Re: [asterisk-users] Skip ./configure when source directory has not changed
On Mon, 24 Mar 2014 21:20:45 +0100 Olivier wrote: > A silly question bouncing in my head for a long time : > when I'm installing-configuring a new Asterisk system, I'm using a > script that issue the usual ./configure, make and make install > commands to install Asterisk from source. > > When installation fails for any reason, I would re-run my installation > script which in turn, among many things, would launch the > above ./configure command. > > Is there a smart way to accelerate things a bit and skip ./configure > when source files have not changed since last configure command was > previously run ? You kind of have it backwards there. You would only be able to skip ./configure if you changed the source files but didn't change anything else on the system (e.g., libraries or utilities). But there's no way for a build script to know if anything on the system changed without running ./configure. That's what ./configure does. So the only sensible thing to do is to run it every time. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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 12.1.1 - Having trouble setting up PJSIP
Dan Cropp wrote: I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001@192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes. I’m not seeing anything in the messages log. I’m sure I’m doing something wrong, just not sure where to look or how to track down the problem. It certainly shouldn't crash no matter what you do. Can you get a backtrace[1] and file an issue[2] so we can take care of this? The information you've provided in this email would also be useful. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [2] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skip ./configure when source directory has not changed
2014-03-25 18:42 GMT+01:00 jg : > I, and possibly others, got some unwanted mail from this thread. Somebody > is abusing the email addresses... Maybe but asking question which relates to asterisk is IMHO, an expensive way to harvest email addresses. > > > jg > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11, Local channels and Queues
Hi, On a 11.8.1 system, I'm trying to configure a queue in which busy agents do no get incoming calls. I dynamically add agents with lines (shamelessly copied from a running Freepbx/Asterisk 1.8 system) such as: AddQueueMember(myqueue,Local/123@agent/n,10,,hint:123@subs) Basically, it works but incoming calls are sent to busy agents. In sip.conf, callcounter is set to yes. With CLI, I can read (ringinuse disabled) for each agent. When an agent is on call, "core show hint 123" would output something like: 123@subs: SIP/mac0123456789AB State:InUse Watchers 0 I would suspect I didn't provide appropriate state interface to AddQueueMember. Suggestions ? Regards -- _ -- 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 12.1.1 - Having trouble setting up PJSIP
Additional information with "pjsip set logger on" - Register succeeds... - <--- Received SIP request (485 bytes) from UDP:192.168.9.142:5063 ---> REGISTER sip:192.168.9.234 SIP/2.0 Via: SIP/2.0/UDP 192.168.9.142:5063;branch=z9hG4bK-deea79e7 From: "7001" ;tag=ee56a5177681851fo3 To: "7001" Call-ID: a93c73c5-83c75033@192.168.9.142 CSeq: 25282 REGISTER Max-Forwards: 70 Contact: "7001" ;expires=3600 User-Agent: Cisco/SPA504G-7.4.8a Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces <--- Transmitting SIP response (469 bytes) to UDP:192.168.9.142:5063 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.9.142:5063;rport;received=192.168.9.142;branch=z9hG4bK-deea79e7 Call-ID: a93c73c5-83c75033@192.168.9.142 From: "7001" ;tag=ee56a5177681851fo3 To: "7001" ;tag=z9hG4bK-deea79e7 CSeq: 25282 REGISTER WWW-Authenticate: Digest realm="asterisk",nonce="1395782973/f72250272122471132aabf25deed1c0b",opaque="110098de72b0d893",algorithm=md5,qop="auth" Content-Length: 0 <--- Received SIP request (740 bytes) from UDP:192.168.9.142:5063 ---> REGISTER sip:192.168.9.234 SIP/2.0 Via: SIP/2.0/UDP 192.168.9.142:5063;branch=z9hG4bK-f5a029e3 From: "7001" ;tag=ee56a5177681851fo3 To: "7001" Call-ID: a93c73c5-83c75033@192.168.9.142 CSeq: 25283 REGISTER Max-Forwards: 70 Authorization: Digest