw transport=ice-udp connection=asterisk context=incoming_xmpp and in xmpp.conf: [asterisk] type=client serverhost=some-host username=asterisk@some-host secret=-- usetls=yes usesasl=yes status=available statusmessage="I may be available" timeout=5 -- _ -- 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] chan_motif / res_xmpp problems
I've now replicated my setup on a host with a single IPv4 address and I am still having trouble with the ICE negotiation. I am trying to call from Jitsi to Asterisk through a Prosody XMPP server. Asterisk successfully registers with the XMPP server and appears to be available in the buddy list in Jitsi. Jitsi is being run with the "-4" command line option to use IPv4 only just in case Asterisk doesn't like to see IPv6 ICE candidates. I try clicking to make an audio-only call from Jitsi. In the Asterisk logging (xmpp set debug on) I see the incoming "session-initiate" XML stanza but Asterisk does not send any XML back. I definitely have "icesupport=yes" in rtp.conf and I've tried it with and without specifying a TURN server from each end. Is this working for anybody? -- _ -- 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] chan_motify / res_xmpp bind address?
I have a multi-homed machine (quite a few IP addresses on one of the interfaces) For SIP I found that using externaddr in sip.conf would make it much more reliable with ICE and RTP using the correct IP Is there an equivalent setting for XMPP / motif.conf? I saw bindaddr in gtalk.conf but it doesn't appear to be mentioned in the source code for chan_motif -- _ -- 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] DTLS setting impacts encryption setting
If I understand correctly, setting encryption=no means that Asterisk will make outgoing calls without encryption, but will be happy to accept incoming calls regardless of whether the caller wants encryption or not If encryption=yes, then Asterisk not only uses encryption for the outgoing calls but it will refuse to accept incoming calls unless they use encryption too If I have encryption=no dtlsenable=yes the DTLS support works but Asterisk will no longer accept incoming calls using regular RTP/AVP. These messages appear in the console and the call is rejected with code 488: [Jan 28 11:08:42] WARNING[24673][C-0009]: chan_sip.c:10496 process_sdp: Processed DTLS [FALSE] [Jan 28 11:08:42] WARNING[24673][C-0009]: chan_sip.c:10529 process_sdp: We are requesting SRTP for audio, but they responded without it! I realise not everybody would set encryption=no in this situation, I'm simply trying to make it work for all possible callers to the SIP5060.net test numbers at http://www.sip5060.net/test-calls -- _ -- 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] Sample config files installed to /etc
The sample config files in the Asterisk distribution and packages are really good for getting the demo up and running quickly, for example, to extend the demo to run behind a WebRTC proxy only required about 6 lines of extra code to define a peer in sip.conf and enable TCP However, I'm not sure that they should be installed by default by packages. Most package managers provide a way to diff the files and merge new config options that appear in a new release However, because a lot of things have to be ripped out of the default config to harden it and disable the demo, a simple diff doesn't really help somebody upgrading to a new version, because usually they've altered the files quite dramatically I'd suggest that the config for the demo could be placed under /usr/share/asterisk/samples while the configs installed to /etc/asterisk should be fairly minimal My own workaround at the moment involves tracking the released configs in a git repository and tracking my changes on a branch. However, working with the package manager diff output would help a lot more people and make it much more like other packages they are familiar with. -- _ -- 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] realtime sip.conf and templates
On 06/06/13 15:51, Daniel Pocock wrote: > Is the template capability in sip.conf compatible with realtime sip.conf > entries such as users in a database? > > I notice that contrib/realtime/mysql/sippeers.sql and the wiki page > don't mention a template column: > > https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure > > while some third-party examples do suggest that a column named > "template" is permitted: > > http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip I have actually tried adding that column "template" into sippeers and setting the value as the name of a template from my sip.conf - on Asterisk 11.4, it seems to ignore the column. If there is a way to do this, it would be useful to have it in the wiki. -- _ -- 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] realtime sip.conf and templates
Is the template capability in sip.conf compatible with realtime sip.conf entries such as users in a database? I notice that contrib/realtime/mysql/sippeers.sql and the wiki page don't mention a template column: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure while some third-party examples do suggest that a column named "template" is permitted: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip -- _ -- 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] md5secret, secret and ha1b hash calculation?
Kamailio has both a ha1 and ha1b column in it's user schema: ha1 = H(A1) = MD5(user:realm:password) ha1b = H(A1b) = MD5(user@realm:realm:password) This is intended to support some devices that append @realm to the user and/or to allow users to put either "user-part only" or "user@domain" into the auth-user field of their UA. Can anybody comment on the following: - if secret is configured, and an auth header comes in with auth_user="user@realm", does Asterisk internally make the H(A1b) calculation instead of H(A1) from the secret it has for the user? - if yes, does that mean it would be relatively easy to add an extra parameter, md5secretb for example, that mimics ha1b and allows cleartext secrets to be abolished? - what has been observed in practice? Are there any devices actively behaving like this or is it purely a legacy thing? In repro, we decided to store both versions of every hash when a user is added/updated, but only ha1 is consulted by the authentication code. The ha1b is simply stored to avoid the hassle of resetting all passwords if support for ha1b is completed in future. Regards, Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] offline builds - mp3 [patch]
On 04/06/13 19:13, Tzafrir Cohen wrote: > On Tue, Jun 04, 2013 at 06:44:43PM +0200, Daniel Pocock wrote: >> On 04/06/13 18:37, Tzafrir Cohen wrote: >>> On Tue, Jun 04, 2013 at 12:49:35PM +0200, Daniel Pocock wrote: >>>> >>>> >>>> As mentioned in the thread about MP3, I found that the rpmbuild process >>>> demands network access, e.g. to access the mp3 code in SVN. >>>> >>>> Some people need to build on isolated networks though >>>> >>>> I've attached a patch that allows the MP3 code to be placed in /tmp >>>> before the build starts, then svn will not be used during the build. If >>>> it finds /tmp/asterisk-contrib-mp3.tar.gz then it will be used instead >>>> of going to SVN >>>> >>>> I'm not sure if there are other build steps that access the network, >>>> this one was more obvious because I was trying to build on a fresh VM >>>> without any svn client >>> >>> I'm sure you're aware of: >>> http://patch-tracker.debian.org/patch/series/view/asterisk/1:1.8.13.1~dfsg-3/mpglib >>> >> >> The notes suggest that MP3 patent issues are a factor so I guessed >> that's why it is excluded from the tarball >> >> When building with rpmbuild the tarball is usually not unpacked >> manually, hence my own proposed patch looks in /tmp for the mp3 code - >> it could just as easily use your the patch from Debian as an input >> though, as long as it can be found in /tmp or some other predefined >> location. > > How would you do that in a proper chrooted build? > > The proper fix would be to applow to use a newer version of mpglib that > is included with some distributions. > I'm not claiming that this was a proper fix - it is just a bare minimum to allow offline builds with rpmbuild. Although it has the feeling of a hack, it doesn't prevent anybody implementing a more elegant solution in future. On the other hand, I was thinking about simply making up my own branch of the code and a repackaged tarball and maybe even publishing some convenient binary RPMs for everybody who wants to try this. I realise that asterisk-11.deb packages are a work in progress too, I didn't want to put pressure on people to finish them, that's why I've just been talking about the RPMs today. -- _ -- 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] offline builds - mp3 [patch]
On 04/06/13 18:37, Tzafrir Cohen wrote: > On Tue, Jun 04, 2013 at 12:49:35PM +0200, Daniel Pocock wrote: >> >> >> As mentioned in the thread about MP3, I found that the rpmbuild process >> demands network access, e.g. to access the mp3 code in SVN. >> >> Some people need to build on isolated networks though >> >> I've attached a patch that allows the MP3 code to be placed in /tmp >> before the build starts, then svn will not be used during the build. If >> it finds /tmp/asterisk-contrib-mp3.tar.gz then it will be used instead >> of going to SVN >> >> I'm not sure if there are other build steps that access the network, >> this one was more obvious because I was trying to build on a fresh VM >> without any svn client > > I'm sure you're aware of: > http://patch-tracker.debian.org/patch/series/view/asterisk/1:1.8.13.1~dfsg-3/mpglib > The notes suggest that MP3 patent issues are a factor so I guessed that's why it is excluded from the tarball When building with rpmbuild the tarball is usually not unpacked manually, hence my own proposed patch looks in /tmp for the mp3 code - it could just as easily use your the patch from Debian as an input though, as long as it can be found in /tmp or some other predefined location. -- _ -- 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] blog about WebRTC + TLS + Asterisk 11
I've now prepared a blog about my experience setting up Asterisk 11 with repro as a SIP proxy for WebSocket clients: http://danielpocock.com/using-resiprocate-to-connect-asterisk-webrtc In particular, the focus is on the use of packages because that makes it faster for more people to deploy identical working systems. To get the demo running for the WebSocket client, I really only needed to change about 5 lines in sip.conf - all other configuration is the default - the more painful step is rebuilding the packages with SRTP support. Regards, Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Google/XMPP and Asterisk/XMPP
Given the recent announcement about Google slimming their support for public interconnection with XMPP, can anybody comment on where this leaves the XMPP support in Asterisk? In particular, I notice many of the references to XMPP on the wiki link to https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google which seems to suggest that XMPP support and Google Talk support are one and the same. Is the XMPP support only tuned for Google variation of XMPP/ICE/TURN, or is it supported for all open Jabber servers? I currently run 1.8 (before chan_motif) against ejabberd -- _ -- 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] offline builds - mp3 [patch]
As mentioned in the thread about MP3, I found that the rpmbuild process
demands network access, e.g. to access the mp3 code in SVN.
