Re: [asterisk-users] dialplan reloading
Hi Jerry, From the Asterisk CLI, enter the command core restart when convenient, this command will restart asterisk only when there is no incoming call, and when it will close all outgoing calls. With a restart of asterisk should reload all the information: extensions, sip, agi, iax, voicemail, etc. Danilo Il 01/11/12 23:30, Jerry Geis ha scritto: If I issue a dialplan reload and some AGI starts as its reloading and directs something into the diaplan that is still reloading what happens I presume my context is not there? What I see is the diaplan is messed up somehow and I goto the default context then after that it is messaged up until I stop and restart. How do i prevent this from happening? Thanks, jerry -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to create channel of type 'DAHDI' (cause 17 - User busy)
Hi,
I have 6 Red FXO with TDM2400p in my PC. I have install asterisk and dahdi
driver.
Scenario is
jitsi- asterisk server- analog PBX landline phone
I configured this scenario as follow
in chan_dahdi.conf file
; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
;FXO Modules
group=2
echocancel=yes
signalling=fxs_ks
context=Incoming
channel=1-20
After loading module in astrisk giving o/p below
module load chan_dahdi.so
Loaded chan_dahdi.so
== Parsing '/etc/asterisk/chan_dahdi.conf': == Found
== Parsing '/etc/asterisk/users.conf': == Found
-- Registered channel 1, FXS Kewlstart signalling
-- Registered channel 2, FXS Kewlstart signalling
-- Registered channel 3, FXS Kewlstart signalling
-- Registered channel 4, FXS Kewlstart signalling
-- Registered channel 5, FXS Kewlstart signalling
-- Registered channel 6, FXS Kewlstart signalling
-- Registered channel 7, FXS Kewlstart signalling
-- Registered channel 8, FXS Kewlstart signalling
-- Registered channel 9, FXS Kewlstart signalling
-- Registered channel 10, FXS Kewlstart signalling
-- Registered channel 11, FXS Kewlstart signalling
-- Registered channel 12, FXS Kewlstart signalling
-- Registered channel 13, FXS Kewlstart signalling
-- Registered channel 14, FXS Kewlstart signalling
-- Registered channel 15, FXS Kewlstart signalling
-- Registered channel 16, FXS Kewlstart signalling
-- Registered channel 17, FXS Kewlstart signalling
-- Registered channel 18, FXS Kewlstart signalling
-- Registered channel 19, FXS Kewlstart signalling
-- Registered channel 20, FXS Kewlstart signalling
-- Automatically generated pseudo channel
[Nov 2 14:38:50] WARNING[1886]: chan_dahdi.c:17278 process_dahdi:
Ignoring any changes to 'userbase' (on reload) at line 23.
[Nov 2 14:38:50] WARNING[1886]: chan_dahdi.c:17278 process_dahdi:
Ignoring any changes to 'vmsecret' (on reload) at line 31.
[Nov 2 14:38:50] WARNING[1886]: chan_dahdi.c:17278 process_dahdi:
Ignoring any changes to 'hassip' (on reload) at line 35.
[Nov 2 14:38:50] WARNING[1886]: chan_dahdi.c:17278 process_dahdi:
Ignoring any changes to 'hasiax' (on reload) at line 39.
[Nov 2 14:38:50] WARNING[1886]: chan_dahdi.c:17278 process_dahdi:
Ignoring any changes to 'hasmanager' (on reload) at line 47.
== Registered channel type 'DAHDI' (DAHDI Telephony Driver)
== Manager registered action DAHDITransfer
== Manager registered action DAHDIHangup
== Manager registered action DAHDIDialOffhook
== Manager registered action DAHDIDNDon
== Manager registered action DAHDIDNDoff
== Manager registered action DAHDIShowChannels
== Manager registered action DAHDIRestart
Loaded chan_dahdi.so = (DAHDI Telephony Driver)
In my extension.conf file i wrote dialplan for user so sandeep is jitsi
user and 81 and 88 is landline number.
