[asterisk-users] DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657
Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over SIP trunk from which calls get routed to third server (C) (1.6.0.9) via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP clients connect. SIP clients can also dial outside and call goes like C -> B -> A -> PSTN. I have an occasional problem where DTMF is not recognized, ie if clients type a digit while in menu the system does not register it. In my C server I saw a log line like this today: DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657 Is the above message an indication of this problem? How can I fix it? with regards, raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IPKall using iax
Is it possible to receive a call via IPKall through IAX connectivity without registration? If so how to set it up. I've run-into and old link; http://forum.voxilla.com/ipkall-support-forum/ipkall-beta-testing-iax-connectivity-without-registration-26728.html -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E65 fails registration, soft phone works
Martin, sounds like the hiccup my E71 had once. I think the symptoms were identical. Changing the transport type from Auto to UDP solved the problem for me. The Auto setting worked, but only sometimes. Maybe the E65 is similar... Luki 2009/9/12 martin f krafft : > Hey folks, > > I am trying to get an E65 to connect to asterisk, and I would really > appreciate a second set of eyes. The SIP dialog completes fine, but > the phone subsequently says "Registration failed". > > I am in a network that has what seems to be a SIP-capable NAT > gateway, but the asterisk is configured nat=yes anyway. Using > a softphone (twinkle), I can connect just fine, SIP and RTP work. > > But when the E65 tries to connect, it seems to complete the SIP > REGISTER dialog, but then it'll say "Registration failed": ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP and line disconnection detection
Hi Tzafrir, Thanks for the hint, I'll check it to see if Sangoma supports this or not. Regards. -- M. Shokuie Nia. On Thu, Sep 17, 2009 at 10:27 AM, Tzafrir Cohen wrote: > On Thu, Sep 17, 2009 at 09:34:56AM +0330, M Shokuie wrote: > > Dear Folks, > > > > Im looking for a way to detect if an analog line is connected to card or > not > > (Im using Sangoma A200). Im using the dialtone detection when dialing but > > need a way to detect the disconnection of the line when it actually > happens. > > I have no idea about the Sangoma drivers, but reecnt in-tree DAHDI > drivers report this by raising a RED channel alarm if there's nothing > connected. This means that Asterisk won't try dialing through it. > > -- > Tzafrir Cohen > icq#16849755 > jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] "Channels got stuck in asterisk 1.4.18.1"
das sandesh wrote: > Hi All, > > Today I faced a problem with channels getting stuck. We use asterisk > 1.4.18.1, and there were 2 extensions (channels) that got stuck. When I > try to do "soft hangup ", it says "Requested for soft hangup" > for that channel, but if we go and check once again those channels are > still stuck. Also even after asterisk restart it did'nt go, finally we > had to kill the asterisk process and then start asterisk to come back to > normal. > > I wanted to know did any one faced such a problem? Is there any way of > getting to know if the channel gets stuck (since in our senario we came > to know since the person at the extension(channel) that got stuck was > not able to receive calls) or is there a way to eradicate the channel > getting stuck? > > Thank you very much. > > Regards > Sandesh Upgrade to a recent version of Asterisk. 1.4.26.2 is the latest 1.4 release. Not much chance you're going to get help when you're using something as 1.4.18.1. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] "Channels got stuck in asterisk 1.4.18.1"
Hi All, Today I faced a problem with channels getting stuck. We use asterisk 1.4.18.1, and there were 2 extensions (channels) that got stuck. When I try to do "soft hangup ", it says "Requested for soft hangup" for that channel, but if we go and check once again those channels are still stuck. Also even after asterisk restart it did'nt go, finally we had to kill the asterisk process and then start asterisk to come back to normal. I wanted to know did any one faced such a problem? Is there any way of getting to know if the channel gets stuck (since in our senario we came to know since the person at the extension(channel) that got stuck was not able to receive calls) or is there a way to eradicate the channel getting stuck? Thank you very much. Regards Sandesh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem in upgrading to 1.6.1.0
Sorry I wasn't more specific. The error message is just the standard 'Can't find that extension'. The problem is, however, that asterisk parses users.conf (and doesn't complain), but none of the users specified therein are loaded into the dialplan or even shown as peers (using sip show peers/users). A downgrade from 1.6.1.6 to 1.6.0.9 promptly fixed it, as with Oguzhan. Regards, Örn 2009/9/18 Benny Amorsen : > Örn Arnarson writes: > >> I'm seeing the same behavior in 1.6.1.6. >> >> Any info on this? > > It would be helpful if you copied the exact error message involving the > username field. > > > /Benny > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI commands not running !!!!!
On Tue, 8 Sep 2009, Mindaugas Kezys wrote: We have a watchdog which sends SIP OPTIONS packet to Asterisk and if it does not respond ? restarts it. Is this something you can share? Is it a "wrapper" around another tool like sipsak or sipp? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] digium fax: is this even close to working?
My set up is 1.6.0.15 with the digium fax modules. I want to capture a
fax from the internal analog fax machine (using an SPA2102), and then
resend it. I know the internal extension of the fax machine, and for now
I'm just testing it to one outside fax machine if I dial 8447.
In particular, I'm completely unfamiliar with the use of "G" in the Dial
app.
exten => 8447,1,Answer()
exten => 8447,n,GoSub(Capture-Fax,s,1(1xxxyyy))
[Capture-Fax]
exten =>
s,1,Set(FAXFILE=${FAX_RX_FOLDER}/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)})
exten => s,n,ReceiveFAX(${FAXFILE}.tif) ;; 1.6 use ReceiveFAX
exten => s,n,Hangup()
exten=>h,1,Dial(${TRUNK}/${ARG1},,G(send^1)) ;ARG1: outside fax number
exten=>send,1,SendFAX(${FaxFile}.tif)
exten=>send,n,Hangup()
exten=>send,n,Return()
Thanks for any reviewing eyes.
