Re: [asterisk-users] Looking for better fax handling
On 5/21/18 1:49 PM, D'Arcy Cain wrote: I am having troubles with sending faxes. I hope someone can help me work out a better method. I have a project that I like to use to send faxes. It might be able to drop into your environment pretty easily. https://github.com/jkister/astelegraph I use samba to get the files from the workstation to the server, but using SSH or email is just as easy -- it'll pick up files dropped in /var/spool/asterisk/fax/raw. -- _ -- 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
Re: [asterisk-users] app_swift w/ Asterisk 14
On 2/3/17 3:16 PM, Brent Davidson wrote: > Trying to compile app_swift with Asterisk 14.2.1 and getting the > following. Can anybody tell me what I'm missing?: app_swift has not been updated for asterisk 14. i've maintained it for a while but haven't done anything with asterisk 14. the project certainly needs more hands on the code. regardless of asterisk14, an important bugfix is at https://github.com/jkister/app_swift but darren has not accepted/rejected the changes. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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
Re: [asterisk-users] recreating extensions.conf from live dialplan ?
On 4/13/2016 1:26 PM, Steve Edwards wrote: This should get you close: sudo asterisk -r -x 'dialplan show' >extensions.wip and then feed extensions.wip through: Ya, that's pretty good! besides the fact that I've never used "same" (i understand where it's coming from) and a few contexts confuzzled (missing general/globals and extra parkedcalls - but again I get it) - it seems to be perfect. One for a wiki, somewhere. thanks, -- Jeremy Kister http://jeremy.kister.net/ -- _ -- 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] recreating extensions.conf from live dialplan ?
On 4/13/16 11:57 AM, A J Stiles wrote: You could try *CLI> dialplan show Between my older backup and dialplan show, I guess that's my best shot. Thanks :D -- _ -- 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] recreating extensions.conf from live dialplan ?
On 4/13/16 11:37 AM, Steve Edwards wrote: Will 'dialplan save' help? I just tried this one. It writes the dialplan, but without the application arguements. Worthless. right, was a good shot. in my case I have writeprotect=yes in general, so that would have been the first hurdle. but asterisk does have my latest-and-greatest code in memory and active in it's dialplan. hoping for something similar... -- _ -- 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] recreating extensions.conf from live dialplan ?
with the slip of a finger, i destroyed by extensions.conf (grep -i > extensions.conf) I have a backup that is dozens of hours of code old. is there a way i can use the asterisk cli (or some other asterisky method) to recreate that extensions.conf ? -- _ -- 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] [announce] astelegraph
i looked over the archives for this list and didnt really see any add-on announcements (and asterisk-announce seems just dev info and tumbleweed) so forgive me if i upset anyone here. I just hooked up a little package i call astelegraph. Unlike a lot of fax solutions that provide a fax <-> email gateway, it makes outbound faxes easier by letting users place a file into a folder that's instantly faxed to whatever the filename is (no cron or every-minute checking ickyness). e.g., 211234.pdf gets faxed to 215-555-1234. other neatness include putting an email address in the filename for reporting and/or putting a timestamp in the filename for scheduling. i find it nice to be able to scp a file to the server and have it faxed. and i set up a samba share to the same directory so i can drag a file right into the window and have it faxed. http://github.com/jkister/astelegraph let me know if you find it useful, -- Jeremy Kister http://jeremy.kister.net/ -- _ -- 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] moving from meetme to confbridge
I'm moving away from meetme to confbridge. the only remaining task i have is to convert (a ton of): exten => x,n,Page(SIP/123/124&...,diqA(mysound)) or exten => x,n,Page(SIP/123/124&...,iq) (both inbound and outbound) I started down a very long road with creating call files and joining a conference but it got complicated very quickly. sometimes I use Page to do one-way intercom or two-way intercom -- got that working, albeit crazy. But other times I use a callfile/AMI to connect to a context that plays TTS -- and i don't see how i can link my TTS into the confbridge like I could with Page. Is there an easier replacement of app_page ? I'd hate to keep dahdi+meetme just for Page. I would post here what I have so far, but it's so complex it would be a headache to explain what I was thinking. -- Jeremy Kister http://jeremy.kister.net/ -- _ -- 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] app_swift crash asterisk 11.20.0-rc1
I found the app_swift module (that I've been helping maintain) makes
asterisk crash on versions higher than 11.19.0 - something that happened
on 11.20.0-rc1 makes asterisk segfault. I realize app_swift is not a
'supported' module -- I'm just having a hard time finding the cause and
am wondering if I could borrow anyone's eyes.
of note, app_swift doesnt /always/ crash asterisk, e.g., when I call
into asterisk from a phone and swift is in the dialplan, all seems fine.
it seems that it's just when I make a callfile that dials out.
a backtrace is at http://pastebin.com/Dfd4P8sK
replication is easy (if you have swift):
echo "testing 1 2 3" > /var/lib/asterisk/tts
cat <<__EOE__ >> /etc/asterisk/extensions.conf
[intercom]
exten => _2XZ,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => _2XZ,n,Page(SIP/${EXTEN},diqA(local/intercom))
[tts]
exten => s,1,Wait(1)
exten => s,n,GotoIf($[0${LEN(${TEXT})} > 1]?text)
exten => s,n,Set(SPEECH=${SHELL(cat /var/lib/asterisk/tts)})
exten => s,n,Goto(swift)
exten => s,n(text),Set(SPEECH=${TEXT})
exten => s,n,NoOp(${SPEECH})
exten => s,n(swift),Swift(${SPEECH})
exten => s,n,Hangup
__EOE__
cat <<__EOS__ > /var/spool/asterisk/tmp/test123
Channel: Local/221@intercom
Callerid: "TTS" <0>
MaxRetries: 2
WaitTime: 45
Context: tts
Extension: s
Priority: 1
__EOS__
mv /var/spool/asterisk/tmp/test123 /var/spool/asterisk/outgoing/test123
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[asterisk-users] Asterisk 13/PJSIP + registration
Using Asterisk 13.3.2 on CentOS7 and pjproject 2.4, I can't make asterisk try to send a register. I have configured my pjsip.conf similar to https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples#res_pjsipConfigurationExamples-ASIPtrunktoyourserviceprovider,includingoutboundregistration my pjsip.conf: http://pastebin.com/raw.php?i=EA0PEcrb using tcpdump, I never even see a packet sent from asterisk trying to register. on the asterisk console: asterisk13*CLI pjsip show registrations No objects found. asterisk13*CLI pjsip show contacts Contact: Aor/ContactUri... Status RTT(ms).. = Contact: provider1/sip:1xxxnnny...@sip.provider1.com Unknown nan asterisk13*CLI pjsip list aors Aor: Aor.. MaxContact = Aor: provider1 0 FYI, I can modify pjsip.conf to add configuration for a softphone to register to asterisk - that works fine. Can someone give me a clue on how to make this outbound registration happen ? -- _ -- 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] Allison Smith AMA
For anyone interested, Allison Smith's AMA (not sure she's still around): http://www.reddit.com/r/IAmA/comments/2rrb7m/iama_professional_telephone_voice_ama/ -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 stun setup , not using public ip returned by stun server
On 10/14/2014 2:25 AM, chandapure shiva wrote: I have put nat =force_rport,comedia in general section , but still not working . I hate to ask, but did you reload sip afterwards? asterisk -rx 'sip reload' -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 stun setup , not using public ip returned by stun server
On 10/13/2014 2:50 AM, chandapure shiva wrote: In above packet VIA and CONTACT SIP-HEADERS contains the asterisk server private IP address which is behind the NAT , as per my understanding it supposed to be the public ip address of my network. do you also have the appropriate nat statement in sip.conf ? since you have the 'stun show status' command, i beleive the correct nat statement is nat=force_rport,comedia in the general section. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 app_swift is live
