[asterisk-users] terrible MeetMe sound with 1.6.2.9
Hi, Using MeetMee(1234,M) on asterisk 1.6.2.9 (debian/testing) we have terrible sound: the MOH is unrecognizable and speakers can't be understood; it sounds ghostly. However the prompts (your are the only one in this conference, etc.) sound fine. Our server has a Digium T410P card with two E1 lines going in and the wct4xxp dahdi module. Any idea? -- _ -- 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] terrible MeetMe sound with 1.6.2.9
On Tue, Feb 08, 2011 at 10:59:47AM -0600, Danny Nicholas wrote: Any idea? I use mpg123 to play my MOH so I can control the volume (my users complain that standard MOH is a bit loud). Forgot to add that our MOH sounds fine when listened to (on the same extension as MeetMe) with MusicOnHold(default). So it's not a MOH problem as speakers in the MeetMe conference are affected too. -- _ -- 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] terrible MeetMe sound with 1.6.2.9
On Tue, Feb 08, 2011 at 11:09:19AM -0600, Warren Selby wrote: On Tue, Feb 8, 2011 at 11:04 AM, Louis-David Mitterrand vindex+lists-asterisk-us...@apartia.org wrote: Forgot to add that our MOH sounds fine when listened to (on the same extension as MeetMe) with MusicOnHold(default). So it's not a MOH problem as speakers in the MeetMe conference are affected too. Do you have DAHDI installed and running? Yes, all our calls come through a dahdi device. The calls sound fine. Only MeetMe is affected it seems. Show us the output of dahdi_test from the command line. Opened pseudo dahdi interface, measuring accuracy... 99.999% 99.998% 99.995% 99.995% 99.999% 99.992% 99.998% 100.000% 100.000% 99.996% ^C --- Results after 10 passes --- Best: 100.000 -- Worst: 99.992 -- Average: 99.997198, Difference: 99.998508 -- _ -- 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] queue with strategy=linear
Hi, Using asterisk 1.6.2.0 I have a queue definition with strategy=linear. How do I jump to the next dialplan item after having tried (unsuccessfully) all queue members? If I use Queue(test,n) then only the first member is contacted. And if I omit the n option then all members are retried indefinitely. Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] best channel driver for 1.4.x and beronet/junghanns 4BRI?
Hi, What is the best channel driver to use asterisk 1.4.x with a 4BRI isdn card from Beronet or Junghanns (same hardware, different pcid)? Are these cards now supported by plain (non-patched) dahdi/zaptel modules? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] after 1.4.26 upgrade: ast_carefulwrite: write() returned error: Broken pipe
Hi, After upgrading a debian/lenny server to 1.4.26 I get this error: == Manager 'munin' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'munin' logged on from 127.0.0.1 [Jul 26 17:45:12] ERROR[12354]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe repeated each time munin logs in. Should I be concerned? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to match no callerid in 1.6 ?
On Fri, Jul 24, 2009 at 11:14:47AM +0200, Philipp Kempgen wrote:
Louis-David Mitterrand schrieb:
On Fri, Jul 24, 2009 at 10:37:38AM +0200, Michiel van Baak wrote:
On 10:17, Fri 24 Jul 09, Louis-David Mitterrand wrote:
This used to work fine in 1.4:
exten = 2131/,1,NoOp(reject3: ${CALLERID(num)})
exten = 2131/,n,Playback(no_unknow_callerid_here)
exten = 2131/,n,Hangup
And now, after upgrading to 1.6.1.x it matches every callerid.
Why remove the elegant and minimal exten/emtpy
notation
Not that need the exten/callerid syntax for anything but I'd say
this is a bug and a regression.
The syntax is exten[/callerid] so the / clearly says that there
is a second argument even if that happens to be an empty string.
Dear asterisk devs: should I file a bug report? (exten/,prio
matching all callerid's)
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[asterisk-users] how to remove MWI from a Polycom phone
Hi, I'd like to disable MWI on certain lines of my IP650 Polycom phone. So I removed the mailbox= parameter from that line's peer section in sip.conf. Yet the envelope still appears in front of that line and the phone MWI keeps blinking. Where should I look to completely disable MWI on a certain line? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to match no callerid in 1.6 ?
Hi,
This used to work fine in 1.4:
exten = 2131/,1,NoOp(reject3: ${CALLERID(num)})
exten = 2131/,n,Playback(no_unknow_callerid_here)
exten = 2131/,n,Hangup
And now, after upgrading to 1.6.1.x it matches every callerid.
Did something change?
Thanks,
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Re: [asterisk-users] how to match no callerid in 1.6 ?
On Fri, Jul 24, 2009 at 10:37:38AM +0200, Michiel van Baak wrote:
On 10:17, Fri 24 Jul 09, Louis-David Mitterrand wrote:
This used to work fine in 1.4:
exten = 2131/,1,NoOp(reject3: ${CALLERID(num)})
exten = 2131/,n,Playback(no_unknow_callerid_here)
exten = 2131/,n,Hangup
And now, after upgrading to 1.6.1.x it matches every callerid.
Did something change?
Yes, it's now working as it supposed to work.
Use something like this:
exten = 2131,1,GotoIf($[${CALERID(num) = ]?nocallerid,1)
exten = 2131,n,Dia(SIP/Something); or whatever you want to do
exten = nocallerid,1,Playback(no_unknown_callerid_here)
exten = nocallerid,n,Hangup()
Thanks for clearing that up.
But this sucks. Why remove the elegant and minimal exten/emtpy
notation in favor of an unwieldy GotoIf?
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[asterisk-users] agent login status visual clue on Polycom?
Hi, Is there a way on Polycom phones to show an agent whether he is logged in or not? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gap between Playback and Queue
On Wed, Jun 17, 2009 at 01:08:33PM -0500, Danny Nicholas wrote: If this is a recorded sound, you might want to truncate it with lame or audacity. It is quite common in my shop as we record using the phones. Thanks for this suggestion. The problem was indeed a silence at the beginning of my musiconhold tracks. Audacity did a fine job and fixed my problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] gap between Playback and Queue
Hi,
I have a 2/3 second gap between the end of a welcome message played with
Playback and the start of the Queue music. Here is the dialplan:
exten = ${EXTEN},1,NoOp($EXTEN)
exten = ${EXTEN},n,SIPAddHeader(Alert-Info: Ring_CCC)
exten = ${EXTEN},n,Set(CALLERID(name)=${MYCID})
exten = ${EXTEN},n,Answer()
exten = ${EXTEN},n,Wait,1
exten = ${EXTEN},n,Playback(/usr/local/share/asterisk/sounds/welcome)
;; slight gap (silence) here -
exten = ${EXTEN},n,Queue(ccc|t|||${QUEUEWAITTIME})
The welcome sound does end correctly after the last word.
Any idea?
Thanks,
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[asterisk-users] optimising asterisk sounds for g722
Hi, After upgrading to 1.6.x and hdvoice (g722) polycome phones I am wondering how to optimize asterisk sounds and music on hold to take advantage of that codec. I often listen to a special music extension on my headset: /usr/bin/wget -q -O - http://music.example.com | /usr/bin/madplay -Q -z -o raw:- --mono -R 8000 - I tried doubling the frequency to 16000 but this slows down the music. What should I do to get better music quality while retaining backwards compatibility to g711? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hfcpci with 1.6 ?