[general]
static=yes
writeprotect=no
clearglobalvars=no
[Incoming]
exten = s,1,Answer
exten = s,2,Dial(DAHDI/g1,20,rt)
exten = s,3,Voicemail(1000,u)
exten = s,103,Voicemail(1000,b)
exten = sandeep,1,Dial(SIP/sandeep)
exten = sandeep,n,Hangup()
exten = 1004,4,Dial(SIP/sandeep)
exten = 1004,n,Hangup()
; Testing extension, prepare to be insulted like a
; Monthy Python knight
exten = 81,1,Dial(DAHDI/1,20,rt)
exten = 81,n,Hangup()
exten = 88,1,Dial(DAHDI/1,20,rt)
exten = 88,n,Hangup()
exten = 8500,1,VoiceMailMain
exten = 8501,1,MusicOnHold
exten = _9.,1,Dial(DAHDI/g2/www${EXTEN:1})
exten = _9.,2,Congestion
exten = 201,1,Answer()
exten = 201,n,Playback(tt-monty-knights)
exten = 201,n,Hangup()
; Echo-test, it is good to test if we have sound in both directions.
; The call is answered
exten = 202,1,Answer()
; Welcome message is played
exten = 202,n,Playback(welcome)
; Play information about the echo test
exten = 202,n,Playback(demo-echotest)
; Do the echo test, end with the # key
exten = 202,n,Echo()
; Plays information that the echo test is done
exten = 202,n,Playback(demo-echodone)
; Goodbye message is played
exten = 202,n,Playback(vm-goodbye)
; Hangup() ends the call, hangs up the line
exten = 202,n,Hangup()
After loading extension and dahdi, i called from jitsi and dialed 81 but
asterisk is giving o/p as below and busy tone is coming on jitsi
-- Executing [81@myphones:1] Dial(SIP/sandeep-,
DAHDI/1,20,rt) in new stack
-- Called 1
[Nov 2 14:45:31] WARNING[2145]: chan_dahdi.c:7536 handle_alarms:
Detected alarm on channel 1: Red Alarm
-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
Re: [asterisk-users] dialplan reloading
Hi Jerry, From the Asterisk CLI, enter the command core restart when convenient, this command will restart asterisk only when there is no incoming call, and when it will close all outgoing calls. With a restart of asterisk should reload all the information: extensions, sip, agi, iax, voicemail, etc. Danilo Danilo, Ok - but then what if a call comes in while it decides to reload - or if an AGI is started while it decides to reload - Sure there is nothing happening at that moment - but lets say right after it decides that its convenient and before its done - something gets started. What to do about that? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan reloading
On Fri, 2012-11-02 at 06:25 -0400, Jerry Geis wrote: Hi Jerry, From the Asterisk CLI, enter the command core restart when convenient, this command will restart asterisk only when there is no incoming call, and when it will close all outgoing calls. With a restart of asterisk should reload all the information: extensions, sip, agi, iax, voicemail, etc. Danilo Danilo, Ok - but then what if a call comes in while it decides to reload - or if an AGI is started while it decides to reload - Sure there is nothing happening at that moment - but lets say right after it decides that its convenient and before its done - something gets started. What to do about that? Jerry -- Sorry to step in here but I think the 2 of you are talking at cropp purposes I initial query was about a dialplan reload, not an asterisk restart. Jerry, how long does your system take to perform a dialplan reload? surely it is under a second. If you look in the logs, at the end of any dialplan reload (well in 1.8 at least) you will get some time stats such as below [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Time to scan old dialplan and merge leftovers back into the new: 0.41 sec [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Time to restore hints and swap in new dialplan: 0.03 sec [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Time to delete the old dialplan: 0.04 sec [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Total time merge_contexts_delete: 0.48 sec As you can see in this example it takes under a ten thousandth of a second. Is that something to really be concerned about? Regards Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with R2D configuration
Hi, Has anybody worked on R2D Brazillian setup. I have configured R2 using OpenR2 with Asterisk. While doing some analysis I found R2D is already included in libopenr2. Have anyone tested the same. Regards, Gopal. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan reloading