sean
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Re: [asterisk-users] Asterisk Web Meetme module not loading
Matt wrote: > On 1/09/09 5:53 PM, Glen wrote: >> Matt Riddell wrote: >>> In the latest readme for WebMeetMe (3.1.0) it states: >>> >>> * Compile and install CBMySQL >>> App_cbmysql is now included in the web-meetme package, >>> located in ./cbmysql. To install just run make; make install >>> >>> Copy the sample cbmysql.conf to /etc/asterisk and create >>> a dialplan similar to the one in cb-extensions.conf.sample >>> Modify the settings to suit your system. The location of the >>> mysql.sock file is likely not correct, check /etc/my.conf for >>> the correct location. >>> >>> >> That fixed it Matt, just compiling in the wrong directory. >> >> Thanks for all your help. > No problems :) I haven't actually used it myself, but it looks pretty cool! Matt- Thanks for jumping in. I have been offline for close to four weeks recovering from oral surgery. Months go by without a single Web-MeetMe question, and as soon as I stop watching email a bunch show up... As was discovered, the app is compiled separately from Asterisk. Due to changes in the AMI interface over the years, there will soon be three versions of WMM- 2.X for Asterisk 1.2 (largely unmaintained, but problem reports are rare) 3.x for Asterisk 1.4 4.X for Asterisk 1.6 (recommend 1.6.0.7 or 1.6.1 or newer) Starting in 1.6 the scheduling logic that was in app_cbmysql is now native to app_meetme when using RealTime, so app_cbmysql has not been updated for 1.6. I need to get 4.X released. I normally like to run a new release in house for a couple of months before release, but between the surgery and changes at work it is not likely to happen. The good news is that the 4.X release has less to test since the scheduling logic moved into app_meetme, so I just need to confirm that nothing is seriously broken in the UI. Thanks, Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call-limit on dahdi channel
On Thu, Sep 17, 2009 at 12:02:16PM +0300, Tzafrir Cohen wrote: > On Thu, Sep 17, 2009 at 08:18:13AM +1000, Alex Samad wrote: > > Hi > > > > how do i set the call-limit on a dahi line - its connected to the pstn > > network - shared fax line. How do i tell asterisk not to send more than > > 1 call there ! > > Asterisk will not send out more than one call on that line. > > You want to avoid calling if someone else is calling on a fax machine > connected to the same line? yes > signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem in upgrading to 1.6.1.0
Örn Arnarson writes: > I'm seeing the same behavior in 1.6.1.6. > > Any info on this? It would be helpful if you copied the exact error message involving the username field. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 order
Pardon my ignorance, but couldn't you set up this provider as a "Static peer" so he would always be first in the hash? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, September 18, 2009 2:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX2 order David A. Bandel wrote: > I've been fighting with this seems like forever now and can't make it > work in 1.6.x. In 1.4.x, I could make sure a particular voip provider > was always first in the list by making him as an #include and putting > it last. > > Now in 1.6.x I can never get this to take. I really don't want to run > TWO asterisk servers just for some IAX trunking. > > Real problem: my voip provider doesn't send username authentication, > just their secret. If they are either the first or the only, then all > works because * checks just the first iax2 entry in the "show iax2 > peers" list. If they are anything but first, all my incoming calls > from them fail (identification failure). > > Is there ANY way to make them first? In 1.6.x they are always 3rd. Unfortunately the answer is no; in Asterisk 1.6.x chan_iax2 stores users/peers in a hash table, which as you have determined means they are not stored in any particularly obvious order... and even if you did determine the order, it could change in the future if the hash table definition is changed. Your provider really needs to use a proper authentication mechanism, or you need to restrict that peer/user entry to their IP address(es) if possible. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dCAP Exam
Jared Smith writes: > In a nutshell, you can pass the test without having any experience on > Polycom IP phones and Digium cards, as long as you know how to use > Asterisk itself. You certainly can, but I think it's worth it to invest ~30 minutes beforehand so you know where you put IP addresses and accounts in Polycom phones, and so you can get basic DAHDI working. It isn't hard, it takes about 30 minutes to learn, and it doesn't even really require that you have the hardware in front of you. I don't think it's unreasonable at all that it is in the test -- if you can't connect SOME kind of phone to Asterisk, you don't deserve certification. They have to pick one brand because it's infeasible to bring 5 different phones for each test taker. So, to all you people who complain that the dCAP is too hardware specific: It isn't. Really, the only tricky thing to know is that you probably want the Address and the Auth User ID fields on a Polycom phone to contain the same value (often the phone extension, if you don't want to be fancy). The Address field should NOT contain an IP address. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 order
David A. Bandel wrote: > I've been fighting with this seems like forever now and can't make it > work in 1.6.x. In 1.4.x, I could make sure a particular voip provider > was always first in the list by making him as an #include and putting > it last. > > Now in 1.6.x I can never get this to take. I really don't want to run > TWO asterisk servers just for some IAX trunking. > > Real problem: my voip provider doesn't send username authentication, > just their secret. If they are either the first or the only, then all > works because * checks just the first iax2 entry in the "show iax2 > peers" list. If they are anything but first, all my incoming calls > from them fail (identification failure). > > Is there ANY way to make them first? In 1.6.x they are always 3rd. Unfortunately the answer is no; in Asterisk 1.6.x chan_iax2 stores users/peers in a hash table, which as you have determined means they are not stored in any particularly obvious order... and even if you did determine the order, it could change in the future if the hash table definition is changed. Your provider really needs to use a proper authentication mechanism, or you need to restrict that peer/user entry to their IP address(es) if possible. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI Caller ID problem
- "Danny Nicholas" wrote:
> Cidstart=polarity or cidstart=ring will probably fix this.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> h...@cfht.hawaii.edu
> Sent: Thursday, September 17, 2009 8:04 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] DAHDI Caller ID problem
>
> Aloha,
>
> I'm finishing up the final touches on this install, and have run into
> an
> odd problem.
>
> I can't seem to get Caller ID on the analog phone lines working. It's
> a
> Digium AEX 410 card.
>
> I have Verbose set and a line to print the CID:
>
> I have usecallerid=yes and callerid=asreceived set in both
> chan_dahdi.conf
> and users.conf
>
Assuming that you have standard US CID i.e. a bell fsk spill between the first
and second rings, then you will need to set the following in chan_dahdi.conf :
usecallerid=yes
cidstart=ring
cidsignalling=bell
callerid = asreceived (For incoming trunks)
> [analog]
> include=>default
> exten => s,1,Verbose(passed id is ${CALLERID(num)})
> exten => s,2,Answer
> exten => s,3,Dial(SIP/100,,)
>
> And this is what I'm getting.