username="7001",realm="asterisk",nonce="1395782973/f72250272122471132aabf25deed1c0b",uri="sip:192.168.9.234",algorithm=MD5,response="e234a6e6abf82aec119d49a413e0a9b1",opaque="110098de72b0d893",qop=auth,nc=0001,cnonce="9c4b3692" Contact: "7001" ;expires=3600 User-Agent: Cisco/SPA504G-7.4.8a Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces -- Added contact 'sip:7001@192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds <--- Transmitting SIP response (442 bytes) to UDP:192.168.9.142:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.9.142:5063;rport;received=192.168.9.142;branch=z9hG4bK-f5a029e3 Call-ID: a93c73c5-83c75033@192.168.9.142 From: "7001" ;tag=ee56a5177681851fo3 To: "7001" ;tag=z9hG4bK-f5a029e3 CSeq: 25283 REGISTER Date: Tue, 25 Mar 2014 21:29:33 GMT Contact: ;expires=3599 Contact: Content-Length: 0 - Dialing 1 from phone below. - *CLI> <--- Received SIP request (898 bytes) from UDP:192.168.9.142:5063 ---> INVITE sip:1@192.168.9.234 SIP/2.0 Via: SIP/2.0/UDP 192.168.9.142:5063;branch=z9hG4bK-9b8d1e07 From: "7001" ;tag=9fa6d06bfc4546d4o3 To: Call-ID: 6353f577-bd7d8538@192.168.9.142 CSeq: 101 INVITE Max-Forwards: 70 Contact: "7001" Expires: 240 User-Agent: Cisco/SPA504G-7.4.8a Content-Length: 393 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 8644 8644 IN IP4 192.168.9.142 s=- c=IN IP4 192.168.9.142 t=0 0 m=audio 16462 RTP/AVP 0 2 8 9 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <--- Transmitting SIP response (455 bytes) to UDP:192.168.9.142:5063 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.9.142:5063;rport;received=192.168.9.142;branch=z9hG4bK-9b8d1e07 Call-ID: 6353f577-bd7d8538@192.168.9.142 From: "7001" ;tag=9fa6d06bfc4546d4o3 To: ;tag=z9hG4bK-9b8d1e07 CSeq: 101 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1395783027/7ae9aaf5d61fc322eac8dec60d9c8dbe",opaque="13d5988e59a920a6",algorithm=md5,qop="auth" Content-Length: 0 <--- Received SIP request (381 bytes) from UDP:192.168.9.142:5063 ---> ACK sip:1@192.168.9.234 SIP/2.0 Via: SIP/2.0/UDP 192.168.9.142:5063;branch=z9hG4bK-9b8d1e07 From: "7001" ;tag=9fa6d06bfc4546d4o3 To: ;tag=z9hG4bK-9b8d1e07 Call-ID: 6353f577-bd7d8538@192.168.9.142 CSeq: 101 ACK Max-Forwards: 70 Contact: "7001" User-Agent: Cisco/SPA504G-7.4.8a Content-Length: 0 <--- Received SIP request (1155 bytes) from UDP:192.168.9.142:5063 ---> INVITE sip:1@192.168.9.234 SIP/2.0 Via: SIP/2.0/UDP 192.168.9.142:5063;branch=z9hG4bK-d1aac763 From: "7001" ;tag=9fa6d06bfc4546d4o3 To: Call-ID: 6353f577-bd7d8538@192.168.9.142 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="7001",realm="asterisk",nonce="1395783027/7ae9aaf5d61fc322eac8dec60d9c8dbe",uri="sip:1@192.168.9.234",algorithm=MD5,response="c0f7e47e6af69559a266c3ec22793ff0",opaque="13d5988e59a920a6",qop=auth,nc=0001,cnonce="9adbf5ea" Contact: "7001" Expires: 240 User-Agent: Cisco/SPA504G-7.4.8a Content-Length: 393 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 8644 8644 IN IP4 192.168.9.142 s=- c=IN IP4 192.168.9.142 t=0 0 m=audio 16462 RTP/AVP 0 2 8 9 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-3
[asterisk-users] Asterisk 12.1.1 - Having trouble setting up PJSIP
I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001@192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes. I'm not seeing anything in the messages log. I'm sure I'm doing something wrong, just not sure where to look or how to track down the problem. Can anyone offer some hints? - pjsip.conf - [transport-udp] type=transport protocol=udp bind=0.0.0.0 [7001] type=endpoint transport=transport-udp context=IS disallow=all allow=ulaw auth=7001 aors=7001 [7001] type=aor max_contacts=1 contact=sip:7001@192.168.9.142:5063; Line 4 on my phone is setup for port 5063. ; I have also tried without this setting and am seeing the exact same scenario [7001] type=auth auth_type=userpass password=1234 username=7001 - extensions.conf - [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=DAHDI/G2 ; Trunk interface TRUNKMSD=1 [IS] exten => 1,1,Verbose(1,Unrouted call handler) exten => 1,n,Answer() exten => 1,n,Wait(1) exten => 1,n,Playback(tt-weasels) exten => 1,n,Hangup() Have a great day! Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skip ./configure when source directory has not changed
I, and possibly others, got some unwanted mail from this thread. Somebody is abusing the email addresses... jg -- _ -- 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] Skip ./configure when source directory has not changed
