Re: [asterisk-users] Opensource Speech recognition for Asterisk
On 8/21/2010 6:09 PM, Zeeshan Zakaria wrote: Then may be these big multi-billion dollar corporations should use one of them, with whom we all deal regarding various services, and who put us through these voice recognition time-wasting activity in a hope that the poor caller will eventually give up, or will wait painfully long until one of their agent will get time to attend call in person. Your experience could be different and better then most, and you have certainly complete right of your own opinion. Zeeshan A Zakaria -- www.ilovetovoip.com http://www.ilovetovoip.com On 2010-08-21 6:57 PM, Paul Belanger paul.belan...@polybeacon.com mailto:paul.belan...@polybeacon.com wrote: On Sat, Aug 21, 2010 at 6:21 PM, Zeeshan Zakaria zisha...@gmail.com mailto:zisha...@gmail.com wrote: I yet have to see ANY... I disagree, while not Open Source like the OP requested, both Nuance and Microsoft Speech Server are nothing to laugh at. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com mailto:paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com http://blog.polybeacon.com -- _ -- Bandwidth and Colocation Pr... Zeeshan, You have to figure, Speech is a complex thing. I work at a company that sells ASR system outsourcing, and from using the products, with my run of the mill accent-less American language use, I haven't seen much of a problem, compared to other systems. It is very hard to make a computer understand long and short vowel and consonant sounds as being the same work as the ones said within the parameters of their dictionaries. It is very difficult to develop these especially in languages that the developers are not fluent in. As a side note, most of the BIG multimillion dollar companies outsource their call center functionality. As for our poster, it depends on how much time you want to dedicate to a dictionary set for recognition. If you are willing to spend a bit though, Nuance, and Holly Connect are good products, as well as the mentioned (in another post) Lumenvox. ~Seann smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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 SIP realtime and realtime DB tools
All, I am contemplating moving static SIP users to SIP realtime, and I am wondering if there is a nice simple tool to be able to do this with? I am not concerned with something that would do all the work for me, just something easier to use for a decent set of changes, than pure sql or phpmyadmin changes for the users. This is also because I am going to try the same trick with my dial plan in the future, but want to start with a few phones first. Thanks, Seann smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] Error of FreePBX after installing from Yum Repository of Asterisk
Steve Edwards wrote:
On Mon, 7 Jun 2010, bruce bruce wrote:
CentOS 5.4 and asterisk does stay running after it's loaded by asterisk
-g. But the chkconfig --add asterisk doesn't work :(
What does chkconfig --list asterisk show?
The add command looks in the asterisk script for a line that looks like:
# chkconfig: 2345 98 98
This says that chkconfig should create the appropriate links in the
/etc/rc{x}.d/ hierarchy so that Asterisk will be started at runlevels 2,
3, 4, 5 with a start priority of 98 (see man chkconfig for details) and
a stop priority of 98. Since CentOS servers should be running at runlevel
3, the 2, 4, and 5 are superfluous.
If there is no such line, chkconfig will not create the appropriate links.
Also, if /etc/init.d/asterisk does not have execute privileges, it will
not be executed on startup and Asterisk will not be running as expected.
I am running centos 5.4 myself. What I have for the chkconfig, mentioned
above is:
Mon Jun 07-12:29:27-r...@eiji.tsukinokage.net:cgi-bin chkconfig --list
asterisk
asterisk0:off 1:off 2:on3:on4:on5:on6:off
You should see the same thing if it is set up correctly. If it is in
there, try service asterisk start, and verify it is still running. If it
isn't, check /var/log/asterisk/messages and that should give you an idea
as to what killed it.
FOP is in an error error state is most likely due to channel's not being
found, and thus the pattern matches in the code, at the lines specified
are not matching anything and causing errors with the perl code that
runs the FOP server. This points back to the PBX's configuration. You
can look into what FOP is looking for in the op_server.cfg file and see
if your manager is allowing connections, etc, for the program to work.
My suggestion is make sure that asterisk by itself works in terms of
starting up cleaning, then verify amportal/FreePBX is configured and
working correctly, then FOP should work correctly after that.
