Re: [asterisk-users] DAHDI Caller ID problem
- Danny Nicholas da...@debsinc.com 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] 1.6.1 + TDM840 FSK MWI problem
- Doug Bailey dbai...@digium.com wrote: - Barry Miller asterisk-us...@notanet.net wrote: On Fri, Sep 04, 2009 at 04:10:43PM -0500, Doug Bailey wrote: - Barry Miller asterisk-us...@notanet.net wrote: Hi, Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? [snip] The only thing I can think of that would be preventing the output would be problems in the interface chip with the On-Hook transfer mode. If you run a dahdi_monitor on the channel that should be sending the FSK spill and look at the results in a program like audacity, you can see if the MWI FSK spill is actually reaching the interface SLIC IC. Something like dahdi_monitor 1 -t spilloutput.raw (Monitors the output going to dahdi channel 1.) Hmm. With both 1.4 1.6, without touching /etc/[asterisk|dahdi], I used a butt-set to go off-hook, then back on. I got: 1.4.26.1: dahdi_monitor captured stutter dialtone, 4.5 seconds of silence, then the FSK spill. And that's what I heard. 1.6.1.6: dahdi_monitor captured stutter dialtone, 1.5 seconds of silence, then the FSK spill. Sounds good with audacity. But all I heard through the butt-in was stutter dialtone. No FSK spill at all. Here's hoping this tells you more than it does me :) Actually it does tell me a lot. The problem appears in how the interface chip is being programmed. For some reason, the interface chip is not being set to on-hook transfer mode which would allow for the mwi spill to go out on the actual fxs port lines. I am looking to see where the problem lies. (It is either in chan_dahdi or in the driver.) I hope to have more information later. The problem lies in a race condition between chan_dahdi making an ioctl call to set the VMWI state (performed in the do_monitor loop ) and a a subsequent call to set the channel in ONHOOK transfer mode (performed in mwi_send_init). This requires the driver to send successive commands to the SLIC interface chip linefeed register. If the command required for the VMWI mode is not completed by the time the ONHOOK transfer mode is requested, the ONHOOK transfer request is thrown away and the MWI spill does not get sent. I will be fixing this in the drivers trunk branch and hope to have it committed soon. I'm not sure when it will be released. Another option is to comment out the ioctl call for VMWI in the do_monitor loop (especially if you do not care about line reversal MWI.). Doug ___ -- 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] 1.6.1 + TDM840 FSK MWI problem
- Doug Bailey dbai...@digium.com wrote: - Doug Bailey dbai...@digium.com wrote: - Barry Miller asterisk-us...@notanet.net wrote: On Fri, Sep 04, 2009 at 04:10:43PM -0500, Doug Bailey wrote: - Barry Miller asterisk-us...@notanet.net wrote: Hi, Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? [snip] The only thing I can think of that would be preventing the output would be problems in the interface chip with the On-Hook transfer mode. If you run a dahdi_monitor on the channel that should be sending the FSK spill and look at the results in a program like audacity, you can see if the MWI FSK spill is actually reaching the interface SLIC IC. Something like dahdi_monitor 1 -t spilloutput.raw (Monitors the output going to dahdi channel 1.) Hmm. With both 1.4 1.6, without touching /etc/[asterisk|dahdi], I used a butt-set to go off-hook, then back on. I got: 1.4.26.1: dahdi_monitor captured stutter dialtone, 4.5 seconds of silence, then the FSK spill. And that's what I heard. 