Sorry to step in here but I think the 2 of you are talking at cropp purposes I initial query was about a dialplan reload, not an asterisk restart. Jerry, how long does your system take to perform a dialplan reload? surely it is under a second. If you look in the logs, at the end of any dialplan reload (well in 1.8 at least) you will get some time stats such as below [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Time to scan old dialplan and merge leftovers back into the new: 0.41 sec [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Time to restore hints and swap in new dialplan: 0.03 sec [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Time to delete the old dialplan: 0.04 sec [2012-10-29 00:10:03] VERBOSE[19096] pbx.c: -- Total time merge_contexts_delete: 0.48 sec As you can see in this example it takes under a ten thousandth of a second. Is that something to really be concerned about? Regards Ish Actually my mistake - looks like based on my code certain things happen and I issue two dialplan reload commands. So the second is killing the first. Then asterisk looses information. So certainly I should not be doing that - but I'm surprised asterisk lets another reload happen before the first is complete. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan reloading
Jerry Geis wrote: Actually my mistake - looks like based on my code certain things happen and I issue two dialplan reload commands. So the second is killing the first. Then asterisk looses information. So certainly I should not be doing that - but I'm surprised asterisk lets another reload happen before the first is complete. What version of Asterisk are you running? There was an issue found in February where this exact behavior could occur, two dialplan reload commands would clobber each other. It was also resolved back then in all supported branches (1.8, 10, and trunk). http://lists.digium.com/pipermail/asterisk-commits/2012-February/053537.html For the 1.8 fix if you are curious. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan reloading
What version of Asterisk are you running? There was an issue found in February where this exact behavior could occur, two dialplan reload commands would clobber each other. It was also resolved back then in all supported branches (1.8, 10, and trunk). http://lists.digium.com/pipermail/asterisk-commits/2012-February/053537.html For the 1.8 fix if you are curious. Josh, I am running 1.4.43, planning on switching to 11 but have not got there... Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different codec for different type of calls
Qasim,
Thank you for your response. I tried it but still doesn't work. This is
what I have:
exten = _XXX.,1,NoOP(Set G711 codec)
exten = _XXX.,n,Set(SIP_CODEC=ulaw)
exten = _XXX.,n,Set(SIP_CODEC_OUTBOUND=ulaw)
exten = _XXX.,n,Dial(DAHDI/g1/$EXTEN)
Then I get this error:
WARNING[12156]: channel.c:5796 ast_request: No translator path exists for
channel type DAHDI (native (ulaw|alaw|slin)) to (h264|silk8)
WARNING[12156]: app_dial.c:2277 dial_exec_full: Unable to create channel of
type 'DAHDI' (cause 58 - Bearer capability not available)
I tried both g711 and ulaw with no luck. I read the SIP_CODEC value and it
is set properly.
Any suggestions/ideas?
Thanks,
Ali Pey
On Thu, Nov 1, 2012 at 12:02 PM, qasimak...@gmail.com
qasimak...@gmail.comwrote:
exten = _X.,1,NoOP(G711 CoDec)
exten = _X.,n,Set(SIP_CODEC=g711)
exten = _X.,n,Dial(...)
*${SIP_CODEC}*: Set the SIP codec for the inbound (=first) call leg (see
channelvariables.txt or README.variables in 1.2); Asterisk 1.6.2 also comes
with SIP_CODEC_OUTBOUND https://issues.asterisk.org/view.php?id=13243for
the remote (=second) call leg.
Hope this helps,
Regards,
Qasim
On Thu, Nov 1, 2012 at 8:49 PM, Ali Pey ali...@gmail.com wrote:
Hello,
Let's say I have a sip client that supports both G711 and G729 codecs and
I have them both enabled in sip.conf and G729 has higher priority.
Can I force the call to choose a different codec based on the dialed
number or other conditions?
For instance I would want to do G711 if the call was routed to T1 card
over Dahdi but G729 if the call was going to another sip client.
Thanks,
Ali Pey
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Re: [asterisk-users] Different codec for different type of calls
SIP_CODEC is only useable on a SIP channel. You can specify DAHDI codecs in
users.conf.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ali Pey
Sent: Friday, November 02, 2012 12:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Different codec for different type of calls
Qasim,
Thank you for your response. I tried it but still doesn't work. This is what
I have:
exten = _XXX.,1,NoOP(Set G711 codec)
exten = _XXX.,n,Set(SIP_CODEC=ulaw)
exten = _XXX.,n,Set(SIP_CODEC_OUTBOUND=ulaw)
exten = _XXX.,n,Dial(DAHDI/g1/$EXTEN)
Then I get this error:
WARNING[12156]: channel.c:5796 ast_request: No translator path exists for
channel type DAHDI (native (ulaw|alaw|slin)) to (h264|silk8)
WARNING[12156]: app_dial.c:2277 dial_exec_full: Unable to create channel of
type 'DAHDI' (cause 58 - Bearer capability not available)
I tried both g711 and ulaw with no luck. I read the SIP_CODEC value and it
is set properly.
Any suggestions/ideas?
Thanks,
Ali Pey
On Thu, Nov 1, 2012 at 12:02 PM, qasimak...@gmail.com qasimak...@gmail.com
wrote:
exten = _X.,1,NoOP(G711 CoDec)
exten = _X.,n,Set(SIP_CODEC=g711)
exten = _X.,n,Dial(...)