Regards,
Seann
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Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk
On 6/7/2010 5:20 PM, bruce bruce wrote:
Thanks for the input Seann and Steve. That is insightful. I did run
chkconfig --list asterisk and following is the output:
*[r...@tel ~]# chkconfig --list asterisk*
*asterisk0:off 1:off 2:on3:on4:on5:on6:off*
In file /usr/sbin/safe_asterisk I have priority for asterisk set at 9.
*# run asterisk with this priority*
*PRIORITY=9*
/var/log/messages doesn't show anything important or related to why
asterisk not starting at startup. I think asterisk should start first
and then amportal will start as well is asterisk does start.
Here is what happens if I do amportal restart:
*[r...@tel ~]# amportal restart*
*
*
*STOPPING ASTERISK*
*Unable to connect to remote asterisk (does
/var/run/asterisk/asterisk.ctl exist?)*
*Asterisk Stopped*
*
*
*STOPPING FOP SERVER*
*SETTING FILE PERMISSIONS*
*chown: cannot access `/dev/tty9': No such file or directory*
*Permissions OK*
*
*
*STARTING ASTERISK*
*Cannot find specified TTY (9)*
*safe_asterisk: no process killed*
*mpg123: no process killed*
*
*
*-*
*Asterisk could not start!*
*Use 'tail /var/log/asterisk/full' to find out why.*
*-*
*[r...@tel ~]#*
*[r...@tel ~]#*
*[r...@tel ~]# asterisk -g*
*[r...@tel ~]# amportal start*
*
*
*
*
*SETTING FILE PERMISSIONS*
*chown: cannot access `/dev/tty9': No such file or directory*
*Permissions OK*
*
*
*STARTING ASTERISK*
*Asterisk is already running*
*
*
*STARTING FOP SERVER*
*FOP Server Started*
I did a tail and here it is:
*[r...@tel ~]# tail /var/log/asterisk/full*
*[Jun 7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'userbase' (on reload) at line 23.*
*[Jun 7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'vmsecret' (on reload) at line 31.*
*[Jun 7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'hassip' (on reload) at line 35.*
*[Jun 7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'hasiax' (on reload) at line 39.*
*[Jun 7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'hasmanager' (on reload) at line 47.*
*[Jun 7 22:17:36] NOTICE[4384] chan_skinny.c: Configuring skinny from
skinny.conf*
*[Jun 7 22:17:36] WARNING[4384] res_musiconhold.c: Cannot open dir
/var/lib/asterisk/moh or dir does not exist*
*[Jun 7 22:17:36] WARNING[4384] res_musiconhold.c: Cannot open dir
/var/lib/asterisk/moh/.nomusic_reserved or dir does not exist*
*[Jun 7 22:17:36] WARNING[4384] res_musiconhold.c: No music on hold
classes configured, disabling music on hold.*
*[Jun 7 22:18:40] ERROR[4479] pbx.c: Did not remove this priority
label (57/vmxopts) from the peer_label_table of context macro-vm,
extension vmx!*
Thanks,
Bruce
On Mon, Jun 7, 2010 at 3:29 PM, Steve Edwards asterisk.org
http://asterisk.org@sedwards.com http://sedwards.com wrote:
On Mon, 7 Jun 2010, bruce bruce wrote:
CentOS 5.4 and asterisk does stay running after it's loaded by
asterisk
-g. But the chkconfig --add asterisk doesn't work :(
What does chkconfig --list asterisk show?
The add command looks in the asterisk script for a line that
looks like:
# chkconfig: 2345 98 98
This says that chkconfig should create the appropriate links in the
/etc/rc{x}.d/ hierarchy so that Asterisk will be started at
runlevels 2,
3, 4, 5 with a start priority of 98 (see man chkconfig for
details) and
a stop priority of 98. Since CentOS servers should be running at
runlevel
3, the 2, 4, and 5 are superfluous.
If there is no such line, chkconfig will not create the
appropriate links.
Also, if /etc/init.d/asterisk does not have execute privileges, it
will
not be executed on startup and Asterisk will not be running as
expected.
