Re: [asterisk-users] Iptables rule help
well that should map incoming packets to 5091 to 5060, but may not rewrite [new] outbound packets from 5060 to 5091, which your isp may be blocking. an iptables SNAT or MASQUERADE might help you there. i'm not positive on if this would be needed or not. more importantly, however, if your isp is blocking all outgoing traffic to 5060, it won't get to your softphone anyway, unless you also configure that end to also not use 5060. and if you're reconfiguring ports on the softphone end anyway, why not just put 5091 in there, 5091 in sip.conf's bindport, and not mess with iptables at all? another option might be that your isp is blocking rtp as well. can you see what the asterisk console is doing when you attempt such calls? and/or tcpdump? -tcl. On Sat, 16 Dec 2006, Mail list wrote: Hello my isp has blocked outgoing and incoming connection for port 5060 . I have ssh access to server so i want to send all traffic from port 5091 to port 5060 of asterisk .so i tried iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to 127.0.0.1:5060 Now my softphone is able to register with asterisk but it isnt able to make any calls . bindport = 5091 in my sip.conf under extensions is not working .. asterisk doesnt listen to port 5091 .. but if i put in general section of sip.confthen it works but then asterisk wont listen on 5060 . How can i use iptables in this situation ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] taskset with asterisk
has anyone tried out using taskset/isolcpus/cpusets in any way with asterisk on linux? any feedback if so? as far as i can tell, aside from mpg123 and agi calls, asterisk doesn't appear to be threaded or fork in any way, so if i've got a multiproc system whose primary responsibility is asterisk, why not set all system processes (sshd, mysql for cdr, monitoring, etc) to the first cpu, and give asterisk full reign of the second cpu? or am i missing something and asterisk actually will take advantage of MP aside from moh and agis? thoughts? i believe i've seen this mentioned on this list before, but can't recall the thread(s). -tcl. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk -x option in 1.2.9.1
i've encountered similar problems with the last line of an: asterisk -rx "iax2 show channels" with 1.2.7.1, i can typically expect the last line, which is "n active IAX channels" to appear. with 1.2.9.1 it's a complete gamble, typically not displaying (but fine if run from interactive console, of course). this threw off our monitoring of in-use channels. -tcl. On Mon, 3 Jul 2006, Douglas Garstang wrote: Two's enough for me! I'll open a bug. Doug. -Original Message- From: Watkins, Bradley [mailto:[EMAIL PROTECTED] Sent: Mon 7/3/2006 4:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] Asterisk -x option in 1.2.9.1 I have definitely run into this on the one production site I have with 1.2.9.1 I haven't tried backrevving in order to see if it affects 1.2.9 or older, but it is very annoying. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Friday, June 30, 2006 12:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk -x option in 1.2.9.1 This really looks like a bug. It seems as though the '-x' option is broken as of 1.2.9.1 Sometimes the output of the -x command will be only a single line: hestia:(pbx1)~ # asterisk -rx 'database show' //Agents/80014054 : [EMAIL PROTECTED];80014054 and sometimes it will display many or all lines. A buffering issue of some sort maybe? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fun with Echo
yeah this post is old and there have been dozens of replies, but here's some feedback for the list, now that i have some. we're using a sangoma a102 card (no hw ec) with 2 pris from sbc. asterisk 1.2.7.1, zaptel 1.2.6 (much testing previously with 1.2.5). we first used: KB1 (not aggressive), echocancel yes (128), echotraining yes (400), rxgain/txgain 0. most calls with this setup were fine, but maybe 5% of pri calls were reported as having anywhere from slight to completely unbearable echo. playing with echocancel yielded worse results. 256 got results similar to Brian's report below. 64 yielded echo on calls from numbers that never had echo before. we're now pretty settled on similar settings to Brian's: KB1 (WITH aggressive), cancel yes (128), training yes (400), gains 0. AGGRESSIVE_SUPRESSOR made a huge difference for us. reports of bad echo on calls dropped from one or two reports per day to 0 since we started using this setup about 2.5 weeks ago. i still receive the occassional report of slight echo on a call or two (or slight echo at the beginning of a call that goes away), from people who i actively ask about it, but for the most part everyone is very content. i also played around with MG2 (non-aggressive), per suggestions on this thread, but aggressive KB1 showed better results in a few tests. i have not as of yet tried aggressive MG2. looking over our cacti logs, load average on the system didn't noticeably