Re: [asterisk-users] SPA504G auto answer
On 23/10/2014 5:43 AM, Leandro Dardini wrote: Hello, I am struggling to have a SPA504G to auto answer (for intercom/paging). I have tried the following SIP headers (not all together), but without luck: SIPAddHeader(Call-Info:\;answer-after=0); SIPAddHeader(Call-Info: answer-after=0); SIPAddHeader(Alert-Info: info=intercom); SIPAddHeader(Alert-Info: Ring Answer); SIPAddHeader(P-Auto-Answer: normal); What does your dialplan look like that makes the paging call? Listing from my Asterisk: '8000' = 1. Set(SIP_CODEC=alaw) 2. Dial(MulticastRTP/linksys/224.168.168.168:45678/224.168.168.168:6061) 3. Hangup() Using the Web Interface on you SPA501, log in as admin and select the advanced view. Select the SIP tab, down the bottom of the page there is a section headed 'Linksys Key System Parameters'. You will want settings much like Linksys Key System: yes Multicast Address: 224.168.168.168:6061 Key System Auto Discovery: no Key System IP Address: leave blank Force LAN Codec: 711a may be set to none, G711a or G711u Auto Ans GrPage On Active Call: no Select the User tab and check Auto Answer Page: yes If you have it all configured much like I have listed hear and it still doesn't work then you need to check the firewall configuration on your Asterisk system and ensure it is allowing outbound Multicast traffic. Larry. -- _ -- 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] SPA504G auto answer
On 23/10/2014 4:57 PM, Larry Moore wrote:
On 23/10/2014 5:43 AM, Leandro Dardini wrote:
Hello,
I am struggling to have a SPA504G to auto answer (for intercom/paging).
I have tried the following SIP headers (not all together), but without
luck:
SIPAddHeader(Call-Info:\;answer-after=0);
SIPAddHeader(Call-Info: answer-after=0);
SIPAddHeader(Alert-Info: info=intercom);
SIPAddHeader(Alert-Info: Ring Answer);
SIPAddHeader(P-Auto-Answer: normal);
What does your dialplan look like that makes the paging call?
Listing from my Asterisk:
'8000' = 1. Set(SIP_CODEC=alaw)
2. Dial(MulticastRTP/linksys/224.168.168.168:45678/224.168.168.168:6061)
3. Hangup()
Using the Web Interface on you SPA501, log in as admin and select the
advanced view. Select the SIP tab, down the bottom of the page there is
a section headed 'Linksys Key System Parameters'.
You will want settings much like
Linksys Key System: yes
Multicast Address: 224.168.168.168:6061
Key System Auto Discovery: no
Key System IP Address: leave blank
Force LAN Codec: 711a may be set to none, G711a or G711u
Auto Ans GrPage On Active Call: no
Select the User tab and check
Auto Answer Page: yes
If you have it all configured much like I have listed hear and it still
doesn't work then you need to check the firewall configuration on your
Asterisk system and ensure it is allowing outbound Multicast traffic.
Larry.
Hmm, my SPA525G doesn't auto-answer a page however my SPA92X do.
The above is for paging, I use a macro to perform an intercom, here is
what I have in my extensions.ael.
context {
_8XXX = {
sip_intercom(${EXTEN:1});
};
};
macro sip_intercom( extension ) {
ChanIsAvail(SIP/${LOCAL(extension)},s);
NoOp( Status : ${AVAILSTATUS} );
switch(${AVAILSTATUS}) {
case 1:
Set(TIMEOUT(absolute)=1920);
SIPAddHeader(Alert-Info: Ring Answer);
SIPAddHeader(Alert-Info: Info=Alert-Autoanswer);
SIPAddHeader(Alert-Info: info=ringAutoAnswer);
SIPAddHeader(Call-Info:\;Answer-After=0);
SIPAddHeader(P-Auto-Answer: normal);
SIPAddHeader(Answer-Mode: Auto);
Dial(SIP/${LOCAL(extension)});
Hangup();
break;
case 2:
Busy();
Hangup();
break;
case 5:
Congestion();
Hangup();
break;
default:
PlayBack(invalid);
Hangup();
break;
};
return;
};
Larry.