1.6.1.6: dahdi_monitor captured stutter dialtone, 1.5 seconds of silence, then the FSK spill. Sounds good with audacity. But all I heard through the butt-in was stutter dialtone. No FSK spill at all. Here's hoping this tells you more than it does me :) Actually it does tell me a lot. The problem appears in how the interface chip is being programmed. For some reason, the interface chip is not being set to on-hook transfer mode which would allow for the mwi spill to go out on the actual fxs port lines. I am looking to see where the problem lies. (It is either in chan_dahdi or in the driver.) I hope to have more information later. The problem lies in a race condition between chan_dahdi making an ioctl call to set the VMWI state (performed in the do_monitor loop ) and a a subsequent call to set the channel in ONHOOK transfer mode (performed in mwi_send_init). This requires the driver to send successive commands to the SLIC interface chip linefeed register. If the command required for the VMWI mode is not completed by the time the ONHOOK transfer mode is requested, the ONHOOK transfer request is thrown away and the MWI spill does not get sent. I will be fixing this in the drivers trunk branch and hope to have it committed soon. I'm not sure when it will be released. Another option is to comment out the ioctl call for VMWI in the do_monitor loop (especially if you do not care about line reversal MWI.). See https://issues.asterisk.org/view.php?id=15875 for more information ___ -- 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] 1.6.1 + TDM840 FSK MWI problem
- Barry Miller asterisk-us...@notanet.net wrote: On Fri, Sep 04, 2009 at 04:10:43PM -0500, Doug Bailey wrote: - Barry Miller asterisk-us...@notanet.net wrote: Hi, Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? [snip] The only thing I can think of that would be preventing the output would be problems in the interface chip with the On-Hook transfer mode. If you run a dahdi_monitor on the channel that should be sending the FSK spill and look at the results in a program like audacity, you can see if the MWI FSK spill is actually reaching the interface SLIC IC. Something like dahdi_monitor 1 -t spilloutput.raw (Monitors the output going to dahdi channel 1.) Hmm. With both 1.4 1.6, without touching /etc/[asterisk|dahdi], I used a butt-set to go off-hook, then back on. I got: 1.4.26.1: dahdi_monitor captured stutter dialtone, 4.5 seconds of silence, then the FSK spill. And that's what I heard. 1.6.1.6: dahdi_monitor captured stutter dialtone, 1.5 seconds of silence, then the FSK spill. Sounds good with audacity. But all I heard through the butt-in was stutter dialtone. No FSK spill at all. Here's hoping this tells you more than it does me :) Actually it does tell me a lot. The problem appears in how the interface chip is being programmed. For some reason, the interface chip is not being set to on-hook transfer mode which would allow for the mwi spill to go out on the actual fxs port lines. I am looking to see where the problem lies. (It is either in chan_dahdi or in the driver.) I hope to have more information later. Doug ___ -- 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] 1.6.1 + TDM840 FSK MWI problem
- Barry Miller asterisk-us...@notanet.net wrote: On Wed, Sep 02, 2009 at 09:44:05AM -0500, Doug Bailey wrote: - Barry Miller asterisk-us...@notanet.net wrote: Hi, Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? The ability to do line reversal MWI was added into the 1.6.2 branch. Looking through the 1.6.1 code base, I don't see anything other than fsk MWI (with and without Ring Pulse Alert Signalling.) In any case, this is set by defining mwisendtype in chan_dahdi. The default for this is fsk spills. It can be set to nofsk if you want to disable the fsk spills. The line reversal is set by specifying mwisendtype=lrev Regards, Doug Bailey Thanks, but that's not the problem. I _want_ FSK. A few ast_debug's in chan_dahdi tell me that after calling vmwi_generate(), it's taking the MWI_SEND_SPILL path through mwi_send_thread(), and happily sending about 9K bytes of spill, 160 bytes at a time. But my phones (and a butt-set) tell me that nothing is being received. The only thing I can think of that would be preventing the output would be problems in the interface chip with the On-Hook transfer mode. If you run a dahdi_monitor on the channel that should be sending the FSK spill and look at the results in a program like audacity, you can see if the MWI FSK spill is actually reaching the interface SLIC IC. Something like dahdi_monitor 1 -t spilloutput.raw (Monitors the output going to dahdi channel 1.) I don't understand the DAHDI ioctls very well. Is it possible that the TDM840 is not in the correct state when the spill is transmitted? Thanks again, --Barry ___ -- 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] Noises on Batphones