--
Thanks in advance,
-
First, I would create the directories that it is missing, and view your
tty's in /dev (ls -Al /dev | grep tty) and validate it is there, and
what permissions it has. Mine, default install, has:
crw-rw 1 root tty 4, 9 Jun 7 17:24 tty9
~Seann
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Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk
The op_server.pl is part of the Flash Operators Panel, which isn't really important to the operation of the PBX, it is just a nice pretty interface showing what lines and what groups are active. What O/S are you using? Are there any errors in the asterisk logs? Does asterisk stay running after it starts? ~Seann On 6/6/2010 5:00 PM, bruce bruce wrote: Reboot like 10 times and the problem still presists. Also, upon reboot despite having done chkconfig --add asterisk asterisk still doesn't load automatically. And amportal start fails. So, I have to do asterisk -g first and then amportal start. Wondering if that might be related? Thanks for the input. On Sun, Jun 6, 2010 at 4:47 PM, dotnetdub dotnet...@gmail.com mailto:dotnet...@gmail.com wrote: On 6 June 2010 19:48, bruce bruce bruceb...@gmail.com mailto:bruceb...@gmail.com wrote: Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI Use of uninitialized value in hash element at /var/www/html/panel/op_server.pl http://op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl http://op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at /var/www/html/panel/op_server.pl http://op_server.pl line 3374. What could be causing that? I searched google and no useful information. Thanks, Bruce Reboot and should go away -- _ -- 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 smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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 Sip Proxies and SIP persistence
All, I am looking into open source idea's for something I play with on the closed source side. What I am thinking is to get two Asterisk PBX's behind a single SIP proxy to load balance calls inbound, and potentially outbound to an external sip provider, with the potential of multiple provider type lines (SPA3102, and a sip provider) that allows the call to persist. What I am looking at is something like what I do at work. Having an F5 with a SIP VIP configured, with persistence set up to follow call-id, or from, or to, or what ever ends up being best for my environment (Typically call-id) between 2 to 30 sip servers/engines. Since this is more dev/research as a way of saving 12,000-38,000 on a device for testing and learning more in depth on sip in transit in a home lab environment, I am after something that can do something like that, though not expecting the performance of an actual F5, or anything like that. Thanks in advance, Seann Clark smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] AGI and Severe Weather Alerts
All, I am toying with an idea of using an AGI to be able to 'call' my phone, or phones, in case of severe weather warnings. I have been tinkering with a script that reads from weather underground for the forecast, based off a PHP version of a weather AGI I found on the net. It seems rather trivial to have the AGI as a script, that does nothing unless a condition is met, and then call out, with a TTS synthesized read out of the warning, or error seen. I would like to know if anyone has done this before and what they used to get the warning for their area's. I haven't a very clear idea of how to parse properly XML data in either python or perl, but I have templates of what did work (until formats changed, StormSiren being a python module I used for sms). Also if I ever get anything to work, and anyone is interested I can share my code. Regards, Seann Clark smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] Strange Error -- ASterisk 1.6
All, I just noticed this in my logs, and am rather lost as to what module it pertains to. I would assume pseudo-realtime priority for the process, but I am looking for a little confirmation from the group: [Apr 28 12:28:36] WARNING[20773] asterisk.c: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority) Thanks, Seann smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] Strange Error -- ASterisk 1.6
Danny Nicholas wrote: We've been here, done this; This is a 1.6 NEW and Specific message to tell you that Asterisk can't start it's canary-monitor thread and that under certain conditions, you might be about to lock up. Look through the earlier posts in April. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Seann Clark Sent: Wednesday, April 28, 2010 2:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Strange Error -- ASterisk 1.6 All, I just noticed this in my logs, and am rather lost as to what module it pertains to. I would assume pseudo-realtime priority for the process, but I am looking for a little confirmation from the group: [Apr 28 12:28:36] WARNING[20773] asterisk.c: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority) Thanks, Seann Danny, Thanks for that response, it gave me just enough to confirm my idea. I can't find the stuff in the earlier threads (yet) but as i have a lot to shuffle through, and see what else I can find from it. Once again, thank you. Regards, Seann smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] Broadvoice inbound fails on Asterisk 1.6.1