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Re: [asterisk-users] SPA504G auto answer
On 23/10/2014 6:41 PM, Larry Moore wrote: snip Listing from my Asterisk: '8000' = 1. Set(SIP_CODEC=alaw) 2. Dial(MulticastRTP/linksys/224.168.168.168:45678/224.168.168.168:6061) 3. Hangup() snip Hmm, my SPA525G doesn't auto-answer a page however my SPA92X do. snip Just upgraded the firmware on my SPA525G from 7.5.4 to 7.5.6 and the paging function is now working! -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
On 10/22/2014 03:55 PM, Tim Nelson wrote: - Original Message - Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in question) - SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2]. Thank you, --Tim [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway [2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html *bump* Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as gateway do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job. :) Thanks! --Tim I can't help with your root problem (maybe check core show function FAXOPT?), but the spandsp site is up. Try using www.spandsp.org. Downloads are available here: http://www.spandsp.org/downloads/spandsp/ -Dave -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
On 23/10/2014 3:55 AM, Tim Nelson wrote: - Original Message - Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in question) - SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2]. Thank you, --Tim [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway [2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html *bump* Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as gateway do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job. :) No thoughts on your problem, I do think you will need a newer version of spandsp through - the site seems to be up now. -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? Larry. -- _ -- 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] Auto video call hangup
Hi, I use a simple scheme: SIP video phone A (h264/Asterisk 1.8.11) ---IAX2 trunk SIP video phone B (h264/Asterisk 11.7.0) When calls from A to B and vice versa drop on pickup. On B side: [Oct 24 16:33:49] DEBUG[15590][C-0012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] DEBUG[15590][C-0012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] DEBUG[15590][C-0012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] WARNING[15202] chan_iax2.c: Received mini frame before first full video frame [Oct 24 16:33:49] DEBUG[15206] chan_iax2.c: Ooh, video format changed to h264 [Oct 24 16:33:49] DEBUG[15590][C-0012] res_rtp_asterisk.c: Ooh, format changed from unknown to h264 [Oct 24 16:33:49] DEBUG[15590][C-0012] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xb7d79c54' [Oct 24 16:33:49] DEBUG[15207] chan_iax2.c: Ooh, voice format changed to 'ulaw' [Oct 24 16:33:49] DEBUG[15590][C-0012] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [Oct 24 16:33:49] DEBUG[15590][C-0012] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [Oct 24 16:33:49] DEBUG[15590][C-0012] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xb69b9894' [Oct 24 16:33:49] DEBUG[15204] chan_iax2.c: Packet arrived out of order (expecting 7, got 5) (frametype = 3, subclass = 24) [Oct 24 16:33:49] DEBUG[15204] chan_iax2.c: Acking anyway [Oct 24 16:33:49] DEBUG[15211] chan_iax2.c: Packet arrived out of order (expecting 7, got 6) (frametype = 2, subclass = 13) [Oct 24 16:33:49] DEBUG[15211] chan_iax2.c: Acking anyway [Oct 24 16:33:49] DEBUG[15590][C-0012] res_rtp_asterisk.c: Difference is 45450, ms is 505 (45450), pred/ts/samples 45450/0/0 [Oct 24 16:33:50] DEBUG[15193][C-0012] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.0.192:5060 [Oct 24 16:33:50] DEBUG[15590][C-0012] res_rtp_asterisk.c: 0xb7d93ce0 -- Probation learning mode pass with source address 192.168.0.192:5004 [Oct 24 16:33:50] DEBUG[15590][C-0012] res_rtp_asterisk.c: 0xb69b5488 -- Probation learning mode pass with source address 192.168.0.192:5006 [Oct 24 16:33:50] WARNING[15590][C-0012] chan_iax2.c: Can't compress subclass 2097217 [Oct 24 16:33:50] DEBUG[15207] chan_iax2.c: Received VNAK: resending outstanding frames [Oct 24 16:33:50] DEBUG[15209] chan_iax2.c: Received VNAK: resending outstanding frames [Oct 24 16:33:50] DEBUG[15208] chan_iax2.c: Received VNAK: resending outstanding frames [Oct 24 16:33:50] DEBUG[15590][C-0012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:50] DEBUG[15204] chan_iax2.c: Received VNAK: resending outstanding frames [Oct 24 16:33:50] DEBUG[15211] chan_iax2.c: Received VNAK: resending outstanding frames [Oct 24 16:33:50] DEBUG[15203] chan_iax2.c: Received VNAK: resending outstanding frames [Oct 24 16:33:50] DEBUG[15202] chan_iax2.c: Received VNAK: resending outstanding frames [Oct 24 16:33:50] DEBUG[15206] chan_iax2.c: Received VNAK: resending outstanding frames [Oct 24 16:33:50] DEBUG[15205] chan_iax2.c: Immediately destroying 1664, having received hangup [Oct 24 16:33:50] DEBUG[15227] manager.c: Examining event: [Oct 24 16:33:50] DEBUG[15590][C-0012] channel.c: Didn't get a frame from channel: IAX2/THNS-1664 [Oct 24 16:33:50] DEBUG[15590][C-0012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:50] DEBUG[15590][C-0012] channel.c: Bridge stops bridging channels IAX2/THNS-1664 and SIP/507-001d [Oct 24 16:33:50] DEBUG[15590][C-0012] channel.c: Soft-Hanging up channel 'IAX2/THNS-1664' [Oct 24 16:33:50] DEBUG[15590][C-0012] pbx.c: Launching 'Macro' [Oct 24 16:33:50] VERBOSE[15590][C-0012] pbx.c: -- Executing [h@macro-dial-one:1] Macro(IAX2/THNS-1664, hangupcall,) in new stack Debug at A side: [Oct 23 17:42:47] WARNING[14880] chan_iax2.c: Received mini frame before first full video frame [Oct 23 17:42:47] WARNING[14887] chan_iax2.c: Received mini frame before first full video frame [Oct 23 17:42:47] WARNING[14881] chan_iax2.c: Received mini frame before first full video frame [Oct 23 17:42:47] WARNING[14885] chan_iax2.c: Received mini frame before first full video frame [Oct 23 17:42:47] WARNING[22489] res_rtp_asterisk.c: Don't know how to send format unknown packets with RTP [Oct 23 17:42:47] WARNING[14886] chan_iax2.c: Received mini frame before first full video frame [Oct 23 17:42:47] WARNING[14879] chan_iax2.c: Received mini frame before first full video frame [Oct 23 17:42:47] WARNING[14878] chan_iax2.c: Received mini frame before first full video frame [Oct 23 17:42:47] WARNING[14880] chan_iax2.c: Received mini frame before first full video frame [Oct 23 17:42:47] WARNING[14887] chan_iax2.c: Received mini frame before first full video frame [Oct 23 17:42:47] WARNING[14881] chan_iax2.c: Received mini