Darren has recently merged my changes to app_swift - now supporting asterisk 12 and 13. If anyone has the Cepstral TTS engine installed and would like to link it with asterisk, app_swift is the way to go. This is the first version that 'configures' to make a Makefile. Please give it a try and report back any issues. git clone 'https://github.com/darrensessions/app_swift' cd app_swift configure make make install make reload -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 help troubleshooting Asterisk Auto dial out problem
On 4/30/2014 7:24 PM, Jesse Thompson wrote: impacted. However new files introduced into /var/spool/asterisk/outgoing/ folder get ignored. No messages spring up on asterisk -rvv console, nothing shows up in the logs, the .call files just get snubbed. We're at a loss to Are the new files being named uniquely ? there are bugs (e.g., jira# 11291) that have to do with files having the same name. my solution was to add .$$ on the end of the filename to ensure it was unique. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Replying to Posts
On 3/13/2014 11:33 AM, A J Stiles wrote: If you need to make a point-by-point argument, split up your reply -- a critical piece to this component is proper quoting. the person replying needs to differentiate between what he is writing and what is is replying to. notice the in front of what I am quoting, above. in addition, clicking reply, quoting 100 lines, and then adding a 1 line response is lazy. trim the quotation to what makes sense. that said, i love a good top-post flame thread, so this should be interesting to watch. I'll start off by saying the biggest whine i hear is that my MUA doesn't support bottom-posting, which holds no water. i dont care that much, though- i don't waste time on top-posted messages a nor messages that are quoted stupidly. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] help with Cepstral 6 and Asterisk 11
On 1/8/2014 9:12 PM, Brandon Coale wrote: However, I am not able to get app_swift to compile. I am running Asterisk 11.6.0 and CentOS 6.4 64-bit. I am wondering if anyone else out there has been able to get app_swift working with Asterisk 11 and could share any tricks they used to get it installed? can you pastie your configure and make ? I don't have Cepstral6 but did submit tweaks to the code that should have made it Cepstral6 compatible. Also since you recently spent money with Cepstral, they'll help you. They've got at least one guy who understands the app_swift code and was working on forking it as an official version. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Recurring SIP problem with asterisk 11.6 11.7
I have regularly (once a week, once per few hundred calls?) been having problems with Asterisk's SIP stack not responding to packets from any of my registered devices. In the past, I could not tolerate the outage, so i would restart asterisk to make things happy. My Asterisk server is currently in this broken state and I can leave it this way for a short while. Devices are registered to it and I can 'sip qualify peer xxx'. 'sip show peer xxx' all show Status OK. but whenever one of the devices tries to make a new call, Asterisk just doesnt respond. 'sip set debug on' shows no packets. from the asterisk server (10.1.0.3), i can see one of my phones (10.1.0.111) trying to make a call: # tcpdump -i eth0 -s 0 -t -n host 10.1.0.111 ARP, Request who-has 10.1.0.3 tell 10.1.0.111, length 46 ARP, Reply 10.1.0.3 is-at 00:0c:29:07:39:8e, length 28 IP 10.1.0.111.5060 10.1.0.3.5060: SIP, length: 926 IP 10.1.0.111.5060 10.1.0.3.5060: SIP, length: 926 IP 10.1.0.111.5060 10.1.0.3.5060: SIP, length: 926 IP 10.1.0.111.123 10.1.0.3.123: NTPv3, Client, length 48 IP 10.1.0.3.123 10.1.0.111.123: NTPv3, Server, length 48 IP 10.1.0.111.5060 10.1.0.3.5060: SIP, length: 926 IP 10.1.0.111.5060 10.1.0.3.5060: SIP, length: 926 IP 10.1.0.111.5060 10.1.0.3.5060: SIP, length: 926 IP 10.1.0.111.123 10.1.0.3.123: NTPv3, Client, length 48 IP 10.1.0.3.123 10.1.0.111.123: NTPv3, Server, length 48 ARP, Request who-has 10.1.0.111 tell 10.1.0.3, length 28 ARP, Reply 10.1.0.111 is-at 00:13:c4:01:da:4a, length 46 IP 10.1.0.111.5060 10.1.0.3.5060: SIP, length: 926 IP 10.1.0.111.5060 10.1.0.3.5060: SIP, length: 926 IP 10.1.0.111.5060 10.1.0.3.5060: SIP, length: 926 any ideas how we can find out what's upset ? -- _ -- 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] Recurring SIP problem with asterisk 11.6 11.7
On 11/12/2013 7:37 PM, Jeremy Kister wrote: any ideas how we can find out what's upset ? more info: when I create a /var/spool/asterisk/outgoing/callfile (with multiple SIP/xxxSIP/yyy), the extensions ring. but when i answer with the handset the call does not connect and the other extensions continue ringing. if i am in the asterisk CLI while the phones are ringing, i can use 'sip show channels' and see the extensions in Init: INVITE. but if i use channel request hangup tab the session hangs. I can strace these hung rasterisk, but nothing's useful: # strace -p 25331 Process 25331 attached - interrupt to quit read(3, ^C unfinished ... Process 25331 detached # strace -p 26727 Process 26727 attached - interrupt to quit read(3, ^C unfinished ... Process 26727 detached # strace -p 26768 Process 26768 attached - interrupt to quit read(3, ^C unfinished ... Process 26768 detached the ringing eventually times out, but still no errors on the console. -- _ -- 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] Recurring SIP problem with asterisk 11.6 11.7
On 11/12/2013 8:46 PM, Duncan Turnbull wrote: Any chance DNS is dying about the same time the problem occurs good idea, but I don't use DNS anywhere in Asterisk. well, except for sip.conf:externhost. it's all IP addresses. -- _ -- 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 11.6 nat problem
using asterisk 11.6.0-rc1 i just converted my nat=yes to nat=auto_force_rport,auto_comedia I have my asterisk box on the same subnet as a cisco 1760 (vgw1). a few times per day, Asterisk thinks vgw1 is dead (by qualify/options). A 'sip reload' always fixes the problem. i left 'sip set debug peer vgw1' on the console. but i dont see what's causing the issue.. http://kister.net/tmp/ast-sip.conf http://kister.net/tmp/ast-console.txt can anyone spot the issue? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] OT: Asterisk loses Oprah on live TV
just thought this was cute enough to pass along, https://www.youtube.com/watch?feature=player_detailpagev=GLwct15X_3g#t=135 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 drop calls after 15 minutes
On 9/10/2013 7:05 AM, Administrator TOOTAI wrote: I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk Just for kicks, I would disable session-timers to see if the problem goes away. in the general section and/or each peer in sip.conf: session-timers=refuse -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Strange issues with newly rebooted machine
On 8/6/13 5:30 AM, Mike Diehl wrote: sip show peer voice12 load This command just returns, with no output. throwing out a random idea since it's early in the morning and you might be in a big jam... assuming the sip isnt working correctly at all (and its not just a console issue), after asterisk is started, perhaps try core set verbose 10, core set debug 10, module unload chan_sip.so, and module load chan_sip.so . if there are any errors loading the module it may be easy to spot them. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] chanstats console errors
On 5/9/2013 3:13 PM, asterisk...@jeremykister.com wrote: I frequently see on the console: WARNING[7832]: chan_sip.c:19134 show_chanstats_cb: Could not get RTP stats bump. (sorry). -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] qualify=yes: OPTIONS: How to Change?: `From: asterisk`
On 5/9/13 8:21 PM, Brian LaVallee wrote: When qualify is enabled on a trunk, the From line shows asterisk. See the SIP message below. I had the same annoyance/issue. fixed it in https://issues.asterisk.org/jira/browse/ASTERISK-17616 the patch was included in 1.8.9 rc1. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] cisco 7940 and asterisk 11
On 2/14/2013 1:20 AM, Julian Lyndon-Smith wrote: this is a real issue for us - anyone got _any_ clues or ideas ? Ever since we upgraded to asterisk 11 we have had audio problems with our cisco 7940 phones. I use all 7940 with my asterisk 1.8 upgraded to asterisk 11. I havent had any issues with call quality whatsoever. i'm running sip image 03-08-12 g711ulaw only. -- Jeremy Kister http://jeremy.kister.net./ -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 AGI
I recently upgraded to asterisk 11 from 1.8.