Hi, Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ? What drivers are available? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hfcpci with 1.6 ?
On Tue, Jun 09, 2009 at 02:45:26PM +0200, Klaus Darilion wrote: Louis-David Mitterrand schrieb: Hi, Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ? What drivers are available? Digium's BRI cards are also based on Cologne Chip - thus you could try Digiums BRI drivers. http://lists.digium.com/pipermail/asterisk-users/2008-April/208806.html You mean the vzaphfc module included in dahdi ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hfcpci with 1.6 ?
On Tue, Jun 09, 2009 at 04:04:29PM +0300, Tzafrir Cohen wrote: On Tue, Jun 09, 2009 at 02:45:26PM +0200, Klaus Darilion wrote: Louis-David Mitterrand schrieb: Hi, Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ? What drivers are available? mISDN 1.1? mISDN2 (chan_lcr)? chan_dahdi? I tried chan_lcr and it works fine. Just one small problem: callerid's arrive without the 'national' prefix (0). How can I fix that? Digium's BRI cards are also based on Cologne Chip - thus you could try Digiums BRI drivers. http://lists.digium.com/pipermail/asterisk-users/2008-April/208806.html The Digium driver is for a slightly different chip: HFC-4S. mISDN (the various versions) include drivers for it. zaphfc should work with Zaptel. zaphfc has been ported to DAHDI and is reported to crash Asterisk successfully (http://bugs.debian.org/532345 ). Does the digium driver also work for Beronet's (or Junghanns) 4BRI and 8BRI cards? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] advice on OrderlyStats (or other cc software)
Hi, Is anyone here using OrderlyStats with asterisk in a call center setting? If so what what is your experience with it? Is that software really free for asterisk users? Or is there a better option out there? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] advice on OrderlyStats (or other cc software)
On Mon, May 04, 2009 at 10:04:53PM +1000, Rob Hillis wrote: Louis-David Mitterrand wrote: Hi, Is anyone here using OrderlyStats with asterisk in a call center setting? If so what what is your experience with it? Is that software really free for asterisk users? Or is there a better option out there? The short answer is OrderlyStats isn't really free for Asterisk. The long answer is that OrderlyStats is free for Asterisk systems with two or less agents. That's really only applicable for the tiniest of call centres. I haven't used OrderlyStats, so I can't speak for the relative merits of it. However, I have used QueueMetrics (which incidentally is /also/ free for call centres of two or less simultaneous agents) and am fairly happy with it. It's not spectacularly pretty - only the latest version has begun to introduce graphs and charts, but it's functional. The price is similar to that of OrderlyStats and the licence you purchase for both of them is time limited - 4 years in the case of QueueMetrics, 5 for OrderlyStats. QueueMetrics will offer a 50% discount for non-profit organisations - I don't know whether OrderlyStats offers the same thing or not. Thank you Rob for the detailed and informative answser. Much appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What do you use? .conf or AEL?
On Tue, Feb 10, 2009 at 01:56:16PM -0800, Mik Cheez wrote: I use them both; my legacy dialplan is all .conf and new stuff is .ael. I find AEL to be the better option when jumping around, but that's just my opinion. But isn't AEL just converted into .conf language anyway? Or has this evolved with 1.4.x ? -- http://www.critikart.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] integration with Microsoft CRM?
Hi, How hard is it to integrate asterisk with Microsoft CRM? Thanks for any suggestions, pointers, etc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] integration with Microsoft CRM?
On Wed, Jan 21, 2009 at 09:02:51AM -0200, David fire wrote: how hard is to integrate whit a virus? sorry ok i read MS CRM but... did you tried VTiger? www.vtiger.com the next release (5.1) will be integrated whit asterisk not only click to dial and popups on incoming calls a queue monitor system too. (Thanks to Wolfgang) I wasn't aware of VTiger. It looks pretty good. Do you know when 5.1 is supposed to be released? What version of asterisk is required for integration with VTiger? Thanks, -- http://www.critikart.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] integration with Microsoft CRM?
On Wed, Jan 21, 2009 at 12:58:51PM -, Andrew Thomas wrote: Try http://forums.vtiger.com/viewtopic.php?t=14314 Thanks, this is a really interesting link. -- http://www.critikart.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] integration with Microsoft CRM?
On Wed, Jan 21, 2009 at 09:00:41AM -0500, Jon Weisman wrote: ok what about people that have no choice but to use MS CRM? That's also my concern, as MS CRM is my customer's choice, not ours, and I may or may not succeed in steering them toward an open-source solution such as vTiger. They already looked at (and dismissed) SugarCRM. I am assuming that MS CRM uses TAPI to interface with a third party PBX. In that cas the TAPI page on voip-info.org gives a few (mostly commercial) solutions. In any cas I'd still welcome any pointers or ideas on Microsoft CRM with asterisk. Thanks, -- http://www.critikart.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE410P alarms stay RED with 1.4.22
Hi, I tried upgrading from debian's 1.4.21.2 package to vanilla 1.4.22 but then my TE410P alarms stay RED and no zap channels can be created, even if they are correctly listed by zap show channels. I tried adding dahdichanname = no to asterisk.conf's [options] to no effect. Going back to 1.4.21.2 brings my alarms back to OK. This is with zaptel 1.4.12.1. -- http://www.critikart.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE410P alarms stay RED with 1.4.22
On Tue, Nov 11, 2008 at 09:49:14AM +0100, Louis-David Mitterrand wrote: Hi, I tried upgrading from debian's 1.4.21.2 package to vanilla 1.4.22 but then my TE410P alarms stay RED and no zap channels can be created, even if they are correctly listed by zap show channels. I tried adding dahdichanname = no to asterisk.conf's [options] to no effect. Going back to 1.4.21.2 brings my alarms back to OK. OK false alarm here: we use an isdnguard device that needs an additional res_watchdog.c file (bristuff patch). Once added it works. Let's hope 1.4.22 will solve our random crashes and system resources hogging... -- http://www.critikart.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] crashes after upgrade from 1.2.16 to 1.4.21.2
On Thu, Nov 06, 2008 at 11:46:48AM +0100, Louis-David Mitterrand wrote: Hi, After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we experience crashes at random intervals with: [Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame read(0, unfinished ... +++ killed by SIGSEGV (core dumped) +++ Process 15755 detached No other crash yet but an asterisk instance eating all of our resources: top - 09:48:32 up 4 days, 12:38, 7 users, load average: 18.06, 16.75, 14.46 Tasks: 142 total, 6 running, 136 sleeping, 0 stopped, 0 zombie Cpu(s): 44.7%us, 50.0%sy, 0.0%ni, 5.1%id, 0.2%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 4057264k total, 3994416k used,62848k free, 279500k buffers Swap:2k total,0k used,2k free, 3044220k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 6537 asterisk -11 0 610m 22m 8760 S 379 0.6 110:33.91 asterisk -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tired of midget packet received warnings
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote: Louis-David Mitterrand wrote: When monitoring an asterisk through its iax2 port I get these warnings at the console: [Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process: midget packet received (1 of 4 min) This is triggered by the monitoring app sending a POKE to the iax port. The warning appears even without any '-v'. Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Could asterisk at least _not_ report this harmless, below-warning event when using a zero-verbose (asterisk -r) level? That would be nice and logical. -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tired of midget packet received warnings
