[asterisk-users] Vestec Tech Support
Hi Richard, Reference your query, Vestec provides free installation support for its speech recognition engine. As stated on our website at http://www.vestec.com/support_overview in order to post a tech support request, you first need to send an email to supp...@vestec.com You will then receive an automated response to open a (free-of-charge) account with us and post your query on our tech support forum. A customer service representative will then respond to your query within 24-48 hours. Please note that Vestec tech support forum is NOT visible to the public; without an account with Vestec, you cannot access it. That is why you have not been able to Google it. Moreover, once you have opened an account with Vestec and posted your tech support query, only you will be able to see the communication between yourself and Vestec tech support, thereby assuring you of privacy as well as a personalized log. Hope this helps. We are looking forward to seeing you on Vestec tech support forum. Best regards, -Kashif Kashif Kahn VP, Business Development Vestec, Inc. Waterloo, ON Canada phone: (519) 885-7615-- _ -- 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] append CID label
ok
thank you
i will try
On Sat, Jun 26, 2010 at 10:31 PM, C F wrote:
> exten => s,n,Set(CALLERID(name)=label${CALLERID(name)})
> put this before the dial command.
>
> On Sat, Jun 26, 2010 at 10:09 PM, Thomas Perron
> wrote:
>> I want a call to connect via my DID to my dialplan.
>> Then, I want to attach a label to the incoming call
>>
>> call arrives
>> starts to dive through the dial plan
>> then rings a trunk/channel via SIP (see below)
>> Question: before answering my 1212111 endpoint I want to see a
>> flag CID that correlates to the DID number that was called. And, then
>> change it to something like the characters "blue"
>> How??? please.
>>
>> exten => s,1,Answer()
>> exten => s,n,Dial(SIP/callwithus/1212111,120,A,(test))
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] append CID label
exten => s,n,Set(CALLERID(name)=label${CALLERID(name)})
put this before the dial command.
On Sat, Jun 26, 2010 at 10:09 PM, Thomas Perron wrote:
> I want a call to connect via my DID to my dialplan.
> Then, I want to attach a label to the incoming call
>
> call arrives
> starts to dive through the dial plan
> then rings a trunk/channel via SIP (see below)
> Question: before answering my 1212111 endpoint I want to see a
> flag CID that correlates to the DID number that was called. And, then
> change it to something like the characters "blue"
> How??? please.
>
> exten => s,1,Answer()
> exten => s,n,Dial(SIP/callwithus/1212111,120,A,(test))
>
> --
> _
> -- 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
>
--
_
-- 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] append CID label
I want a call to connect via my DID to my dialplan. Then, I want to attach a label to the incoming call call arrives starts to dive through the dial plan then rings a trunk/channel via SIP (see below) Question: before answering my 1212111 endpoint I want to see a flag CID that correlates to the DID number that was called. And, then change it to something like the characters "blue" How??? please. exten => s,1,Answer() exten => s,n,Dial(SIP/callwithus/1212111,120,A,(test)) -- _ -- 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] Up-to-date list of Asterisk appliances?
On Sat, 26 Jun 2010 13:35:12 -0400, Paul Belanger wrote: >Might get better results on asterisk-biz, and posting your budget price range. I'll check it out. Thanks Paul -- _ -- 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] Support from Vestec
Does it exist? Sending email to their support address appears to be a black hole. They reference a forum, but Google can't find it. I keep having problems in any grammar than has a an "o" for "zero": it breaks recognition anywhere NEAR it. For example, if I say "two o five", it gets recognized as "one o five". -- _ -- 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] Where can I find a minimal set of empty configuration files (SIP only)?
On Sat, 26 Jun 2010 21:09:54 +0300, "Eyal Goltzman" wrote: >After installing and learning Asterisk I found myself with a need for a >minimal set of empty configuration files with only the "must have" stuff in >order to setup a SIP only machine, is there a place to find it? The experts might chime in, but I would say: /etc/asterisk/asterisk.conf /etc/asterisk/sip.conf /etc/asterisk/extensions.conf /etc/asterisk/codecs.conf /etc/asterisk/rpt.conf If using a PCI card /etc/dahdi/* -- _ -- 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] [voice mail] Estimating file size?