>
> *CLI> core set verbose 10
> Verbosity was 1 and is now 10
> -- Starting simple switch on 'DAHDI/1-1'
> [Sep 17 14:44:05] NOTICE[15308]: chan_dahdi.c:7542 ss_thread: Got
> event 18
> (Ring Begin)...
> [Sep 17 14:44:07] NOTICE[15308]: chan_dahdi.c:7542 ss_thread: Got
> event 2
> (Ring/Answered)...
> [Sep 17 14:44:07] NOTICE[15308]: chan_dahdi.c:7706 ss_thread: MWI:
> Channel
> 1 message waiting!
The fact that you get "MWI: Channel 1 message waiting!" indicates that a fsk
spill was processed and contained a message waiting indicator packet.
Unfortunately, it does not indicate that a standard CID packet was included as
well.
> -- Executing [...@analog:1] Verbose("DAHDI/1-1", "passed id is ") in
> new
> stack
> passed id is
> -- Executing [...@analog:2] Answer("DAHDI/1-1", "") in new stack
> -- Executing [...@analog:3] Dial("DAHDI/1-1", "SIP/100,,") in new
> stack
> == Using SIP RTP CoS mark 5
> -- Called 100
> -- SIP/100-b6a22338 is ringing
> -- SIP/100-b6a22338 answered DAHDI/1-1
> == Spawn extension (analog, s, 3) exited non-zero on 'DAHDI/1-1'
> -- Hungup 'DAHDI/1-1'
>
> I'm also getting these errors:
> [Sep 17 14:01:06] ERROR[14462]: callerid.c:562 callerid_feed: No start
> bit
> found in fsk data.
> [Sep 17 14:01:06] WARNING[14462]: chan_dahdi.c:7582 ss_thread:
> CallerID
> feed failed: Success
> [Sep 17 14:01:06] WARNING[14462]: chan_dahdi.c:7686 ss_thread:
> CallerID
> returned with error on channel 'DAHDI/1-1'
>
The error you are seeing ("No start bit found in fsk data") indicates that the
fsk processing code cannot lock onto the fsk spill. You may want to adjust the
gain applied to the incoming signal while it processes cid. This can be
adjusted by setting:
cid_rxgain=x.x
This value is in dB and defaults to +5 dB if it is not specified. (You may want
to test both higher and lower values.)
Regards,
Doug Bailey
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Re: [asterisk-users] dCAP Exam
On Thu, 2009-09-17 at 15:12 -0400, jon pounder wrote: > Not that I would ever consider taking an exam like that, but I have been > using/configuring asterisk since nearly the beginning of this mailing > list, and I have never touched dahdi or polycom. Someone should still be > able to pass an exam without knowing about specific hardware where there > is more than one alternative to use in real configurations. Let me try to clarify things a bit here... The dCAP test is primary a test of Asterisk skills, not your familiarity with the configuration of a particular brand of phone or with the Digium line of hardware cards. Part of the test does require you to get an IP phone registered and talking to Asterisk, but the instructor should be more than happy to walk you through the web interface of the phone and say "Put the SIP username here" and "Put the SIP password here" and "Put the IP address of your Asterisk server here". If you'd rather use a softphone on Linux, you're free to use that instead of or in addition to the IP phone. Another part of the test asks you to get an analog phone connected to Asterisk. If you don't get this part working, it doesn't have a huge effect on your score. (Less than 5% of the total score comes from configuring the analog phone correctly.) In addition, you are also asked to connect Asterisk to a (emulated) telco. We do give you the choice, however, of using *either* PSTN or VoIP connectivity to do so. In a nutshell, you can pass the test without having any experience on Polycom IP phones and Digium cards, as long as you know how to use Asterisk itself. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Records for MeetMe
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony
Sent: Friday, September 18, 2009 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR Records for MeetMe
Andy Rosen wrote:
> ... figure out a good way to log which conference ID that is being
used.
The only way I have found to do this is in the events, the conference
enter event has the unique id of the call, which will tie it to the cdr,
and the conference number.
Hope this helps!
Anthony
---
This may not be exactly what you're looking for but it's one possible
solution. I use a wrapper around MeetMe to emulate what my users were
used to doing. I playback an intro file which prompts the user for a pin
number then use it to query a SQL db for the conference id. I then
record the entered pin in the CDR userfield. You could easily modify it
to prompt for the conference ID then store it instead. Here's an example
from my dialplan.
Read(PIN,globeop/bridge_greeting-nancy,7,,3,5);
AGI(go-meetme.agi|${PIN});
Set(CDR(UserField)=${PIN});
MeetMe(${CONFROOM}|${OPTIONS}|${PIN});
${OPTIONS} and ${CONFROOM} are populated in my AGI script.