- Jason Martin jmar...@metrixmatrix.com wrote: Hello, The company I work for recently purchased 2 Rhino CB24s and a Rhino PCI-E R4T1. The channel banks are plugged into the R4T1, as well as 2 PRIs from our telco. The CB24s are for all internal analog phones. Most of the phones are setup in batphone mode, which is immediate=on in the DAHDI config. They are set up this way because we are an outgoing call center, and the context that the batphones go to a database table to pull the phone number they are calling. Along with this new hardware, we changed to a new server (just a Dell E520 workstation with 4 gigs of RAM and 2 250 gig SATA drives software RAIDed) with the following software: Ubuntu 9.04 Asterisk 1.6.1.4 Asterisk-addons 1.6.1.1 (for the cdr-mysql plugin) dahdi-linux-complete 2.2.0.2 + 2.2.0 libpri 1.4.10.1 rhino drivers 0.99.2 Since day one, all batphones have had a weird noise at the very beginning of the call. I contacted Rhino about it and the support tech told me that it's fsk tones that have caller ID and MWI information and advised me to turn off advanced features like mailboxes. The phones already didn't have mailboxes, but I put in mwisendtype=nofsk in chan_dahdi.conf anyway, and all features like faxdetect and transfer are turned off. If the fsk spill that you hear is at the beginning of the call, it is due to the transmission of CID information. (This is usually transmitted between the first and second rings.) You should be able to set usecallerid=no in chan_dahdi.conf to disable the sending of the caller id fsk spill. Regards Doug Bailey ___ -- 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] 1.6.1 + TDM840 FSK MWI problem
- Barry Miller asterisk-us...@notanet.net wrote: Hi, Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? The ability to do line reversal MWI was added into the 1.6.2 branch. Looking through the 1.6.1 code base, I don't see anything other than fsk MWI (with and without Ring Pulse Alert Signalling.) In any case, this is set by defining mwisendtype in chan_dahdi. The default for this is fsk spills. It can be set to nofsk if you want to disable the fsk spills. The line reversal is set by specifying mwisendtype=lrev Regards, Doug Bailey ___ -- 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] Message Waiting Indicator on DAHDI line
- Mike asterisk-us...@norgie.net wrote:
Folks,
I have recently installed Asterisk 1.6.1.1. I have two PSTN lines
connected to a TDM400 and two VoIP lines using SIP. I have a CISCO
7940
using SIP as my desk phone. Calling any of the four lines should
ring
the desk phone. This works fine, except that when ringing the PSTN
lines, it activates the MWI on the 7940. I can see this happening on
the console:
[Aug 4 16:48:47] NOTICE[2964]: chan_dahdi.c:7669 ss_thread: MWI:
Channel 3 message waiting!
Looking at the offending piece of code, it seems to suggest from the
comment that it is getting the MWI from the CLID.
/* If the CID had Message waiting payload, assume that this for MWI
only
and hangup the call */
if (flags CID_MSGWAITING) {
ast_log(LOG_NOTICE, MWI: Channel %d message waiting!\n,
p-channel);
notify_message(p-mailbox, 1);
/* If generated using Ring Pulse Alert, then ring has been answered
as
a call and needs to be hungup */
if (p-mwimonitor_rpas) {
ast_hangup(chan);
return NULL;
}
}
I have set usecallerid=no on both interfaces and globally but I
still
cannot get it to stop.
I have failed to turn anything up on Google regarding this.
Does anyone have any suggestions please?
Mike.
This code is designed to handle Message Waiting Indication (MWI) incoming on FXO
line. This data could very well be embedded in your CID spill as part of an
MDMF message that also contains the caller id information. (See main/callerid.c
in the callerid_feed function.)
If your incoming line has a mailbox associated with it, the MWI information will
be pushed to that mailbox. You may want to look at how your mailboxes are
defined
and the channels to which they are associated.