${SIP_CODEC}: Set the SIP codec for the inbound (=first) call leg (see
channelvariables.txt or README.variables in 1.2); Asterisk 1.6.2 also comes
with SIP_CODEC_OUTBOUND https://issues.asterisk.org/view.php?id=13243 for
the remote (=second) call leg.
Hope this helps,
Regards,
Qasim
On Thu, Nov 1, 2012 at 8:49 PM, Ali Pey ali...@gmail.com wrote:
Hello,
Let's say I have a sip client that supports both G711 and G729 codecs and I
have them both enabled in sip.conf and G729 has higher priority.
Can I force the call to choose a different codec based on the dialed number
or other conditions?
For instance I would want to do G711 if the call was routed to T1 card over
Dahdi but G729 if the call was going to another sip client.
Thanks,
Ali Pey
--
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[asterisk-users] Outgoing Google Motif Calls connect but continue ringing on outgoing side
I upgraded from Asterisk 10 to 11 and switched from gtalk.conf and jabber.conf to use motif.conf and xmpp.conf. I disabled gtalk and jabber from loading in modules.conf noload = res_jabber.so noload = chan_gtalk.so After copying my settings to the new conf files and restarting Asterisk I had no errors, but making outgoing calls from clients just kept ringing even though the other side picks up and hears nothing. I played with my settings for days and have no idea what I changed that got it working so I'm hoping someone can tell me what caused this and maybe what I changed that fixed it. Now it works but I don't know why so I'd like some feedback. My Asterisk Server is NOT behind a NAT but my Clients are and I'm using Google Voice for incoming and outgoing calls. Here is what I have done. I completely removed my [general] section from motif.conf and added a [default](!) and transport=google-v1 like the example states. The [general] section was needed in gtalk.conf to get it working but seems to not be of any use now. [general] ;context=incoming;;Context to dump call into ;bindaddr=0.0.0.0 ;;Address to bind to ;bindaddr=76.12.113.228 ;externip=76.12.113.228 ;disallow=all ;allow=ulaw ;allowguest=yes ;;Allow calls from people not in peer list [default](!) disallow=all allow=alaw allow=ulaw allow=h264 transport=google-v1 ;Using google or google-v1 didn't make a difference context=incoming [asterisk](default) connection=asterisk I removed the /Talk suffix from my xmpp.conf username fields and changed timeout=5. It took me a while to notice the /Talk was not needed anymore. [asterisk] type=client ;;Client or Component connection serverhost=talk.google.com ;;Route to server for example, talk.google.com username=aster...@gmail.com;;Username with optional resource. secret=secret ;;Password priority=1 ;;Resource priority port=5222 ;;Port to use defaults to 5222 usetls=yes ;;Use tls or not usesasl=yes ;;Use sasl or not status=available;;One of: chat, available, away, xaway, or dnd statusmessage=Asterisk Server ;;Have custom status message for Asterisk. timeout=5 I changed my sip settings for my google clients to: [asterisk] host=dynamic type=friend nat=force_rport,comedia canrevinvite=no qualify=yes dtmfmode=rfc2833 context=home disallow=all allow=ulaw;h263 Can someone tell me if these settings are correct? I have no idea but it works now. I also made sure port 5060 and 5222 was open in iptables I also had to change rtp.conf to add icesupport=yes. I use my own rtp port range that is opened on the firewall. [general] icesupport=yes rtpstart=15000 rtpend=2 ;rtpchecksums=no ;dtmftimeout=3000 ;rtcpinterval = 5000 ; Milliseconds between rtcp reports ; strictrtp=yes I also had to add icesupport=no in sip.conf[general]section to stop failed to extend errors happening for SIP calls. -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing Google Motif Calls connect but continue ringing on outgoing side
Roy Abshire wrote: I upgraded from Asterisk 10 to 11 and switched from gtalk.conf and jabber.conf to use motif.conf and xmpp.conf. I disabled gtalk and jabber from loading in modules.conf noload = res_jabber.so noload = chan_gtalk.so After copying my settings to the new conf files and restarting Asterisk I had no errors, but making outgoing calls from clients just kept ringing even though the other side picks up and hears nothing. Did you follow the guide at https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google or just move the configuration files over and tweak them? I played with my settings for days and have no idea what I changed that got it working so I'm hoping someone can tell me what caused this and maybe what I changed that fixed it. Now it works but I don't know why so I'd like some feedback. So, you changed lots of settings and then it started working or did you give up after a failed call, come back, and it started working? If it started working without any changes in between it could have been a temporary problem with the Google Voice gateway you were being connected to. I've seen this a few times during testing. My Asterisk Server is NOT behind a NAT but my Clients are and I'm using Google Voice for incoming and outgoing calls. Here is what I have done. I completely removed my [general] section from motif.conf and added a [default](!) and transport=google-v1 like the example states. The [general] section was needed in gtalk.conf to get it working but seems to not be of any use now. snip Can someone tell me if these settings are correct? I have no idea but it works now. Your settings seem fine. I also made sure port 5060 and 5222 was open in iptables I also had to change rtp.conf to add icesupport=yes. I use my own rtp port range that is opened on the firewall. Yes, this is indeed a requirement. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing Google Motif Calls connect but continue ringing on outgoing side