Some people need to build on isolated networks though
I've attached a patch that allows the MP3 code to be placed in /tmp
before the build starts, then svn will not be used during the build. If
it finds /tmp/asterisk-contrib-mp3.tar.gz then it will be used instead
of going to SVN
I'm not sure if there are other build steps that access the network,
this one was more obvious because I was trying to build on a fresh VM
without any svn client
--- contrib/scripts/get_mp3_source.sh.orig 2013-06-04 12:41:08.222602824 +0200
+++ contrib/scripts/get_mp3_source.sh 2013-06-04 12:40:45.218602846 +0200
@@ -9,6 +9,15 @@
exit 1
fi
+LOCAL_COPY=/tmp/asterisk-contrib-mp3.tar.gz
+if [ -f ${LOCAL_COPY} ]; then
+echo "***"
+echo "Found ${LOCAL_COPY} - unpacking it, not downloading"
+echo "***"
+tar xzf ${LOCAL_COPY}
+exit 0
+fi
+
svn export http://svn.digium.com/svn/thirdparty/mp3/trunk addons/mp3 $@
exit 0
--
_
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Re: [asterisk-users] RHEL6 packages - SRTP support? [patch]
On 03/06/13 23:04, Daniel Pocock wrote: > On 03/06/13 19:18, Jason Parker wrote: >> >> On 06/03/2013 12:03 PM, Daniel Pocock wrote: >>> I tried building manually from the source RPM >>> >>> Before running rpmbuild, I installed libsrtp-devel and I notice that >>> res_srtp.so is generated during the build >>> >>> However, the rpmbuild fails for other reasons (see the other email I >>> sent to the list about mISDNutils-devel and other spec file errors) >>> >>> Can you confirm the exact procedure you recommend for rpmbuild on a >>> CentOS6 system >> rpmbuild --rebuild --without tds --without misdn somepackage.src.rpm >> > Now trying it on a fresh VM (CentOS 6 + EPEL6, freshly built) > > I have these errors during installation of build dependencies: > > > Installing : kmod-dahdi-linux-fwload-vpmadt032-2.6.2-1_centos6.2.6.32 > 20/73 > WARNING: > /lib/modules/2.6.32-358.6.2.el6.x86_64/weak-updates/dahdi-linux-fwload-vpmadt032/dahdi_vpmadt032_loader.ko > needs unknown symbol vpmadtreg_unregister > WARNING: > /lib/modules/2.6.32-358.6.2.el6.x86_64/weak-updates/dahdi-linux-fwload-vpmadt032/dahdi_vpmadt032_loader.ko > needs unknown symbol vpmadtreg_register > WARNING: > /lib/modules/2.6.32-358.el6.x86_64/weak-updates/dahdi-linux-fwload-vpmadt032/dahdi_vpmadt032_loader.ko > needs unknown symbol vpmadtreg_unregister > WARNING: > /lib/modules/2.6.32-358.el6.x86_64/weak-updates/dahdi-linux-fwload-vpmadt032/dahdi_vpmadt032_loader.ko > needs unknown symbol vpmadtreg_register > > > > and then rpmbuild fails with: > > checking for gcc... no > checking for cc... no > configure: error: in `/root/rpmbuild/BUILD/asterisk-11.4.0': > configure: error: no acceptable C compiler found in $PATH > See `config.log' for more details > error: Bad exit status from /var/tmp/rpm-tmp.isB97h (%build) > > > RPM build errors: > Bad exit status from /var/tmp/rpm-tmp.isB97h (%build) > > > > so I think to be added to the build dependencies in the spec file. > > > Then there is a more cryptic failure: > > checking how to run the C++ preprocessor... /lib/cpp > configure: error: in `/root/rpmbuild/BUILD/asterisk-11.4.0': > configure: error: C++ preprocessor "/lib/cpp" fails sanity check > See `config.log' for more details > error: Bad exit status from /var/tmp/rpm-tmp.q7wr5n (%build) > > > > config.log reveals that g++ is missing. The build dependency is gcc-c++ > - I install that and it fails due to missing make > > Later, there is another failure due to missing subversion. > > Altogether, these are the missing lines for the spec file: > > BuildRequires: gcc > BuildRequires: gcc-c++ > BuildRequires: make > BuildRequires: subversion > > > although I would recommend not having a build dependency on SVN or > network access, some people like to build on secured machines without > network access. > > > Eventually, I end up with the same failure I had before: > > > RPM build errors: > File not found: > /root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/cdr_adaptive_odbc.so > File not found: > /root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/cdr_odbc.so > File not found: > /root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/cel_odbc.so > File not found: > /root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/func_odbc.so > File not found: > /root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/res_config_odbc.so > File not found: > /root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/res_odbc.so > > > > Now I've tried: > > > rpmbuild --rebuild \ > --without tds \ > --without misdn \ > --without odbc \ > asterisk-11.4.0-1_centos6.src.rpm > > and on the rebuild, I get > > gzip: ./usr/share/man/man8/autosupport.8 already exists; do you wish to > overwrite (y or n)? > gzip: ./usr/share/man/man8/astgenkey.8 already exists; do you wish to > overwrite (y or n)? > > which suggests that `make clean' didn't really clean up after the last > attempt > > Finally, I get: > > Checking for unpackaged file(s): /usr/lib/rpm/check-files > /root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64 > error: Installed (but unpackaged) file(s) found: >/usr/lib64/asterisk/modules/func_speex.so >/usr/lib64/asterisk/modules/res_srtp.so > > > RPM build errors: > Installed (but unpackaged) file(s) found: >/usr/lib64/ast
[asterisk-users] Asterisk 11 + repro WebRTC tested
I've just done a test with a WebRTC client connecting to the repro proxy with the SIP messages relayed over TCP to Asterisk Asterisk successfully answers the call using SAVPF, SRTP and ICE. The client is greeted by the demo This was tested in the Asterisk 11 environment described in my earlier email about SRTP build issues on the asterisk-users list. This is quite useful because it proves that Asterisk doesn't have to be exposed as the HTTP WebSocket server: all the WebSocket handshake and message parsing is done by the proxy. Specific versions tested: - Asterisk 11.4 built from SRPM on CentOS 6 + EPEL6 - repro 1.9.0~alpha0 package from Debian experimental - JsSIP `tryit' client - Google Chrome Just some more notes about problems encountered with the Asterisk SRPM: it doesn't seem to know anything about /usr/share/asterisk/sounds - even though I install both the gsm and ulaw sounds RPMs, it always gives errors such as file.c:701 ast_openstream_full: File demo-congrats does not exist in any format I manually edited extensions.conf to include the full absolute paths and then it works, e.g: BackGround(/usr/share/asterisk/sounds/demo-congrats) -- _ -- 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] RHEL6 packages - SRTP support? [patch]
On 03/06/13 19:18, Jason Parker wrote:
>
>
> On 06/03/2013 12:03 PM, Daniel Pocock wrote:
>> I tried building manually from the source RPM
>>
>> Before running rpmbuild, I installed libsrtp-devel and I notice that
>> res_srtp.so is generated during the build
>>
>> However, the rpmbuild fails for other reasons (see the other email I
>> sent to the list about mISDNutils-devel and other spec file errors)
>>
>> Can you confirm the exact procedure you recommend for rpmbuild on a
>> CentOS6 system
> rpmbuild --rebuild --without tds --without misdn somepackage.src.rpm
>
Now trying it on a fresh VM (CentOS 6 + EPEL6, freshly built)
I have these errors during installation of build dependencies:
Installing : kmod-dahdi-linux-fwload-vpmadt032-2.6.2-1_centos6.2.6.32
20/73
WARNING:
/lib/modules/2.6.32-358.6.2.el6.x86_64/weak-updates/dahdi-linux-fwload-vpmadt032/dahdi_vpmadt032_loader.ko
needs unknown symbol vpmadtreg_unregister
WARNING:
/lib/modules/2.6.32-358.6.2.el6.x86_64/weak-updates/dahdi-linux-fwload-vpmadt032/dahdi_vpmadt032_loader.ko
needs unknown symbol vpmadtreg_register
WARNING:
/lib/modules/2.6.32-358.el6.x86_64/weak-updates/dahdi-linux-fwload-vpmadt032/dahdi_vpmadt032_loader.ko
needs unknown symbol vpmadtreg_unregister
WARNING:
/lib/modules/2.6.32-358.el6.x86_64/weak-updates/dahdi-linux-fwload-vpmadt032/dahdi_vpmadt032_loader.ko
needs unknown symbol vpmadtreg_register
and then rpmbuild fails with:
checking for gcc... no
checking for cc... no
configure: error: in `/root/rpmbuild/BUILD/asterisk-11.4.0':
configure: error: no acceptable C compiler found in $PATH
See `config.log' for more details
error: Bad exit status from /var/tmp/rpm-tmp.isB97h (%build)
RPM build errors:
Bad exit status from /var/tmp/rpm-tmp.isB97h (%build)
so I think to be added to the build dependencies in the spec file.