Doug
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Re: [asterisk-users] Asterisk on AVR32
When you run configure, you need to spec the host parameter for the architecture and environment you will be running under. For example, a 1.4 distro being built for a blackfin running uclinux would run ./configure host=bfin-uclinux This will imply that the compiler being used to build the code will be named host cpu-os-gcc and is accessible in the PATH that you provide to the make system. In the example above, the compiler is named: bfin-uclinux-gcc Doug - Kyle Kienapfel doctor.w...@gmail.com wrote: why is CROSS_ARCH=Linux? is this something the AVR32 distro is doing, or something you did? it should be something line avr or avr32 On Thu, Jun 18, 2009 at 3:08 AM, Paulo Santos paulo.r.san...@sapo.pt wrote: Greetings everyone, I'm trying to compile asterisk for an AVR32 (Atmel NGW100). Buildroot for AVR32 already has the asterisk package, though it has bugs. Firstly it tries to apply a patch for 1.2 on a 1.6, but deleting the contents of the patch file did the trick. Now, the problem is making asterisk. The first error is because asterisk needed to be ./configure:ed. Trying to just do ./configure, make gives an error [1]. Trying to do ./configure with the same args as make plus --host it can't even configure [2] I don't know much about cross-compiling, or even regular compiling for that matter. Does any one have any idea on how to do this? Thanks in advance, Best regards, Paulo Santos [1] menuselect/menuselect --check-deps menuselect.makeopts /bin/bash: menuselect/menuselect: cannot execute binary file make[1]: *** [menuselect.makeopts] Error 126 make[1]: Leaving directory `/home/psantos/br/buildroot-avr32-v2.3.0/build_avr32/asterisk-1.6.0-beta6' make: *** [/home/psantos/br/buildroot-avr32-v2.3.0/build_avr32/asterisk-1.6.0-beta6/asterisk] Error 2 [2] configure: WARNING: If you wanted to set the --build type, don't use --host. If a cross compiler is detected then cross compile mode will be used. checking build system type... i686-pc-linux-gnu checking host system type... Invalid configuration `CROSS_ARCH=Linux': machine `CROSS_ARCH=Linux' not recognized configure: error: /bin/bash ./config.sub CROSS_ARCH=Linux failed ___ -- 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
Re: [asterisk-users] pstn hangs up: MWI no message waiting ??
- sean darcy seandar...@gmail.com wrote: OK. Calmer now. If fact a 410 would have the same problem. I'll make the fix on our machines. Should I file a bug, or does the 169154 commit already fix it? sean The issues has been corrected in trunk and the 1.6.1 branch. Sicne we have addressed the issue and if it works for you, then I don't see the need for a bug report. If you have any other concerns or have any other problems with it, then go ahead and submit a bug report. Doug ___ -- 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] pstn hangs up: MWI no message waiting ??
- sean darcy seandar...@gmail.com wrote: Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! As part of the implementation of issue 8587, a check was incldued for MWI messages preceded by Ring Pulse Alert Signals (RPAS). The RPAS is answered by chan_dahdi as a standard call and the MWI message is processed. As part of the implementation, if the MWI message was included, the channel was hung up. This did not take into account that possibility of MWI messages included into the to standard CID spills. I believe this is the case here and the MWI portion of the CID spill is causing the channel to hang up. You can look at commit 169154 for a fix or simply remove the ast_hangup calls immediately after the message MWI: channel %d no message waiting!\n and MWI: Channel %d no message w ___ -- 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] pstn hangs up: MWI no message waiting ??
- sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! This message occurs when the pstn sends a FSK spill indicating the message waiting status of the FXO port in question. This may encoded in the caller ID indicator or may be contained in its own message spill. This is output as a NOTICE logging message. Regards, Doug Bailey ___ -- 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] Line 0005 cannot be answered?