All, I have been fighting with my dialplan for hours now, and google searches talk lots but offer nothing in terms of explication for this. I have my SIP peer set up and working with Broadvoice: [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=551234 secret=password defaultuser=551234 insecure=port,invite context=broadvoice authname=551234 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no monitor=yes qualify=yes disallow=all allow=ulaw nat=yes register = 551...@sip.broadvoice.com:password:551...@sip.broadvoice.com in extensions.conf: [broadvoice] exten = 551234,1,Set(CDR(accountcode)=44) exten = 551234,n,AppendCDRUserField(BroadVoice) exten = 551234,n,7090093,1,Goto(112,1) and Asterisk is still giving me this error in the logs (while playing a number does not exist sound clip): [Apr 27 18:11:19] NOTICE[12179] chan_sip.c: Call from '551234' to extension '551234' rejected because extension not found. I have played with the register settings, I have played with the sip context settings, I have tried an 's' extension in the broadvoice context, and I am out of ideas. Does anyone have an idea of what is going on with this? Thanks, Seann smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] Broadvoice inbound fails on Asterisk 1.6.1
The hidden number is no different from what I posted. This is inbound, I pick up my cell phone, dial 551234, which then hits my * box, which then the * box barfs that error. On 4/27/2010 8:35 PM, Peder wrote: Is this an inbound call to that number? Or are you calling out from that number? I understand the need to obfuscate the numbers, but it says Call from '551234' to extension '551234', so are you calling yourself? Or did you just change both numbers to the same number. Maybe just change the first 6 digits, so we can read it easier. And more debug info would help. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Seann Clark Sent: Tuesday, April 27, 2010 7:15 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Broadvoice inbound fails on Asterisk 1.6.1 All, I have been fighting with my dialplan for hours now, and google searches talk lots but offer nothing in terms of explication for this. I have my SIP peer set up and working with Broadvoice: [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=551234 secret=password defaultuser=551234 insecure=port,invite context=broadvoice authname=551234 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no monitor=yes qualify=yes disallow=all allow=ulaw nat=yes register = 551...@sip.broadvoice.com:password:551...@sip.broadvoice.com in extensions.conf: [broadvoice] exten = 551234,1,Set(CDR(accountcode)=44) exten = 551234,n,AppendCDRUserField(BroadVoice) exten = 551234,n,7090093,1,Goto(112,1) and Asterisk is still giving me this error in the logs (while playing a number does not exist sound clip): [Apr 27 18:11:19] NOTICE[12179] chan_sip.c: Call from '551234' to extension '551234' rejected because extension not found. I have played with the register settings, I have played with the sip context settings, I have tried an 's' extension in the broadvoice context, and I am out of ideas. Does anyone have an idea of what is going on with this? Thanks, Seann smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] Broadvoice inbound fails on Asterisk 1.6.1
On 4/27/2010 8:07 PM, Richard Kenner wrote:
[sip.broadvoice.com]
...
[broadvoice]
exten = 551234,1,Set(CDR(accountcode)=44)
and Asterisk is still giving me this error in the logs (while playing a
number does not exist sound clip):
[Apr 27 18:11:19] NOTICE[12179] chan_sip.c: Call from '551234' to
extension '551234' rejected because extension not found.
I have played with the register settings, I have played with the sip
context settings, I have tried an 's' extension in the broadvoice
context, and I am out of ideas. Does anyone have an idea of what is
going on with this?
The register is irrelevant for incoming calls and an 's' extension
won't get reached in this situation. MOST LIKELY what's happening is
that the SIP call isn't maching the security parameters in
[sip.broadvoice.com] and thus being put into the default context.
To test this theory, add
exten = _X.,1,NoOp(${EXTEN})
in both the default and broadvoice contexts and see which one gets hit
and what the extension is when you make the incoming call.
If it's going to default, then turn SIP debugging on and then make
another call and see if the parameters in the INVITE match what you
expect in the [sip.broadvoice.com] clause.
Did that, didn't get very far with the dialplan route. Checked the
invite settings and realized, duh on me, that the domain wasn't matching
up. It was passing my public IP address, and * was looking for the
asterisk box IP. Changed that setting and tested and it works.
Thanks for the idea's and helping rattle a bit more sense into what I
was doing.