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote: Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Could asterisk at least _not_ report this harmless, below-warning event when using a zero-verbose (asterisk -r) level? That would be nice and logical. I'd take this warning seriously. It means that your monitoring app isn't monitoring what you think it is. Granted, the monitoring app is simple minded: it only checks if a port is open. In that respect is does a hell of a good job: I hear a beeping alarm as soon as an asterisk instance goes south. So what you are saying is that all monitoring apps should speak native iax, else they are bad? Simply checking if a port is open means it's misconfigured or badly written? I wouldn't go so far. Small generic port-monitoring apps should be allowed to check on asterisk without raising such spurious warnings. You know what happens when crying wolf to often, no one listens after a while. A midget packet is not corrupted, I do have a stateful firewall (fiaif) to intercept those. rant AFAIK the onus is on asterisk to adapat: I've suffered too long of the infamous iax2 port-clogging bug that would and render a server 'unreachable' for no good reason. So much so that I went off iax2 entirely and use SIP exclusively for inter-asterisk communication. So much for the muched touted new and advanced pbx communication protocol the iax2 was sold for! This deal-breaker bug went unfixed for years until recently, despite numerous asterisk users reporting iax2 anomalies month after month. A I bitter? yes. Do I trust Digium folks to know their stuff about what is correct or not in networking protocols? I'll let you guess the answer. /rant I always want to know when I get malformed protocol packets in. It is always bad news, mostly either a misconfiguration (your case), an attack, (ie my firewall is not protecting this service) or a sign of a switch port going bad. Fix the cause not the symptom. T. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tired of midget packet received warnings
On Sat, Nov 08, 2008 at 02:33:18PM +1100, Rob Hillis wrote: Tzafrir Cohen wrote: On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote: I'd take this warning seriously. It means that your monitoring app isn't monitoring what you think it is. I always want to know when I get malformed protocol packets in. It is always bad news, mostly either a misconfiguration (your case), an attack, (ie my firewall is not protecting this service) or a sign of a switch port going bad. Fix the cause not the symptom. Maybe it's me, but I think that warning should be regarding a problem I can fix. Malformed network content does not neceserily fall under that definition. notice? Absolutely it does. Warnings of malformed packets are often (as mentioned above) symptomatic of network problems. Fix the network problem, fix the warning. C'mon, even firewalls give you the option of _not_ logging malformed packets! fiaif does. Else your logfile would be the weak point of your system. And what if you can't fix the source of these packets? And what if friendly peers outside of your realm (likely to iax-call you, so can't block them) sends these packets? There are holes in your logic. So asterisk has to be puritan of the lot? Holier than thou? Pro-life with malformed packets? I see where this is going and I don't like it one bit. -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] crashes after upgrade from 1.2.16 to 1.4.21.2
Hi, After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we experience crashes at random intervals with: [Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame read(0, unfinished ... +++ killed by SIGSEGV (core dumped) +++ Process 15755 detached On a second sister-machine with a mirror install we have the same problem. So it doesn't seem to be a hardware problem. This is with a TE410P card. Any idea? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] tired of midget packet received warnings
Hi, When monitoring an asterisk through its iax2 port I get these warnings at the console: [Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process: midget packet received (1 of 4 min) This is triggered by the monitoring app sending a POKE to the iax port. The warning appears even without any '-v'. Is there a way to avoid these warnings? Or at least turn them off when at the console in non-verbose mode? Thanks, -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tired of midget packet received warnings
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote: Louis-David Mitterrand wrote: When monitoring an asterisk through its iax2 port I get these warnings at the console: [Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process: midget packet received (1 of 4 min) This is triggered by the monitoring app sending a POKE to the iax port. The warning appears even without any '-v'. Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Hi, Is POKE a generic udp thing or specific to iax? In the former case I'll probably be able to submit a patch to wmnetmon (great dockable applet I'm using). Thanks, -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mismatched callerid on phone and CDR ?
On Wed, Oct 15, 2008 at 11:30:49AM -0500, Tilghman Lesher wrote: On Wednesday 15 October 2008 10:26:50 Louis-David Mitterrand wrote: For some calls (usally telemarketers) entering through a BRI zap channel I somtimes notice the callerid on my polycom 601 phone and the CDR's 'src' field don't match. They are even totally different. And the displayed callerid is nowhere to be seen in the CDR record. Is there a rational explanation? The ANI and CallerID do not necessarily have to match; they just generally do. The src field reflects the ANI, if set, with a fallback to CallerID number, if not. Thanks for your explanation. Is it then possible to record both informations in the CDR as well? And is there a way to display both fields on my phone's display? -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mismatched callerid on phone and CDR ?
Hi, Using asterisk 1.4.21.2. For some calls (usally telemarketers) entering through a BRI zap channel I somtimes notice the callerid on my polycom 601 phone and the CDR's 'src' field don't match. They are even totally different. And the displayed callerid is nowhere to be seen in the CDR record. Is there a rational explanation? -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] wireless headphone that can answer a call?
Hello and sorry for the OT, Is it possible for a wireless headset of which the base is connected to a Polycom IP601 to remotely answer a call? In the same way as a bluetooth headset. thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] discrepancy between CDR clid and Polycom IP601 clid
Hi, Returning to my office I find two missed calls (from autodialers) that my IP601 displays as originating from 011. However the CDR database recorded the call this way: calldate: 2008-04-04 14:18:16+02 clid: 0172752780 src:0172752780 dst:2131 dcontext: default channel:Zap/1-1 dstchannel: SIP/0146472131-007a7e80 lastapp:VoiceMail lastdata: 2131|su duration: 55 disposition:ANSWERED amaflags: 3 accountcode: uniqueid: asterisk-4208-1207311496.129 userfield: How can the phone display a different clid than the CDR database? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quickfix for building zaptel with 2.6.24?
On Thu, Feb 28, 2008 at 11:10:37AM -0600, Kevin P. Fleming wrote: Louis-David Mitterrand wrote: zenon:~# module-assistant -t build zaptel make[3]: Entering directory `/usr/src/linux-2.6.24.3' scripts/Makefile.build:46: *** CFLAGS was changed in /usr/src/modules/zaptel/Makefile. Fix it to use EXTRA_CFLAGS. Stop. Is there a quickfix out there? Yes, use Zaptel 1.4.9.1 or wait for the release of 1.4.10 later today or first thing tomorrow. If you decide to use 1.4.9.1, please note that if you are using analog cards with FXO modules, there is a known bug in DTMF generation that will affect your ability to dial out on those ports. That has been fixed in Subversion (see issue 11855 on bugs.digium.com) and will be in the next release. Thanks for your answer Kevin, but I need the debian'ized bristuff'ed version to be able to package and deploy it. I'll just patiently wait for Tzafrir (thanks for your work!) to release them for debian. Cheers, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] quickfix for building zaptel with 2.6.24?