[asterisk-users] logger.conf
with the below defined in logger.conf on 11.6 cert 6 I am not getting any log message other than notice and warning in any files when doing module reload logger - queue log is the only one that says it restarts *CLI module reload logger == Parsing '/etc/asterisk/logger.conf': Found Asterisk Queue Logger restarted built fresh box with make samples - added 2 stations, dialing from station a to station b on console (core set verbose 15) get -- x=0, open writing: /var/spool/asterisk/voicemail/default/20545/tmp/9mrzpn format: wav49, 0x33b9d68 -- x=1, open writing: /var/spool/asterisk/voicemail/default/20545/tmp/9mrzpn format: gsm, 0x33c7938 -- x=2, open writing: /var/spool/asterisk/voicemail/default/20545/tmp/9mrzpn format: wav, 0x33c0f18 [Oct 23 11:03:33] WARNING[8832][C-0003]: file.c:830 ast_readaudio_callback: Failed to write frame -- User hung up == Spawn extension (default, stdexten-NOANSWER, 1) exited non-zero on 'SIP/20321-0001' == Using SIP RTP CoS mark 5 -- Executing [20545@default:1] Gosub(SIP/20321-0002, 20545,stdexten(SIP/20545)) in new stack -- Executing [20545@default:5] NoOp(SIP/20321-0002, Start stdexten) in new stack -- Executing [20545@default:50001] Set(SIP/20321-0002, LOCAL(ext)=20545) in new stack -- Executing [20545@default:50002] Set(SIP/20321-0002, LOCAL(dev)=SIP/20545) in new stack -- Executing [20545@default:50003] Set(SIP/20321-0002, LOCAL(cntx)=) in new stack -- Executing [20545@default:50004] Set(SIP/20321-0002, LOCAL(mbx)=20545) in new stack -- Executing [20545@default:50005] Dial(SIP/20321-0002, SIP/20545,20) in new stack [Oct 23 11:03:34] WARNING[8839][C-0004]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [20545@default:50006] Goto(SIP/20321-0002, stdexten-CHANUNAVAIL,1) in new stack -- Goto (default,stdexten-CHANUNAVAIL,1) -- Executing [stdexten-CHANUNAVAIL@default:1] Goto(SIP/20321-0002, stdexten-NOANSWER,1) in new stack -- Goto (default,stdexten-NOANSWER,1) -- Executing [stdexten-NOANSWER@default:1] VoiceMail(SIP/20321-0002, 20545,u) in new stack 0x7f66440060b0 -- Probation passed - setting RTP source address to 10.128.1.47:7078 -- SIP/20321-0002 Playing 'vm-theperson.ulaw' (language 'en') -- SIP/20321-0002 Playing 'digits/2.ulaw' (language 'en') -- SIP/20321-0002 Playing 'digits/0.ulaw' (language 'en') -- SIP/20321-0002 Playing 'digits/5.ulaw' (language 'en') expecting all this detail to exist in logs it does not any suggestions on how to get this functionality back - broke after upgrade 1.8.9.2 to 11.6 cert 6 [general] [logfiles ] console = notice, warning, error, debug, verbose(15) messages = notice, warning, error, debug, error full = notice, warning, error, debug, verbose(15) debug = debug syslog.local1 = notice, warning, error,debug, verbose fax = notice, warning, error, debug, fax, dtmf, fax -- Jared Terrell Network Telecommunications Mott Community College 1401 E. Court Flint, MI 48503 810-762-0545 810-577-7510 -- _ -- 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] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)
On Oct 22, 2014, at 3:27 PM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: From: Paul Albrecht palbre...@glccom.com Here’s a link to the minutes: https://wiki.asterisk.org/wiki/ display/AST/AstriDevCon+2014 It has you saying: Leif: we're in a transition, moving from dialplan model to external control model. Probably need external application to be built for us to move completely away from AMI/AGI. So you’re saying Asterisk is moving away from the dial plan or were you misquoted? Paul, I think you are getting worked up way too early in this process. This is one comment with only a little bit of context surrounding it. Such a major change would take quite awhile to make and there would be plenty of warning before it happens, with plenty of opportunities to discuss. The dial plan isn't going away tomorrow and if it does ever go away, there will be plenty of time to work out a transition plan. Seems like now is as good a time as any to raise these issues, in fact, sooner is better than later because once developers start down a path it’s very difficult to get them change their minds no matter how much sense it makes. The fact that developers are even considering taking away user functionality like the dial plan is in of itself a very serious problem because it demonstrates they don’t see Asterisk from the user perspective. Looking at the path development has taken, it seems pretty clear that they have been working towards enabling greater external control of what Asterisk does, making it the engine that can drive other media applications. Doesn't mean it can't and won't be used as a traditional pbx, but to grow what it does will require some changes. If being a mature part of Asterisk means that something shouldn't be changed, we should also protest the move from the current SIP stack to pjsip. There are any number of reasons to deprecate mature code. It