Bob
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[asterisk-users] No more room in scheduler
Hi, I running into the following problem on my Asterisk setup: --snip-- [Sep 3 01:40:59] NOTICE[9170] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 3 [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:09] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:09] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:09] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:09] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? --snip-- This happens once a week, at same about the same time (give or take a couple minutes). Always from "span 3" too. It just continually spits out those messages until I restart Asterisk. I've seen others post about this, but haven't seen a real answer. Someone said to run a 'dahdi_test -v' when this happens; I did and I get 99% every time. Someone else said this is usually caused by the telco. running some type of test on the line, and I would agree since it happens every week at pretty much the same time and same day. So, yes, lets say the telco. is sending some type of signal that freaks out Asterisk/DAHDI. I could call them and ask them to stop, but it would seem more appropriate for Asterisk/DAHDI to just "handle" this and not cry. A short term fix would be to just have a cron run around 2:00 a.m. weekly that will restart Asterisk. Should I open a bug for this? asterisk-1.6.1.1 dahdi-linux-2.2.0.2 dahdi-tools-2.2.0 Linux jekyll.mcc.edu 2.6.18-128.1.1.el5 #1 SMP Mon Jan 26 13:58:24 EST 2009 x86_64 x86_64 x86_64 GNU/Linux Red Hat Enterprise Linux Server release 5.3 (Tikanga) Dell PowerEdge 2950 (2) Wildcard TE220 (4th Gen) [r...@jekyll ~]# cat /etc/dahdi/system.conf # 20090801 MAS # Span 1 span=1,1,0,esf,b8zs bchan=1-23 dchan=24 echocanceller=mg2,1-23 # Span 2 span=2,2,0,esf,b8zs bchan=25-47 dchan=48 echocanceller=mg2,25-47 # Span 3 span=3,3,0,esf,b8zs bchan=49-71 dchan=72 echocanceller=mg2,49-71 # Span 4 span=4,4,0,esf,b8zs bchan=73-95 dchan=96 echocanceller=mg2,73-95 # Global loadzone= us defaultzone = us [r...@jekyll ~]# cat /etc/asterisk/chan_dahdi.conf [general] [channels] ; Span 1 group = 1 context = from_pstn switchtype = qsig signalling = pri_net channel => 1-23 context = default ; Span 2 group = 2 context = from_avaya switchtype = qsig signalling = pri_net channel => 25-47 context = default ; Span 3 group = 7 context = from_pstn switchtype = qsig signalling = pri_cpe channel => 49-71 context = default ; Span 4 group = 7 context = from_pstn switchtype = qsig signalling = pri_cpe channel => 73-95 context = default [r...@jekyll ~]# cat /etc/dahdi/modules # 20090801 MAS wct4xxp wctc4xxp Let me know if any more information is needed. Any help is greatly appreciated! Thanks, Marc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 order
Folks, I've been fighting with this seems like forever now and can't make it work in 1.6.x. In 1.4.x, I could make sure a particular voip provider was always first in the list by making him as an #include and putting it last. Now in 1.6.x I can never get this to take. I really don't want to run TWO asterisk servers just for some IAX trunking. Real problem: my voip provider doesn't send username authentication, just their secret. If they are either the first or the only, then all works because * checks just the first iax2 entry in the "show iax2 peers" list. If they are anything but first, all my incoming calls from them fail (identification failure). Is there ANY way to make them first? In 1.6.x they are always 3rd. TIA, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto Visit my blog at: http://www.pananix.com/cgi-bin/blosxom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi + SIP Realtime
Good afternoon gentlemen (and ladies).
A costumer of mine has many servers and each one maps their SIP extensions to
the others via DUNDi. It works like a charm. SIP extensions can only register
at one server, the one they "belong" to. In case one extension wants to call
other that is registered in another server, DUNDi takes care of that by calling
the other server using IAX2 and G.729 codec. It's important to clarify that
each server today works as a completely independent PBX, talking to each other
using IAX2 that is routed via a MPLS network. Also, the servers are very
distant from each other physically.
Today the extensions are being mapped in DUNDi like this:
[dundi-internal]
exten => _70XX,1,Noop()
And all extensions configurations are done in sip.conf. No realtime is being
used, yet.
Now the customer wants to take a step further and make it possible that *every*
SIP extension could be able to register in *every* server. That would make
possible for them to use DNS to automatically "find" the closest PBX and make
the extension register on that one.
So far I considered the following for this project:
- Moving all SIP extensions from individual sip.confs to one MySQL database,
and point all servers to that one
- Configure sip.conf on each machine like this:
regcontext=dundi-internal
rtcachefriends=yes
rtsavesysname=yes
rtupdate=no
rtautoclear=yes
ignoreregexpire=no
That way each time an extension registers, Asterisk would add an extension to
the dundi-internal context, which as you guessed, is the one being mapped to
the other servers. So instead of mapping extensions using wildcards, the
extensions will be mapped individually.
extensions.conf would be something like this:
[internal]
;Tries to make the call using SIP, in the case
;the extension is registered in this server
;If it's not, switches to DUNDi
exten => _,1,Dial(SIP/${EXTEN},60)
exten => _,n,NoOp(DIALSTATUS = ${DIALSTATUS})
exten => _,n,NoOp(FROM_DUNDI = ${FROM_DUNDI})
exten => _,n,GotoIf($["${FROM_DUNDI}" = "1"]?end:start)
exten => _,n(start),Answer()
exten => _,n,Playback(vm-dialout)
exten => _,n,Goto(dundi-internal-helper,${EXTEN},1)
exten => _,n(end),Noop(Loop detected. Hanging up.)
exten => _,n,Hangup()
[dundi-internal-helper]
switch => DUNDi/dundi_internal
[from-dundi]
exten => _,1,Set(FROM_DUNDI=1)
exten => _,n,Dial(SIP/${EXTEN},60)
So far it's working fine in a test lab with 2 servers running Asterisk 1.6.0.15.
For the gurus out there: is there something that I'm doing terribly wrong, that
would break everything and make the universe collapse into itself when I apply
the same principle on production?