I had VXML working via AGI in 1.8 - from extensions.conf:
[VXML]
exten = s,1,Answer
exten = s,n,Set(ENCODED=${URIENCODE(${ARG1})})
exten = s,n,AGI(agi://localhost/url=${ENCODED})
exten = s,n,Hangup
Using asterisk 11 on the same host with the same config in extensions.conf:
-- Executing [s@VXML:1] Answer(SIP/143-0043, ) in new stack
-- Executing [s@VXML:2] Set(SIP/143-0043,
ENCODED=http%3A%2F%2Fexample.com%2Fcgi-bin%2Favxml) in new stack
-- Executing [s@VXML:3] AGI(SIP/143-0043,
agi://localhost/url=http%3A%2F%2Fexample.com%2Fcgi-bin%2Favxml) in new
stack
[Feb 11 16:28:45] ERROR[28501][C-0012]: utils.c:1187
ast_carefulwrite: write() returned error: Connection refused
[Feb 11 16:28:45] WARNING[28501][C-0012]: res_agi.c:1528
launch_netscript: Connect to
'agi://localhost/url=http%3A%2F%2Fexample.com%2Fcgi-bin%2Favxml' failed:
Connection refused
-- Executing [s@VXML:4] Hangup(SIP/143-0043, ) in new stack
== Spawn extension (VXML, s, 4) exited non-zero on 'SIP/143-0043'
however, my daemon listening on port 4573 never sees activity.
so i set up a super-simple server* on port 4573 and saw that Asterisk is
not attempting the connection.
can someone replicate this behavior ? Or is this just my config ?
* http://jeremy.kister.net/code/asterisk/simple_agid.pl
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Jeremy Kister
http://jeremy.kister.net./
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Re: [asterisk-users] asterisk 11 AGI
On 2/11/2013 11:13 PM, Jeremy Kister wrote: [Feb 11 16:28:45] ERROR[28501][C-0012]: utils.c:1187 ast_carefulwrite: write() returned error: Connection refused [...] can someone replicate this behavior ? Or is this just my config ? opening issue in jira; this is a bug. https://issues.asterisk.org/jira/browse/ASTERISK-21065 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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's app_page options
On 1/26/2013 4:00 PM, Richard Mudgett wrote: features. You have found two bugs in confbridge: Issues created in jira. thanks for your input! -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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's app_page options
I have just upgraded to asterisk 11 from 1.8 I have noticed that my Page command: exten = 1,1,Page(SIP/101,diqA(local/intercom)) does not play the local/intercom sound to the conference. according to the doc at https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Page , it seems like it still should. is there something i need to do to make this work how i expect it? my confbridge.conf is vanilla; i dont see anything that needs changing. also, when the conference ends, the CLI shows: [Jan 25 23:50:52] ERROR[3746][C-000a]: confbridge/conf_state.c:47 conf_invalid_event_fn: Invalid event for confbridge user '' [Jan 25 23:50:52] ERROR[3745][C-000a]: confbridge/conf_state.c:47 conf_invalid_event_fn: Invalid event for confbridge user '' any way to hush/fix that? Thanks, -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Case-sensitivity of Dialplan variables.
On 10/1/2012 5:15 PM, Mark Michelson wrote: (HASH would be evaluated properly but hash would not). My personal opinion is that all variable evaluations should be case-sensitive. +1 case insensitivity to accommodate carelessness is evil. much easier for NoOp to tell us SIP_CODEC is unset, regardless of misspellings. I could be convinced to vote up 1s for I, 0s for O, and 3 for E. So SIP_CODEC, S1P_C0D3C, and SiP_cOdEC would all evaluate equally. The next step would be to appease the English spelling reform people by allowing SIP_KODEK too. :p -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] accept email and make phone call?
On 9/20/2012 1:31 PM, Joseph Acquisto wrote: Any ideas on how asterisk could accept an email (such as an email to SMS or num...@mybox.org sort of thing) and make a phone call to a specific number and make an announcement? that's actually what my jkSMS package does. i don't know if it'd be useful out of the box, depending on what you're trying to do. http://jeremy.kister.net/code/asterisk/jkSMS -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Trouble with call pickup using RPID with Cisco
I have a Cisco 1760V with FXO ports hooked to POTS lines talking SIP to asterisk 1.8.15.0. imagining in extensions.conf: exten = 1,1,Dial(SIP/121) exten = 2,1,Dial(SIP/121SIP/122) When a caller dials extension 2 /and/ I have trustrpid=yes generaterpid=yes sendrpid=yes in sip.conf and I use the pickup exten, the caller is disconnected. see: http://jeremy.kister.net/tmp/ast/group-with-rpid if i set the rpid generate/send = no for the cisco peer, the user is connected. see: http://jeremy.kister.net/tmp/ast/group-without-rpid calls to exten 1 work regardless of rpid settings. i have replication configs at http://jeremy.kister.net/tmp/ast/ Can someone help me determine if this is a problem with asterisk or ios ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 on Solaris/sparc
On 7/19/2012 3:50 AM, Hans Witvliet wrote: Perhaps system too busy, disk not fast enough? before doing a play-back, run iostat 1 in another window interesting. the stutter certainly correlates to minor amounts of disk i/o. when there is no stutter, there is nothing to report. but a minor amount of wait/busy lines up with the stutter. # iostat -znx 1 extended device statistics r/sw/s kr/s kw/s wait actv wsvc_t asvc_t %w %b device 0.0 72.00.0 456.0 0.1 0.11.00.9 2 4 c0t0d0 0.0 72.00.0 456.0 0.1 0.11.20.9 2 4 c0t2d0 extended device statistics r/sw/s kr/s kw/s wait actv wsvc_t asvc_t %w %b device extended device statistics r/sw/s kr/s kw/s wait actv wsvc_t asvc_t %w %b device Incase iowait is too high, try moving the files with the playback sound/speech upon tmpfs (thus eliminating the hard disk) That's worth a shot. I dont have big enough tmpfs to copy the whole sounds spool, so i: # cd /var/lib/asterisk/sounds/en/ # mkdir /tmp/sounds # ln -s /tmp/sounds tmpfs # cp mysound.ulaw tmpfs Playback(tmpfs/mysound) But it didnt help, still randomish stutter lining up with the disk. this is a great help, at least i can start hacking at things now. Thanks, -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 1.8 on Solaris/sparc
I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10. The system itself is happy and phone calls (between two parties) seem fine. Unfortunately, when a caller listens to a Playback recording, there seems to be moments of stutter - perhaps 1 second of stutter for every 10 seconds of Playback. The stutter is not consistent at the same point of the playback file. To eliminate encoding as an issue, I have only codec_ulaw/format_pcm loaded and the recording is ulaw. I've niced down the asterisk process to -19 even though I don't see asterisk taking more than 3% cpu. Is this behavior indicative of a timing problem? loading res_timing_pthread.so makes things horribly worse. i don't believe any other software timer is available for Solaris/sparc, right ? other thoughts ? Thanks, -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 on Solaris/sparc
On 7/18/2012 2:27 AM, Jeremy Kister wrote: I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10. .. ok, if the system weren't Solaris - let's say it was Debian Linux, what would be on the list of things to check for ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] # button behavior
On 6/27/2012 3:44 PM, khalid touati wrote: #, this happened: -- Started music on hold, class 'default', on SIP/USPBX2-07d5 -- SIP/8425-07d4 Playing 'pbx-transfer.gsm' (language 'en') and it gets disconnected. Anyone has a clue? do you have # assigned in /etc/asterisk/features.conf ? perhaps to put the caller on hold ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close
On 6/22/2012 10:39 PM, Darren Sessions wrote: both would be appreciated. if you can send me a backtrace, that'd be great http://jeremy.kister.net/tmp/swift/ -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close
On 6/20/2012 8:24 AM, Darren Sessions wrote: I just finished replying to your direct email (which you can disregard now as this seems to be a different problem). I'm pretty sure I know what the issue is, but I'll have to get back to you later this evening (my time). I have a different problem- i just compiled app_swift 3 from the new git repo for asterisk 1.8.13.0 asterisk loads the module fine, but as soon as i try to swift anything, asterisk core dumps. i'll be glad to post the corefile or sample extensions.conf if desired. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] using Wifi smartphones as SIP clients
On 5/7/2012 4:24 AM, Bart Coninckx wrote: has anyone any experience in using Wifi smartphones as SIP clients? Does this work properly? What models/brands are optimal for this (in terms of ease of use, battery life etc)? www.acrobits.cz has Acrobits and Groundwire, which are both great on iPhone. They also ahve software for Android, but I cant attest. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Fwd: Re: Authentication: username and password, also to be from the LAN
On 3/26/2012 1:11 PM, bilal ghayyad wrote: If it possible, then is it possible to be a configuration per user? Just expanding on Jim's answer- to allow user example with password secret from 192.168.0.*, do something like: in /etc/asterisk/sip.conf: [example] type=friend secret=secret host=dynamic deny=0.0.0.0/0 permit=192.168.0.0/24 then asterisk -rx sip reload -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Executing Script after MixMonitor is called
On 1/25/2012 10:29 AM, Faraj Khasib wrote: I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan what version of asterisk are you using ? if it's an older version of 1.8 ( 1.8.4) and you're also recording the call, you may be encountering a known bug. https://issues.asterisk.org/jira/browse/ASTERISK-17346 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] ConfBridge details
On 1/24/2012 5:32 PM, Kevin P. Fleming wrote: In essence, I would suggest not spending too much time trying to work the Asterisk 1.8 version of ConfBridge into your dialplan/repertoire, unless you really need it. The version in Asterisk 10 is much, much better. good stuff. thanks for the heads-up. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] ConfBridge details
running Asterisk 1.8.9.0-rc2, what are the ways to interface with
ConfBridge ?