may not be needed or something better may come along. Don’t object to extending the Asterisk user interface or changing Asterisk internals. Do object to is taking away functionality that users expect, are familiar with, and has made the Asterisk project successful. All I can say is that having experience with a few versions of Asterisk, it seems to get better and more stable as new versions come along. Perhaps a bit of faith that they are not trying to kill off their product simply by having a discussion at a dev conference is in order. Then your experience is atypical. Asterisk has been unstable for several years as developers have continually shoveled new features into the code base over several releases. That’s not necessary objectionable, it’s even to be expected; however, at some point developers need to turn their attention to less glamorous less exciting things like stability and performance. Kevin Larsen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)
On Oct 22, 2014, at 3:39 PM, Matthew Jordan mjor...@digium.com wrote: On Wed, Oct 22, 2014 at 1:55 PM, Paul Albrecht palbre...@glccom.com wrote: On Oct 22, 2014, at 11:31 AM, Matthew Jordan mjor...@digium.com wrote: On Wed, Oct 22, 2014 at 11:14 AM, Paul Albrecht palbre...@glccom.com wrote: On Oct 22, 2014, at 10:33 AM, Joshua Colp jc...@digium.com wrote: Paul Albrecht wrote: Really? Shouldn’t something this major affecting the entire Asterisk community get discussed on the lists? Any idea what Leif is talking about when he says the community is in transition, moving from dial plan model to external control. It was something Ben Klang brought up and wanted to talk about - it's not something that has been decided 'nor does anyone know what the future entails. Any further discussions will naturally occur on the mailing list and in fact some things have explicit action items to bring them up on here. The suggestion that Asterisk should consider deprecating AMI/AGI is “crazy talk.” It doesn’t merit discussion and shouldn’t be on the agenda in the first place. It’s completely impractical and can never happen. Moreover, Leif seems to think we (the asterisk community) are in transition. What does that mean? Are we abandoning the dial plan? Seriously? That’s never gonna happen either. ARI isn’t easier to use than dial plan scripting. I guess one could hope that what happens in Vegas stays in Vegas”, but I don’t think the Asterisk community has that kind of luck. Just because someone decided to bring up a radical idea does not mean we refuse to discuss it. So you agree that deprecating AMI/AGI is “crazy talk” but you’ll discuss it because of your open-mindedness? I didn't say that the idea of deprecating AMI/AGI is crazy talk. I did say that radical ideas - and even ones that some folks think are crazy - are all fine to discuss at AstriDevCon. The whole point of AstriDevCon is to have a large, free, and open conversation about Asterisk Development. I fundamentally disagree with the notion that that should be discouraged. The problem with AstriConDev is there is no user input so what you have is a developer echo chamber and what you get is groupthink. This is an open source project. Communication is done in an open, transparent manner. People should feel like they can bring up interesting, radical, and yes - even crazy - ideas. By the same token, when you propose ideas, you must be prepared for honest criticism and accept it in graciously rather than simply resorting to argument ad hominem. You didn't have honest criticism. You labelled a discussion point as crazy talk and said we shouldn't have even discussed it. There was no ad hominem attack. I never attacked you. I never even attacked your statements. I simply defended the free exchange of ideas in AstriDevCon. I have no problem doing that. On the other hand, you did callously label an Asterisk Developer's admittedly ambitious idea as crazy talk. In the future, you may want to choose your language more carefully if you wish for others to have a more open discussion with you. If you don't like that, you don't have to participate in the discussion. You haven’t really responded to the substance of my post, that is, is asterisk abandoning the dial plan? There are Asterisk users (who also happen to develop) who would like to minimize the dialplan necessary in their systems, to the point where they may no longer even need the dialplan. This is a fundamentally sound idea for some systems, particularly those that require scaling Asterisk out to many machines. There are also some Asterisk users who build complex applications on top of Asterisk, and who find having to use multiple interfaces cumbersome. They like ARI, and would like to see it able to do more than what it currently does today. Don’t have a problem with extending the Asterisk user interface or changing Asterisk internals that are not visible to users. Do object to taking away taking away user functionality like the dial plan that users expect, are familiar with, and has made the Asterisk project successful. Fully deprecating a feature in Asterisk is non-trivial. You must have: (1) A logical and full replacement for the feature (2) Buy-off from the developer community (3) Several major versions of the project in which the deprecated feature must remain Even in the case of point #3, deprecated features have often lasted in *many* versions of Asterisk. We are enormously conservative in what we choose to remove from the project. Not interested in what rules or process steps need to be followed to deprecate features. The fact of the matter is you’re not starting with a blank sheet of paper and you can’t simply abandon the existing user interface because what will really happen is your users will abandon you and your project.
Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)
On Oct 23, 2014, at 1:55 AM, Olle E Johansson o...@edvina.net wrote: It is critical that a group of developers ask themself questions along these lines - what if??? - What if we removed AGi and AMI? - What if we made a pluggable PBX? - What if we restarted working on a SIP channel? - What if we made a whole new bridge architecture? - What if we skip the idea of making a PBX? Good development quite frequently starts with these kind of ideas and questions that may see crazy but results in really good changes. Brainstorms needs to be open and not restricted, that is what the astridevcons are for. We need to go wild and see what comes out of it. A lot of the great changes we see in Asterisk 13 comes from many years of wild discussions. Pinemango anyone? The unacknowledged problem we’re dealing with is the fact that we’re not starting with a blank sheet of paper, but rather a mature user interface that users expect, are familiar with, and has made project successful. Extending the the user interface is one thing, throwing it away is another entirely different thing. But hey, you have the user's community attention at Astricon, why not have the courage of your convictions and announce to the the unwashed masses you’re planning to do away with the dial plan? /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- 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] logger.conf
On Thu, Oct 23, 2014 at 10:07 AM, Jared Terrell jared.terr...@mcc.edu wrote: with the below defined in logger.conf on 11.6 cert 6 I am not getting any log message other than notice and warning in any files when doing module reload logger - queue log is the only one that says it restarts *CLI module reload logger == Parsing '/etc/asterisk/logger.conf': Found Asterisk Queue Logger restarted built fresh box with make samples - added 2 stations, dialing from station a to station b on console (core set verbose 15) get -- x=0, open writing: /var/spool/asterisk/voicemail/default/20545/tmp/9mrzpn format: wav49, 0x33b9d68 -- x=1, open writing: /var/spool/asterisk/voicemail/default/20545/tmp/9mrzpn format: gsm, 0x33c7938 -- x=2, open writing: /var/spool/asterisk/voicemail/default/20545/tmp/9mrzpn format: wav, 0x33c0f18 [Oct 23 11:03:33] WARNING[8832][C-0003]: file.c:830 ast_readaudio_callback: Failed to write frame -- User hung up == Spawn extension (default, stdexten-NOANSWER, 1) exited non-zero on 'SIP/20321-0001' == Using SIP RTP CoS mark 5 -- Executing [20545@default:1] Gosub(SIP/20321-0002, 20545,stdexten(SIP/20545)) in new stack -- Executing [20545@default:5] NoOp(SIP/20321-0002, Start stdexten) in new stack -- Executing [20545@default:50001] Set(SIP/20321-0002, LOCAL(ext)=20545) in new stack -- Executing [20545@default:50002] Set(SIP/20321-0002, LOCAL(dev)=SIP/20545) in new stack -- Executing [20545@default:50003] Set(SIP/20321-0002, LOCAL(cntx)=) in new stack -- Executing [20545@default:50004] Set(SIP/20321-0002, LOCAL(mbx)=20545) in new stack -- Executing [20545@default:50005] Dial(SIP/20321-0002, SIP/20545,20) in new stack [Oct 23 11:03:34] WARNING[8839][C-0004]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [20545@default:50006] Goto(SIP/20321-0002, stdexten-CHANUNAVAIL,1) in new stack -- Goto (default,stdexten-CHANUNAVAIL,1) -- Executing [stdexten-CHANUNAVAIL@default:1] Goto(SIP/20321-0002, stdexten-NOANSWER,1) in new stack -- Goto (default,stdexten-NOANSWER,1) -- Executing [stdexten-NOANSWER@default:1] VoiceMail(SIP/20321-0002, 20545,u) in new stack 0x7f66440060b0 -- Probation passed - setting RTP source address to 10.128.1.47:7078 -- SIP/20321-0002 Playing 'vm-theperson.ulaw' (language 'en') -- SIP/20321-0002 Playing 'digits/2.ulaw' (language 'en') -- SIP/20321-0002 Playing 'digits/0.ulaw' (language 'en') -- SIP/20321-0002 Playing 'digits/5.ulaw' (language 'en') expecting all this detail to exist in logs it does not any suggestions on how to get this functionality back - broke after upgrade 1.8.9.2 to 11.6 cert 6 [general] [logfiles ] console = notice, warning, error, debug, verbose(15) messages = notice, warning, error, debug, error full = notice, warning, error, debug, verbose(15) debug = debug syslog.local1 = notice, warning, error,debug, verbose fax = notice, warning, error, debug, fax, dtmf, fax Remove the extraneous space you have in the [logfiles ] line. It must be [logfiles]. Richard -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
On 23/10/2014 10:07 PM, Larry Moore wrote: On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? Larry. Have you had a look at https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance As an exercise you could disable T.38 on 'Asterisk calling system', if you have an ATA which is originating the call to 'Asterisk calling system' disable T.38 on that device too and disable in your sip.conf using t38pt_udptl=no. If you are using SendFax() on 'Asterisk calling system' ensure T.38 is not able to be used. If using an ATA connecting to 'Asterisk calling system' ensure you have set in your peer's configuration canreinvite=no or directmedia=no, depending on the version of Asterisk you are running on this system. On Asterisk system in '(box in question)' set directmedia=no for the peer which is connecting to 'SIP Provider' and also to 'Asterisk calling system', you may want to set setvar=FAXOPT(gateway)=yes in your peer config to 'SIP Provider' otherwise it will need to be set in your dialplan. Set your verbose debug to at least 3 on '(box in question)', possibly a little higher and send a fax - you may now see the Fax Gateway detect CED. Not sure if this is suppressed in You may want enable udptl debugging on '(box in question)'. Larry. -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
- Original Message - On 10/22/2014 03:55 PM, Tim Nelson wrote: - Original Message - Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in question) - SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2]. Thank you, --Tim [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway [2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html *bump* Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as gateway do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job. :) Thanks! --Tim I can't help with your root problem (maybe check core show function FAXOPT?), but the spandsp site is up. Try using www.spandsp.org. Downloads are available here: http://www.spandsp.org/downloads/spandsp/ It is up now, thanks! --Tim -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