I'll be happy to provide more details in case there are any doubts. I really
appreciate your feedback, no matter what is it. :)
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
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Re: [asterisk-users] console color
On Fri, 18 Sep 2009, Damon Estep wrote: > after the execution of '#service asterisk restart' we no longer have > color in the console I consider that a "good thing." Color confuses me -- I'm a binary kind of guy :) Seriously though, all those escape sequences are hell if you capture the console output using something like "script." > additionally, when executing the restart of the service we get a message > that asterisk exited on signal 9, but I have not been able to find a > definition for signal 9. Try "kill -l" and "man kill" Signal 9 (SIGKILL) is used to kill a process that doesn't take a hint -- signal 15 (SIGTERM). -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DeadAgi
>> On Thu, 17 Sep 2009, Anahi Ludue?a wrote: >> >>> Thanks for the answers! The file didn't have the first line! >>> #!/usr/bin/php > Steve Edwards wrote: >> Glad you found the answer. However... >> >>> The command ls -l returns: >>> >>> -rwxrwxrwx 1 root root 140 Sep 17 15:42 finconf.php >> >> Having an executable with 777 permissions is a very bad idea. Think >> about somebody (or some program) executing something like: >> >> echo "rm -f -r /whatever-they-want" \ >>> /var/lib/asterisk/agi-bin/finconf.php On Fri, 18 Sep 2009, Ishfaq Malik wrote: > Agreeing with the above here, really you want the script owned by > asterisk.asterisk and permissions of 0755 If the file has the permissions set to 755, the ownership is irrelevant to the execution. 755 (-rwxr-xr-x) means: 1) 7 (rwx) -- read, write, and execute by the user that owns the file, 2) 5 (r-x) -- read and execute by members of the group that owns the file, 3) 5 (r-x) -- read and execute by users not in the group that owns the file. Thus, anybody can execute the AGI. You could have the permission set to 500 (-r-x--) meaning only the owner can read (which is a prerequisite to execute) and execute the file, assuming that the user executing Asterisk is the owner of the file. You could even have the permission set to 1 (-x) if you execute Asterisk as root. Personally, since I am usually the "lone developer and admin," I set the ownership of /var/lib/asterisk/ and below to my username so I don't have to use sudo every time I update an AGI or a sound file. I set the permissions of the AGIs to 755 so the user running Asterisk can execute them as well. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem in upgrading to 1.6.1.0
I'm seeing the same behavior in 1.6.1.6. Any info on this? On Wed, Apr 29, 2009 at 12:49 PM, Oguzhan Kayhan wrote: > Hello, > I just tried to upgrade to 1.6.1.0 from 1.6.0.9 and i had problems in > registering users. > As i see from debug it successfully reads from users.conf but later,when a > user tries to logon it say peer not found > And there were an error msg about mysql about the username field..Smthing > changed in mysql tables??? > > Now i downgraded to 1.6.0.9 again and everything is working.. > > > > ___ > -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio Files
On Fri, 18 Sep 2009, Anahi Ludue?a wrote: What can I use to transfer the audio files to and from Asterisk? I was searching and I found the following commands: PUT SOUNDFILE and GET SOUNDFILE They are new commands of AGI, but is there another way to do that? From a shell command line, I use rsync, scp, mv, or cp. What are you trying to accomplish? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Records for MeetMe
Andy Rosen wrote: > ... figure out a good way to log which conference ID that is being used. The only way I have found to do this is in the events, the conference enter event has the unique id of the call, which will tie it to the cdr, and the conference number. Hope this helps! Anthony ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] console color
thanks! I will research that.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Friday, September 18, 2009 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] console color
On Friday 18 September 2009 09:28:24 Damon Estep wrote:
> about once a month we issue a "service asterisk restart" via a cron
job,
> and this is where we lose the color.
Most likely, your TERM environmental variable is not set when the cron
job
runs. This environmental variable should be set to the name of a
terminal
which supports color ("linux" is a good choice, except on non-Linux
systems,
where "xterm-color" might be better).
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] console color
On Friday 18 September 2009 09:28:24 Damon Estep wrote:
> about once a month we issue a "service asterisk restart" via a cron job,
> and this is where we lose the color.
Most likely, your TERM environmental variable is not set when the cron job
runs. This environmental variable should be set to the name of a terminal
which supports color ("linux" is a good choice, except on non-Linux systems,
where "xterm-color" might be better).
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org
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[asterisk-users] Queue Call Disconnection
There is an environment Setup uses Asterisk 1.2 and doesn't want to upgrade.
There is an issue while a call goes to any queue we create, the call is being
disconnected after 20 seconds and it is hangup.
The following is the configuration:
- vi /etc/asterisk/queues_additional.conf
[8]
wrapuptime=0
timeout=30
strategy=ringall
servicelevel=5
retry=4
reportholdtime=No
queue-youarenext=
queue-thereare=
queue-callswaiting=
periodic-announce-frequency=0
periodic-announce=periodic-announce
music=default
monitor-join=yes
monitor-format=
member=Local/8...@from-internal/n,0
member=Local/8...@from-internal/n,0
member=Local/8...@from-internal/n,0
member=Local/8...@from-internal/n,0
member=Local/8...@from-internal/n,0
member=Local/8...@from-internal/n,0
member=Local/8...@from-internal/n,0
member=Local/8...@from-internal/n,0
maxlen=0
leavewhenempty=no
joinempty=Yes
context=
announce-holdtime=no
announce-frequency=30
- The Verbosity logs:
after ringing on the available extensions for 20 seconds, it goes to macro
"hangupcall"
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("Local/8...@from-internal-80d5,2", "SIP/808|30|Ttr") in
new stack
-- Called 808
-- Local/8...@from-internal-80d5,1 is ringing
-- dialparties.agi: Checking CW and CFB status for extension 811
-- dialparties.agi: DbSet CALLTRACE/811 to 809
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("Local/8...@from-internal-3d60,2", "SIP/811|30|Ttr") in
new stack
-- Called 811
-- Local/8...@from-internal-3d60,1 is ringing
-- SIP/808-b7b038f8 is ringing
-- SIP/811-0936e868 is ringing
-- Stopped music on hold on SIP/809-09397eb0
== Spawn extension (from-internal, 8, 6) exited non-zero on 'SIP/809-09397eb0'
-- Executing Macro("SIP/809-09397eb0", "hangupcall") in new stack
-- Executing ResetCDR("SIP/809-09397eb0", "w") in new stack
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/8...@from-internal-3d60,2' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/8...@from-internal-3d60,2' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/8...@from-internal-3d60,2'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/8...@from-internal-80d5,2' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/8...@from-internal-80d5,2' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/8...@from-internal-80d5,2'
-- Executing NoCDR("SIP/809-09397eb0", "") in new stack
-- Executing Wait("SIP/809-09397eb0", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/809-09397eb0' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/809-09397eb0'
localhost*CLI>
- vi /etc/asterisk/extensions.conf
[macro-hangupcall]
exten => s,1,ResetCDR(w)
exten => s,2,NoCDR()
exten => s,3,Wait(5)
exten => s,4,Hangup
I tested multiple queues, and all of them are doing the same..
so please can anyone tell me how to resolve this issue.
Thanks
Torintino
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Re: [asterisk-users] Reproducible crash - known bug?