2014-03-25 10:35 GMT+01:00 jg : > This wasn't a technical question. It's scam to get some fresh email > addresses. > what for ? ;-)) > > jg > > > -- > _ > -- 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] Skip ./configure when source directory has not changed
2014-03-25 10:18 GMT+01:00 A J Stiles : > On Monday 24 Mar 2014, Olivier wrote: > > Is there a smart way to accelerate things a bit and skip ./configure when > > source files have not changed since last configure command was previously > > run ? > > That probably isn't what you really want to do. > > The most common reason why a build fails is because a -dev package is > missing. > Alternatively, some required component might be present, but located > somewhere > away from the usual search path. > > Once either of these situations is resolved, you are going to need to > re-run > the configure script anyway. > That's true but if you compare this with Makefile's ability to check files last modification date ... Maybe I'll just leave it to the man running the script to decide when to run configure and it's not needed. Thanks for sharing on this anyway. > -- > AJS > > Answers come *after* questions. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Spammer direct replying to those posting on the users list
We apparently have a spam bot subscribed to the list and replying *directly* to anyone who posts on the list. The E-mails, generally use the name Alyssa or a Katie in the mail and have images attached. They come from a variety of addresses that so far don't appear subscribed to the list. However spammers don't typically subscribe to lists at the addresses they send from or appear to send from. This is just a notice that we are working on it and doing what we can. -- Digium's Asterisk Development Team 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] IAXModem or T38Modem?
I would not say "happy", since there is no happiness in a world with T.38, but Level 3 supports T.38.Level 3 is wholesale only as far as I know. Vitelity has some fax service stuff too. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, March 25, 2014 11:19 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] IAXModem or T38Modem? On 03/25/2014 10:13 AM, Steven Howes wrote: > On 25 Mar 2014, at 15:00, Jeff LaCoursiere wrote: >> On 03/24/2014 05:50 PM, Thorolf Godawa wrote: >>> But your carrier has to support T38, when we began to evaluate this >>> some years ago, this was not true for all. >> Would you share the provider you are using? I have had almost zero luck so >> far. > What country are you in? Every carrier we've tried in the UK supported T38 > without problem (so far.. I expect there is a few that don't). Not sure I can > name specific carriers as this is the non-commercial list though. > > Steve Sorry - should have mentioned USA. I say "almost" zero luck, because I have managed to get a few faxes out with a handful of providers tested, but none consistently. If anyone is very happy with their T.38 provider, please email me off list. I got the spam message. At least she is cute. Cheers, j -- _ -- 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] Asterisk 11.8.0 and 11.8.1
On Tue, Mar 25, 2014 at 10:13 AM, Jerry Geis wrote: > OK back in the office so I can get some files > > in my confbridge.conf file > [MessageNetConfUserMuted] > type=user > quiet=yes > startmuted=yes > announce_only_user=no > announce_user_count_all=no > announce_join_leave=no > > This is from the console so 410 and 411 are joining as MUTED. > This all worked fine 11.0 -> 11.7. I have only encountered problems with > 11.8+ > dropping back to 11.7 works again. > > So how do I found out if its something I have wrong or was this introduced > in 11.8+ Hey Jerry, when something like this occurs suddenly between minor release versions, you can always check issues.asterisk.org/jira to see if it has been reported. A search for the words muted and confbridge, then ordering the results by creation date will show this issue: https://issues.asterisk.org/jira/browse/ASTERISK-23461 Which looks like the same issue that you are having. If you click on the source tab you can see the commits it was fixed in. Looks like it was after 11.8.1, so you'll have to wait until the next release, or grab 11 from SVN. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] IAXModem or T38Modem?