Then there is a more cryptic failure:
checking how to run the C++ preprocessor... /lib/cpp
configure: error: in `/root/rpmbuild/BUILD/asterisk-11.4.0':
configure: error: C++ preprocessor "/lib/cpp" fails sanity check
See `config.log' for more details
error: Bad exit status from /var/tmp/rpm-tmp.q7wr5n (%build)
config.log reveals that g++ is missing. The build dependency is gcc-c++
- I install that and it fails due to missing make
Later, there is another failure due to missing subversion.
Altogether, these are the missing lines for the spec file:
BuildRequires: gcc
BuildRequires: gcc-c++
BuildRequires: make
BuildRequires: subversion
although I would recommend not having a build dependency on SVN or
network access, some people like to build on secured machines without
network access.
Eventually, I end up with the same failure I had before:
RPM build errors:
File not found:
/root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/cdr_adaptive_odbc.so
File not found:
/root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/cdr_odbc.so
File not found:
/root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/cel_odbc.so
File not found:
/root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/func_odbc.so
File not found:
/root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/res_config_odbc.so
File not found:
/root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/res_odbc.so
Now I've tried:
rpmbuild --rebuild \
--without tds \
--without misdn \
--without odbc \
asterisk-11.4.0-1_centos6.src.rpm
and on the rebuild, I get
gzip: ./usr/share/man/man8/autosupport.8 already exists; do you wish to
overwrite (y or n)?
gzip: ./usr/share/man/man8/astgenkey.8 already exists; do you wish to
overwrite (y or n)?
which suggests that `make clean' didn't really clean up after the last
attempt
Finally, I get:
Checking for unpackaged file(s): /usr/lib/rpm/check-files
/root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64
error: Installed (but unpackaged) file(s) found:
/usr/lib64/asterisk/modules/func_speex.so
/usr/lib64/asterisk/modules/res_srtp.so
RPM build errors:
Installed (but unpackaged) file(s) found:
/usr/lib64/asterisk/modules/func_speex.so
/usr/lib64/asterisk/modules/res_srtp.so
so I added those two items to the spec file and finally I have a build.
I attach a diff for fixing the spec file, it fixes all these issues
except the `make clean'
Regards,
Daniel
--- SPECS/asterisk.spec.orig 2013-06-03 22:52:12.302227936 +0200
+++ SPECS/asterisk.spec 2013-06-03 23:02:33.899224528 +0200
@@ -70,6 +70,10 @@
Requires: %{name}-doc = %{actversion}
Requires: %{name}-voicemail = %{actversion}-%{release}
Requires: asterisk-sounds-core-en-gsm
+BuildRequires: gcc
+BuildRequires: gcc-c++
+BuildRequires: make
+BuildRequires: subversion
BuildRequi
Re: [asterisk-users] RHEL6 packages - SRTP support?
On 03/06/13 18:46, Jason Parker wrote: > The packages currently do not support SRTP. > I tried building manually from the source RPM Before running rpmbuild, I installed libsrtp-devel and I notice that res_srtp.so is generated during the build However, the rpmbuild fails for other reasons (see the other email I sent to the list about mISDNutils-devel and other spec file errors) Can you confirm the exact procedure you recommend for rpmbuild on a CentOS6 system? -- _ -- 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] missing build dependency / mISDNutils-devel and other errors
Building from the source RPM I get an error mISDNuser-devel is needed I was able to obtain all the other build dependencies from EPEL 6, but that one doesn't appear to existing in EPEL or in packages.asterisk.org I then tried adding --nodeps to the rpmbuild command: rpmbuild --rebuild --nodeps asterisk-11.4.0-1_centos6.src.rpm Running as a normal user, the build fails on the line mv asterisk-sources-11.4.0-1_centos6.make.err /var/log/ due to the permissions on /var/log/ As a hack, I set the perms on /var/log/ to 0777 and try again and then it fails with some "file not found" messages at the end of the build, missing: cdr_adaptive_odbc.so cdr_odbc.so cel_odbc.so func_odbc.so res_config_odbc.so res_odbc.so I checked my system, unixODBC-devel is present, so it appears to be a build config issue with the spec file -- _ -- 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] RHEL6 packages - SRTP support?
I tried installing the Asterisk 11 RHEL6 packages from packages.asterisk.org I followed this guide: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages The SRTP support appears to be missing though. I notice libsrtp was not automatically installed as a dependency, and no srtp module exists under /usr/lib64/asterisk/modules Is it still necessary to do a source build every time SRTP is needed? Or is the srtp module distributed in some other rpm? -- _ -- 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] asterisk debian package and digium repository
On 07/08/12 23:11, Rusty Newton wrote: > On 8/7/2012 7:27 AM, Paul Belanger wrote: >> On 12-08-07 03:31 AM, ml asterisk wrote: >>> Hi, >>> >>> I used to install asterisk on debian squeeze with digium repository. >>> The last build of asterisk available is 1.8.11.1. >>> Is this repository discontinued ? >>> >> Since leaving Digium they have become unmaintained. If you are >> interested in helping out, you might want to reach out to >> #asterisk-dev or asterisk-dev mailing list. >> > Matt Jordan has been working on getting it sorted out, but we only > have so much time and resources. Anyone who wants to step up and help > out, don't be shy! As Paul said: #asterisk-dev or asterisk-dev mailing > list. > Can anybody clarify the current situation with packages? The wiki says they stopped being supported in March 2012: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages but a) it's not clear if it is talking about the asterisk.org hosted packages or the debian.org hosted packages b) the asterisk.org hosted .deb packages appear to be updated in November 2012 c) Debian 7 was released with Asterisk 1.8 at the beginning of May 2013, that means it is supported as long as Debian 7 is current + 1 year (approximately 3 years total) d) Ubuntu appears to be carrying 1.8 in their best-efforts supported "universe" catalog Furthermore, I notice that on packages.asterisk.org RHEL6 has Asterisk 11 packages but Debian/Ubuntu (pool directory) still has 1.8 Having a look at Debian's PTS, I found that Asterisk 11 packaging is a work-in-progress, it may be possible for people to obtain it from Debian's SVN or git and build packages with dpkg-buildpackage -- _ -- 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] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
On 01/04/13 22:06, Joshua Colp wrote: > Daniel Pocock wrote: >> Thanks for the fast reply. I agree backporting full support for AVPF >> would not be justified for many use cases (including my own). What I >> was more curious about is whether the F can be tolerated (in other >> words, ignored or silently removed), as described here: > > From a code perspective, it could. Still not something I would be > comfortable with putting in Asterisk 1.8. > >> http://www.ietf.org/mail-archive/web/rtcweb/current/msg01145.html >> "1) RTCWEB end-point will always signal AVPF or SAVPF. I signalling >> gateway to legacy will change that by removing the F to AVP or SAVP." >> >> and whether such behavior is possible even without setting avpf=yes on a >> per-peer basis? > > This is fine for incoming but what about outgoing to a device? > Excellent question... I've seen one of my Polycom devices reboot itself each time it receives a raw SDP from WebRTC, so if such a hack is implemented, I'd guess that stripping the F is better than ignoring it. -- _ -- 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] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
On 31/03/13 23:43, Joshua Colp wrote: > Daniel Pocock wrote: >> I'm trying to call from DruCall to Asterisk and I get this error: >> >> WARNING[11021]: chan_sip.c:8687 process_sdp: Error in codec string 'F >> 103 104 111 0 8 107 106 105 13 126' >>== Problem setting up ssl connection: >> error::lib(0):func(0):reason(0) >> >> >> I'm guessing my Asterisk is too old (it is 1.8 from Debian). Can you >> confirm which version is needed to parse a media descriptor with SAVPF? >> Do I need to upgrade all the way to v11 with WebRTC support, or was >> avpf support added in some intermediate version? > > Asterisk 1.8 does not have any knowledge of AVPF, and since it's a new > feature it was only added to Asterisk 11. You could try to backport the > changes but chan_sip has changed quite a bit, so it could be rough. Thanks for the fast reply. I agree backporting full support for AVPF would not be justified for many use cases (including my own). What I was more curious about is whether the F can be tolerated (in other words, ignored or silently removed), as described here: http://www.ietf.org/mail-archive/web/rtcweb/current/msg01145.html "1) RTCWEB end-point will always signal AVPF or SAVPF. I signalling gateway to legacy will change that by removing the F to AVP or SAVP." and whether such behavior is possible even without setting avpf=yes on a per-peer basis? -- _ -- 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] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