On 09/16/2009 08:53 AM, Jared Smith wrote: > Please open a report on our issue tracker at http://issues.asterisk.org/ Will do. Thanks! -- Ian Pilcher arequip...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] console color
I agree, but what is odd is that technically the same init script should be used in both cases, since heartbeat initially starts asterisk with the service start command, and service restart asterisk is used to restart it. perhaps safe_asterisk is restarting asterisk after the stop portion of the restart and before the start portion? and perhaps when this happens is when I get the "exited on signal 9", because the ports are already in use as the result of the safe_asterisk recovery, so service asterisk start (the other half of restart) fails? any ideas on how to modify safe_asterisk to have the same console parameters as asterisk? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, September 18, 2009 8:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] console color Just a wild guess, but your "service" probably runs two "flavors" of asterisk. Flavor 1 is /usr/sbin/asterisk (executable) which provides a console as you expect. Flavor 2 is /usr/sbin/safe_asterisk (shell-bash) which turns off the console color. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Damon Estep Sent: Friday, September 18, 2009 9:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] console color Hoping someone can help me understand what is happening here; we start asterisk as a service at boot (actually, with heartbeat) on CentOS using the asterisk init script installed with "make config" upon reboot of the server (when the asterisk service is first started by heartbeat) we get color in the console when we connect to it using asterisk -r after the execution of '#service asterisk restart' we no longer have color in the console this appear to be the case in all versions tested (1.2, 1.4, and 1,6) additionally, when executing the restart of the service we get a message that asterisk exited on signal 9, but I have not been able to find a definition for signal 9. I assume this is normal because we force an unconditional restart. we do the restart periodically due to some processes that don't always clean up after themselves, and the fact that a reload does not clean them up either (zombie channels, zombie manager connections). these are very heavily loaded servers, and the idea that a full restart should never be needed has been proven inaccurate over several years of experience :) I do not think this is heartbeat related, but just in case, here are the heartbeat details; these are heartbeat version 1 clusters the asterisk init script that is used is derived from "make config" we chkconfig --add asterisk, then chkconfig asterisk off (heartbeat starts it) we then define the asterisk service as a heartbeat managed resource about once a month we issue a "service asterisk restart" via a cron job, and this is where we lose the color. Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] console color
Just a wild guess, but your "service" probably runs two "flavors" of asterisk. Flavor 1 is /usr/sbin/asterisk (executable) which provides a console as you expect. Flavor 2 is /usr/sbin/safe_asterisk (shell-bash) which turns off the console color. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Damon Estep Sent: Friday, September 18, 2009 9:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] console color Hoping someone can help me understand what is happening here; we start asterisk as a service at boot (actually, with heartbeat) on CentOS using the asterisk init script installed with "make config" upon reboot of the server (when the asterisk service is first started by heartbeat) we get color in the console when we connect to it using asterisk -r after the execution of '#service asterisk restart' we no longer have color in the console this appear to be the case in all versions tested (1.2, 1.4, and 1,6) additionally, when executing the restart of the service we get a message that asterisk exited on signal 9, but I have not been able to find a definition for signal 9. I assume this is normal because we force an unconditional restart. we do the restart periodically due to some processes that don't always clean up after themselves, and the fact that a reload does not clean them up either (zombie channels, zombie manager connections). these are very heavily loaded servers, and the idea that a full restart should never be needed has been proven inaccurate over several years of experience :) I do not think this is heartbeat related, but just in case, here are the heartbeat details; these are heartbeat version 1 clusters the asterisk init script that is used is derived from "make config" we chkconfig --add asterisk, then chkconfig asterisk off (heartbeat starts it) we then define the asterisk service as a heartbeat managed resource about once a month we issue a "service asterisk restart" via a cron job, and this is where we lose the color. Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] console color
Hoping someone can help me understand what is happening here; we start asterisk as a service at boot (actually, with heartbeat) on CentOS using the asterisk init script installed with "make config" upon reboot of the server (when the asterisk service is first started by heartbeat) we get color in the console when we connect to it using asterisk -r after the execution of '#service asterisk restart' we no longer have color in the console this appear to be the case in all versions tested (1.2, 1.4, and 1,6) additionally, when executing the restart of the service we get a message that asterisk exited on signal 9, but I have not been able to find a definition for signal 9. I assume this is normal because we force an unconditional restart. we do the restart periodically due to some processes that don't always clean up after themselves, and the fact that a reload does not clean them up either (zombie channels, zombie manager connections). these are very heavily loaded servers, and the idea that a full restart should never be needed has been proven inaccurate over several years of experience :) I do not think this is heartbeat related, but just in case, here are the heartbeat details; these are heartbeat version 1 clusters the asterisk init script that is used is derived from "make config" we chkconfig --add asterisk, then chkconfig asterisk off (heartbeat starts it) we then define the asterisk service as a heartbeat managed resource about once a month we issue a "service asterisk restart" via a cron job, and this is where we lose the color. Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Blind Transfer Won't Hangup
ew stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/8678-c876bf18' in macro 'hangupcall'
== Spawn extension (macro-dial, h, 1) exited non-zero on 'SIP/8678-c876bf18'
-- Executing [...@macro-dial:8] Set("SIP/8678-c876bf18",
"DIALSTATUS=ANSWER") in new stack
-- Executing [...@macro-dial:9] GosubIf("SIP/8678-c876bf18",
"0?ANSWER,1") in new stack
-- Executing [8...@from-internal:19] Goto("SIP/8678-c876bf18",
"nextstep") in new stack
-- Goto (from-internal,8688,21)
-- Executing [8...@from-internal:21] Set("SIP/8678-c876bf18",
"RingGroupMethod=") in new stack
-- Executing [8...@from-internal:22] GotoIf("SIP/8678-c876bf18",
"0?nodest") in new stack
-- Executing [8...@from-internal:23] Set("SIP/8678-c876bf18",
"__NODEST=") in new stack
-- Executing [8...@from-internal:24] DBdel("SIP/8678-c876bf18",
"BLKVM/8688/SIP/8678-c876bf18") in new stack
-- DBdel: family=BLKVM, key=8688/SIP/8678-c876bf18
-- DBdel: Error deleting key from database.