change between any of our echo handling changes. anyway, i know there's no silver bullet for software ec, and anyone reading this looking for ideas should be sure to keep that in mind, but this thread gave us a number of ideas to try, so i wanted to report some of the results. good luck, all! -tcl. On Thu, 8 Jun 2006, Brian Swan wrote: I've spent the last week or so troubleshooting echo problems at my Wife's business, and I've been able to clear up about 99% of the echo, but there is still a little residual echo that I can't seem to "tweak out". The users describe it as "buzzing or crackling", but what it sounds like to me is a slight echo, but just of one syllable of the word. I've followed the numerous suggestions in the mailing list archives which is what has enabled me to get this far. After trying all the echo cancelers, and all the settings on each I settled on: - KB1 (with AGGRESSIVE_SUPRESSOR) - echocancel=128 - echotraining=600 - rxgain=0 - txgain=0 If I turned up the echocancel to 256 I'm able to eliminate the "buzzing" echo, however, on inbound calls I get a HUGE very loud echo that takes a good 20 seconds to go away (which I don't get on echocancel=128). Does anyone have any suggestions on how troubleshoot/tune this to eliminate the "buzz"? Here's my setup: Asterisk 1.2.9.1 Zaptel 1.2.6 TDM400P (3xFXO, 1xFXS) 6 Cisco 7960's running chan_sccp2 Along the same lines, in my hunt for echo, I was rather disappointed that I couldn't really find any tools to help troubleshoot the issue. It seemed like the general consensus was "Play with all the settings until it sounds good". I did manage to compile the Zaptel driver with the Zaptel "preload" patch, which helped me visualize my echo problems a bit, but preloading didn't get rid of the buzz. Also, I've eliminated inside wiring by moving the Asterisk box to my demarc to test, I also tried the TDM400P on different hardware to ensure it's not a motherboard issue. But, at the end of the day, those are all just stabs in the dark... If anyone has any suggestions, I'd sure appreciate it! Thanks! Brian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems compiling Zaptel
Try shutting off asterisk/zaptel and unloading any zaptel modules (rmmod zaptel, wcfxo, etc) before doing the make install, so udev removes any /dev/ entries associated with them (ie /dev/zap/transcode). If not using udev/devfs, then perhaps unload all zaptel modules, rm -fr /dev/zap, then make install. -tcl. On Wed, 28 Jun 2006, Mark Davies wrote: Hi guys, I'm getting the following error when trying to compile zaptel on a debian machine running 2.4.27-3-386. gcc -g -c -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -o zttool.o zttool.c gcc -lnewt zttool.o -o zttool gcc -shared -Wl,-soname,libtonezone.so.1.0 -lm -o libtonezone.so zonedata.lo tonezone.lo make[1]: Leaving directory `/usr/src/zaptel' mkdir -p /dev/zap rm -f /dev/zap/ctl rm -f /dev/zap/channel rm -f /dev/zap/pseudo rm -f /dev/zap/timer rm -f /dev/zap/253 rm -f /dev/zap/252 rm -f /dev/zap/251 rm -f /dev/zap/250 mknod /dev/zap/ctl c 196 0 mknod /dev/zap/transcode c 196 250 mknod: `/dev/zap/transcode': File exists make: *** [devices] Error 1 And when I try to modprobe the X100P card, I get the following. asterisk:/usr/src/zaptel# modprobe wcfxo /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol remove_wait_queue_R5dbd8645 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol __pollwait_R43c77cc3 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol create_proc_entry_Ra52db232 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol proc_mkdir_Rba727c62 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol add_wait_queue_Rf89d8ae0 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol remove_proc_entry_Rf2afedc2 /lib/modules/2.4.27-3-386/misc/zaptel.o: insmod /lib/modules/2.4.27-3-386/misc/zaptel.o failed /lib/modules/2.4.27-3-386/misc/zaptel.o: insmod wcfxo failed Any help is much appreciated. Regards, Mark. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI - Ring requested on channel errors - inbound & outbound stop working.
this problem seems to occur in 1.2.9.1 (1.2.9 also? dunno about 1.2.8)
with users of chan_agent and agents making transfers. Kevin P. Fleming
<[EMAIL PROTECTED]> was looking at the issue last i read on this list.
check out the thread "1.2.9.1 crashed today" on this list over the last
~1.5 weeks.
we had this same problem. rolling back to 1.2.7.1 "fixed" it for us.
-tcl.
On Tue, 27 Jun 2006, Dan Sully wrote:
A few days ago, I started getting these errors on my Asterisk (1.2.9.1)
console:
-- Executing Queue("Zap/1-1", "sales|tT|||3600") in new stack
-- Channel 0/2, span 1 got hangup
-- Channel 0/1, span 1 got hangup request
Jun 27 10:53:27 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested
on channel 0/2 already in use on span 1. Hanging up owner.
Jun 27 10:53:29 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested
on channel 0/2 already in use on span 1. Hanging up owner.