On 17/12/12 13:34, Joshua Colp wrote: > Barco You wrote: >> Dear All, > > Hola, > >> I use sipml5 to register two users from browser and the two clients >> are successfully connected. But when I made a call from one of the >> users, the other user doen'st have call notification and for a while the >> calling process ended. I check the /var/log/asterisk/messages got the >> following log: >> >> [Dec 17 14:54:11] WARNING[11471][C-] chan_sip.c: Received SAVPF >> profle in audio offer but AVPF is not enabled: audio 52760 RTP/SAVPF 103 >> 104 0 8 107 106 105 13 126 >> [Dec 17 14:54:11] WARNING[11471][C-] chan_sip.c: Received SAVPF >> profle in video offer but AVPF is not enabled: video 52760 RTP/SAVPF 100 >> 101 102 >> [Dec 17 14:54:11] WARNING[11471][C-] chan_sip.c: Insufficient >> information in SDP (c=)... > > As the warning states - you haven't enabled AVPF support. This is > generally done on a per-peer basis using "avpf=yes" in the configuration. > > I would suggest you follow > https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support since > there may be other things you have missed. > I'm trying to call from DruCall to Asterisk and I get this error: WARNING[11021]: chan_sip.c:8687 process_sdp: Error in codec string 'F 103 104 111 0 8 107 106 105 13 126' == Problem setting up ssl connection: error::lib(0):func(0):reason(0) I'm guessing my Asterisk is too old (it is 1.8 from Debian). Can you confirm which version is needed to parse a media descriptor with SAVPF? Do I need to upgrade all the way to v11 with WebRTC support, or was avpf support added in some intermediate version? Also, I'm using a SIP proxy and it takes care of handling all the WebRTC connections and proxying the requests into a normal TCP/TLS connection to Asterisk. I was hoping to avoid opening up WebRTC access directly on Asterisk. One effect this has is that I can't control the `avpf=yes' setting on a per-peer basis, as the proxy is carrying requests from various types of peer, some public, some private. Is there any outright reason Asterisk can't support (S)AVPF on demand? -- _ -- 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] gtalk only working with ulaw???
On 14/01/13 23:31, Joshua Colp wrote: > Daniel Pocock wrote: >> >> I've set up a peer to use G.722 only and tried to make it talk to an >> Asterisk box >> >> Asterisk always rejects the call with the following error: > > chan_gtalk was written to only support a limited number of codecs, not > the full set that Asterisk is capable of. chan_motif does not have this > limitation. > Thanks for the fast feedback, I think I found it here http://blogs.digium.com/2012/07/24/asterisk-11-development-the-motive-for-motif/ I normally run the Debian packages, is there any chance you will provide this module to be built standalone with 1.8? -- _ -- 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] gtalk only working with ulaw???
I've set up a peer to use G.722 only and tried to make it talk to an Asterisk box Asterisk always rejects the call with the following error: [Jan 14 22:20:16] WARNING[32653]: chan_gtalk.c:1343 gtalk_newcall: Capabilities don't match : us - 0x4 (ulaw), peer - 0x1000 (g722), combined - 0x0 (nothing) Yet I've set gtalk.conf to only allow G.722, is there some other place where chan_gtalk could be getting it's configuration? gtalk.conf: [general] context=jingle_guest bindaddr=A.B.C.D allowguest=yes disallow=all allow=g722 [guest] ; special account for options on guest account disallow=all allow=g722 context=jingle_guest connection=asterisk and here is jabber.conf: [general] debug=yes autoprune=no autoregister=yes [asterisk] type=client serverhost=jabber.example.org username=u...@example.org/asterisk secret=1234 priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage="Daniel" timeout=100 -- _ -- 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] Paris - mini-DebConf - VoIP - 24 November
For those using Debian/Ubuntu (and anybody else is welcome of course), there is a mini-DebConf in Paris this weekend: http://fr2012.mini.debconf.org/ There is a presentation at 16:00 about Debian's role in establishing an alternative to Skype, this will look at some of the packages available on the upcoming Debian 7 (wheezy), and strategic ways of deploying them to build a genuinely free and open cloud for real-time communications. There is no registration fee - all welcome -- _ -- 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] Asterisk as TLS server as well as TLS client
On 20/08/12 22:53, Danny Nicholas wrote: > I'm fond of the tar-config-make method that Asterisk uses. Is this possible > for reSIPprocate? If so can you provide a link? > http://www.resiprocate.org/ReSIProcate_1.8_Release You can access the download directory (use the 1.8.5 tarball) or SVN from there Any feedback is welcome, there is a link there to the reSIProcate community email lists -- _ -- 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] Asterisk as TLS server as well as TLS client
On 20/08/12 21:11, Danny Nicholas wrote: > This is all "nice and good" but the documentation all assumes that you are on > a Debian box and use MYSQL. What about us SUSE/Postgresql folks? They are both good questions, and there are partial answers: SUSE: reSIProcate can be built from source on a large number of platforms. I recently converted the upstream project to autotools, this should make it straightforward to build (and even package it) for SUSE. There has been some mention of RPM packaging on the resiprocate dev email list. I'm even working on it for OpenCSW at the moment. Postgresql: This is a bigger challenge. - Scott recently added the MySQL support for the 1.8 release, before that there was no working DB support, just BDB files. - It should probably be generalised for UNIXODBC or something like that, I actually used that approach in dynalogin. However, it will probably need someone to volunteer or present a commercial opportunity to enhance it like that. As for the guides: to make it easy, they talk about what exists today. Once the RPM packages appear in Fedora or SUSE, I will definitely update the guides, there is no hidden agenda to force people onto Debian. -- _ -- 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] Asterisk as TLS server as well as TLS client
On 20/08/12 16:23, Administrator TOOTAI wrote: > Hi, > > I have to connect 3 asterisk servers,each of them being TLS server for > his clients and connected in both way in TLS with both others asterisk, > each having hi own Common Name. Is this possible? > > I set up 2 asterik's , one server and the other client, this is OK. But > I can't deal with certificats generated on both servers. > > I tried to put tlscertfile ans tlscafile in the peer definition, each > pointing to the certificate generated by the server, but thatś not working. > > Thanks for any hint. > Asterisk doesn't seem to implement mutual TLS authentication, see the comments in this thread: http://java.net/projects/jitsi/lists/users/archive/2012-08/message/37 People who want strong TLS typically use a SIP proxy as a front-end to Asterisk, either repro or Kamailio stand out as leaders in TLS support http://www.opentelecoms.org/use-a-sip-proxy-instead-of-asterisk At the bottom, there are links to some practical guides how to use either repro or Kamailio with 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
[asterisk-users] new How-to guide: using repro SIP proxy for TLS with Asterisk
Given the limitations around Asterisk's TLS support, and all the benefits of using a SIP proxy, I've put together a rough guide about how to use the repro SIP proxy as a front-end for Asterisk connectivity with TLS peers: http://www.opentelecoms.org/using-repro-with-asterisk-or-freeswitch It works for TLS from phones, but also for full federated SIP with any other SIP-enabled domain on the public Internet. * repro does all the connectivity work (certificate validation, etc) and registration service * Asterisk sits in the background and provides applications (voicemail, queues, etc) Any feedback, questions or discussion about this is very welcome. -- _ -- 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] Debian 7/Asterisk TLS bug and others