-- Executing [8...@from-internal:25] Goto("SIP/8678-c876bf18",
"ext-local,vmb8688,1") in new stack
-- Goto (ext-local,vmb8688,1)
-- Executing [vmb8...@ext-local:1] Macro("SIP/8678-c876bf18",
"vm,8688,BUSY,") in new stack
-- Executing [...@macro-vm:1] Macro("SIP/8678-c876bf18",
"user-callerid,SKIPTTL") in new stack
-- Executing [...@macro-user-callerid:1] Set("SIP/8678-c876bf18",
"AMPUSER=8678") in new stack
-- Executing [...@macro-user-callerid:2] GotoIf("SIP/8678-c876bf18",
"0?report") in new stack
-- Executing [...@macro-user-callerid:3] ExecIf("SIP/8678-c876bf18",
"0?Set(REALCALLERIDNUM=8678)") in new stack
-- Executing [...@macro-user-callerid:4] Set("SIP/8678-c876bf18",
"AMPUSER=8678") in new stack
-- Executing [...@macro-user-callerid:5] Set("SIP/8678-c876bf18",
"AMPUSERCIDNAME=Ryan Wagoner") in new stack
-- Executing [...@macro-user-callerid:6] GotoIf("SIP/8678-c876bf18",
"0?report") in new stack
-- Executing [...@macro-user-callerid:7] Set("SIP/8678-c876bf18",
"AMPUSERCID=8678") in new stack
-- Executing [...@macro-user-callerid:8] Set("SIP/8678-c876bf18",
"CALLERID(all)="Ryan Wagoner" <8678>") in new stack
-- Executing [...@macro-user-callerid:9] Set("SIP/8678-c876bf18",
"REALCALLERIDNUM=8678") in new stack
-- Executing [...@macro-user-callerid:10]
ExecIf("SIP/8678-c876bf18", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [...@macro-user-callerid:11]
GotoIf("SIP/8678-c876bf18", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [...@macro-user-callerid:20] NoOp("SIP/8678-c876bf18",
"Using CallerID "Ryan Wagoner" <8678>") in new stack
-- Executing [...@macro-vm:2] GotoIf("SIP/8678-c876bf18", "0?4") in new
stack
-- Executing [...@macro-vm:3] SIPAddHeader("SIP/8678-c876bf18",
"Diversion: \;reason=no-answer\;screen=no\;privacy=off") in
new stack
-- Executing [...@macro-vm:4] Dial("SIP/8678-c876bf18",
"SIP/exchange-vm") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called exchange-vm
-- Got SIP response 302 "Moved Temporarily" back from 10.9.1.13
-- Now forwarding SIP/8678-c876bf18 to
'SIP/t...@10.9.1.13:5067' (thanks to SIP/exchange-vm-ac968658)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- SIP/10.9.1.13:5067-ac80fdc8 is ringing
-- Executing [...@macro-record-enable:3]
StopMonitor("Local/28...@from-internal-d4cc;1", "") in new stack
-- Executing [...@macro-record-enable:4]
AGI("Local/28...@from-internal-d4cc;1",
"recordingcheck,20090918-101319,1253283198.1299") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
-- SIP/10.9.1.13:5067-ac80fdc8 answered SIP/8678-c876bf18
-- Packet2Packet bridging SIP/8678-c876bf18 and SIP/10.9.1.13:5067-ac80fdc8
-- AGI Script recordingcheck
completed, returning 0
-- Executing [...@macro-record-enable:5]
MacroExit("Local/28...@from-internal-d4cc;1", "") in new stack
-- Executing [8...@from-internal-xfer:13]
GotoIf("Local/28...@from-internal-d4cc;1", "1 ?skipsimple") in new
stack
-- Goto (from-internal-xfer,8532,15)
-- Executing [8...@from-internal-xfer:15]
Set("Local/28...@from-internal-d4cc;1", "RingGroupMethod=ringall") in
new stack
-- Executing [8...@fro
[asterisk-users] calls drop during attended transfer with PRI line
Hi all, I'm using Asterisk 1.2.18 with a PRI card. My problem is during attended transfer which is working fine with some telco but it isn't with some others. The problem seems to be a notify message sent back to the caller containing a REMOTE HOLD command How can I tell Asterisk not to send it? I can send you the telco log if you need it. Thank you. Giorgio Incantalupo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI TDM440E still has echo on bridged connections
" Hello, Strangely i purchase a TDM440E with the echo canceller onboard and I still receive a horrible echo and i'm only using bridged connections between DAHDI/4 and DAHDI/1. I turned of echo cancellation on bridged connections which seemed to help alittle bit. I ran fxotune -i5 and setup fxotune -s to apply settings on startup, which has helpped but there is still echo on the begining of each call. Any idea's as to why there would be an echo at the beginning of a bridged conversation with echo cancellation turned off? Also this only happens on incomming calls not outgoing calls Thank you, Brad Finberg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI Caller ID problem
Cidstart=polarity or cidstart=ring will probably fix this.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
h...@cfht.hawaii.edu
Sent: Thursday, September 17, 2009 8:04 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DAHDI Caller ID problem
Aloha,
I'm finishing up the final touches on this install, and have run into an
odd problem.
I can't seem to get Caller ID on the analog phone lines working. It's a
Digium AEX 410 card.
I have Verbose set and a line to print the CID:
I have usecallerid=yes and callerid=asreceived set in both chan_dahdi.conf
and users.conf
[analog]
include=>default
exten => s,1,Verbose(passed id is ${CALLERID(num)})
exten => s,2,Answer
exten => s,3,Dial(SIP/100,,)
And this is what I'm getting.
*CLI> core set verbose 10
Verbosity was 1 and is now 10
-- Starting simple switch on 'DAHDI/1-1'
[Sep 17 14:44:05] NOTICE[15308]: chan_dahdi.c:7542 ss_thread: Got event 18
(Ring Begin)...
[Sep 17 14:44:07] NOTICE[15308]: chan_dahdi.c:7542 ss_thread: Got event 2
(Ring/Answered)...
[Sep 17 14:44:07] NOTICE[15308]: chan_dahdi.c:7706 ss_thread: MWI: Channel
1 message waiting!