On Sat, Jun 26, 2010 at 05:53:27PM +0100, Gordon Henderson wrote: > On Sat, 26 Jun 2010, Gilles wrote: > > > Hello > > > > To run Asterisk on an embedded appliance, ie. where RAM and > > non-volatile memory is an issue (respectively 64MB and 256MB), I need > > to check how much space voice messages take to save and play back. > > > > Is there a document that shows the different options, with/without > > compression, so I can make an informed choice? > However, you can trivially test it for this and other codecs: Or, if you're too lazy to try that way, grab the moh files from http://downloads.asterisk.org/pub/telephony/sounds/ . They are included in various formats. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Codec negotiation
I have Polycom phones that support the g722 codec. Adding allow=g722 to the [general] section of sip.conf works great and I can make calls between the phones using g722. However Asterisk is negotiating g722 for calls going out my voip provider and transcoding these to ulaw. In sip.conf for the provider I have deny=all and allow=ulaw. This can cause potential audio degrading and wastes cpu cycles. If Asterisk knows the trunk only supports ulaw why would it offer g722 to the phone. Ryan -- _ -- 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] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)
On Wed, Jun 23, 2010 at 12:57 PM, James Lamanna wrote: > On Tue, Jun 22, 2010 at 8:57 PM, Andres wrote: >> >>> completely as well. >>> >>> Below I've posted a patch that responds with a 200 OK to these >>> keep-alive requests, and I believe >>> also solves the temporary loss of registration problem, though more >>> testing in different environments >>> for those who experience this problem would be greatly appreciated. >>> >>> The patch is against 1.4.32. >>> >>> >> A workaround we have used for a long time is to simply change the config >> on the Linksys phones to send an empty packet as a keep-alive. There is >> obviously no response from asterisk but it keeps the NAT bindings alive >> and well on every router we have tested. > > Hi Andres, > I have noticed that on Linksys phones that have a short REGISTER time, > the lack of > NAT keep alive responses can cause the phone to no longer be able to register. > That's why I've spent a lot of effort to hopefully make these > keep-alives supported. > >> >> Andres >> http://www.neuroredes.com >>> -- James > > -- James > > -- Is anybody running 1.6.2 with Linksys phones that would be willing to help test the patch on https://issues.asterisk.org/view.php?id=17379 Ryan -- _ -- 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] I look ARI (Asterisk Recording Interface)
Hello, I use cri http://www.tikalnetworks.com/voip/index.php?cid=38 Best regards On Thu, Jun 24, 2010 at 3:22 AM, Mickael Monsieur < mickael.monsi...@gmail.com> wrote: > Hello Bruce, > > This module is not reliable on FreePBX? > You know if there is a open source web-voicemail for Asterisk? > > Best regards, > Mickael. > > 2010/6/23 bruce bruce > > It's one of the bad modules that goes with FreePBX anyhow. The moment you >> go over 3000 recordings you are already in trouble. It's about time someone >> come up with a better moduel. >> >> On Wed, Jun 23, 2010 at 11:05 AM, Mickael Monsieur < >> mickael.monsi...@gmail.com> wrote: >> >>> Hello, >>> I look ARI (Asterisk Recording Interface) >>> the publisher site is closed... >>> >>> http://www.littlejohnconsulting.com/ari >>> >>> Thank you, >>> Mickael >>> >>> -- >>> _ >>> -- 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 >>> >> >> >> -- >> _ >> -- 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 >> > > > -- > _ > -- 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 > -- _ -- 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] Where can I find a minimal set of empty configuration files (SIP only)?
On Sat, Jun 26, 2010 at 2:09 PM, Eyal Goltzman wrote: > Hello, > > After installing and learning Asterisk I found myself with a need for a > minimal set of empty configuration files with only the "must have" stuff in > order to setup a SIP only machine, is there a place to find it? Depends on how you 'installed'. If source package, then... 'make samples' sip.conf It's full of comments. -- _ -- 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] Where can I find a minimal set of empty configuration files (SIP only)?
Hello, After installing and learning Asterisk I found myself with a need for a minimal set of empty configuration files with only the "must have" stuff in order to setup a SIP only machine, is there a place to find it? Thanks, Eyal -- _ -- 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 - Failed to extend from xxx to xxx
On Saturday 26 June 2010 04:28:18 Stuart Elvish wrote: > I have a problem with Asterisk 1.6.1.6 realtime (MySQL databases > hosted on a separate machine). When Asterisk is in verbose mode, it > prints messages saying "failed to extend from 512 to 664" (quite a few > lines in a block) and then the last message is mostly "failed to > extend from 512 to 663". The number of lines varies unpredictably. 1.6.1.6 is pretty old, and you should, at a minimum, upgrade to the latest 1.6.1 release, which is 1.6.1.20. There are a myriad of bugs that have been corrected in that time. That said, the 1.6.1 branch is in security mode. Even if you found a legitimate bug, it is not going to be fixed in branch, and there will be no more releases of the 1.6.1 branch, other than for security fixes. The exact error that you're looking at is a memory management issue. A dynamic buffer, which started as size 512, needed to be expanded, but the memory allocation failed in some way, and this warning message was the result. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) 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] [voice mail] Estimating file size?