Roy Abshire wrote: I copied my settings over, and looked at the guide over and over to change the settings. But what the guide doesn't tell you is what you don't need anymore. So I didn't know if /Talk was ok or needed to be omitted and externip or bindaddr was ok still because I had to have it for gtalk. I also don't know if transport=google or google-v1 is required and tried both. The guide isn't written for migrations, it's for configuring from scratch. That's why. Treating it as a migration document may have caused stuff to go wonky. I kept tweaking sip, motif, and xmpp settings until it started working. I also tried reload at the console and finally started trying asterisk restart so I have no idea what helped. Okay, so it sounds like something you did solved it. Without recreating the exact scenario and going through nothing stands out immediately. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing Google Motif Calls connect but continue ringing on outgoing side
Roy Abshire wrote: I do have one thing I'm really unsure about. I'm using my Google Account for Asterisk and I'm also logged into it from my Desktop Computer. Am I not supposed to be logged into this account and strictly use it for the Asterisk Server only? Does Asterisk have a problem knowing what Google Talk Login to use? There's nothing explicit to prevent you from doing this but Google decides what client gets incoming calls, so it may go to your desktop when you don't want it to. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI got event HDLC Abort
hi folks. recently some of our customers complained about bad voice quality on the phone system. i looked at the logs and found a lot of these: [2012-11-03 08:26:38] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:45] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:54] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 i upgraded Asterisk/dahdi/libpri. tried turn on/off echo canceller etc. nothing seems to help. call the phone company to check out the line (which they said it's working fine) any idea? do i have a hardware issue here? i've check syslog there was no dahdi errors. here's my system.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 span=3,0,0,esf,b8zs bchan=49-71 dchan=72 span=4,0,0,esf,b8zs bchan=73-95 dchan=96 and here's my chan_dahdi.conf: [channels] switchtype=national pridialplan=unknown prilocaldialplan=unknown internationalprefix = 001 nationalprefix = unknownprefix = signalling=pri_cpe usecallerid=yes usecallingpres=yes echocancel=no echocancelwhenbridged=no group=1 callgroup=1 pickupgroup=1 faxdetect=incoming context=defaultspan1 channel = 1-23 -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI got event HDLC Abort
is there static on the line?? is there timing slips and crc4 errors? are they increasing throughout the day? are you getting timing slips during the day when users are using the phones and not off-peak hours? are you getting hdlc abort erros when you hear a static noises?? is the card sharing irq? is your system plugged directly into an outlet without ups? On Fri, Nov 2, 2012 at 8:40 PM, Edwin Lam edwin@officegeneral.comwrote: hi folks. recently some of our customers complained about bad voice quality on the phone system. i looked at the logs and found a lot of these: [2012-11-03 08:26:38] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:45] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:54] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 i upgraded Asterisk/dahdi/libpri. tried turn on/off echo canceller etc. nothing seems to help. call the phone company to check out the line (which they said it's working fine) any idea? do i have a hardware issue here? i've check syslog there was no dahdi errors. here's my system.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 span=3,0,0,esf,b8zs bchan=49-71 dchan=72 span=4,0,0,esf,b8zs bchan=73-95 dchan=96 and here's my chan_dahdi.conf: [channels] switchtype=national pridialplan=unknown prilocaldialplan=unknown internationalprefix = 001 nationalprefix = unknownprefix = signalling=pri_cpe usecallerid=yes usecallingpres=yes echocancel=no echocancelwhenbridged=no group=1 callgroup=1 pickupgroup=1 faxdetect=incoming context=defaultspan1 channel = 1-23 -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/**pks/lookup?op=getsearch=**0xD6506D20http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