-- Executing [...@analog:1] Verbose("DAHDI/1-1", "passed id is ") in new
stack
passed id is
-- Executing [...@analog:2] Answer("DAHDI/1-1", "") in new stack
-- Executing [...@analog:3] Dial("DAHDI/1-1", "SIP/100,,") in new stack
== Using SIP RTP CoS mark 5
-- Called 100
-- SIP/100-b6a22338 is ringing
-- SIP/100-b6a22338 answered DAHDI/1-1
== Spawn extension (analog, s, 3) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
I'm also getting these errors:
[Sep 17 14:01:06] ERROR[14462]: callerid.c:562 callerid_feed: No start bit
found in fsk data.
[Sep 17 14:01:06] WARNING[14462]: chan_dahdi.c:7582 ss_thread: CallerID
feed failed: Success
[Sep 17 14:01:06] WARNING[14462]: chan_dahdi.c:7686 ss_thread: CallerID
returned with error on channel 'DAHDI/1-1'
I have tried calleridsignal=dtmf & ring, as well as calleridstart=ring &
polarity. No love.
I searched on google for info, but nothing I found had a solution for my
problem.
I know that there's something I missing, but I can't seem to figure it
out. Can you all help me?
Thanks in advance!
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Re: [asterisk-users] dCAP Exam
"Danny Nicholas" writes: > Since Digium's contribution to Asterisk (hardware-wise) is Analog DAHDI > cards, this makes sense (to me). They make quite a few digital DAHDI cards too (PRI and BRI). Analog is a bit 80's. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audio Files
Hi people, What can I use to transfer the audio files to and from Asterisk? I was searching and I found the following commands: PUT SOUNDFILE and GET SOUNDFILE They are new commands of AGI, but is there another way to do that? Thanks, Anahi Ludueña _ Descubre todas las formas en que puedes estar en contacto con amigos y familiares. http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help sending call to local server
I have used the SIPPEER function to find if a phone is local and
available before.
PaulH
Asterisk User wrote:
> Hi,
>
> I have a generalized syntax for dial application in my dialplan where
> I send calls to particular server.
> Here is my dial sysntax...
> exten =>
> _x.,1,Dial(${Dial_technology}/${extension_to_ca...@${server_ip},30,r)
>
> I can send a call to remote server using register statement in
> sip.conf or iax.conf and it works as calls get landed in particular
> context of remote server.
>
> Would you please suggest me changes to be made in .conf file(s) if I
> want the calls to be landed in context of local server if Server_ip is
> the IP of a server running asterisk?
>
> Thanking you
>
>
> --ASTERISK USER
>
>
>
>
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Re: [asterisk-users] DeadAgi
Thanks guys, I'll take it into account!... Anahi Ludueña > Date: Fri, 18 Sep 2009 10:13:12 +0100 > From: i...@pack-net.co.uk > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] DeadAgi > > > > Steve Edwards wrote: > > On Thu, 17 Sep 2009, Anahi Ludue?a wrote: > > > >> Thanks for the answers! > >> The file didn't have the first line! > >> #!/usr/bin/php > > > > Glad you found the answer. However... > > > >> The command ls -l returns: > >> > >> -rwxrwxrwx 1 root root 140 Sep 17 15:42 finconf.php > > > > Having an executable with 777 permissions is a very bad idea. Think > > about somebody (or some program) executing something like: > > > > echo "rm -f -r /whatever-they-want" \ > > >/var/lib/asterisk/agi-bin/finconf.php > > > Agreeing with the above here, really you want the script owned by > asterisk.asterisk and permissions of 0755 > > Ish > > > > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > > Register Now: http://www.astricon.net > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > Ishfaq Malik > Software Developer > PackNet Ltd > > Office: 0161 660 3062 > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users _ Chatea sin límites en Messenger con la tarifa plana de Orange http://serviciosmoviles.es.msn.com/messenger/orange.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DeadAgi
Steve Edwards wrote: > On Thu, 17 Sep 2009, Anahi Ludue?a wrote: > >> Thanks for the answers! >> The file didn't have the first line! >> #!/usr/bin/php > > Glad you found the answer. However... > >> The command ls -l returns: >> >> -rwxrwxrwx 1 root root 140 Sep 17 15:42 finconf.php > > Having an executable with 777 permissions is a very bad idea. Think > about somebody (or some program) executing something like: > > echo "rm -f -r /whatever-they-want" \ > >/var/lib/asterisk/agi-bin/finconf.php > Agreeing with the above here, really you want the script owned by asterisk.asterisk and permissions of 0755 Ish > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] limit concurrent calls on trunk supporting multiple DID
This sounds like a clever way to solve my problem.
But, in the meanwhile, I've already implemented this programatically
from agi script, solving also another requirement I had which was to
limit the number of calls during a certain period (to avoid malicious
sip uri calls flooding the pbx)
Thanks anyway, I'll definately put this useful command on top of my
head. I'm sure I'll need it one day
Patrick
On Fri, Sep 18, 2009 at 02:11, C. Chad Wallace
wrote:
>
> At 7:16 AM on 17 Sep 2009, Patrick wrote:
>
>> I've one SIP trunk that support multiple DID. Only the trunk is
>> documented in sip.conf (called DID is taken from the sip-header in
>> real time).
>> I would like to limit the number of simultaneous calls on each DID. Is
>> there a way to achieve this ?
>
> I think you could use GROUP() and GROUPCOUNT() for that. I do that for
> Queue calls currently, so each agent only gets one call at a time. It
> would go something like this (entirely untested):
>
> [incoming]
> exten => _X.,1,Set(DID=${EXTEN})
> exten => _X.,n,GotoIf($[GROUP_COUNT(${DID})=0]?accept)
> exten => _X.,n,Busy()
>
> exten => _X.,n(accept),Set(GROUP()=${DID})
> ; Now let the call through as usual...
> exten => _X.,n,Goto(mainmenu,s,1)
>
> That puts each call into a group named by the DID, and returns Busy
> if there is another call on the same DID.
>
> --
>
> C. Chad Wallace, B.Sc.
> The Lodging Company
> http://www.skihills.com/
> OpenPGP Public Key ID: 0x262208A0
>
>
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>
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