On 11/08/12 01:26, Paul Belanger wrote: >> Is Digium officially endorsing 1.8.13 for wheezy in any way? >> > No. Digium nor the Asterisk Project has anything to do with the package > within Debian. In fact, most of the work is done by Tzafrir. I'm not referring to the actual packaging processes, but just the general strategy For example, if wheezy is released at Christmas, it could be the current version for 2 years (until end of 2014) and then another year of security updates (until end of 2015). Anyone using Debian during that period will come across Asterisk v1.8.13 It raises various issues: - with TLS use likely to grow over that time, will the problems in the current version become noticed by many more people? - will general security updates for 1.8.x continue up to at least 2015? I've raised a bug report in Debian about the general state of the TLS support and to see if it is appropriate for the long lifespan of packages in Debian - any comments on this would be really welcome http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=684649 -- _ -- 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] Debian 7/Asterisk TLS bug and others
>> Debian is very conservative about accepting updates during the `freeze' >> process - they will most likely want to see a 1.8.13.2 release with ONLY >> the most essential fixes >> >> a) is anyone else aware of these bugs? >> >> b) what essential changes should go into 1.8.13.2 for Debian? >> > We don't need to release a 1.8.13.2 release of Asterisk. Once the issue > has been fixed in the 1.8 release branch, it would just be back-ported > into a Debian patch for the package. My impression was that a 1.8.13.2 release would be as conservative as any patches back-ported for the Debian package. It's not necessary, but it might be a convenient way to achieve the same goal. Is Digium officially endorsing 1.8.13 for wheezy in any way? Is anyone officially working on this particular problem already? I was tempted to have a closer look at it, but don't want to duplicate an effort that is already underway elsewhere. -- _ -- 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] Debian 7/Asterisk TLS bug and others
Debian 7 is currently in the `freeze' status with 1.8.13 - that means Debian 7 is very likely to release 1.8.13 and be carrying it for the next 2-3 years (typical lifetime of a Debian release) I run 1.8.8. TLS has a bug: it fails to receive BYE over the TLS connection from my Polycom phone. I tried 1.8.13, the version in Debian 7, and found a more severe bug: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=683956 The TLS clients can't connect at all, this looks like a really bad regression from 1.8.8 I've looked at 1.8.(14, 15, 16-rc1) and their changelogs don't mention any fix. Debian is very conservative about accepting updates during the `freeze' process - they will most likely want to see a 1.8.13.2 release with ONLY the most essential fixes a) is anyone else aware of these bugs? b) what essential changes should go into 1.8.13.2 for Debian? -- _ -- 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] Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration Tutorial
On 06/08/12 13:48, Daniel-Constantin Mierla wrote: > * http://asipto.com/u/68 > > The tutorial focuses on how to use Asterisk's database structure to > perform authentication in Kamailio SIP server, along with user location, > nat traversal, instant messaging, presence, a.s.o., offloading > processing from Asterisk. Asterisk will still handle all the calls, > enabling rich telephony such as MoH, transcoding, ring back, IVR, etc. This is a good tutorial, but can you clarify the scope of what Kamailio will do in this configuration? - just scalability and protocol conversion (e.g. UDP with Asterisk, TLS with phones)? - does it mean Kamailio is also intended to add other services, e.g. presence and IM functionality? - any comments on using the Jabber gateway module? - is it intended for fully federated SIP, e.g. someone sets this up for example.org, and somebody else in example.com can make a call to u...@example.org, routed over the public Internet, using DNS SRV and mutual TLS? If it is intended that someone can turn on the mutual TLS mode and use it to federate their Asterisk server, then I'd like to link to it from http://www.opentelecoms.org/use-a-sip-proxy-instead-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] sip tls problem
On 06/08/12 02:59, Vladimir Mikhelson wrote: > Have you tried 1.8.15? I'm trying 1.8.13 because that is the versions currently scheduled for release in Debian 7 (wheezy) http://packages.debian.org/wheezy/asterisk If 1.8.15 contains definite solutions for TLS problems, then either a) they can be applied as patches on the Debian package of 1.8.13 b) there could be some attempt to get 1.8.15 accepted into Debian (the catalog for wheezy is technically frozen now for final testing before release, so they are not keen to accept whole new versions of packages) > SIP TLS with self-signed certificate seems to be working fine here. The > OS is CentOS 5.8 and there are no chained certificates in my environment. > > -Vladimir > The original poster was also using self-signed certs I've observed the problem using chained certs (with 1 root, 2 intermediate, and then my server cert) -- _ -- 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] sip tls problem
Package: asterisk Version: 1:1.8.13.0~dfsg-1+b1 Severity: important On 05/03/12 10:47, Wolfgang Pichler wrote: > Hi all, > > i have had sip TLS with an own signed certificate (using the > ast_tls_cert script) running on asterisk-1.8.8 - i then have updated > to 1.8.9.3 - and now i get the message "FILE * open failed!" > > I have already recreated the certificates with the script - but still no > luck... > > Does anyone here know the source of the problem ? > I'm seeing similar problems with the 1.8.13 package in Debian [Aug 5 19:05:16] WARNING[6169]: tcptls.c:235 handle_tcptls_connection: FILE * open failed! 1.8.8 was working (although it had other severe problems, for example, closing the TLS connection and not receiving a BYE, keeping channels open forever) My cert is a Thawte 123 cert, there are actually 4 certs in the chain, root at the top The log claims it loads successfully: SIP channel loading... == Parsing '/etc/asterisk/sip.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found == SIP Listening on 192.168.100.1:5060 == Using SIP CoS mark 4 SSL certificate ok With 1.8.8, this was fine With 1.8.13, I connect to the server using `openssl s_client', and it only shows the text of ONE of the certificates - it seems to repeat the same certificate four times though. This is a very bad sign. With 1.8.8, I would see ALL four certificate in the output below. $ openssl s_client -connect 192.168.100.1:5061 -showcerts CONNECTED(0003) depth=0 /O=/OU=Go to https://www.thawte.com/repository/index.html/OU=Thawte SSL123 certificate/OU=Domain Validated/CN= verify error:num=20:unable to get local issuer certificate verify return:1 depth=0 /O=/OU=Go to https://www.thawte.com/repository/index.html/OU=Thawte SSL123 certificate/OU=Domain Validated/CN= verify error:num=27:certificate not trusted verify return:1 depth=0 /O=/OU=Go to https://www.thawte.com/repository/index.html/OU=Thawte SSL123 certificate/OU=Domain Validated/CN= verify error:num=21:unable to verify the first certificate verify return:1 --- Certificate chain 0 s:/O=/OU=Go to https://www.thawte.com/repository/index.html/OU=Thawte SSL123 certificate/OU=Domain Validated/CN= i:/C=US/O=Thawte, Inc./OU=Domain Validated SSL/CN=Thawte DV SSL CA -BEGIN CERTIFICATE- MIIETDCCAzSgAwIBAgIQWppejHk2XLkg+v70FfjEujANBgkqhkiG9w0BAQUFADBe .. xlRmMVj1hUPeE+83S05bqB6mI09P3IGWUf0LfljDT5bmU/BFM0OhXaRe42sNHy1Y -END CERTIFICATE- --- Server certificate subject=/O=/OU=Go to https://www.thawte.com/repository/index.html/OU=Thawte SSL123 certificate/OU=Domain Validated/CN= issuer=/C=US/O=Thawte, Inc./OU=Domain Validated SSL/CN=Thawte DV SSL CA --- No client certificate CA names sent --- SSL handshake has read 1273 bytes and written 447 bytes --- New, TLSv1/SSLv3, Cipher is AES256-SHA Server public key is 2048 bit Secure Renegotiation IS supported Compression: NONE Expansion: NONE SSL-Session: Protocol : TLSv1 Cipher: AES256-SHA Session-ID: 0DAB4C1A6E2AC5D4A86769E8F00B469810F679CAC26CACEFC9F902F267E3490F Session-ID-ctx: Master-Key: 42C512C4D1C2AA32136F79F45A98A7D6AC99FD1579734728A9AC5C213424B2D1CEAA3749CCD22D2F4CB3400853E5EC93 Key-Arg : None Start Time: 1344190380 Timeout : 300 (sec) Verify return code: 21 (unable to verify the first certificate) -- _ -- 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] dlz-ldap-enum - expose LDAP data to Asterisk via ENUM
I've recently released a dlz ENUM module for the bind9 nameserver: http://www.opentelecoms.org/dlz-ldap-enum Basically, it handles ENUM queries from Asterisk, FreeSWITCH, repro, Kamailio, Lumicall, searches for the phone number in ENUM, and if found, returns the email address as both a SIP address and Jabber address This should make it even easier than ever before to get federated VoIP up and running using email addresses interchangeably with phone numbers. If the data already exists in LDAP as an address book, then just install bind9, install the module and you're up and running. Regards, Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding to 0.0.0.0 a security risk?