I see the CLI command 'confbridge' documented for asterisk 10, but i
dont see how to interface with confbridge on 1.8
What I'm trying to do is keep track of conferences that are used.
I tried something like the below, but not only does Confbridge not
return, but i'd need something that erases the database entry after the
conference is empty, not after 1 particular user leaves.
[macro-confbridge-setup]
exten = s,1,Set(NUM=$[0${NUM} + 1]);
exten = s,n,Set(CONFNO=99${NUM})
exten = s,n,GotoIf(${DB_EXISTS(confbridge:${CONFNO})}?1)
exten = s,n,Set(DB(confbridge/${CONFNO})=1)
[foo]
exten = s,1,Macro(confbridge-setup)
exten = s,n,ConfBridge(${CONFNO})
exten = s,n,NoOp( ${DB_DELETE(confbridge/${CONFNO})} )
--
Jeremy Kister
http://jeremy.kister.net./
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Re: [asterisk-users] Asterisk 1.6.2.22 Now Available
On 12/19/2011 4:08 PM, Asterisk Development Team wrote: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.22 or for the non-404-version: http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/ChangeLog-1.6.2.22 ;p -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Trying to send customer mwi updates
On 12/9/2011 12:55 AM, Mike Diehl wrote: What am I doing wrong? perhaps try: http://jeremy.kister.net/code/asterisk/mwi.pl -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] t.38 interop with metaswitch
I'm trying to receive a t.38 fax from a Metaswitch 7.3. I have full control over the metaswitch, but it is in production. I have the Metaswitch hooked to Asterisk 1.8.7.0 (middleman named s3). Then I have the target Asterisk 1.8.7.0 with res_fax_digum 1.8.4_1.3.0 (named pbx1) registered to s3. attempts to receive fax over t.38 always error in res_fax with fax session timed-out i have debug output at: http://jeremy.kister.net/tmp/t38/pbx1.txt http://jeremy.kister.net/tmp/t38/s3.txt is the UDPTL debug on pbx1.txt (near line 474) interesting in that LOG_TAG(s) is evaluated to 'SIP/' ? I don't think my (sip|udptl|extensions).conf are interesting, but i'd be happy to post them. the only interesting tidbit is that when i changed 't38pt_udptl=yes' to 'yes,none' or 'yes,redundancy' the fax would fail with 't38 negotiation failed. fyi, g711/rtp audio detected faxes are working fine. anyone have suggestions on what i can try next? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] t.38 interop with metaswitch
On 10/11/2011 11:48 AM, Kevin P. Fleming wrote: Well, as a starting point, I'd suggest disabling directmedia (canreinvite) on s3. It should be possible for directmedia to be enabled for RTP and not interfere with UDPTL, but there could still be lingering problems there. yep, you hit the nail on the head. setting directmedia=no on s3 allows me to receive t38 faxes on pbx1. debug for successful faxes in this case are at: http://jeremy.kister.net/tmp/t38/no-reinvite-on-s3/pbx1.txt http://jeremy.kister.net/tmp/t38/no-reinvite-on-s3/s3.txt is there further changes that can be done to allow reinvite on s3? or is this something that should go to the tracker ? thanks, -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Maybe slightly OT but..
On 10/10/2011 10:08 PM, Andres wrote: I would recommend Acrobits. Not free but only a few bucks. It works fine with ATT 3G. +1 only thing i like better is it's big brother, Groundwire -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 not working for me
On 9/4/2011 10:48 PM, Joseph wrote: [globals] DYNAMIC_FEATURES=automon = not = exten = 11,1,GotoIfTime(*,*,1,jan?holiday,s,1) ; new years day hmm, the syntax seems ok. is func_logic.so loaded? asterisk -rx 'module show like logic' -- Executing [74791270@internal:1] Dial(SIP/218-000e, SIP/77804791270@pstn-5665,60,tr) in new stack == Using UDPTL CoS mark 5 [Sep 4 20:22:33] WARNING[27543]: app_dial.c:2196 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) can you post the relevant parts of your dialplan and sip.conf ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] any iLBC folks around?
since www.ilbcfreeware.org is broken, asterisk installs that want ilbc are failing. I have no idea how to contact them since the site is offline. It's been offline at least 12 hours - I can't imagine they *don't* know but at the same time it's still offline.. pbx1 dig +norecurse @a0.org.afilias-nst.info ilbcfreeware.org ns ; DiG 9.6-ESV-R1 @a0.org.afilias-nst.info ilbcfreeware.org ns ; (2 servers found) ;; global options: +cmd ;; Got answer: ;; -HEADER- opcode: QUERY, status: NOERROR, id: 58731 ;; flags: qr rd; QUERY: 1, ANSWER: 0, AUTHORITY: 2, ADDITIONAL: 0 ;; QUESTION SECTION: ;ilbcfreeware.org. IN NS ;; AUTHORITY SECTION: ilbcfreeware.org. 86400 IN NS ns84.worldnic.com. ilbcfreeware.org. 86400 IN NS ns83.worldnic.com. ;; Query time: 34 msec ;; SERVER: 199.19.56.1#53(199.19.56.1) ;; WHEN: Fri Sep 2 16:03:38 2011 ;; MSG SIZE rcvd: 84 pbx1 dig +norecurse @ns83.worldnic.com ilbcfreeware.org ns ; DiG 9.6-ESV-R1 +norecurse @ns83.worldnic.com ilbcfreeware.org ns ; (1 server found) ;; global options: +cmd ;; Got answer: ;; -HEADER- opcode: QUERY, status: SERVFAIL, id: 50898 ;; flags: qr; QUERY: 1, ANSWER: 0, AUTHORITY: 0, ADDITIONAL: 0 ;; QUESTION SECTION: ;ilbcfreeware.org. IN NS ;; Query time: 15 msec ;; SERVER: 205.178.190.42#53(205.178.190.42) ;; WHEN: Fri Sep 2 16:03:41 2011 ;; MSG SIZE rcvd: 3 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 needs iLBC fixing [was: any iLBC folks around?]