~Seann
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Re: [asterisk-users] hardware clock drift and CDR
On 4/26/2010 7:33 AM, Vieri wrote: --- On Sun, 4/25/10, Gordon Hendersongordon+aster...@drogon.net wrote: Hi, I've noticed that one of my new servers (new mobo) if drifting slowly backwards in time (in aprox. 24 hours, system time drifts back 5 minutes). I have an ntpd process which is supposed to sync with a lan time server but it's not quite working. So I'm launching a manual ntpdate or ntp-client once an hour and that seems to work. If you can run ntpdate and it sets the time, then you are not running ntpd. The 2 can not run at the same time. Hi Gordon, Are you sure about this? ntpd is a daemon and adjusts the time in a continuous manner. ntp-client or ntpdate or whatever are one-time clients that reset the system clock. I don't see why an ntp-client can't be run while ntpd is working (it shouldn't be necessary but may come in handy when the time difference is big and ntpd refuses to sync). Anyway, I've noticed that my ntpd log messages don't say anything when trying to sync to my Windows PDC LAN time server. Curiously, ntp-client DOES sync to this Windows server. So I decided to sync to pool.ntp.org and now I see syslog messages that actually show that the system time gets adjusted by ntpd. I'd rather sync to my LAN time server but this is off-topic on this ML. How does Asterisk CDR count the duration/billsec values? Does it rely on system time ONLY for call start or also for call end? What Asterisk-related side-effects should I expect from a drifting clock? Who cares. Just fix ntpd then your worys are gone. Well, I still have doubts about that. I could look at * source code but I'd rather hear from someone here. My ntp log shows this: 26 Apr 13:06:30 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:21:24 ntpd[534]: time reset +2.318647 s 26 Apr 13:21:44 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:37:46 ntpd[534]: time reset +2.325417 s 26 Apr 13:38:06 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:54:11 ntpd[534]: time reset +2.327974 s 26 Apr 13:55:19 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:09:16 ntpd[534]: time reset +2.177572 s 26 Apr 14:10:08 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:26:07 ntpd[534]: time reset +2.357017 s That kind of scares me because if I'm not mistaken it means that about every 20 seconds, my ntpd adjusts the system time by about 2 seconds forward. So my clock is going back 2 seconds every 20... That's a significant drift. And it would definitely make a difference in my CDR records IF Asterisk were to compare the start and end system times. Should I worry about this? Vieri If it is NTP that you are worried about, you can see what your servers look like by doing an ntpq -p which should show you the clocks in the pool, which ones it is using etc. Example: remote refid st t when poll reach delay offset jitter == *clock.trit.net 192.12.19.20 2 u 385 512 377 50.2203.094 0.558 +blue.nonexiste. 91.189.94.4 3 u 339 512 377 49.154 -16.663 4.596 +216.45.57.38216.218.254.202 2 u 155 512 377 50.2381.419 0.481 With my system synchronized to clock.trit.net. That is off my master clock, and everything else is synced to it by +/- 1 second. To fix this the easiest way, that I have seen at least, stop ntpd, do an ntpdate to your primary chosen clock (ntpdate clock.trit.net in my example) and restart ntpd and verify that your clock is sync'ed accurately. Also verify that it isn't hitting your hardware dummy clock in ntpd.conf, and if it is, and you can't force it out, you can remove it temporarily. Your CDR's will be screwy in terms of timestamps based on the system time constantly changing, as well as your log files being slightly off, and if you are doing anything remote to another box in terms of logging or database, it will be even more screwy. ~Seann smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] Linksys/Sipura SPA-3201 FXO/FSA with Asterisk
All,
I am looking at a little support on this, as I haven't found it on
google yet. I have had this work on Callweaver, but am moving to
Asterisk for a variety of reasons. My dial plans, and everything else
transferred perfectly, though I am not sure they are 'correct' for
Asterisk 1.6.1, with simple things like SIP users outlined in the
sip.conf file, not in the users file, and my dialplan syntaxes don't
appear to be liked by the asterisk-gui program (not a big deal, was just
something shiny to look at for me, to try to figure out a way to get
this going).
What my problem is with Asterisk is my SPA-3201 is my primary voice
gateway, as I do not own any Digium hardware, and currently do not have
a SIP provider outside of my PBX at home. When I restart Asterisk,
everything works perfectly. I let Asterisk sit for an hour or so, and it
stops allowing calls to be routed into the assigned extension. I do see
stuff from the communications, at the time the call lands on the
Asterisk server:
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
The logic is that the SPA is registered as an extension on my system,
and incoming calls are routed into the system VIA that extension. The
dialplan that the SPA connects to is:
[gw8028]
exten = 8028,1,Answer
exten = 8028,n,Set(CallerNum=${CALLERID(num)})
exten = 8028,n,Set(CallerName=${CALLERID(name)})
exten = 8028,n,Set(CDR(accountcode)=8203)
exten = 8028,n,Set(CDR(UserField)=POTS)
exten = 8028,n,Goto(from-internal,111,1)
exten = 8028,n,Hangup
the 'from-internal' is my current call filtering/processing subsystem.