Have you tried to contact Digium Support about this question? They are very good at answering AA50 configuration questions. Regards, Doug Bailey - Fidel Garcia [EMAIL PROTECTED] wrote: I have a Digium Appliance AA50 configure with 8 lines and two dial plans. Each dial plan takes care of a particular location. In Dialplan2 we have 4 lines. xxx xxx 0333 xxx xxx 0005 xxx xxx 0006 xxx xxx 0007 When a call gets to line 0005 you pick up the phone but the call does not get connected, you can still hear the phone ringing in a noisy weird way on the handset - the caller’s phone never stops ringing. Is this a problem related to the way the line was taken out of the punch panel or does I have to do anything with configuration? Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] increase ring time out
- Fidel Garcia [EMAIL PROTECTED] wrote:
Where exactly do I have to change it?
The GUI on the AA50 generates users via users.conf. These users are added into
the dialplan automatically and are placed into the default context. Calls to
the users are made via the stdexten macro. In that macro is a Dial statement
with a timeout of 20. You would have to adjust that timeout manually and save
it off (Run the save_config script)
One caveat is that the AA50 is not supported when you manually modify the dial
plan. The changes you make are at your own risk.
- Doug Bailey
This is the extensions.conf file:
;! Automatically generated configuration file
;! Filename: extensions.conf (/etc/asterisk/extensions.conf)
;! Generator: Manager
;! Creation Date: Tue Jul 22 15:14:28 2008
;!
[general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
priorityjumping = no
[globals]
trunk_1 = Zap/g1
trunk_1_cid = asreceived
[dundi-e164-canonical]
[dundi-e164-customers]
[dundi-e164-via-pstn]
[dundi-e164-local]
include = dundi-e164-canonical
include = dundi-e164-customers
include = dundi-e164-via-pstn
[dundi-e164-switch]
switch = DUNDi/e164
[dundi-e164-lookup]
include = dundi-e164-local
include = dundi-e164-switch
[macro-dundi-e164]
exten = s,1,Goto(${ARG1},1)
include = dundi-e164-lookup
[macro-trunkdial]
exten = s,1,set(CALLERID(all)=${IF(${LEN(${CALLERID(num)})} 6 ?
${CALLERID(al
l)} : ${ARG2})})
exten = s,n,Dial(${ARG1})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Hangup
exten = _s-.,1,NoOp
[iaxtel700]
exten =
_91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
[iaxprovider]
[trunkint]
exten = _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten = _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
exten = _91NXXNXX,1,Macro(dundi-e164,${EXTEN:1})
exten = _91NXXNXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunktollfree]
exten = _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[international]
ignorepat = 9
include = longdistance
include = trunkint
[longdistance]
ignorepat = 9
include = local
include = trunkld
[local]
ignorepat = 9
include = default
include = parkedcalls
include = trunklocal
include = iaxtel700
include = trunktollfree
include = iaxprovider
[macro-stdexten]
exten = s,1,Dial(${ARG2},20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[macro-stdPrivacyexten]
exten = s,1,Dial(${ARG2},20|p)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = s-DONTCALL,1,Goto(${ARG3},s,1)
exten = s-TORTURE,1,Goto(${ARG4},s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[macro-page]
exten = s,1,ChanIsAvail(${ARG1}|js)
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
exten = s,n(autoanswer),Set(_ALERT_INFO=RA)
exten = s,n,SIPAddHeader(Call-Info: Answer-After=0)
exten = s,n,NoOp()
exten = s,n,Dial(${ARG1}||)
exten = s,n(fail),Hangup
[demo]
exten = s,1,Wait(1)
exten = s,n,Answer
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n(restart),BackGround(demo-congrats)
exten = s,n(instruct),BackGround(demo-instruct)
exten = s,n,WaitExten
exten = 2,1,BackGround(demo-moreinfo)
exten = 2,n,Goto(s,instruct)
exten = 3,1,Set(LANGUAGE()=fr)
exten = 3,n,Goto(s,restart)
exten = 1000,1,Goto(default,s,1)
exten = 1234,1,Playback(transfer,skip)
exten = 1234,n,Macro(stdexten,1234,${CONSOLE})
exten = 1235,1,Voicemail(u1234)
exten = 1236,1,Dial(Console/dsp)
exten = 1236,n,Voicemail(u1234)