On 9/2/2011 4:15 PM, Jeremy Kister wrote: since www.ilbcfreeware.org is broken, asterisk installs that want ilbc are failing. it appears this was done on purpose since Google bought them. Asterisk is going to need fixing. I'll probably hook something up. http://www.webrtc.org/ilbc-freeware https://issues.asterisk.org/jira/browse/ASTERISK-18412 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 needs iLBC fixing [was: any iLBC folks around?]
On 9/2/2011 8:33 PM, Jeremy Kister wrote: Asterisk is going to need fixing. I'll probably hook something up. https://issues.asterisk.org/jira/browse/ASTERISK-18412 a patch and brief instructions are now available at the above URL. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Outgoing calls fail in chan_gtalk
On 8/20/2011 12:46 PM, Paul Belanger wrote: Anybody able to confirm the patch in ASTERISK-18301[1] fixes outbound google voice calls? confirmed on asterisk 1.8.6.0-rc1 pre-patch behavior: ring-no-answer post-patch behavior: expected -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Multiple Asterisk Sessions on same machine
On 7/19/2011 2:07 PM, Michael wrote: We would like Asterisk to listen on port 5060 and on an additional port. From what we read online, it's not really possible, so is it possible to if you're running iptables, you can set up a pretty simple rule to forward your additional port to 5060. http://www.cyberciti.biz/faq/linux-port-redirection-with-iptables/ remember UDP vs TCP. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] SoftHangup on asterisk 1.8.2.3
On 7/7/2011 9:32 AM, Ishfaq Malik wrote:
I'm having the same issue on 1.8.3.2 (with a couple of patches)
exten = s,1,Set(CHAN=${SHELL(asterisk -rx core show channels | awk
'/^SIP\/vgw1-/ { print $1 }' | head -1)})
This turned out to be a PEBKAC error. A newline was attached to the
$CHAN variable.
adding | tr -d '\n' to the end of the command fixed it right up.
--
Jeremy Kister
http://jeremy.kister.net./
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[asterisk-users] issues/jira
anyone from digium around ? https://issues.asterisk.org/jira/ Oops - an error has occurred System Error Cause: java.lang.NoClassDefFoundError: Could not initialize class org.codehaus.xfire.util.STAXUtils -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] issues.asterisk.org
i'm trying to review issues that i'm monitoring and/or have reported at http://issues.asterisk.org when I click on 'My View' or 'View Issues' I get an error: APPLICATION ERROR #401 Database query failed. Error received from database was #1142: DELETE command denied to user 'mantisreadonly'@'localhost' for table 'mantis_tokens_table' for the query: DELETE FROM mantis_tokens_table WHERE '2011-06-06 00:03:56' expiry. Are tickets that I had set up for monitoring on mantis going to be automatically monitored in jira ? similarly, are tickets that I reported in mantis going to show as me being the reporter in jira? or are the tickets going to stay in mantis until they are resolved and never make it into jira ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] issues.asterisk.org
On 6/6/2011 1:08 AM, Jeremy Kister wrote: similarly, are tickets that I reported in mantis going to show as me being the reporter in jira? or are the tickets going to stay in mantis until they are resolved and never make it into jira ? after some more clicking, i see the answer to this one; nevermind. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] iptables for Asterisk - Any good guides out there?
On 5/14/2011 7:51 PM, Bruce B wrote: and then rebuild everything from the beginning with a very limited scope and then without locking myself block all other traffic. Can you suggest what I should put in the shell that would get me this: you may want to start with: http://jeremy.kister.net/code/asterisk/iptables.init modify RTPRANGE and the trusterd array at the top, add in your DID providers to the siprtp array at the top, that should get you near there. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] iptables for Asterisk - Any good guides out there?
On 5/14/2011 9:45 PM, Jeremy Kister wrote: http://jeremy.kister.net/code/asterisk/iptables.init oops, that's: http://jeremy.kister.net/code/iptables/iptables.init -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 1.8 + google voice
somewhere along the way, i noticed incoming calls from google voice are no longer working on my asterisk 1.8.3.2 system. When the call comes in, asterisk immediately prints on the console: == Spawn extension (google-in, s, 2) exited non-zero on 'Gtalk/+12153930924-f947' [May 12 22:47:18] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote peer reported an error, trying to establish the call anyway the calling side just hears ringing. i have plenty of debug info, but nothing too interesting. anyone else having this problem ? or is it time for bug report ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 + google voice
On 5/12/2011 11:08 PM, Jeremy Kister wrote: [May 12 22:47:18] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote peer reported an error, trying to establish the call anyway I found the problem, and I am sending in a bug report :) if anyone is interested, the issue is 19286 (i'll be completing it shortly) -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 1.8.4 Now Available
On 5/10/2011 10:38 AM, Asterisk Development Team wrote: Below is a sample of the issues resolved in this release: [...] For a full list of changes in this release candidate, please see the ChangeLog: I'm a bit confused about this release (and previous releases on the 1.8 track) so please bare with me. I viewed the ChangeLog, but I don't see any of the 'sample issues' listed. why is that ? I would expect to see the 'sample issues' listed after 1.8.4-rc3. Also, is there a reason/procedural error that patches such as: https://issues.asterisk.org/view.php?id=18382 https://issues.asterisk.org/view.php?id=18742 didnt make it into this 1.8.4 release ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] The new ConfBridge application is now in Asterisk Trunk!