The outbound side of this works just fine though, as well as my ATA's
and Cisco 7960's are able to make and receive calls when this is
happening. I can include any additional details if requested, as I don't
know exactly what would be helpful to others with this. The SPA itself
hasn't been changed in seven months, and is stable with Callweaver.
Thanks in Advance,
Seann Clark
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Re: [asterisk-users] Linksys/Sipura SPA-3201 FXO/FSA with Asterisk
Yes, the SPA-3201 is set as: (S0:8028) on dialplan 8, which is what I
have the device set to use. My bare bones working dialplan from
Callweaver works nearly perfectly with Asterisk, and takes all the calls
and works just as it did in Callweaver (making adjustments for the
differences in dialplan syntaxes as Callweaver still uses Asterisk 1.2
syntax). It is just after an hour I can't get calls inbound to Asterisk.
If I stop Asterisk, and start Callweaver, it can sit for months and
handle calls no problem, with a like dialplan. SIP users and settings
aren't changed between the systems either, and my Cisco phones, and the
other Linksys ATA I have plays well. I am a little stumped on that. I
will include a SIP dump when I get that back up in test mode (Since it
is my home telephone system and I need it for work, which I am doing
right now, I can't afford the downtime right this moment, but tomorrow I
should have time for this).
Thanks in advance,
Seann Clark
On 4/9/2010 12:08 AM, Jose Flores Galicia wrote:
Hi.
On the Spa 3102 is set as Dialplan s0:8028 on PSTN line tab, since
other way the incoming call will try to be routed to a non set
extension on [gw8028] context
Best Regards
Jose Flores Galicia
floj...@gmail.com mailto:floj...@gmail.com
BriefCode Code Based Training
2010/4/8 Seann Clark nombran...@tsukinokage.net
mailto:nombran...@tsukinokage.net
All,
I am looking at a little support on this, as I haven't found it
on google yet. I have had this work on Callweaver, but am moving
to Asterisk for a variety of reasons. My dial plans, and
everything else transferred perfectly, though I am not sure they
are 'correct' for Asterisk 1.6.1, with simple things like SIP
users outlined in the sip.conf file, not in the users file, and my
dialplan syntaxes don't appear to be liked by the asterisk-gui
program (not a big deal, was just something shiny to look at for
me, to try to figure out a way to get this going).
What my problem is with Asterisk is my SPA-3201 is my primary
voice gateway, as I do not own any Digium hardware, and currently
do not have a SIP provider outside of my PBX at home. When I
restart Asterisk, everything works perfectly. I let Asterisk sit
for an hour or so, and it stops allowing calls to be routed into
the assigned extension. I do see stuff from the communications, at
the time the call lands on the Asterisk server:
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
The logic is that the SPA is registered as an extension on my
system, and incoming calls are routed into the system VIA that
extension. The dialplan that the SPA connects to is:
[gw8028]
exten = 8028,1,Answer
exten = 8028,n,Set(CallerNum=${CALLERID(num)})
exten = 8028,n,Set(CallerName=${CALLERID(name)})
exten = 8028,n,Set(CDR(accountcode)=8203)
exten = 8028,n,Set(CDR(UserField)=POTS)
exten = 8028,n,Goto(from-internal,111,1)
exten = 8028,n,Hangup
the 'from-internal' is my current call filtering/processing subsystem.
The outbound side of this works just fine though, as well as my
ATA's and Cisco 7960's are able to make and receive calls when
this is happening. I can include any additional details if
requested, as I don't know exactly what would be helpful to others
with this. The SPA itself hasn't been changed in seven months, and
is stable with Callweaver.
Thanks in Advance,
Seann Clark
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_
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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
smime.p7s
Description: S/MIME Cryptographic Signature
--
_
-- 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