Hi, I am trying to build zaptel 1.4.8 with kernel 2.6.24 on debian/sid: zenon:~# module-assistant -t build zaptel make[3]: Entering directory `/usr/src/linux-2.6.24.3' scripts/Makefile.build:46: *** CFLAGS was changed in /usr/src/modules/zaptel/Makefile. Fix it to use EXTRA_CFLAGS. Stop. Is there a quickfix out there? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] flooded by Maximum trunk data space exceeded messages
On Wed, Oct 31, 2007 at 04:53:49PM +0400, Arun Kumar wrote: try to reduce number of calls on trunk or create multiple trunks. The flood happens when I have only one call on the trunk. On 10/31/07, Louis-David Mitterrand [EMAIL PROTECTED] wrote: Hi, Using 1.4.13 and trunking a single iax channel to a similar box my asterisk console is flooded with: [Oct 31 10:49:34] WARNING[5195] chan_iax2.c: Maximum trunk data space exceeded to xx.xx.xx.xx:4569 Known issue? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] flooded by Maximum trunk data space exceeded messages
Hi, Using 1.4.13 and trunking a single iax channel to a similar box my asterisk console is flooded with: [Oct 31 10:49:34] WARNING[5195] chan_iax2.c: Maximum trunk data space exceeded to xx.xx.xx.xx:4569 Known issue? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queues without 302 redirects?
Hi, Using 1.4.13 is it possible to ignore 302 redirects from sip devices belonging to a queue? For a queue that rings the whole office it doesn't seem very useful to obey a redirect programmed on a phone. It seems this was the default behaviour in 1.2. Thanks, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queues without 302 redirects?
On Wed, Oct 31, 2007 at 06:11:47PM +0100, Louis-David Mitterrand wrote: Hi, Using 1.4.13 is it possible to ignore 302 redirects from sip devices belonging to a queue? For a queue that rings the whole office it doesn't seem very useful to obey a redirect programmed on a phone. It seems this was the default behaviour in 1.2. For the record and google the answer is the 'i' option in Queue(). Thanks again to Strom_M on #asterisk! god I love IRC... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Thomson ST2030 firmware upgrade
Hello, I'm trying to upgrade a Thomson ST2030 phone froms its default 1.42 firmware to the latest version (1.56) through tftp. The phone loads the .inf file, then the correct firmware file (as stated in the ST2030S.inf), then it reboots and loops doing these same things again and again. The firmware version on the phone stays at 1.42. Is there a special intermediate firmware version to use before going to the latest? Something special to include in the .inf file? I looked everwhere on the Net (including voip-info). Thanks, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.2.x - 1.4.x upgrade: dialplan block no longer works
Hi,
a block of my extensions.conf no longer works after upgrading from
1.2.17 to 1.4.4. I have:
[macro-dialout]
exten = s,1,Gosub(s-${ARG1},1)
exten = s,n,Congestion
;; default
exten = _s-!,1,Gosub(s-NET,1)
When calling that macro whith no argument ($ARG1 empty):
exten = _0[1-9],1,Macro(dialcapi)
The call is not routed. Apparently _s-! does not match s-:
-- Executing [EMAIL PROTECTED]:1] Macro(SIP/0146472130-0821fe08,
dialcapi) in new stack
-- Executing [EMAIL PROTECTED]:5] Gosub(SIP/0146472130-0821fe08,
s-|1) in new stack
== Auto fallthrough, channel 'SIP/0146472130-0821fe08' status is
'UNKNOWN'
Any idea why?
___
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Re: [asterisk-users] bad case of buzzing
On Wed, Apr 18, 2007 at 01:04:31PM +0200, Tim Koehler wrote: Hi, are you using PoE or power supplies? As power supllies usually are not grounded it could be that it's comming from the power source. We are using PoE You could try using a grounded PoE switch or probably a power backup to test if this is the case. The problem was solved by changing the server and installing a fresh OS image on 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
Re: [asterisk-users] polycom random reboots
On Mon, Apr 16, 2007 at 12:25:55PM +0200, Bas van der Veen wrote: Hello, Did you find anything while testing the LAN? Also, can you confirm that switching the switch, cabling, etc. did NOT solve the problem? It did not. We finally changed the server itself and reinstalled from a known-working installation at another of our sites. We also removed a 4BRI card percieved to be flaky (not needed on this 100% voip site). No more reboots since. I have spontaneous reboots with IP600's. ___ --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] bad case of buzzing
Hello, We are at wit's end on this. One (and only one) of our five asterisk installation is giving us real headaches. Buzzing and/or choppy sound interfere with conversations. I recorded some conversations with monitor() and no problem whatsoever appear in the recording, while the local user was hearing the buzz and half my words. This is a 1.2.16 installation with mISDN but mostly using SIP to our central PRI-equipped asterisk. Phones are Polycom 430, 601, Cisco 7960, 7912 all to the latest firmware. We tried everything: changing the switch, network cards, auditing every network drop with fluke, re-certifying our wan, swapping some phones to no effect. Has anyone gone through that ordeal? ___ --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] polycom random reboots
On Wed, Mar 21, 2007 at 05:53:27AM -0500, Bruce Reeves wrote: Yes, I recently saw this with a 501, in my case the network drop was the problem. If you have a good tester then run it on the connection. I had another drop near by and just swicthed to it. Was that phone using POE ? ___ --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] no incoming dad with mISDN 1.1.1 and asterisk?