- Original Message - On 23/10/2014 3:55 AM, Tim Nelson wrote: - Original Message - Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in question) - SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2]. Thank you, --Tim [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway [2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html *bump* Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as gateway do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job. :) No thoughts on your problem, I do think you will need a newer version of spandsp through - the site seems to be up now. The version of SpanDSP is not in question at this point. The problem lies in I need a way to use the T38 Gateway function, but *also* initiate the reinvite to T.38 on the call as the provider will not do this, saying it is the *caller*'s responsibility. This is contrary to past experience however... --Tim -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
- Original Message - On 23/10/2014 10:07 PM, Larry Moore wrote: On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? Larry. Have you had a look at https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance As an exercise you could disable T.38 on 'Asterisk calling system', if you have an ATA which is originating the call to 'Asterisk calling system' disable T.38 on that device too and disable in your sip.conf using t38pt_udptl=no. If you are using SendFax() on 'Asterisk calling system' ensure T.38 is not able to be used. If using an ATA connecting to 'Asterisk calling system' ensure you have set in your peer's configuration canreinvite=no or directmedia=no, depending on the version of Asterisk you are running on this system. On Asterisk system in '(box in question)' set directmedia=no for the peer which is connecting to 'SIP Provider' and also to 'Asterisk calling system', you may want to set setvar=FAXOPT(gateway)=yes in your peer config to 'SIP Provider' otherwise it will need to be set in your dialplan. Set your verbose debug to at least 3 on '(box in question)', possibly a little higher and send a fax - you may now see the Fax Gateway detect CED. Not sure if this is suppressed in You may want enable udptl debugging on '(box in question)'. I do *not* want to disable reinvites or udptl media as it is required for T.38 operation. All testing shows (via packet capture) no reinvite for T.38 is happening on the call leg with the ITSP. Thank you for the idea however on setting the FAXOPT for gateway in the provider SIP peer definition, I will test that shortly. --Tim -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
- Original Message - On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? The originating endpoint is an IAXmodem controlled by Hylafax. Actual call flow is IAXmodem --G.711u via localhost-- Asterisk (old version with no T.38 support) --G.711u-- Asterisk 11.x --G.711u/T.38-- ITSP The problem lies on the Asterisk 11.x system not being able to reinvite to T.38 on the call leg with the ITSP, and given the ITSP does not do this either, the call is stuck in G.711u with varying performance. :/ --Tim -- _ -- 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] 11.13.1: unable to load sip.conf (or iax )
On 10/23/2014 11:26 AM, sean darcy wrote: Running 11.13.1 on Fedora. This is a new install, but a copy of a previous - working -install. module load chan_sip Unable to load module chan_sip Command 'module load chan_sip' failed. SIP channel loading... [Oct 23 14:46:08] NOTICE[669]: chan_sip.c:31438 reload_config: Unable to load config sip.conf I don't think it's permissions: ls -ld /etc/asterisk /etc/asterisk/sip* drwxr-x---. 4 asterisk asterisk 4096 Oct 23 00:34 /etc/asterisk -rw-r-. 1 asterisk asterisk 3588 Oct 22 18:37 /etc/asterisk/sip.conf -rw-r-. 1 asterisk asterisk 91033 Oct 23 00:28 /etc/asterisk/sip.conf.rpmnew -rw-r-. 1 asterisk asterisk 790 Oct 23 00:28 /etc/asterisk/sip_notify.conf ps aux | grep asterisk asterisk 294 0.1 5.5 1076736 33364 ? Ssl 14:36 0:03 /usr/sbin/asterisk -f -C /etc/asterisk/asterisk.conf The sip module itself is loaded: module show like chan_sip Module Description Use Count chan_sip.soSession Initiation Protocol (SIP) 0 1 modules loaded I've tried my old config, and just the sip.conf.sample. Same result. FWIW: ls -l /usr/lib64/asterisk/modules/chan* -rwxr-xr-x. 1 root root 72808 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_agent.so -rwxr-xr-x. 1 root root 16032 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_bridge.so -rwxr-xr-x. 1 root root 347920 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_iax2.so -rwxr-xr-x. 1 root root 41888 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_local.so -rwxr-xr-x. 1 root root 118144 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_mgcp.so -rwxr-xr-x. 1 root root 67424 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_motif.so -rwxr-xr-x. 1 root root 11936 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_multicast_rtp.so -rwxr-xr-x. 1 root root 44392 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_phone.so -rwxr-xr-x. 1 root root 755296 Oct 23 00:29 /usr/lib64/asterisk/modules/chan_sip.so Any help appreciated. sean Weirdness: made iax.conf.simple: [general] autokill=yes [idefisk] type=friend host=dynamic context=phones (extra credit for remembering the source) module unload chan_iax2.so Unable to unload resource chan_iax2.so Command 'module unload chan_iax2.so' failed. [Oct 23 16:53:26] WARNING[669]: loader.c:571 ast_unload_resource: Unload failed, 'chan_iax2.so' is not loaded. module load chan_iax2.so Unable to load module chan_iax2.so Command 'module load chan_iax2.so' failed. [Oct 23 16:53:36] ERROR[669]: chan_iax2.c:13488 set_config: Unable to load config iax.conf But then: cp -a iax.conf.simple iax.conf cp: overwrite ‘iax.conf’? y ls -l iax* -rw-r-. 1 asterisk asterisk 74 Oct 23 16:52 iax.conf -rw-r-. 1 asterisk asterisk 652 Oct 22 18:37 iax.conf.real -rw-r-. 1 asterisk asterisk 74 Oct 23 16:52 iax.conf.simple module load chan_iax2.so Loaded chan_iax2.so cp iax.conf.real iax.conf cp: overwrite ‘iax.conf’? y module unload chan_iax2.so Unloaded chan_iax2.so module load chan_iax2.so Loaded chan_iax2.so So the simple config will load. Then if I unload it, and the real config will load !! This approach also works for sip.conf, but now have another problem : it won't recognize any of the #includes. For instance: module load chan_sip.so Unable to load module chan_sip.so Command 'module load chan_sip.so' failed. SIP channel loading... [Oct 23 17:13:43] ERROR[669]: config.c:1549 process_text_line: The file '/etc/asterisk/exts/droid.sip.conf' was listed as a #include but it does not exist. [Oct 23 17:13:43] ERROR[669]: chan_sip.c:31461 reload_config: Contents of sip.conf are invalid and cannot be parsed grep exts/droid.sip sip.conf #include /etc/asterisk/exts/droid.sip.conf ls -l /etc/asterisk/exts/droid.sip.conf -rw-r--r--. 1 asterisk asterisk 316 Oct 22 18:37 /etc/asterisk/exts/droid.sip.conf I also tried relative addressing, exts/droid.sip.conf , same problem. And, of course, all this works on the 11.10.2 server. sean -- _ -- 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] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)