On 07/02/12 05:29, Gordon Messmer wrote: > On 02/06/2012 03:27 PM, Josh wrote: >>> Why do you see binding to 0.0.0.0 to be a security risk? >> Purely because a response from Asterisk can be received as a result of a >> connection on *any* interface on the system/machine. If I have Asterisk >> confined to, say, 2 interfaces - eth0 (10.1.1.1) and eth1 (10.2.1.1) >> then a request over a third/subsequent interface cannot be served - it >> is not normally possible. >> >> When Asterisk binds to 0.0.0.0 that is not the case and request over a >> third/subsequent interface *can* be served by Asterisk (provided the >> routing is setup properly, that is). > > All of that is true, but none of it appears to be a security concern, > specifically. If you are connecting to the public internet, then it is much more important to think about a) do you really expose your Asterisk directly, or hide it behind a SIP router such as Kamailio? b) should you be using TLS (which is connection oriented and secured with certificates) rather than UDP? Everyone who connects with a cert has been screened in some way by a CA. c) if using TLS (or even just TCP), why not have the extra security of a port-forwarding from a firewall to the Asterisk TLS port? Then no other ports or addresses on the Asterisk box are exposed. -- _ -- 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] CA Issued Certificates / TLS + SRTP
On 01/02/12 10:58, Stuart Elvish wrote: > Thanks for the clarification. I have looked at Polycom's website and > saw which phones have the latest firmware (or at least a firmware that > supports TLS) available. > > Didn't get around to the testing with the chained certificate but will > try again this evening. > > One thing that frustrates people about Polycom is the very limited list of root CAs they support - it was probably OK when they first started doing SSL, but things have changed a lot now The latest phones (e.g. IP321) have more memory than those they replace (e.g. IP320) and so they should be able to handle a larger list of built in root CAs (which Polycom can distribute through the firmware update). The obvious ones that are missing are the budget CAs: - CaCert.org (all certs are free) - startssl.com (which has some free certs) - GoDaddy These budget CAs are now supported by the various Linux distributions and Android phones, so they are clearly above a certain threshold of stability Polycom phones should also be able to handle 4096 bit certs with the extra memory, but that appears to need remediation in the firmware (I tried installing a custom 4096 bit cert and it didn't accept it) If anyone is registered with Polycom as a reseller, they can quote these issue numbers: EXT-3192 GoDaddy root CA cert https://jira.polycom.com:8443/browse/EXT-3192 EXT-3193 CACert root CA cert https://jira.polycom.com:8443/browse/EXT-3193 EXT-3238 Support for 4096 bit keys https://jira.polycom.com:8443/browse/EXT-3238 As in most commercial enterprises, the more customers who request fixes on these issues, the higher it will go on their priority list -- _ -- 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] CA Issued Certificates / TLS + SRTP
> * And, is it necessary to use both my server specific certificate and > the intermediate certificate on the telephones or will the telephones > only require the server specific certificate? The phones should already have the root certificate for Geotrust, you should not deploy intermediate roots into the phones if you can avoid it >>> If I understand this correctly (and the other emails you sent), the >>> Polycom does not need any preloaded certificates / keys, it will ask the >>> CA and then evaluate the certificate provided by Asterisk during TLS >>> setup; is that correct? Makes it much easier. (Unfortunately my Polycom >>> is a bit old so I will have to see if I can upgrade it.) By `preloaded', I mean you should not have to load any certificates or key pairs manually into the phones The phones should have the default CA certs that come in the firmware Most recent Polycom phones also have a client certificate and private key built in. This allows you to secure the provisioning process. Some of the older Polycoms went end-of-life, some don't have client certs built in, so you'll have to research all that carefully on their support site. E.g. the IP 300, IP 430 and IP 500 are too old for proper TLS, while the IP321, IP 450 and IP550 are good -- _ -- 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] SRV record for non-standard SIP port?
On 31/01/12 16:16, Gilles wrote: > Hello > > To cut down on the number of hackers trying to break into an Asterisk > server, I'd like to simply move the SIP port from the standard UDP > 5060 to something non-standard. Something more appropriate for your goal might be a move to TLS, it is definitely needed for any external connectivity This RFC provides some details: http://tools.ietf.org/html/rfc5922 The bottom line is that external SIP peers must send you their cert when they connect. SIP hackers will need to identify themselves (e.g. with credit card) to get a certificate, or they just won't be able to talk to your server. Obviously, this cuts out about 99% of the script kiddies. As a further safety measure, you could use something like repro or Kamailio as a SIP router to isolate your Asterisk from the public internet. All DNS SRV records would point at the SIP router, not Asterisk. Phones would register with the SIP router. Calls would be selectively routed to Asterisk (e.g. for voicemail) -- _ -- 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] CA Issued Certificates / TLS + SRTP
On 30/01/12 17:12, Stuart Elvish wrote: > Hi all, > > Firstly, apologies if the answer to this question should be obvious. > > I have just started working with SRTP and had a self-signed > certificate working perfectly. I have now purchased a CA signed > certificate but can't get it to work properly with Asterisk. I think I > have a configuration error. No, you've found a bug - I just posted an update about this issue yesterday, predicting people would get stuck on this issue: http://lists.digium.com/pipermail/asterisk-users/2012-January/269856.html > The certificate is a GeoTrust Rapid SSL certificate. I have received > the my server specific crt file and also an intermediate certificate. Intermediate certificates work for some user agents (e.g. my Polycom). There has been speculation that they won't work with some older UAs Ultimately, most of the budget priced certificates are signed with an intermediate cert, and OpenSSL supports it, so there is no reason Asterisk shouldn't support this. > I am not sure of the following and would greatly appreciate if someone > could give me some guidance: > * Can I specify the intermediate and .crt files separately in the > sip.conf file? (I am thinking of a process similar to Apache where you > specify three different files; server specific certificate, chain file > and key file.) No, for OpenSSL-based code (such as Asterisk), it works like this: http://lists.sip-router.org/pipermail/sr-users/2012-January/071771.html However, Asterisk needs to be patched first, as in bug 17727 > * Should the intermediate and server specific certificates be combined > into one certificate file? Yes, in the correct order Currently, Asterisk expects the key and cert together in the same file: I think that is bad, but that is the way it is: https://issues.asterisk.org/jira/browse/ASTERISK-19267 > * And, is it necessary to use both my server specific certificate and > the intermediate certificate on the telephones or will the telephones > only require the server specific certificate? The phones should already have the root certificate for Geotrust, you should not deploy intermediate roots into the phones if you can avoid it -- _ -- 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] TLS problems - patch in Jira
I've just come across this issue: https://issues.asterisk.org/jira/browse/ASTERISK-17727 I am strongly in support of TLS and I believe this issue will be a stumbling block for more and more users - because more and more CAs are using the intermediate certificate chains For example, the free startssl.com certs are trusted by Android phones now. I have a UA running on my phone against a SIP proxy with Kamailio. I have the free cert and the intermediate cert in a single pem file. It all works. As noted in the bug, there may be phones that don't supported chain certs - but that shouldn't prevent the rest of us using them. People with such phones (which are becoming the minority) can just not use chained certs. There is no reason not to apply the supplied patch - that patch for Asterisk just makes it use the same OpenSSL function that Kamailio is using to load the chain -- _ -- 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] RFC 5922 (TLS Certificates) and Asterisk
I've raised a bug report about this here: https://issues.asterisk.org/jira/browse/ASTERISK-19268 I'm just wondering who else has been investigating RFC 5922 style certificate practices? Which CAs have been able to provide appropriate certificates? What kind of interoperability testing has been done between the major products (e.g. Asterisk, Kamailio, OpenSIPS, reSIProcate/repro)? -- _ -- 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] Asterisk 1.8 Packages for Debian and Ubuntu
>> - upgrade policy - is it intended that someone who has Debian 6 with >> the existing Asterisk 1.6 packages (from Debian's maintainer) can just >> upgrade to the Digium package without moving or changing any config? > > There is nothing specific about the packages that is going to make this > situation any better or worse than any method of upgrading from Asterisk > 1.6.X to Asterisk 1.8. Issues related to version compatibility can be found > in the UPGRADE*.txt files in the Asterisk source. > > http://svn.asterisk.org/view/asterisk/trunk/UPGRADE-1.8.txt?view=markup > Apart from the 1.8 release notes though, there is no need to do any specific changes when going from the Debian-maintained 1.6 package to the Digium-maintained 1.8? I tried the packages (clean install) on one machine yesterday and I noticed that they depend on some of the asterisk packages within the Debian archive, while other packages get pulled down from the Digium archive. Is that intended? I tried to do another machine today and found that your key has gone missing from the key server: # apt-key adv --keyserver subkeys.pgp.net --recv-keys 175E41DF Executing: gpg --ignore-time-conflict --no-options --no-default-keyring --secret-keyring /etc/apt/secring.gpg --trustdb-name /etc/apt/trustdb.gpg --keyring /etc/apt/trusted.gpg --primary-keyring /etc/apt/trusted.gpg --keyserver subkeys.pgp.net --recv-keys 175E41DF gpg: requesting key 175E41DF from hkp server subkeys.pgp.net gpgkeys: key 175E41DF not found on keyserver gpg: no valid OpenPGP data found. gpg: Total number processed: 0 It was definitely there when I tried it yesterday - has it been revoked or something? -- _ -- 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] Asterisk 1.8 Packages for Debian and Ubuntu