Hello, After upgrading my kernel to mISDN-1.1.1 while keeping asterisk-1.2.16 I no longer match any extension. Apparently the dad is empty. However I can see the number just before it (146472130): P[ 4] I IND :SETUP oad:!?145201798p ¡146472130 dad: ¡146472130 pid:2 state:none P[ 4] EXPORT_PID: pid:2 Mar 23 09:35:28 WARNING[6725]: chan_misdn.c:4750 chan_misdn_log: Extension can never match, so disconnecting P[ 4] I SEND:RELEASE oad:!?145201798p ¡146472130 dad: ¡146472130 pid:2 P[ 4] -- bc_state:BCHAN_CLEANED P[ 4] I IND :RELEASE_COMPLETE oad: dad: pid:2 state:EXTCANTMATCH P[ 4] hangup_chan P[ 4] - hangup P[ 4] * IND : HANGUPpid:2 ctx:default dad: ¡146472130 oad:!?145201798p ¡146472130 State:EXTCANTMATCH P[ 4] -- cause:2 P[ 4] -- out_cause:2 P[ 4] -- state:EXTCANTMATCH P[ 4] Channel: mISDN/4-u0 hanguped new state:CLEANING P[ 4] release_chan: bc with l3id: 40001 With mISDN-1.0.4 and the same asterisk it works fine: P[ 4] I IND :SETUP oad:145201798 dad:146472130 pid:2 state:none P[ 4] EXPORT_PID: pid:2 P[ 4] I SEND:PROCEEDING oad:0145201798 dad:0146472130 pid:2 P[ 4] -- bc_state:BCHAN_CLEANED -- Executing Goto(mISDN/4-1, 2130|1) in new stack -- Goto (default,2130,1) -- Executing NoOp(mISDN/4-1, ) in new stack -- Executing Macro(mISDN/4-1, queue) in new stack -- Executing NoOp(mISDN/4-1, 0145201798) in new stack -- Executing Monitor(mISDN/4-1, gsm|20070323-093814-0145201798-2130|mb) in new stack -- Executing Queue(mISDN/4-1, 2130|rntT|||10) in new stack P[ 4] * IND : Indication [3] from s P[ 4] -- * IND : ringing pid:2 P[ 4] I SEND:ALERTING oad:0145201798 dad:0146472130 pid:2 P[ 4] -- bc_state:BCHAN_CLEANED P[ 4] -- * SEND: State Ring pid:2 P[ 4] -- incoming_early_audio off -- Called SIP/0146472130 -- Called SIP/ekiga -- SIP/0146472130-08199d18 is ringing I didn't touch to the mISDN installation other than upgrade the kernel and its modules (compiled on another machine). Should I also upgrade mISDNuser to 1.1.1 on that server? Thanks, ___ --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] polycom random reboots
Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that? ___ --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] polycom random reboots
On Wed, Mar 21, 2007 at 07:07:00AM -0400, joe a. wrote: Did you swap the power module as well? If POE, did you swap the patch cord? If the power module plugs into a power strip did you change that? or at least the position in the strip? Thanks for the tought, but the IP430 has no external power strip or module, it's fully integrated like the IP601. We changed the cable, the wall socket and the switch (was due for an upgrade). Now on to testing the LAN. ___ --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] polycom random reboots
On Wed, Mar 21, 2007 at 04:07:34AM -0700, Henry Cobb wrote: On 3/21/07, Louis-David Mitterrand [EMAIL PROTECTED] wrote: Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that? Our Polycom 3s and 5s ship with flaky power supplies and tend to reboot all of the time (especially in India...), so we found replacement non-Polycom power supplies and they are much more stable. I should have added that we use POE with a 3com PWR-class switch. ___ --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] polycom random reboots
On Wed, Mar 21, 2007 at 05:53:27AM -0500, Bruce Reeves wrote: Yes, I recently saw this with a 501, in my case the network drop was the problem. If you have a good tester then run it on the connection. I had another drop near by and just swicthed to it. What kind of test tool would you suggest? Usually we rely on the cabling guys for that but that entails a delay and I'd be interested in knowing how to do it myself. Thanks, ___ --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] preventing voicemail pickup after SIP redirect ?
Hello, I'm using the classic [stdexten-macro] in extensions.conf whereby a call is picked up by voicemail after a certain ringing time. When programming a SIP phone to redirect calls (SIP 302 redirect) to another extension I'd like to avoid that voicemail pickup so that the call goes into the new destination's voicemail (if applicable). How can I detect that a call has been redirected and should no longer be intercepted by vm? Thanks ___ --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] Re: preventing voicemail pickup after SIP redirect ?
On Tue, Mar 06, 2007 at 07:18:08AM -0600, Eric ManxPower Wieling wrote: How can I detect that a call has been redirected and should no longer be intercepted by vm? That should happen by default. The call should get sent to the new place and it should act like the call was directly dialed to that extension. Actually no. When a call coming in through Zap, Capi or mISDN is redirected by a SIP phone with a 302, then asterisk creates a Local/xx channel to the new destination, while the original channel is still open. So after $RINGTIME is reached, [stdexten-macro] answers the original call and sends it to the original extension's vm. ___ --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] SPA 3000 won't relay DTMF to doorphone
On Fri, Jan 12, 2007 at 02:33:54PM -0500, Doug Crompton wrote: I am using spa3000 hardware - 2.0.1(5673) firmware - 3.1.3(GWa) I have used newer firmwares but find that 3.1.3 had less echo problems. Thanks again Doug for that detailed explanation. As for the DTMF playback level and DTMF playback length settings, what do you use? ___ --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] SPA 3000 won't relay DTMF to doorphone
Hello, Before throwing in the towel with my Sipura 3000 has anyone had much success with that adapter connected to a door phone? In our setup a doorphone is connected to the SPA's fxs port. When a visitor rings, asterisk calls a group of Polycoms and the person who answers has to enter *1 to trigger the door opening. However it seems the SPA doesn't relay the DTMF's to the doorbell. Any suggestions more than welcome, thanks, ___ --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] SPA 3000 won't relay DTMF to doorphone
On Fri, Jan 12, 2007 at 10:58:16AM -0500, Doug Crompton wrote: The spa3000 does not play well with Asterisk with dtmf rfc2833 signaling. Set BOTH the sip.conf AND the spa3000 to inband for DTMF. That would be the line1 tab on spa3000. This applies to the fxo (pstn) also if you are using it for such things as ivr's. Thanks for your suggestion. We tried that without success (using firmware 3.1.7(GWc)) Do you think an upgrade to 3.1.10 might be warranted? ___ --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] better handling of calls forwarded by SIP phones
Hello,
When a user forwards his SIP phone to another extension (say an absent
boss to his secretary) I'd like the unanswsered forwarded call to end up
in the new destination's voicemail. With my current diaplan the call is
handled by the original recipient's voicemail:
[macro-stdexten]
exten = a,1,VoicemailMain(${MACRO_EXTEN})
exten = s,1,Dial(SIP/014647${MACRO_EXTEN}|${RINGTIME}|t|)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(su${MACRO_EXTEN})
exten = s-NOANSWER,n,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(sb${MACRO_EXTEN})
exten = s-BUSY,n,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
Ideally the dialplan would need to detect that the call was forwarded
and not Goto voicemail.
Any idea?