On 03/25/2014 10:13 AM, Steven Howes wrote: On 25 Mar 2014, at 15:00, Jeff LaCoursiere wrote: On 03/24/2014 05:50 PM, Thorolf Godawa wrote: But your carrier has to support T38, when we began to evaluate this some years ago, this was not true for all. Would you share the provider you are using? I have had almost zero luck so far. What country are you in? Every carrier we’ve tried in the UK supported T38 without problem (so far.. I expect there is a few that don’t). Not sure I can name specific carriers as this is the non-commercial list though. Steve Sorry - should have mentioned USA. I say "almost" zero luck, because I have managed to get a few faxes out with a handful of providers tested, but none consistently. If anyone is very happy with their T.38 provider, please email me off list. I got the spam message. At least she is cute. Cheers, j -- _ -- 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] Spammer direct replying to those posting on the users list
On Tuesday 25 Mar 2014, Digium's Asterisk Development Team wrote: > We apparently have a spam bot subscribed to the list and replying > *directly* to anyone who posts on the list. The e-mail address I use for this mailing list is asterisk_l...@earthshod.co.uk ; so I used the following procmail recipe. This filters out anything being sent to that address *without* a Received: header mentioning lists.digium.com: :0 * ^To.*asterisk_list * !^Received.*lists.digium.com asterisk_unwanted (when I am satisfied that it does not lose anything legitimate, I probably will change the last line to /dev/null .) -- AJS Note: Originating address only accepts e-mail from list. If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 11.8.0 and 11.8.1
OK back in the office so I can get some files
in my confbridge.conf file
[MessageNetConfUserMuted]
type=user
quiet=yes
startmuted=yes
announce_only_user=no
announce_user_count_all=no
announce_join_leave=no
This is from the console so 410 and 411 are joining as MUTED.
^[[Kdevgeis*CLI> ^M^[[0K > Channel SIP/411-001c was answered
^M^[[Kdevgeis*CLI> ^M^[[0K-- Executing
[smvoice_pa_app_confbridge_intercom@smvoice-transfers:1]
ConfBridge("SIP/411-001c",
"PA0014,MessageNetConfBridge,MessageNetConfUserMuted") in new s
^M^[[Kdevgeis*CLI> ^M^[[0K > 0x7f343c0137c0 -- Probation passed -
setting RTP source address to 192.168.1.38:8000
^M^[[Kdevgeis*CLI> ^M^[[0K > Channel SIP/410-001d was answered
^M^[[Kdevgeis*CLI> ^M^[[0K-- Executing
[smvoice_pa_app_confbridge_intercom@smvoice-transfers:1]
ConfBridge("SIP/410-001d",
"PA0014,MessageNetConfBridge,MessageNetConfUserMuted") in new s
^M^[[Kdevgeis*CLI> ^M^[[0K > 0x7f345c013c30 -- Probation passed -
setting RTP source address to 192.168.1.39:8000
^M^[[Kdevgeis*CLI> ^M^[
yet I hear feedback on the speaker.
I start the call by calling a Local channel, with variables of what file
(wave) to play and what devices
to bring in conference.
This all worked fine 11.0 -> 11.7. I have only encountered problems with
11.8+
dropping back to 11.7 works again.
So how do I found out if its something I have wrong or was this introduced
in 11.8+
Thanks,
Jerry
On Mon, Mar 24, 2014 at 7:58 PM, Jerry Geis wrote:
> I have used every asterisk 11.8.X version.