From: Paul Albrecht palbre...@glccom.com Seems like now is as good a time as any to raise these issues, in fact, sooner is better than later because once developers start down a path it’s very difficult to get them change their minds no matter how much sense it makes. The fact that developers are even considering taking away user functionality like the dial plan is in of itself a very serious problem because it demonstrates they don’t see Asterisk from the user perspective. Don’t object to extending the Asterisk user interface or changing Asterisk internals. Do object to is taking away functionality that users expect, are familiar with, and has made the Asterisk project successful. Then your experience is atypical. Asterisk has been unstable for several years as developers have continually shoveled new features into the code base over several releases. That’s not necessary objectionable, it’s even to be expected; however, at some point developers need to turn their attention to less glamorous less exciting things like stability and performance. I don't think anyone is objecting to you bringing this up, as it has been mentioned at the dev con. Perhaps it is just that the tone doesn't come across properly in an email, but you are coming across as confrontational and alarmist and it seems to be setting people on edge. Matt has already chimed in that he doesn't see how it would be possible to deprecate the dial plan at this time and even if it were possible, the process would take on the order of years, giving you plenty of time to enact any contingency plans you might need. Scott G. from Digium even posited that if it were to be removed from the core, it would likely end up as a loadable module so that it wouldn't burden those who don't need it and could be loaded for those who do. These developers do not exist in a vacuum, nor do they have total control over where Asterisk goes. Influence, sure, but there is still a corporate structure out there that finds it necessary to be customer oriented. They would have to be monumentally stupid (something which I haven't seen previous evidence of) to kill off the dial plan without providing a path forward for those who depend on it. Furthermore, even if they did pull a stunt so bad as to alienate half their users, the open source code would be forked so fast as to make your head spin or people would migrate to other similar packages (Freeswitch comes to mind). Digium sells their own PBX hardware that I am sure depends on these technologies that you are afraid will go away. They have direct skin in this game too. I would be interested to know just how atypical my experience is. I have found that on my 1.6 systems I would have random crashes over time. After upgrading over multiple sites, my 11.x systems have been rock solid for the most part. I did have a case where I did a store and forward of a fax that if I tried to forward the fax and it had no file to forward would cause a crash, but other than that, I haven't seen any problems in normal day to day usage. I always thought that the general consensus was that the 11.x series was quite a bit more stable than the older versions. Kevin Larsen -- _ -- 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] Ast 13 beta 3 - Segfault when calling on pjsip trunk with directmedia=yes
Hello all,
I'm setting up a couple of test boxes and I'm running into a problem.
What I need help with is determining whether I'm going something wrong
or if I need to post a bug report. I have two asterisk 13.0-beta 3
machines set up with extensions connected to each as such:
3700 AST-A -- AST-B 3800 3801
When I place a call from 3800 to 3700 or the other way around , asterisk
seg faults on both machines at roughly the same time. All connections
are done using PJSIP. The crash occurs when the ringing extension is
answered.
If I set (directmedia=no) OR (directmedia=yes t38_udptl=yes) on the
trunk then the call completes fine. All phones and servers are on the
same LAN with no firewalls active.
The trunk between AST-A and AST-B is configured like this in pjsip.conf
and is identical on both machines:
[transport-lan]
type=transport
protocol=udp
bind=0.0.0.0
tos=af31
[pbxbeta]
type=endpoint
disallow=all
allow=g722
allow=ulaw
transport=transport-lan
context=phone-level3
aors=pbxbeta
send_rpid=no
send_pai=yes
trust_id_inbound=yes
trust_id_outbound=yes
direct_media=yes
direct_media_glare_mitigation=outgoing
;direct_media_method=update
tos_audio=46
tos_video=34
t38_udptl=no
t38_udptl_nat=no
[pbxbeta]
type=aor
contact=sip:{remote IP address}:5060
[pbxbeta]
type=identify
endpoint=pbxbeta
match={remote IP address}
The phones have the following set in pjsip.conf (snippet):
type=endpoint
disallow=all
allow=g722
allow=ulaw
transport=transport-lan
send_rpid=no
send_pai=yes
direct_media=yes
tos_audio=46
tos_video=34
Is there something I'm doing wrong here?
Thanks
-Dave
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Re: [asterisk-users] Ast 13 beta 3 - Segfault when calling on pjsip trunk with directmedia=yes
On Thu, Oct 23, 2014 at 3:32 PM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Hello all,
I'm setting up a couple of test boxes and I'm running into a problem.
What I need help with is determining whether I'm going something wrong or
if I need to post a bug report. I have two asterisk 13.0-beta 3 machines
set up with extensions connected to each as such:
3700 AST-A -- AST-B 3800 3801
When I place a call from 3800 to 3700 or the other way around , asterisk
seg faults on both machines at roughly the same time. All connections are
done using PJSIP. The crash occurs when the ringing extension is answered.