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Re: [asterisk-users] Re: Loosing IAX connection between offices
On Tue, Dec 05, 2006 at 08:02:35AM -0600, Eric ManxPower Wieling wrote: Louis-David Mitterrand wrote: Short story: IAX is still crap in 1.2.13 (haven't tested 1.4), it's unreliable and perfectly good hosts will become UNREACHABLE for no apparent reason, while SIP connections keep going through. Is this with or without the qualify= option in IAX2. With or without 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
[asterisk-users] SetCallingPres propagation
Hello, We have several regional asterisk's connected to a central one making the the PRI calls through a TE410P card. When using SetCallingPres(prohibited) on a call at the regional level, that setting it not forwarded to the central asterisk and the call is made as if no callerid had been sent: the telco substitutes the network number. Using SetCallingPres(prohibited) on the central asterisk works though: the call is received with no callerid at all. How can I suppress callerid presentation at the regional level and keep that setting when trunking the call from regional to central asterisk's? ___ --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] Re: Loosing IAX connection between offices
On Thu, Nov 30, 2006 at 08:52:50AM -0600, DM wrote: Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) Office A is set up with refresh dns and cron job for iax2 reload every 5 minutes. It rarely looses connection to Office B. Short story: IAX is still crap in 1.2.13 (haven't tested 1.4), it's unreliable and perfectly good hosts will become UNREACHABLE for no apparent reason, while SIP connections keep going through. For trunking, avoid IAX and use SIP. ___ --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] chan_misdn on a junghanns card
Hello, I am trying to use chan_misdn on a junghanns QuadBRI card. Using the latest install-misdn-mqueue from beronet, all installation went well apparently. However when I try to load the card it is not recognized: # modprobe hfcmulti type=0x04 protocol=0x12,0x12,0x2,0x2 layermask=0x3,0x3,0xf,0xf Loading only hfcmulti - Loading module(s) for your misdn-cards: - modprobe --ignore-install hfcmulti type=0x4 protocol=0x12,0x12,0x2,0x2 layermask=0x3,0x3,0xf,0xf poll=64 debug=0 Nov 29 11:42:45 pyrrhus kernel: 0 devices registered Trying the same thing on a hfcpci card works and I can receive call with chan_misdn. Is there something specific to junghanns cards? ___ --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] Re: chan_misdn on a junghanns card
On Wed, Nov 29, 2006 at 11:45:50AM +0100, Louis-David Mitterrand wrote: Hello, I am trying to use chan_misdn on a junghanns QuadBRI card. Using the latest install-misdn-mqueue from beronet, all installation went well apparently. However when I try to load the card it is not recognized: This card is a new-style QuadBRI v 2.0 (with hardware watchdog according to KP Junghanns). Apparently this new card is not recognized by mISDN drivers. I tried using an old QuadBRI and the modules load fine. ___ --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] bristuff error: received SETUP message for call that is not a new call
Hello, With the following setup: - asterisk 1.2.13, - zaptel 1.2.10 - bristuff 0.3.0-PRE-1v - quadbri card, after a few hours of normal operation incoming calls suddenly fail to enter with the following message: received SETUP message for call that is not a new call restarting asterisk cures the issue but then it creeps back, making the system unusable. Downgrading asterisk to 1.2.10, zaptel to 1.2.7 and bristuff to 0.3.0-PRE-1q solves the problem durably. ___ --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] Re: Junghanns Bristuff PRI indication
On Mon, Nov 27, 2006 at 09:44:08AM +0200, Kevin Boddy wrote:
I've got a few 8 port Junghanns BRI ISDN cards. Dialling in and out is
working fine but the Telco's busy or invalid number indications are not
being passed through to the user. I have priindication=passthrough in my
zapata.conf but this doesn't seem to help. I'm using Asterisk 1.2.13,
Zaptel 1.2.10 and Bristuff 0.3.0-PRE-1v. This is happening on three
different boxes that I've setup with the same ISDN cards at three
different sites. If the number is busy or invalid the Asterisk box
responds with all-circuits-are-busy indication instead of what the
Telco's is actually sending. This is causing major frustration with the
users as they think the lines are always busy or broken on the Asterisk
box. Any help would be greatly appreciated.
This is what I do:
exten = s,1,Dial(Zap/g1/${MACRO_EXTEN})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s,n,Congestion
exten = s-ANSWER,1,Hangup
exten = s-CONGESTION,1,Playback(invalid)
exten = s-CONGESTION,n,Congestion
exten = s-CANCEL,1,Hangup
exten = s-BUSY,n,Busy
exten = s-CHANUNAVAIL,1,Playback(invalid)
exten = s-CHANUNAVAIL,n,Congestion
___
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[asterisk-users] asterisk sip doesn't see other asterisk-sip
Hello, Here is our setup: asterisk-A --LAN-- nat-router --Internet-- asterisk-B A and B have appropriate friend entries in their sip.conf with a qualify=yes. The router forwards anything on sip,iax and sip/rtp ports to A. The problem: SIP/A remains UNREACHABLE for SIP/B, however A sees B. No problem with iax2. What did I miss in my configuration? Thanks, ___ --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] can't hear MusicOnHold when zap answers
Hello, Using 1.2.13 with bristuff: exten = 8599,1,Answer() exten = 8599,n,Wait(1) exten = 8599,n,MusicOnHold(default) Whan the call comes through a zap (telco) channel I can't hear the music, but through a sip/iax channels I hear it. Any idea why? Thanks, ___ --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] no sound when bridging 2 asterisk SIP connections
Hello, here is our layout: asterisk-A --- WAN --- asterisk-HQ --- WAN --- asterisk-B calls are routed with SIP between asterisk's (found IAX to unreliable). When asterisk-HQ attempts to native-bridge OR simply forward calls between A and B no sound is sent. If either leg (A - HQ or B - HQ) is converted to IAX, then sound flows normally. We are using 1.2.13. What could be the problem? ___ --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] how to indicate an non-existent number?
On Mon, Nov 06, 2006 at 06:47:01PM -0600, Eric ManxPower Wieling wrote: Louis-David Mitterrand wrote: Hello, Using a PRI (E1) with the euroisdn protocol, I don't seem to get any specific message from the telco when attempting to dial a non-existent number. Asterisk returns a busy/congested code, but nothing indicating the number's real status. How do you guys manage that issue? Do you record a message (sorry, the number dialed can't be completed) and play it when the PRI or BRI returns a specific code? And what code is that? We check the value of HANGUPCAUSE. DIALSTATUS is a VERY generic indication of the disposition of the call. It seems PRI and BRI here always return 3 as HANGUPCAUSE From the wiki: #define AST_CAUSE_NO_ROUTE_DESTINATION 3 This is less than explicit regarding an unallocated number (basically I testes by dialling impossibles numbers). ___ --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] HANGUPCAUSE for unalocated number?
Hello, On your BRI or PRI's what do you guys get as HANGUPCAUSE when dialing an unalocated number? I always get 3 (no route) which is less than helpful. ___ --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] how to indicate an non-existent number?
Hello, Using a PRI (E1) with the euroisdn protocol, I don't seem to get any specific message from the telco when attempting to dial a non-existent number. Asterisk returns a busy/congested code, but nothing indicating the number's real status. How do you guys manage that issue? Do you record a message (sorry, the number dialed can't be completed) and play it when the PRI or BRI returns a specific code? And what code is that? Thanks in advance for any insight, ___ --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] polycom's don't register with 2.6.18
Hello, Our Polycom's 601 can no longer register or communicate with the asterisk server when using kernel 2.6.18.x. Cisco 79XX and other phones still work though. Downgrading back to latest 2.6.17.x solves the problem for Polycoms, but I'd really like to understand what's going on there... Any idea? ___ --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] polycom's don't register with 2.6.18
On Fri, Oct 27, 2006 at 12:15:15PM -0400, Doug Lytle wrote: Louis-David Mitterrand wrote: Hello, Our Polycom's 601 can no longer register or communicate with the asterisk server when using kernel 2.6.18.x. Cisco 79XX and other phones still work though. I'm running just 2.6.18 fine Under 1.2 Branch without issue. Connected to Asterisk SVN-branch-1.2-r44580 currently running on livonia (pid = 7349) Are you running Polycom's on this setup? ___ --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] Re: polycom's don't register with 2.6.18
On Fri, Oct 27, 2006 at 05:11:24PM +0200, Louis-David Mitterrand wrote: Our Polycom's 601 can no longer register or communicate with the asterisk server when using kernel 2.6.18.x. Cisco 79XX and other phones still work though. Downgrading back to latest 2.6.17.x solves the problem for Polycoms, but I'd really like to understand what's going on there... After disabling SIP NAT support in 2.6.18 kernel, Polycoms work again. ___ --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] cisco 7960 not registering after * restart
Hello, When I restart asterisk the cisco 7960/7940 phones (sip fw 7.5) fail to re-register themselves with asterisk, even though I put timer_register_expires: 60 in SIPDefault.cnf Is there a way to have these phones register themselves every 60 seconds? Alternatively, can asterisk be made to remember its dynamic sip hosts' registration after a restart? Thanks, ___ --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] cisco 7960 not registering after * restart
On Wed, Oct 11, 2006 at 09:11:43AM -0500, Aaron Daniel wrote: That's a bug with the 7.5 firmware. I would suggest upgrading to the 8.4 version, we've been running it for a few weeks in a test environment and everyone's been pretty satisfied with the new firmware (read: nobody's complained). If the server goes out, they re-register after the timeout without problems. Thanks for your helpful answer, What is the cisco part number for the appropriate smartnet contract required to obtain 79XX firmware? ___ --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] bristuff problem?