On 4/25/2011 9:38 AM, David Vossel wrote: If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, forget everything you know. This is a completely revamped, highly optimized, and feature rich conferencing application capable Can you give a quick lesson on how to use ConfBridge with app_page ? then i could disable meetme dahdi_dummy all together. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 error logging
On 4/17/2011 3:16 AM, Sherwood McGowan wrote: This may sound like a stupid question, but what are your verbosity and debug levels set at currently? nope, thats exactly the type of thing i'm wondering if i'm missing :) but, i tried with verbose 3/debug 0 (which worked in 1.6), and i also tried with verbose 10/debug 10 before posting. no dice. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] sip error logging
bumping once before sending it to the tracker. Original Message Subject: [asterisk-users] sip error logging Date: Fri, 15 Apr 2011 03:39:23 -0400 I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. the last entry I have is: /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: Registration from 'sip:22942@10.0.0.3' failed for '10.0.0.228:5060' - No matching peer found my logger.conf looks like: # grep -v '^;' /etc/asterisk/logger.conf [general] [logfiles] console = notice,warning,error,dtmf messages = notice,warning,error,verbose,dtmf,fax if i send 'options' or 'register' from a non-configured sip peer, i dont see anything in the log. am I missing something ? * i can replicate this behavior on 1.8.2.3 and 1.8.3.2 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] bayardo.sanchez probably doesnt know he is autoresponding to lists
On 4/16/2011 8:20 PM, bayardo.sanc...@gmail.com wrote: I listened to your email using DriveCarefully and will respond as soon as I can. Download DriveCarefully for free at www.drivecarefully.com stop it. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] sip error logging
I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. the last entry I have is: /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: Registration from 'sip:22942@10.0.0.3' failed for '10.0.0.228:5060' - No matching peer found my logger.conf looks like: # grep -v '^;' /etc/asterisk/logger.conf [general] [logfiles] console = notice,warning,error,dtmf messages = notice,warning,error,verbose,dtmf,fax if i send 'options' or 'register' from a non-configured sip peer, i dont see anything in the log. am I missing something ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 error logging
On 4/15/2011 3:39 AM, Jeremy Kister wrote: I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. forgot to mention that I can replicate this behavior on 1.8.2.3 and 1.8.3.2 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] securing sip with iptables [was: asterisk and fail2ban]
On 3/30/2011 4:25 PM, vip killa wrote: could you please elaborate on how you have iptables setup to work that way? I have my config at: http://jeremy.kister.net/code/iptables/ if you already have an iptables config and you just want to make it more secure, the magic happens in the if [ $THROTTLE ] section. if not, just: # make-non-na.pl # vi iptables ## change the MYLAN=10.0.0.0 to whatever you use ## change the RTPRANGE to whatever you have in rtp.conf # mv iptables.init /etc/init.d/iptables # /etc/init.d/iptables start -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] wrong from URI in options message
I recently configured a SIP peer which i must specify my fromuser as my ten digit DID. I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is asterisk. Is this a bug? Or is there some other config I must make ? register = 211941:123456@10.0.138.226/211941~600 [peer](!) type=peer context=inbound qualify=yes qualifyfreq=300 insecure=port,invite nat=yes outgoinglimit=4 incominglimit=4 [mypeer](peer) host=10.0.138.226 defaultuser=211941 fromuser=211941 md5secret=023f30a320a5781e8ffd1af9888012af incominglimit=10 IP (tos 0x0, ttl 64, id 9242, offset 0, flags [none], proto UDP (17), length 555) 10.0.1.3.5060 10.0.138.226.5060: SIP, length: 527 OPTIONS sip:10.0.138.226 SIP/2.0 Via: SIP/2.0/UDP 10.0.83.61:5060;branch=z9hG4bK6abb74e3;rport Max-Forwards: 70 From: asterisk sip:asterisk@10.0.83.61;tag=as7444eb08 To: sip:10.0.138.226 Contact: sip:asterisk@10.0.83.61:5060 Call-ID: 20afd7e40fb31362355eae245dae1fd6@10.0.83.61:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.2.3 Date: Tue, 29 Mar 2011 17:43:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 IP (tos 0xb8, ttl 250, id 0, offset 0, flags [none], proto UDP (17), length 411) 10.0.138.226.5060 10.0.1.3.5060: SIP, length: 383 SIP/2.0 403 From: URI not recognized Via: SIP/2.0/UDP 10.245.83.61:5060;received=10.0.83.61;branch=z9hG4bK6abb74e3;rport=5060 From: asterisk sip:asterisk@10.0.83.61;tag=as7444eb08 To: sip:10.0.138.226;tag=metaswitch+1+0+e288612a Call-ID: 20afd7e40fb31362355eae245dae1fd6@10.0.83.61:5060 CSeq: 102 OPTIONS Server: DC-SIP/2.0 Organization: Content-Length: 0 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] wrong from URI in options message
On 3/29/2011 1:56 PM, Sherwood McGowan wrote: [mypeer](peer) host=10.0.138.226 defaultuser=211941 fromuser=211941 md5secret=023f30a320a5781e8ffd1af9888012af incominglimit=10 IIRC, you need to define the fromuser in the peer in order for the qualify checks (options packets) to contain the user you want uhm, didn't I ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] wrong from URI in options message
On 3/29/2011 2:29 PM, Warren Selby wrote: It looks like you did to me. Is it just OPTIONS packets that are showing the wrong fromuser field? In other words, when you send call traffic over this peer, does it properly create the SIP packets? For some reason, I'm correct - when i actually invite a call or do the register, the from uri is correct. it's just the options packet that is broken. sip development may be able to better tell you. Perhaps open a ticket on the bug tracker? yep, that was the next step - just wanted to run it by a few more eyes before i bothered the devs. https://issues.asterisk.org/view.php?id=19036 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Notify me when the call is answered
On 3/17/2011 8:52 AM, Eric Smith wrote:
How would I achieve a notification this way or another way?
I would use the same premise- below is not tested,
exten = _0031.,n,Dial(SIP/foobar2/${EXTEN},60,wM(notifymobile))
[macro-notifymobile]
exten = s,1,Set(STRIPPED=${CHANNEL:4})
exten = s,n,Set(XTN=${CUT(STRIPPED,-,1)})
exten = s,n,Set(TEXT=Extension $XTN answer call from ${CALLERID(num)})
exten = s,n,System(/opt/swift/bin/swift -o /tmp/$XTN.wav $TEXT)
exten = s,n,Page(SIP/foobar,iqA(/tmp/$XTN.wav))
alternatively, what I actually do:
[macro-AnswerLog]
exten = s,1,Set(STRIPPED=${CHANNEL:4})
exten = s,n,Set(XTN=${CUT(STRIPPED,-,1)})
exten = s,n,Macro(Jabber,x${XTN} answered ${CALLERID(num)})
[macro-Jabber]
; ${ARG1} - message
exten = s,1,Jabbersend(m...@example.com,m...@example.org,${ARG1})
exten = s,n,Jabbersend(m...@example.com,m...@example.net,${ARG1})
--
Jeremy Kister
http://jeremy.kister.net./
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Re: [asterisk-users] Some errors
On 3/15/2011 11:18 AM, Paul Belanger wrote: Theses are leftover issue with the IPv6 conversion for Asterisk 1.8. Collect a complete debug log[1] and open a new issue on the tracker. I believe one was entered a few months ago- https://issues.asterisk.org/view.php?id=18514 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] [announce] jkSMS
For those interested, I have released a first version of jkSMS, which is a simple package that lets cell phones text messages to asterisk. Note it's not real SMS, it makes heavy use of email-to-sms gateways, but it seems to work well. I have had the code running 12 hours, but haven't found any issues. it's not for the faint-of-heart and might require a bit of hacking (really minimal though) if you're not running the same tools that i'm running (like editing the code's DSN if you dont have sqlite installed) http://jeremy.kister.net/code/asterisk/jkSMS/ enjoy, -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] can anyone tell me how to set asterisk to record all phonecall
On 3/4/2011 9:49 PM, John Wu wrote:
I need to use asterisk to record all phonecall I have test using
mixmonitor to record a call.
this is one way it can be done
make sure you have 'lame' installed.
- in your extensions.conf:
[global]
VSA=/var/spool/asterisk
[outbound-or-wherever-you-dial]
exten = _XXX,1,Macro(Snoop,${EXTEN})
exten = _XXX,n,Dial(SIP/${EXTEN},${TIMEOUT})
exten = _XXX,n,StopMixMonitor
; above in case you're in some loop Dial fails,
; e.g., swift+monitor crash asterisk
[macro-Snoop]
; ${ARG1} channel
exten = s,1,GotoIf($[${SNOOPING} = 1]?snooping)
exten = s,n,Set(SNOOPING=1)
exten = s,n,Set(=${STRFTIME(${EPOCH},,%Y)})
exten = s,n,Set(MM=${STRFTIME(${EPOCH},,%m)})
exten = s,n,Set(DD=${STRFTIME(${EPOCH},,%d)})
exten = s,n,Set(HMS=${STRFTIME(${EPOCH},,%H%M%S)})
exten = s,n,Set(FILENAME=${HMS}-${CALLERID(num)}-${ARG1}-${UNIQUEID})
exten = s,n,Set(MIXMON_ARGS=mkdir -p ${VSA}/monitor/${}/${MM}/${DD}
nice -n 19 /usr/local/bin/lame --silent --resample 11.025 -b 16 -t -m
m ${VSA}/monitor/${FILENAME}.wav
${VSA}/monitor/${}/${MM}/${DD}/${FILENAME}.mp3 rm -f
${VSA}/monitor/${FILENAME}.wav)
exten = s,n,MixMonitor(${FILENAME}.wav,,${MIXMON_ARGS})
exten = s,n(snooping),NoOp(snooping on ${CHANNEL})
that'll end up putting a mp3 of the call in
/var/spool/asterisk/monitor//MM/DD/HHMMSS-CALLERID.mp3
don't forget any legal issues you might have to work around, recording
the fact that you declared the message is being recorded.