> This effort is not intended to replace packaging of Asterisk in the > official Debian or Ubuntu repositories. Our repositories are for > providing access to major versions of Asterisk that are newer than what > is included. We are exploring ways to work as closely as possible with > the Debian and Ubuntu package maintainers to ensure that we do not > duplicate efforts and that we provide the best possible result for users > of Asterisk. Thanks for providing these - can you just clarify your policy on the following: - file locations - same layout as the regular Debian packages? - upgrade policy - is it intended that someone who has Debian 6 with the existing Asterisk 1.6 packages (from Debian's maintainer) can just upgrade to the Digium package without moving or changing any config? -- _ -- 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] Jabber/Jingle to Google users via local XMPP server
Hi all, All the examples I've come across seem to suggest configuring jabber.conf/jingle.conf/gtalk.conf for a real Google account. What about the scenario where the Asterisk server should connect to an account on a private Jabber server and using Jingle (voice calling over Jabber)? e.g. for the domain widgets.com: - there is a copy of ejabberd running on the same box as Asterisk, and Asterisk registers to it using the jabber ID aster...@widgets.com - DNS is configured so that u...@widgets.com can chat to u...@gmail.com (already working, testing with a chat client such as Empathy or Psi) Google user frie...@gmail.com wants to make a voice call to aster...@widgets.com - is it possible? For this scenario, is gtalk.conf needed at all? Is gtalk.conf needed for any Jabber server, such as the ejabbard instance described above? Regards, Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Open Source VoIP at FOSDEM
For those of you coming to FOSDEM on 24/25 Feb, there'll be a session in the Debian devroom on Open Source VoIP. http://www.fosdem.org/2007/schedule/speakers/daniel+pocock Several VoIP projects will be represented in various ways throughout the weekend, and there will be some of the following: - hardware giveaways from leading VoIP companies - launch of new open source VoIP product during the session in the Debian devroom - integration of VoIP features into other applications (e.g. OpenGroupware) will also be discussed and demonstrated I look forward to seeing some of you there. Regards, Daniel ___ --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] ztdummy installed but choppy audio warning on load
zap show status will tell you if Asterisk is really using ztdummy Make sure you have chan_zap.so enabled in modules.conf (or that it isn't disabled with a noload declaration) Nigel Godfrey wrote: On a new set up Centos 4.4, kernel 2.6.9-42.0.2.EL, yum updated, 2 BRI-HFC cards, no digium hardware. modprobe zaptel and modprobe ztdummy are both in rc.local, and lsmod gives: [EMAIL PROTECTED] ~]# lsmod Module Size Used by ztdummy 3924 0 zaptel206852 5 ztdummy When asterisk starts it logs warnings: Sep 9 20:28:00 WARNING[2645] res_musiconhold.c: Unable to open pseudo channel for timing... Sound may be choppy. Sep 9 20:28:02 WARNING[2645] chan_iax2.c: Unable to open IAX timing interface: No such file or directory I've Googled the error message, but to no avail. Any thoughts, please? nigel. ___ --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 ___ --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] Intel Based G.729 and SVN-trunk-r42453
Jason Lee wrote: I recompiled with debuging options... both bt and btfull outputs http://pastebin.ca/165250 Before I recompiled it gave me a second of audio then I got nothing but distortion for 5 seconds then asterisk would crash. I retested after compiling it with just a call between two local devices one using ulaw and the other using g729 and I'm getting nothing but distortion. I then tried calling music on hold and it took 3 minutes to crash the whole time I got nothing but distortion. This suggests that someone/something gave the command `stop now' Can you send the backtrace from a segfault? On 9/9/06, Daniel Pocock <[EMAIL PROTECTED]> wrote: Jason Lee wrote: > Hi, > > I was testing the intel based G729 codec on SVN-trunk-r42453 following > the > new instructions for compiling with SVN trunk and it in preliminary > tests it > works ok for some calls but I found when one end of the call is an IVR or > Music On Hold the sound gets all distorted and asterisk segfaults. You > can > view the backtrace at http://pastebin.ca/165220 > > Any assistance on this would be appreciated. > Have you compiled with debugging symbols instead of CPU optimization? Can you type `bt' after the segfault, to give us some more detail? How long into the call does this happen? > > >___ >--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 > > ___ --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 ___ --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 ___ --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] Intel Based G.729 and SVN-trunk-r42453
Jason Lee wrote: Hi, I was testing the intel based G729 codec on SVN-trunk-r42453 following the new instructions for compiling with SVN trunk and it in preliminary tests it works ok for some calls but I found when one end of the call is an IVR or Music On Hold the sound gets all distorted and asterisk segfaults. You can view the backtrace at http://pastebin.ca/165220 Any assistance on this would be appreciated. Have you compiled with debugging symbols instead of CPU optimization? Can you type `bt' after the segfault, to give us some more detail? How long into the call does this happen? ___ --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 ___ --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] Re: FW: Peter Dicks Chairman ofSportingbet PLC isarrested at JFK!!
Brandon Galbraith wrote:
Steve,
Forgive my ignorance, but why does India institute that policy?
Why does France blow up bombs in the south pacific? Each country can do
as it pleases - unfortunately - but that is also good for us VoIP
carriers because it creates and protects high retail prices that we can
easily compete with using our superior technology and better customer
service.
-brandon
On 9/8/06, Steven <[EMAIL PROTECTED]> wrote:
Even in India, you can use VOIP for overseas calls coming from your own
company.
You just can't sell services that allow people to call a PSTN number and
then have their call sent over VOIP to another location.
--
--
Steven
http://www.glimasoutheast.org
"Alex Robar" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
The biggest problem with your argument is that VoIP is not illegal
anywhere. Voice over _internet_ is, but voice over the Internet
protocol is
not. Anyone in China is free to setup Asterisk to use within their
offices.
There's nothing illegal about it. It's using VoIP, but it's not
transmitting
it outside of your office. It's the calls that leave your office that
matter... They're the ones that governments can tax.
...And if what Mark Spencer is doing isn't illegal, your post it most
certainly OT... And definitely flame bait. Why troll about the US?
There's
US hating forums in a lot of places, go post your rants there.
Alex
On 9/8/06, Dean Collins <[EMAIL PROTECTED]> wrote:
>
> Yes I am aware he is the second executive to be arrested..the
first
> is still yet to be charged and is still awaiting trial and has fallen
> off the face of the general media which is why I'm 'motivated' to
draw
> attention and outrage to this second case.
>
> Yes you are right it does belong off this list but with so many people
> on this list doing international business on the internet (including
> yourself by the looks of your own voip carrier website).
>
> So when someone from China places an order on your website for a voip
> service, you agree that it would be ok for the Chinese government
to let
> the Chinese customer go free but for them to arrest you and any other
> directors muWare?
>
> (also lets not forget that the WTO has already ruled that USA
government
> is in breach with their court case back in March).
>
> I'm very curious about your thoughts or will you prefer to stick your
> head in the sand and pretend that the USA lives in a bubble on the
> planet earth and would prefer that Walmart not do business
> internationally or that the Ford motor car you drive not use
Australian
> steel etc etc.
>
>
>
>
> Cheers,
>
> Dean
>
>
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:
asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Jay Milk
> > Sent: Friday, 8 September 2006 1:22 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: {Fraud?} RE: [asterisk-users] FW: Peter Dicks Chairman
> ofSportingbet
> > PLC isarrested at JFK!!
> >
> > Why don't you keep political diatribe to your blog? This is OT, and
> > quite frankly it displays that you have less than perfect grasp on
> reality.
> > Mark Spencer makes a software product that is perfectly legal to use
> > anywhere in the world, even in India (as long as it stays within a
> > building and used as a PBX only). It's the user's responsibility to
> > understand and follow local law in utilizing this software. By the
> same
> > token, a vehicle is legal to use for most, but it becomes a weapon
> when
> > you intentionally or neglectfully run someone over.
> > Peter Dicks was operating a business that was offering certain
> services
> > that are illegal in the place where he offered them. If he was
> arrested
> > the moment he entered the US, then this was investigated before, and
> he
> > was asked to cease offering these illegal services in the US, but
has
> > refused to do so -- if you followed the news as enthusiastically as
> you
> > post OT messages, then you would have realized that Dicks was the
> > *second* executive that was arrested. This should have come as no
> > surprise to Dicks or his lawyers.