Hi Kape, With latest asterisk 1.2.12.1, zaptel 1.2.9.1 and bristuff 0.3.1s after a while calls become stuck: either the caller or callee can't hear the other party, or heavy static is heard. An asterisk restart fixes it for a short while only. This doesn't happen with our older installs (asterisk 1.2.9, zaptel 1.2.7, bristuff 0.3.1q). Are you aware of that problem? Thanks, ___ --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] corrupt faxes
Hello, Since our telco messed with our PRI in some way, we get corrupt faxes like these: http://zenon.apartia.fr/stuff/corrupt_fax.pdf We use the lastest asterisk with a TE410P and spandsp. (for some strange reason, our neighbour company has a traditional pbx fed by 7 BRI's and sees the same problem) Now the telco is trying to racket us with some audit to solve the problem. They are claiming our pbx clockrate might be responsible. What could interefere with faxing in such a way? Could the telco have enabled some echo cancellation on their side? Thanks in advance for any insight, ___ --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] importance of crc4 in zaptel.conf?
Hello, We have a TE410P connected to an EuroISDN E1 with these span definitions: span=1,1,0,ccs,hdb3 span=2,1,0,ccs,hdb3 span=3,1,0,ccs,hdb3 span=4,1,0,ccs,hdb3 Why should we add crc4 to these definitions? What does it do? Thanks, ___ --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] Polycom IP430 sound level too low?
Hello, Has anyone noticed that the Polycom IP430 has a low incoming/outgoing sound level? Is it a firmware issue or should I adjust my zap's tx/rxgain? ___ --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] Polycom 1.6.7 firmware?
Hello, I am looking for the latest 1.6.7 Polycom firmware? Is it available somewhere? Thanks, ___ --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] Re: Polycom 1.6.7 firmware?
On Tue, Aug 08, 2006 at 11:42:01AM -0500, Eric ManxPower Wieling wrote: Louis-David Mitterrand wrote: Hello, I am looking for the latest 1.6.7 Polycom firmware? Is it available somewhere? What issues are you experiencing that 1.6.7 fixes? Flaky buddy watch with 1.6.6. ___ --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] stuck/phantom zap channels
Hello, Using 1.2.9.1 with bristuff and a QuadBRI card, phantom/zombie channels accumulate throughout the day and end up blocking all incoming calls. It's the first time we have this problem and several similar installations work fine. We suspect bad cabling between the telco and the QuadBRI card. Has anyone dealt with this before? Thanks, ___ --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] cheapest Cisco Smartnet contract?
Hello, I've got a few Cisco phones to maintain and need access to firmware files. Dealers here in .fr want unreasonable prices for a Smartnet subscription. Where can I get a better deal on the Net? Thanks, ___ --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] no ring from zap channel
Hello,
I have a TE410P connected to a telco on port1 and legacy Matra pbx on
port2.
When calling an extension managed by the legacy pbx through the telco (with a
normal pots phone), I get ringing. However when calling that same extension
through a SIP phone, no ringing is heard.
Here is the output:
-- Executing Dial(SIP/tmm1-9463, Zap/g2/4530||) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g2/4530
-- Zap/33-1 is proceeding passing it to SIP/tmm1-9463
-- Zap/33-1 is ringing
and the exten:
exten = _45XX,1,Dial(Zap/g2/${EXTEN}||)
g2 being port2 on the TE410P.
I tried adding R to the Dial() options to no avail.
What could the problem be?
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[Asterisk-Users] Re: Linksys SRW224P POE Switch
On Thu, Jun 08, 2006 at 02:04:43PM -0500, Andres wrote: We are currently considering the Linksys POE switch for a small Asterisk office deployment. There will be no separate wiring closet to put it in. Can anybody tell me if this switch has a loud fan? Yes, this switch is loud. It only belongs in a server room, trust me. ___ --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] Re: GXP-2000 (steer clear)
On Tue, Jun 06, 2006 at 11:26:20PM -0400, Daniel Salama wrote: Well, these are encouraging words :) You're basically telling me that I should tell my client to buy other phones. I agree that you cannot compare these phones with Cisco or Polycom. After all, like you said, what do you expect for under $90. However, the fact is that my client just recently invested in these and it will be hard, if not impossible, for me to tell my client to swap them for Polycoms or something else at a much higher cost. I have heard complaints from my client about the speakerphone and they are now, I guess, getting used to picking up the handset :). I have heard any echo problems so far. What bothers me the most is that the phone stops working often (multiple times per day). By this I mean that my client won't be able to dial anything successfully. As soon as 3 or 4 digits are entered, they get a fast busy. To solve it, they need to reboot it. It sounds as if these phones were running Windows instead of Linux :) Anyway, what firmware did you use that solved so many of your problems? I've only had bad experiences with these phones and steer clear of them. In the same price range you can now get the Thomson ST-2030 or Polycom 430 for a much, much better user experience. ___ --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] Re: GXP-2000 (steer clear)
On Wed, Jun 07, 2006 at 08:27:28AM -0400, Daniel Salama wrote: While I would agree with you, the price difference between a GXP-2000 and a Polycom 430 or a Thomson ST-2030. These latter units are, at least, twice as expensive as the GXP-2000. BTW, I never heard of the Thomson ST-2030, but it looks _really_ nice. I get the ST-2030 from a french reseller for ~ 95 EUR/unit. The Polycom IP430 is more in the 140 EUR range however, but it has a real speakerphone and integrated POE (unlike the IP300). ___ --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] ST-2030 reseller (was: Re: GXP-2000 (steer clear))
On Wed, Jun 07, 2006 at 01:55:04PM +0100, asterisk wrote: Any chance of the resellers details ? For the ST-2030 I use this reseller: http://www.hl2d.com Sales contact: Jehan-Philippe Le Roy Responsable des Ventes Partenaires [EMAIL PROTECTED] Tel: +33 1 39 51 60 32 Fax: +33 1 39 51 86 91 49 rue Lamartine 78000 Versailles France, fadge -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Louis-David Mitterrand Sent: 07 June 2006 13:36 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: GXP-2000 (steer clear) On Wed, Jun 07, 2006 at 08:27:28AM -0400, Daniel Salama wrote: While I would agree with you, the price difference between a GXP-2000 and a Polycom 430 or a Thomson ST-2030. These latter units are, at least, twice as expensive as the GXP-2000. BTW, I never heard of the Thomson ST-2030, but it looks _really_ nice. I get the ST-2030 from a french reseller for ~ 95 EUR/unit. The Polycom IP430 is more in the 140 EUR range however, but it has a real speakerphone and integrated POE (unlike the IP300). ___ --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
[Asterisk-Users] very slow network from GXP-2000 switch port
Hello, At a client site yesterday I installed a dozen GrandStream GXP-2000's with 1.1.0.13 firmware but I had to backtrack and reactivate the old PBX and phones: network access for users windoze PC's through the phone's switch port was unbearably slow, making it almost impossible to work. When plugging back PC's directly to the LAN speed was normal again. On my test setup with a single phone here a the office I don't have that problem. Is there a known issue with that firwmare version? Could the switch be playing foul? (a Netgear FSM-7326-PEU) ___ --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] no extension from ISDN phone with bristuff
Hello, I have a Gigaset S44 connected to a quadBRI NT port. Receiving calls works phone, however when dialing out from the phone the call is dropped to the 's' extension, as if no extension had been dialed: -- Accepting voice call from '492389990' to 's' on channel 0/2, span 4 -- Executing Directory(Zap/11-1, default) in new stack -- Playing 'dir-intro' (language 'fr') etc... My zapata.conf contains: [channels] language=fr musiconhold=default switchtype=euroisdn priindication=outofband callerid=asreceived busydetect=no callwaiting=yes callwaitingcallerid=yes pridialplan=unknown nationalprefix=0 internationalprefix=00 callgroup=1 pickupgroup=1 hidecallerid=no usecallerid=yes echocancel=yes context=default ;; for TE ports signalling=bri_cpe_ptmp group=1 channel=1-2 channel=4-5 channel=7-8 ;; for NT ports signalling=bri_net_ptmp echocancel=no pridialplan=local prilocaldialplan=dynamic priindication=passthrough context=international group=2 channel=10-11 And currently using Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1n When the phone is connected directly to the telco ISDN plug, outgoing calls work fine. What did I forget? Thanks, ___ --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] Re: voicemail access on the Thomson ST2030 ?