--
Jeremy Kister
http://jeremy.kister.net./
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Re: [asterisk-users] asterisk behind nat
On 3/2/2011 9:46 AM, Leif Neland wrote: Some of the phones are being disconnected with Asterisk saying no reply to critical packet What kind of phones are they? I might have nothing to do with your network configuration; try adding to sip.conf [general]: session-timers=refuse -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Hide the plain text password
On 2/14/2011 4:36 PM, Jian Gao wrote: Now in my asterisk config files, there are lines like: secret=some_password_in_plain_text Is it possible to hide these plain text password? I think 'md5secret' is what you're looking for. http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] SoftHangup on asterisk 1.8.2.3
I am trying to use SoftHangup in my dialplan, but it's either not
working or I'm not using it correctly.
when i'm on the console, i see:
pbx1*CLI core show channels
Channel Location State Application(Data)
SIP/vgw1-00a2 2156181505@inbound:1 Up AppDial((Outgoing Line))
SIP/143-009f s@macro-SaferSIPDial Up Dial(SIP/99302156181505@vgw1,,
2 active channels
1 active call
194 calls processed
pbx1*CLI
in my dialplan, i have:
exten = s,1,Set(CHAN=${SHELL(asterisk -rx core show channels | awk
'/^SIP\/vgw1-/ { print $1 }' | head -1)})
exten = s,n,SoftHangup(${CHAN})
exten = s,n,Wait(2)
When I dial the extension to invoke the above dialplan code, the console
shows:
-- Executing [s@nineoneone:10] SoftHangup(SIP/111-00a3,
SIP/vgw1-00a2) in new stack
but the SIP/vgw1-00a2 is still active. If I use 'channel request
hangup SIP/vgw1-00a2', the call is dropped instantly.
Am I using SoftHangup incorrectly?
--
Jeremy Kister
http://jeremy.kister.net./
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Re: [asterisk-users] Asterisk 1.8.2.3 Now Available
On 1/26/2011 3:18 PM, Asterisk Development Team wrote: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users mailing list. Patched by mnicholson) I can confirm that this resolves the issue I was having. Thanks to all who were involved, -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] res_fax_digium.so crashing
Since digium is apparently blind to users of their Free Fax for Asterisk, does anyone have advice on how to report a crashing problem with res_fax_digium and Asterisk 1.8.2 ? I have detailed logs/reports and a backtrace ready, but I have no idea who can help. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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_digium.so crashing
On 1/16/2011 4:13 PM, Paul Belanger wrote: I don't believe Digium is blind to its users: Users of Free Fax For Asterisk are not entitled to any Digium technical support [1]. I'm not looking for technical support; I'm just looking for a way to report a bug and possibly help debug/resolve it. But as you know, Digium's website gives FFA users no clear to contact them - even to report problems. issues.asterisk.org has no selection for res_fax_digium since it is not bundled with Asterisk. I call that willful blindness. Don't get me wrong, I'm grateful for FFA and Asterisk in general - I have several running 1.8.2 working correctly. Alternatively, you can generating an unoptimized backtrace [2] and posting the results to the mailing list, seeing if any member of the community has also had an issue. I didnt expect anyone on this list to be interested, but I suppose you're right. This weekend, i set up a new system running Asterisk 1.8.2 on Debian 5.0.7 where the benchmark told me to use res_fax_digium-1.8.0_1.2.1-core2_32 (i also tried generic_32 but that crashed as well). Asterisk correctly detects the fax and transfers to the fax context. But moments after ReceiveFax is called, asterisk crashes, with no tif written where I've directed it to. I have several files including backtraces and config files at http://jeremy.kister.net/tmp/fax/ -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Base memory usage
On 12/31/2010 9:11 AM, Larry Wimble wrote: Removing modules one by one seemed to have virtually no effect until I got to chan_iax2.so. Removing this module dropped memory consumption from 209mb to 16mb (looking at the RES column in the output of `top'). Apparently, it's a known issue: https://issues.asterisk.org/view.php?id=18194 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Base memory usage
On 12/30/2010 9:59 PM, Larry Wimble wrote: I just installed asterisk 1.8.1.1 along with FreePBX on a fairly small VPS (512mb standard, 512mb burst). I note that the asterisk process is using about 209mb of memory just doing nothing (not configured to do anything yet) I'm running 1.8.1 rc1 + some patches (nothing to do with memory) and i'm at 42MB resident (73 virt/8shared) I've got just about everything turned on via menuselect, but then i have a bunch of modules turned off via modules.conf I doubt that's your issue, but if you're interested to see my modules.conf, it's temporarily at http://jeremy.kister.net/tmp/modules.conf -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 - Dial problem if SIP friend is UNREACHABLE.
On 12/20/2010 4:41 AM, Jarek Jarzebowski wrote: Now, when I Dial extension 1050, and there is no 1050 peer registered I got: [Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len 843) to 0.0.4.26:5060 returned -1: Invalid argument You haven't done anything wrong; I have the same issue. Just add it to the list of things to fix in 1.8.. Do you want to add it to http://issues.asterisk.org ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Firewalling and Asterisk
On 11/29/2010 11:03 AM, Jeff LaCoursiere wrote: If I am digesting it correctly, this set of iptables rules does exactly what fail2ban would do, minus the logging, and without the overhead of a scripting language, correct? Very similar to fail2ban, but not quite the same: * this'll block hosts based on X authentication attempts (good OR bad) (fail2ban only counts bad attempts) * this cannot detect encrypted attempts (SIPS), fail2ban can -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Firewalling and Asterisk
On 11/28/2010 12:03 PM, Silver Thorne wrote: So, I am wondering if anyone has a firewall/IP tables statement that keep out unauthorised users? No one seems to get in as we use really http://jeremy.kister.net/code/iptables/ if you already have an iptables configuration, the throttle section is important. if not, the iptables.init script can likely drop in place. if you only need north-american ip addresses to talk to your asterisk box, i suggest you also run the make-non-na.pl from cron every week. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] SIP calls destroyed after 1:20
After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP calls are being destroyed after 1 minute and 20 seconds (80 seconds). Asterisk is sending a BYE message - I have no idea why. http://jeremy.kister.net/tmp/20101115/sip.txt for a sip debug. can anyone suggest how i can further deal with this? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 1.8 fax woes
I upgraded from a perfectly working 1.6.2 asterisk installation (including fax via app_fax_digium) to 1.8.0 this evening. All my custom modules (including swift thanks darren!) are working fine except for fax. When a caller connects, asterisk switches to the fax context and hangs up the call. i've captured with: core set verbose 10 core set debug 10 fax set debug on sip set debug peer vgw1 (vgw1 is my cisco 1760 ata) http://jeremy.kister.net/tmp/fax/console.txt http://jeremy.kister.net/tmp/fax/messages.txt http://jeremy.kister.net/tmp/fax/sip.txt I've tried using the packaged app_fax_spandsp and also Digium's app_fax_digum for 1.8.0-rc1 -- no difference in behavior. Anyone have any ideas how I can get this fixed? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 fax woes
On 11/13/2010 4:36 AM, Jeremy Kister wrote: When a caller connects, asterisk switches to the fax context and hangs up the call. I was wrong, asterisk does not even switch to the fax extension- i added a noop, and it's not making it: exten = fax,1,NoOp( in fax extension ) exten = fax,n,Goto(fax,rx,1) the call ends before the noop. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] useless mpg123 processes hanging around
Running Asterisk 1.6.2.11 on debian 5.0.6 with mpg123 1.4.3
when i start asterisk, i immediately see two mpg123 processes spawned
which sit there forever. I can't imagine it's normal behavior, but if
it is, please explain :)
# /etc/init.d/asterisk stop
stopping asterisk.
#[...]
# /etc/init.d/asterisk start
starting asterisk.