Jun 27 10:53:31 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested
on channel 0/2 already in use on span 1. Hanging up owner.
Jun 27 10:53:33 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested
on channel 0/2 already in use on span 1. Hanging up owner.
Jun 27 10:53:35 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested
on channel 0/2 already in use on span 1. Hanging up owner.
Jun 27 10:53:36 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested
on channel 0/2 already in use on span 1. Hanging up owner.
Jun 27 10:53:38 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested
on channel 0/2 already in use on span 1. Hanging up owner.
-- Channel 0/1, span 1 got hangup
Jun 27 10:53:46 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested
on channel 0/2 already in use on span 1. Hanging up owner.
Jun 27 10:53:49 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested
on channel 0/2 already in use on span 1. Hanging up owner.
At this point, any inbound and outbound calls fail. If I restart asterisk,
things work fine for a few days until it happens again.
Any thoughts on this issue?
Running zaptel 1.2.6 with libpri 1.2.3 and a TE110P, with the wcte11xp driver
on Linux.
Thanks.
-D
--
seriously, first there was the circle, then sliced bread, then tivo
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Re: [Asterisk-Users] 1.2.9.1 crashed today
On Thu, 22 Jun 2006, BJ Weschke wrote: On 6/22/06, Matt <[EMAIL PROTECTED]> wrote: We're now back on 1.2.6 and running stable. Been running for over 17 hours. Something is wrong with 1.2.9.1 Sorry. I may have asked this already, but are you running the tarball releases or checkouts from SVN? I've seen some similar behavior on some of our client systems and I'm trying to get a better read of which date/commit things went wrong. i was using the released tar, NOT svn. -tcl. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.2.9.1 crashed today
we're also seeing a similar problem with 1.2.9.1 (previously 1.2.7.1, without this error). our manager interface is used a fair amount from FOP, and reloads occur staggered throughout the day on changes (but not super often). when the problem occurs, we see a lot of agent and queue functions held up in "show channels", such as AgentCallbackLogin(). the cli is still responsive to commands, but all incoming zap channels (2 pri's worth) become unresponsive except for zap channel 1. our agents also notice strangeness with our queues. outgoing zap calls still work fine (same pris as we use for incoming), and non-zap calls are still good. already-connected zap calls are also good, but new incoming zap calls fail. "soft hangup"s on those agent/queue/related-sip-channels don't respond. reloading some modules work, but a full "reload" errors on either app_queue or chan_agent (i forget which), and further reloads of anything fail after that, due to the queue/agent module never finishing its reload. at this point cli responsiveness to various commands becomes sporadic -- many no longer work, such as "show channels". i didn't think to get a gdb backtrace, blah. the eventual "fix" each time is a full asterisk stop/start. unfortunately, it's happened too often (about 4 times in a week) to continue gathering data with on our production system, so we've rolled back from 1.2.9.1, but kept zaptel, asterisk-addons, and wanpipe [sangoma card] at the latest versions (they were all upgraded simultaneously). if the above info helps any debugging efforts, great. if not, oh well. i'll be seeing what else i can dig up as well (from a test system, logs, etc). -tcl. On Tue, 20 Jun 2006, Matt wrote: Correct.. still running.. just doesn't respond to anything I do, and I have to kill it with the /etc/rc.d/init.d/asterisk stop script On 6/20/06, Steve Totaro <[EMAIL PROTECTED]> wrote: I guess that was a yes to both my scenarios, reloads and manager interface. It is still running just not responsive? Matt wrote: > Arg... ok it just crashed again. Lasted about 7 hours this time. > > On 6/20/06, Matt <[EMAIL PROTECTED]> wrote: > > I use FOP... I believe that that uses manager fairly extensively. > > > > Also... about 2-4 hours prior to the crash I had been playing around > > with getting new MOH working.. .and had reload res_musiconhold several > > times (5 or 6) > > > > > Just curious, do you use the manager interface extensively? > > > >> > ___ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe greeting message.
You may need an: allow=gsm for the appropriate channel config, ie sip.conf, or iax.conf. -tcl. On Mon, 16 Jan 2006, Ken D'Ambrosio wrote: Hi, all. My president wants to have a custom greeting for our bridge. So, I had it recorded (as foo.gsm), modified app_meetme.c to reflect the new filename, compiled, installed... and now get Jan 16 12:53:05 WARNING[14859] file.c: File foo does not exist in any format Jan 16 12:53:05 WARNING[14859] file.c: Unable to open foo (format ulaw): No such file or directory It's in the same directory (/usr/share/asterisk/sounds/) as the other greeting, with the same permissions and ownership. Is there something I'm missing? Thanks, -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