On Mon, May 22, 2006 at 12:25:34PM +0200, picciuX wrote: for provisioning files to be taken, you have to change the config_sn parameter each time you modify the file, otherwise the phone assumes nothing has changed. Even after a factory reset of the phone? (ie: power-cycle with speaker+mute buttons pressed) Thanks, ___ --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] Re: Office to Office via IAX2 problems
On Mon, May 22, 2006 at 10:11:30AM -0500, [EMAIL PROTECTED] wrote: I'm going to try and lay out all the relevant information I have here in this one post. I can provide more info if necessary. ISSUE 1: Office A routinely looses connection to Office B. When typing IAX2 Show Peers, it will show as Unreachable. I issue IAX2 Reload and it will work again for 1-3 days (haven't narrowed the time down yet). My theory is that the DSL at Office2 is changing IP addresses regularly and this is the cause of the problem??? This has been going on since I set up Office B (2-3 weeks). I never had to touch Office B box. Office B seemed to maintain connection, until now (see Issue 2). ISSUE 2: Office B will not connect to Office A via IAX2 any more. The command IAX2 Show Peers shows Office A as Unreachable. IAX2 Reload won't fix it. I even rebooted the box (MS tricks never die). Up until yesterday, Office B always remained connected to Office A (or at least since I set up Office B - 2-3 weeks ago). Each office has port 4569 forwarded to its * box. I even moved Office A box into DMZ, no help. Note, Office A extensions can call extensions at Office B. IAX2 networking is seriously broken in all asterisk versions. Short story: use SIP for asterisk-asterisk instead. ___ --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] voicemail access on the Thomson ST2030 ?
Hello, After reading all the docs and going through the menus, I still can't find the voicemail access button or menu sequence on the ST2030 (http://www.voip-info.org/wiki/view/Thomson+ST2030) Also I can't get phone provisionning through tftp to work. Configuration files are loaded but the phone seems to ignore them. Any idea? ___ --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] Re: poor state of IAX2 code? (was: why a perfectly fine iax2 host becomes UNREACHABLE?)
On Thu, May 04, 2006 at 12:51:52PM -0700, Tom Engleward wrote: --- Vahan Yerkanian [EMAIL PROTECTED] wrote: Andrew Kohlsmith wrote: On Thursday 04 May 2006 11:31, Louis-David Mitterrand wrote: I've got this low-ping 100%-up dsl connection between two asterisk 1.2.7.1 servers. And oftentimes one of them would declare its opposite UNREACHABLE. Same, here, two asterisk 1.2.7.1 boxes connected to the same switch... Over a week I see at least one case of one of the boxes becoming unavailable for the other... simple iax2 reload fixes the problem. Been like this for ages. rant From this thread today I've learned that the problems I've been having the entire time I've been using asterisk (about two weeks) stem not from NAT, as I originally thought, but from asterisk itself, so that if I were to move my asterisk box to a public IP address, my iax2 connection to my PSTN originator (which also runs asterisk) would _still_ be unreliable. This makes iax2 on asterisk useless for receiving calls. No matter how many spiffy features asterisk has, there is one simple nonnegotiable requirement: it must always answer incoming calls. If it can't do that, then it can't be relied on. And over iax2, it can't do that. Right now I have: CLI iax2 show peers tmm1 192.168.0.1 (S) 255.255.255.255 4569 (T) UNREACHABLE CLI sip show peers tmm1 192.168.0.1 A 5060 OK (40 ms) And it has been so for 15 minutes. Why is IAX2 so flaky? If I did not resort to SIP for inter-asterisk communications I would be out of a job at this time. ___ --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] why a perfectly fine iax2 host becomes UNREACHABLE?
I've got this low-ping 100%-up dsl connection between two asterisk 1.2.7.1 servers. And oftentimes one of them would declare its opposite UNREACHABLE. Why can this happen? The host stanzas in iax.conf have raw IP's, so no DNS monkey business here.. An inquiring mind wants to know. ___ --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] why a perfectly fine iax2 host becomes UNREACHABLE?
On Thu, May 04, 2006 at 10:31:17PM +0500, Vahan Yerkanian wrote: Andrew Kohlsmith wrote: On Thursday 04 May 2006 11:31, Louis-David Mitterrand wrote: I've got this low-ping 100%-up dsl connection between two asterisk 1.2.7.1 servers. And oftentimes one of them would declare its opposite UNREACHABLE. Same, here, two asterisk 1.2.7.1 boxes connected to the same switch... Over a week I see at least one case of one of the boxes becoming unavailable for the other... simple iax2 reload fixes the problem. Is SIP between two asterisk boxes more reliable? Has someone tried 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
[Asterisk-Users] brittle IAX connections ?
Hello, I have several asterisk 1.2.7.1 servers connected through iax2 and often the local asterisk would no longer see the remote one, even thought the link is high quality and the ping is perfect. Is there some issues to take into account about IAX2 connections? Is asterisk's DNS resolution too fragile and should I use raw IP's in my configs? Thanks for any help (I'm in hot water with this issue, client expects _quick_ improvement of call quality) ___ --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