> > So, now we're seeing the guy not as a clueless tourist, but a
> > law-defying visitor. He's done something that is illegal, has
refused
> to
> > stop, and was dumb enough to step into the jurisdiction of his
crime.
> Is
> > there a problem with that? Because if there is, you need to start
> > defending Columbian drug-lords and terrorists, too.
> > This is fulfills my quote of OT posts for the day. Just had to say
> > something in the face of such obvious stupidity.
> >
> > Dean Collins wrote:
> > >
> > > Exactly so why aren't they trying to arrest the 50 million
people in
>
> > > the USA who have gambled online?
> > >
> > > Mark (as far as I know) isn't actively checking with asterisk
users
> > > for what country they are in so therefore in the reciprocal
eyes of
> > > the indian government he is si
Re: [asterisk-users] Asterisk hangs up after 10-15 minutes when SIPPhone is on mute
Check sip.conf parameters: rtptimeout rtpholdtimeout David Gagnon wrote: I would recommend you to call Unlimitel as they have a very good support. Or just send a copy of your post to : [EMAIL PROTECTED] David _ De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Mike Envoyé : 7 septembre 2006 11:32 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : [asterisk-users] Asterisk hangs up after 10-15 minutes when SIPPhone is on mute Hi, I have a Polycom 501 connected to Asterisk 1.2.4 (and then connected to a VOIP provider, Unlimitel in my case). My job requires me to attend conference calls regularly, and I am usually there as a silent listener. Therefore, I mute my phone. I`ve noticed that if I mute my phone, after 10-15 of being muted, the line hangs up. I had the same problem with my GXP-2000 before, so I dismissed the phone as being the problem. If I unmute regularly (or the entire time), the line doesnt hang up (until it reaches max timeout of course, which is much more than 15 minutes). So the problem is my phone is muted. I have observed that about 6 times (out of 6 tries) in the last 4 months. It`s a reccuring issue for sure. What I am left with is Asterisk (or my VoIP provider) as the issue. Since I only have control on my own Asterisk server, I thought I should start there. What setting could cause this? I have a fairly fancy dialplan, but I havent changed anything else than the diaplan. All system-wide Asterisk settings are default as far as I know. Thanks, Mike ___ --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 ___ --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
[asterisk-users] Open source G.729 and G.723.1 release for 1.2 and 1.4
The Intel IPP based open source release of G.729 and G.723.1 have now been updated to compile with the following versions of Asterisk: - Asterisk 1.2.11 - Asterisk trunk - tested with SVN r 42264 The code is at the usual location: http://www.readytechnology.co.uk/open/ipp-codecs/ If you intend to do anything other than study this code, I would encourage you to purchase a legitimate license. Please feel free to submit bug reports if this code causes any trouble for you. Regards, Daniel ___ --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
[Asterisk-Users] Open G.729 / G.723.1 update, fixed memory leak
A new release of the open source G.729 patch has been issued. The new URL is: http://www.readytechnology.co.uk/open/ipp-codecs The memory leak in codec_g729 is now fixed. This was due to a problem in a section of code copied from the Intel example. Thanks to those who assisted in locating this bug. If you are still running the old version of the codec, your Asterisk process will run out of memory after several thousand calls (for some people, that might be every 3 months, for other people, it could be more than once a day). Therefore, updating to this latest release is highly advisable. This release combines the G.729 and G.723.1 patches into a single patch against Intel's IPP sample. They are still built as separate modules, so you don't have to install both. I've also included - command line G.729 encoder for converting your WAV files into pre-recorded G.729 files - scripts for generating a Debian package. See the documentation for details. Finally, the donations page has now been fixed. Making a donation is one way to encourage programmers to contribute commercial quality code to the open source community. Regards, Daniel smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP PSTN numbers in Australia?
Hi, I've tried to request VoIP PSTN numbers from a couple of Australian companies who are advertising on Google, but neither of them was able to fulfil despite advertising the numbers on their sites. In fact, I was disappointed that both of them actually asked me to complete their online order form, and then just didn't do anything, wasting more time. Does anyone know of a business that does offer inbound PSTN numbers in Australia? I would be most interested in getting a number for the Gold Coast or Brisbane, but Melbourne and Sydney are fine too. I'm looking to test how successfully this type of number can be used with an Asterisk server half way around the world in the UK, so would prefer to be using a service that offers fairly constant performance. Regards, Daniel smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729 for Asterisk: new version released
Download site: http://www.readytechnology.co.uk/open/g729 Major enhancements: - accepts packets with VAD stuff at the end (please test this with your hardware and give me feedback if it still doesn't work with some devices, you will probably see error messages on the console if bad sized frames are received) - various compiler optimizations offered in the Makefile - 20% boost in performance noticed by several users including myself - static linking, so you don't have to copy IPP libraries to every Asterisk where you install this code Thanks to Arkadi for the Makefile improvements This code is for educational purposes only. If you live in a country where this technology is patented, you may not be allowed to execute this code or distribute a binary version without the permission of the patent holder(s). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium and mailing lists
I was somewhat concerned reading Mark's posting earlier today. Obviously, things are very bad in the US at the moment. Their Government even deported Cat Stevens the other day (check http://news.bbc.co.uk/1/hi/england/london/3686992.stm ). Clearly, given the fact that Digium contributes so much to Asterisk, they shouldn't be forced to risk their company's future by hosting these mailing lists in such an unstable environment where they could get sued for any ridiculous reason. Even an unjustified, ambit claim could generate huge defence costs on Digium's part, and cripple their ability to contribute to Asterisk. Therefore, it seems to be in the best interests of Asterisk's `security' to have the mailing lists hosted by someone other than Digium and maybe in a country that doesn't prohibit freedom of expression. I would certainly be willing to organise hosting through another company that wouldn't be at risk from vexatious legal claims. This would allow genuinely open discussion on the lists and would mean that no messages would need to be censored from the archives. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729 and Asterisk intellectual property issues
-- snip -- Had the patch been against the actual g729 libraries the case would have been clear. Now, the patch is against asterisk to make it interoperate with the g729 libarary and this may or may not be non-infringing. However, the distribution of the g729 libraries themselves are almost certainly infringing. There is also the possibility that the patch to asterisk may be ruled a contribuatory infringement. -- snip -- The patch is not against Asterisk - it is against Intel's sample code. No parts of Asterisk are modified in order to run this code. Nor am I requesting that Asterisk be modified in any way to support this. The code produced by running the build script is a shared library that can be added to Asterisk. The shared library could be used independently of Asterisk, and Asterisk can still be used without the shared library. It is completely optional whether people choose to integrate this code with Asterisk. However, I understand that it probably can't be added to the main distribution and I am happy to continue making it available in source form as an add-on module for those who would like to evaluate it. I certainly never expected that it would be adopted as an official inclusion in Asterisk, and I certainly won't take offence if it isn't. The relevent terms from Intel's license are below. (B) says that I have the right to modify the source code and (C) says that I can combine portions of the sample source into a product and then distribute the resulting application. B. Subject to all of the terms and conditions of this Agreement, Intel grants to you a non-exclusive, non-assignable copyright license to modify the Materials, or any portions thereof, that are (i) provided in source code form or, (ii) are defined as Redistributables and are provided in text form. C. Subject to all of the terms and conditions of this Agreement, Intel grants to you a non-exclusive, non-assignable copyright license to distribute (except under an Evaluation License as specified below) the Redistributables and Sample Source, or any portions thereof, as part of the product or application you developed using the Materials. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Intel IPP licensing and G.729
I'm interested in the g729 diff you posted... I've applied the patch, but I don't seem to have the prerequisites to compile it... I tried downloading the other code available from Intel, but even the 'eval' version won't install without a FlexLM license (damn license managers...). Am I heading the right direction, or is there somewhere else I should be looking? Thanks, Rob I've added more details to the documentation. You may need to a) register on Intel's site under the 'free non-commercial use' section b) obtain the license number and a license key file c) put the license key file in the place specified by Intel's documentation d) run the installer Please read Intel's terms and conditions carefully to establish whether your usage qualifies for the free evaluation download or the 'non-commercial use' download. They both give you the same code, but under different terms. Regards, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Free G.729 ready for download
DISCLAIMER: This code is free (I am not charging you to use it), but you might have to pay royalty fees to the G.729 patent holders for using their algorithm. I finished this last Saturday and have had it on an Asterisk machine for 5 days without a crash, so I'm hoping that means it's safe to release into the real world. This code has also been released on the -dev list. As it is still somewhat new, I would invite anyone with feedback to forward it to me personally or to the -dev list, as I don't monitor the -users list very often. http://www.readytechnology.co.uk/open/g729 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