If I set (directmedia=no) OR (directmedia=yes t38_udptl=yes) on the
trunk then the call completes fine. All phones and servers are on the same
LAN with no firewalls active.
The trunk between AST-A and AST-B is configured like this in pjsip.conf
and is identical on both machines:
[transport-lan]
type=transport
protocol=udp
bind=0.0.0.0
tos=af31
[pbxbeta]
type=endpoint
disallow=all
allow=g722
allow=ulaw
transport=transport-lan
context=phone-level3
aors=pbxbeta
send_rpid=no
send_pai=yes
trust_id_inbound=yes
trust_id_outbound=yes
direct_media=yes
direct_media_glare_mitigation=outgoing
;direct_media_method=update
tos_audio=46
tos_video=34
t38_udptl=no
t38_udptl_nat=no
[pbxbeta]
type=aor
contact=sip:{remote IP address}:5060
[pbxbeta]
type=identify
endpoint=pbxbeta
match={remote IP address}
The phones have the following set in pjsip.conf (snippet):
type=endpoint
disallow=all
allow=g722
allow=ulaw
transport=transport-lan
send_rpid=no
send_pai=yes
direct_media=yes
tos_audio=46
tos_video=34
Is there something I'm doing wrong here?
Thanks
Asterisk shouldn't crash.
Please file a bug report ASAP at issues.asterisk.org, with a properly
generated backtrace:
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)
On Thu, Oct 23, 2014 at 10:45 AM, Paul Albrecht palbre...@glccom.com wrote: On Oct 23, 2014, at 1:55 AM, Olle E Johansson o...@edvina.net wrote: It is critical that a group of developers ask themself questions along these lines - what if??? - What if we removed AGi and AMI? - What if we made a pluggable PBX? - What if we restarted working on a SIP channel? - What if we made a whole new bridge architecture? - What if we skip the idea of making a PBX? Good development quite frequently starts with these kind of ideas and questions that may see crazy but results in really good changes. Brainstorms needs to be open and not restricted, that is what the astridevcons are for. We need to go wild and see what comes out of it. A lot of the great changes we see in Asterisk 13 comes from many years of wild discussions. Pinemango anyone? The unacknowledged problem we’re dealing with is the fact that we’re not starting with a blank sheet of paper, but rather a mature user interface that users expect, are familiar with, and has made project successful. Extending the the user interface is one thing, throwing it away is another entirely different thing. But hey, you have the user's community attention at Astricon, why not have the courage of your convictions and announce to the the unwashed masses you’re planning to do away with the dial plan? This will be the last time I respond to any of your e-mails on the Asterisk mailing lists or engage with you in any fashion. Your tone, language, and rhetoric are all indicative of someone who is not interested in having a discussion or being a productive member of this open source community. Good luck with your endeavors. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
On 24/10/2014 12:49 AM, Tim Nelson wrote: - Original Message - On 23/10/2014 10:07 PM, Larry Moore wrote: On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? Larry. Have you had a look at https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance As an exercise you could disable T.38 on 'Asterisk calling system', if you have an ATA which is originating the call to 'Asterisk calling system' disable T.38 on that device too and disable in your sip.conf using t38pt_udptl=no. If you are using SendFax() on 'Asterisk calling system' ensure T.38 is not able to be used. If using an ATA connecting to 'Asterisk calling system' ensure you have set in your peer's configuration canreinvite=no or directmedia=no, depending on the version of Asterisk you are running on this system. On Asterisk system in '(box in question)' set directmedia=no for the peer which is connecting to 'SIP Provider' and also to 'Asterisk calling system', you may want to set setvar=FAXOPT(gateway)=yes in your peer config to 'SIP Provider' otherwise it will need to be set in your dialplan. Set your verbose debug to at least 3 on '(box in question)', possibly a little higher and send a fax - you may now see the Fax Gateway detect CED. Not sure if this is suppressed in You may want enable udptl debugging on '(box in question)'. I do *not* want to disable reinvites or udptl media as it is required for T.38 operation. All testing shows (via packet capture) no reinvite for T.38 is happening on the call leg with the ITSP. Thank you for the idea however on setting the FAXOPT for gateway in the provider SIP peer definition, I will test that shortly. The canreinvite= option is an old setting, this is replaced by the directmedia= option in newer versions of Asterisk, it doesn't prevent a re-invite, it keeps the audio going through asterisk rather than negotiating an audio channel directly with the other endpoint. The reason I suggested disabling udptl at that end is because my understanding of how the implementation of T.38 Gateway works on Asterisk is; 1) it does not utilise any of the T.38 gateway features in spandsp 2) the gateway will not step in if the originator negotiates T.38 Considering the other post you sent, are you suing IAX between the two Asterisk boxes? To test the T.38 Gateway can work on your box in question set up an IAX modem and configure HylaFAX modem to use the iaxmodem on the box in question, test the gateway functionality. When I tested Asterisk 11 a little while back I configured HylaFAX on my current system to communicate with an IAX modem on my Asterisk 11 test box and was able to observe the T.38 gateway function. I can't tell from the information you've provided if the old Asterisk box is on the same network or having to traverse a WAN link to make the connection out through to your SIP provider. Perhaps you could provide more information about your set up such as entries from your sip.conf, iax.conf, udptl.conf etc. Larry. -- _ -- 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