exten = #,1,Playback(demo-thanks)
exten = #,n,Hangup
exten = t,1,Goto(#,1)
exten = i,1,Playback(invalid)
exten = 500,1,Playback(demo-abouttotry)
exten = 500,n,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten = 500,n,Playback(demo-nogo)
exten = 500,n,Goto(s,6)
exten = 600,1,Playback(demo-echotest)
exten = 600,n,Echo
exten = 600,n,Playback(demo-echodone)
exten = 600,n,Goto(s,6)
exten = 76245,1,Macro(page,SIP/Grandstream1)
exten = _7XXX,1,Macro(page,SIP/${EXTEN})
exten = 7999,1,Set(TIMEOUT(absolute)=60)
exten =
7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED]/n
|d)
exten = 8500,1,VoicemailMain
exten = 8500,n,Goto(s,6)
[page]
exten = _X.,1,Macro(page,SIP/${EXTEN})
[default]
exten = 6050,1
Re: [asterisk-users] delay when rinigng asterisk
- Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: Tell your box to not expect Caller*ID information. You set that with usercallerid=no in /etc/asterisk/zapata.conf Since you are using the Asterisk Appliance you would have to contact Digium for support. Sydney Web Hosting wrote: Hi All, I have just setup an asterisk box (AA50) and all is running well. however when I ring the phone number (analog lines) there seems to be a delay. I'm ringing from my mobile phone - It rings 4 times on my mobile before I can hear it in the office. Any ideas on how to shorten this time? Thanks Dave. I would avoid using zapata.conf on the AA50 as it is manipulated during the boot process. The place to set these parameters is in the general section of users.conf (which is where the gui places them.) How is your unit connected to the PSTN. If you are using an analog input, you will see its associated led flash when it detects ringing from the PSTN. This may help you in determining when the ring is actually occurring on the AA50. I'd also encourage you to get in touch with the support group at Digium as they are quite adept at getting over these types of issues. - Doug ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] restart asterisk daily
If you want to flush your disk cache to see how much memory is being eaten cache pages, try this: echo 3 /proc/sys/vm/drop_caches - ast erisk [EMAIL PROTECTED] wrote: ___ -- 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
Re: [asterisk-users] Telco MWI Detection on TDM400 Interface?
The MWI detection is done using fsk modem detection within chan_zap itself. (It does not support neon MWI detection.) The driver plays no real part in the detection. Doug Bailey - Original Message - From: Jim Duda [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, February 3, 2008 11:09:22 AM (GMT-0600) America/Chicago Subject: [asterisk-users] Telco MWI Detection on TDM400 Interface? I've upgraded to asterisk-1.6.0-beta2. I'm trying to get the new Telco MWI detection function working. It doesn't appear to be working. I have this in zapata.conf ; PSTN connected here ;immediate=no ;busydetect=yes ;busycount=8 ;musiconhold=default mwimonitor=yes ;mwilevel=512 mwimonitornotify=/usr/local/sbin/zapnotify.sh faxdetect=incoming signalling=fxs_ks context=incoming channel = 4 However, the zapnotify.sh script in the /usr/local/sbin directory is never getting called. Do I need a new version of the zaptel drivers? I'm currently using version 1.4.6 Is there anyway to get debugging into the log files? Am I missing something? Jim ___ -- 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
Re: [asterisk-users] Polycom Phone and bitmaps
Shaun wrote: I've been trying to get the polycom 550 phones to show a idle display bitmap but have not been successful. Anybody have any experience with this? The manual gives instructions (http://www.polycom.com/common/documents/support/setup_maintenance/products/voice/soundpoint_ip_soundstation_ip_administrators_guide_v2_2.pdf) but they do not seam to work. So far i've done the following in my sip.conf Also beware that there is enough bitmap quota allocated to your machine. (See quotas in the admin guide) I believe the 550 phone provides 10KB of bitmap space. You should still see the bitmap served to the phone. The phone throws the image away if it is too big. - Doug ___ --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