# psg aster
root 14573 1 0 16:29 pts/200:00:00 /bin/sh
/usr/sbin/safe_asterisk
root 14577 14573 0 16:29 pts/200:00:00 /usr/sbin/asterisk -f
-vvvg -c
root 14665 12726 0 16:33 pts/200:00:00 grep aster
# psg mpg123
root 14605 14577 0 16:29 pts/200:00:00 mpg123 -q -s --mono -r
8000 -b 2048 -f 4096 file1.mp3 file2.mp3 file3.mp3
root 14606 14605 0 16:29 pts/200:00:00 mpg123 -q -s --mono -r
8000 -b 2048 -f 4096 file1.mp3 file2.mp3 file3.mp3
root 14609 12726 0 16:29 pts/200:00:00 grep mpg
they look rather worthless:
# strace -p 14605
Process 14605 attached - interrupt to quit
select(0, NULL, NULL, NULL, {0, 152000}) = 0 (Timeout)
select(0, NULL, NULL, NULL, {0, 17}) = 0 (Timeout)
select(0, NULL, NULL, NULL, {0, 17}) = 0 (Timeout)
select(0, NULL, NULL, NULL, {0, 17}) = 0 (Timeout)
select(0, NULL, NULL, NULL, {0, 17}) = 0 (Timeout)
killing them off doesnt seem to affect anything, except when stopping
asterisk:
Beginning asterisk shutdown
Executing last minute cleanups
== Destroying musiconhold processes
[Nov 4 16:29:39] WARNING[14554]: res_musiconhold.c:1508
moh_class_destructor: Unable to send a SIGHUP to MOH process?!!: No such
process
Asterisk cleanly ending (0).
# egrep -v '^$|^\;' musiconhold.conf
[general]
; decrease consumable cpu cycles and memory
; disabled by default
[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3
sort=random
Anyone have ideas if/how I can change this behavior?
--
Jeremy Kister
http://jeremy.kister.net./
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Re: [asterisk-users] useless mpg123 processes hanging around
On 11/4/2010 5:07 PM, Warren Selby wrote: It is because you're using quietmp3 as your mode. Can you explain what the processes are doing? killing them doesn't affect music on hold or any other mp3 playback. strace shows that their behavior doesnt change during a call. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] useless mpg123 processes hanging around
On 11/4/2010 5:30 PM, Warren Selby wrote: I've never really looked that closely at them, sorry. Are they causing some kind of issue on your box, or are you just curious? just curious; i didnt think it was the expected behavior and wanted to fix it. It actually appears that the child mpg123 does do something, and the parent is in a continuous loop making sure the child is alive. So for the archives, i suppose asterisk spawns these off once so that it can be used as a single source for all channels, rather than spawning off one mpg123 for each channel. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] being bombarded with SIP packets
On 10/28/2010 3:41 AM, Per Jessen wrote: 2) if you've got some iptables rules for limiting inbound SIP by rate? exactly what i was going through; here's how i reacted (throttles both SSH and SIP Register: First, I completely blocked all non-North American Amazon EC2 networks - I won't be registering my sip phone in Nigeria nor from within EC2* any time soon. Then in my iptables startup script: iptables -N THROTTLE iptables -A INPUT -i eth0 -p udp --dport 5060 \ -m string --string REGISTER sip: --algo bm --to 65 -j THROTTLE iptables -A INPUT -i eth0 -p tcp --dport 22 \ -m state --state NEW -j THROTTLE iptables -A THROTTLE -m recent --set --name ABUSE iptables -A THROTTLE -m recent --update --seconds 86400 \ --hitcount 15 --name ABUSE -j LOG $LOGOPTS $PREh15_ iptables -A THROTTLE -m recent --rcheck --seconds 86400 \ --hitcount 15 --name ABUSE -j DROP iptables -A THROTTLE -m recent --update --seconds 3600 \ --hitcount 12 --name ABUSE -j LOG $LOGOPTS $PREh12_ iptables -A THROTTLE -m recent --rcheck --seconds 3600 \ --hitcount 12 --name ABUSE -j DROP iptables -A THROTTLE -m recent --update --seconds 60\ --hitcount 6 --name ABUSE -j LOG $LOGOPTS $PREh6_ iptables -A THROTTLE -m recent --rcheck --seconds 60\ --hitcount 6 --name ABUSE -j DROP iptables -A INPUT -i eth0 -p udp --dport 5060 \ --sport 1024:65535 -j ACCEPT iptables -A INPUT -i eth0 -p tcp --dport 22 \ --sport 1024:65535 -j ACCEPT Note that some SIP clients send more than one register per startup -- e.g.: Siphon on the iPhone registers without credentials first, asterisk sends back unauthorized, then Siphone tries again with the configured username and password. For exactly how i'm using it: mkdir /usr/local/script cd /usr/local/script wget http://jeremy.kister.net/code/iptables/make-non-na.pl wget http://jeremy.kister.net/code/iptables/iptables.init mv iptables.init /etc/init.d/iptables # vi iptables # change the MYLAN to your lan network # change the RDPRANGE to the range defined in /etc/asterisk/rdp.conf ln -s /etc/init.d/iptables /etc/rc2.d/iptables ln -s /etc/init.d/iptables /etc/rc3.d/iptables crontab -e # put in something to run the make-non-na.pl run once per week /usr/local/script/make-non-na.pl /etc/init.d/iptables start * = if you use the Acrobits softphone, you'll need to let EC2 through for push notifications. Currently, I just put 184.72.221.84 in the siprtp section of the iptables script. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] evil disconnect of call with cisco 1760
On 9/4/2010 1:31 AM, Jeremy Kister thought: On 8/29/2010 3:25 AM, Jeremy Kister wrote: whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a critical packet being missed. hmm, either no one has any clue/suggestions or they just don't care about the issue - I better figure it out myself. I wonder if it has to do do with progress indicators on the 1760. Thanks Jeremy, that was it! I ended up putting: progress_ind progress enable 8 progress_ind connect enable 8 on each dial-peer pots, and then: progress_ind setup enable 3 progress_ind connect enable 8 on each dial-peer voip. Problem seems solved. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] evil disconnect of call with cisco 1760
I have asterisk 1.6.2.11 talking to a cisco 1760 running ipvoicek9-mz.124-25b. whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a critical packet being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is on the same subnet as the c1700; there are no nat/firewalls/sbcs in the middle. at ~15:15 the asterisk console reads: WARNING[2492]: chan_sip.c:3778 retrans_pkt: Maximum retries exceeded on transmission cb674a02-b25c11df-b6d5a08d-652fe...@10.9.1.9 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. WARNING[2492]: chan_sip.c:3805 retrans_pkt: Hanging up call cb674a02-b25c11df-b6d5a08d-652fe...@10.9.1.9 - no reply to our critical packet (see doc/sip-retransmit.txt). I have a full sip debug at: http://jeremy.kister.net/tmp/sip_debug/ast.txt A running config of the c1760 is at http://jeremy.kister.net/tmp/sip_debug/c1760.txt Important parts of sip.conf are at http://jeremy.kister.net/tmp/sip_debug/most_of_sip.conf I have verified the same behavior with asterisk 1.6.1.12. Ideas? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] app_swift.c:338 engine: Failed to set voice
On 7/28/2010 6:22 PM, Landy Landy wrote: [Jul 28 18:29:16] NOTICE[5191]: app_swift.c:304 engine: Text to Speak : Hello this is ceptral [Jul 28 18:29:16] ERROR[5191]: app_swift.c:338 engine: Failed to set voice. Do you have cepstral installed and have the voice(s) registered ? try: swift --voices assuming swift is installed an a valid voice is registered, what happens when you type: swift Test Message -o /tmp/file.wav is /tmp/file.wav created ? does it play ? what is the output of: grep ^[a-z] /etc/asterisk/swift.conf somewhere should say voice=X. Is that voice installed as per the above swift --voices command ? also, if you're going to be dialing digits with swift, you'll probably run into detection issues unless you use my patch at http://jeremy.kister.net/code/app_swift-1.6.2.patch -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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