> Have not had an issue till 11.8.0 and 11.8.1
> When I use ConfBridge I am attempting to put all
> participants in MUTE mode and just one talker...
>
> However, since 11.8.0 I am hearing feedback in the
> announcement like the channel is not really muted.
> I dropped back to 11.7.0 and I hear no feedback.
>
> Has something changed - or - am I not correctly setting
> up the confbridge? How do I tell whats up?
>
> Thanks
>
> Jerry
>
--
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Re: [asterisk-users] IAXModem or T38Modem?
On 25 Mar 2014, at 15:00, Jeff LaCoursiere wrote: > On 03/24/2014 05:50 PM, Thorolf Godawa wrote: >> >> But your carrier has to support T38, when we began to evaluate this some >> years ago, this was not true for all. > Would you share the provider you are using? I have had almost zero luck so > far. What country are you in? Every carrier we’ve tried in the UK supported T38 without problem (so far.. I expect there is a few that don’t). Not sure I can name specific carriers as this is the non-commercial list though. Steve -- _ -- 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] IAXModem or T38Modem?
On 03/24/2014 05:50 PM, Thorolf Godawa wrote: Hi, I'm installing Hylafax on my Asterisk system. From what I've read, I can either use IAXModem or T38Modem to provide the virtual fax device. we are using T38modem, it was a long way to get it stable, but finally it works quite good with 10 parallel running T38modems. But your carrier has to support T38, when we began to evaluate this some years ago, this was not true for all. Would you share the provider you are using? I have had almost zero luck so far. Thanks, j -- _ -- 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] Spammer direct replying to those posting on the users list
On 25 Mar 2014, at 14:16, Digium's Asterisk Development Team wrote: > We apparently have a spam bot subscribed to the list and replying > *directly* to anyone who posts on the list. There’s plenty of people harvesting the list archives too, I get loads of spam about gateways etc :( S -- _ -- 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] asteriskdocs.org says 3rd ed. is latest
Le 2014-03-24 13:05, Paul Belanger a écrit : On Mon, Mar 24, 2014 at 8:54 AM, Matt Behrens wrote: I made myself look a little silly recently in a talk regarding asteriskdocs.org. I didn't realize the 4th ed. of the Definitive Guide was apparently actually out (http://shop.oreilly.com/product/0636920025894.do), because I went by asteriskdocs.org's claim that it was being worked on in OFPS (now retired, apparently.) Is the 4th ed. available to read online like the 3rd ed. was? Is someone on this list able to update asteriskdocs.org with current info regardless? I pinged Leif Madsen on this and he's updating the site now. Currently only the 3rd edition is published online. You can find it here : http://it-ebooks.info/book/2332/ -- _ -- 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] Need more meetme users -- hitting some limit
2014-03-21 18:54, Steve Totaro skrev: I found below here: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe If you have too many conferences, one CPU may not be able to mix all the audio and you will have audio problems even if there are 7+ other CPUs that are essentially idle while waiting for one CPU to mix everything. You should be able to handle 512 conference participants on a modern server system without problem. The current trunk of *DAHDI linux limits the number of open pseudo channels to 512 for this reason*. [1] Thanks, Steve T I would check /proc/interrupts also. On some distros irq's are not balanced by default and are all hitting the same core. On Debian I had to install the irqbalance package and the load was spread across the cores. Dahdi is still a single thread thought. -- Johan Wilfer -- _ -- 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] Skip ./configure when source directory has not changed
This wasn't a technical question. It's scam to get some fresh email addresses. jg -- _ -- 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] Skip ./configure when source directory has not changed
On Monday 24 Mar 2014, Olivier wrote: > Is there a smart way to accelerate things a bit and skip ./configure when > source files have not changed since last configure command was previously > run ? That probably isn't what you really want to do. The most common reason why a build fails is because a -dev package is missing. Alternatively, some required component might be present, but located somewhere away from the usual search path. Once either of these situations is resolved, you are going to need to re-run the configure script anyway. -- AJS Answers come *after* questions. -- _ -- 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