On Sat, Jun 26, 2010 at 9:21 AM, Gilles wrote: > To run Asterisk on an embedded appliance, ie. where RAM and > non-volatile memory is an issue (respectively 64MB and 256MB), I need > to check how much space voice messages take to save and play back. > Doing some simple math and using the settings below. You could easy estimate the max amount of disk space one mailbox would use. voicemail.conf [general] format=gsm ; Maximum number of messages per folder. maxmsg=10 ; Maximum length of a voicemail message in seconds maxsecs=180 -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] unwanted entries created in dialplan from users.conf, how can I get rid of it?
On Sat, Jun 26, 2010 at 12:43 PM, Eyal Goltzman wrote: > I don't want those 2 first line to be there only the _1XX. How do I get rid > of it? Why it is added to the dial plan automatically? > Stop using users.conf. You will have more control using sip.conf and extension.conf. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Up-to-date list of Asterisk appliances?
On Sat, Jun 26, 2010 at 8:25 AM, Gilles wrote: > Is there an up-to-date list of Asterisk appliances, ideally broken > down by price (ie. not just entreprise stuff, but also SOHO)? > Might get better results on asterisk-biz, and posting your budget price range. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Detecting hook flash in asterisk
On Sat, Jun 26, 2010 at 7:33 AM, Deepesh D wrote: > Is it possible to do this action on hook flash? > Currently no. You would need to add logic to the channel driver. Or use DTMF to initiate the hookflash: extensions.conf [globals] DYNAMIC_FEATURES=>zapflash features.conf [applicationmap] zapflash => *0,callee,flash,() -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Truth in advertising
Honest-to-god CNAM dip: /CNAM/9162198187 : TELEMARKET SPAM (This is the actual CallerID Name as dipped from AT&T CNAM this morning) ROTFL -Karl -- _ -- 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] [voice mail] Estimating file size?
On Sat, 26 Jun 2010, Gilles wrote: > Hello > > To run Asterisk on an embedded appliance, ie. where RAM and > non-volatile memory is an issue (respectively 64MB and 256MB), I need > to check how much space voice messages take to save and play back. > > The appliance is connected to a landline in Europe (in case that makes > a difference as far as codecs are concerned). > > Is there a document that shows the different options, with/without > compression, so I can make an informed choice? As far as I'm aware, voicemail just stores the files in the format specified in the config file without any overhead.. So if storing using G711, then the length of the file in seconds is the length of the file in bytes / 8000. GSM is 13Kb/sec, or 1625 bytes/second. However, you can trivially test it for this and other codecs: Dial an extension that answers and stores to voicemail, say blah blah into it for one minute and check the resulting file size. divide it by 60 and you'll get a good estimate of the number of bytes per second of recording for your chosen format. Of-course the size of each voicemail is stored in the .txt file associated with each one, and I guess what you're probably after is how much storage space is left in seconds, so that's easy too, once you have the conversion factor - worked out as above and using whatever functions your scripting language has for working out how much disk space is left (e.g. disk_free_space() and disk_total_space() in PHP) Gordon -- _ -- 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] unwanted entries created in dialplan from users.conf, how can I get rid of it?
Hello,
When I call "dialplan reload" I can see the following lines:
== Parsing '/etc/asterisk/extensions.conf': == Found
-- Registered extension context 'default' (0x8a72410) in local table
0x8a679d0; registrar: pbx_config
-- Added extension '_1XX' priority 1 to default (0x8a72410)
.
.
.
== Parsing '/etc/asterisk/users.conf': == Found
-- Added extension '100' priority -1 to default (0xb731b4e8)
-- Added extension '100' priority 1 to default (0xb731b4e8)
-- Added extension '101' priority -1 to default (0xb731b4e8)
-- Added extension '101' priority 1 to default (0xb731b4e8)
That result in a dialplan that look like that:
[ Context 'default' created by 'pbx_config' ]
'100' => hint: SIP/100&IAX2/100 [pbx_config]
1. Dial(${HINT})[pbx_config]
'_1XX' => 1. Playback(digits/4) [pbx_config]
I don't want those 2 first line to be there only the _1XX. How do I get rid
of it? Why it is added to the dial plan automatically?
extention.conf look like this:
[default]
exten => _1XX,1,Playback(digits/4)
users.cong look like this:
[100]
username = 100
transfer = yes
mailbox = 100
call-limit = 100
type = peer
fullname = Polycom
registersip = no
host = dynamic
callgroup = 1
type = peer
context = default
cid_number = 100
hasvoicemail = no
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = no
secret = 100
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
autoprov = no
macaddress =
linenumber = 1
LINEKEYS = 1
disallow = all
allow = ulaw,gsm
Thanks,
Eyal
--
_
-- 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] [voice mail] Estimating file size?
Hello To run Asterisk on an embedded appliance, ie. where RAM and non-volatile memory is an issue (respectively 64MB and 256MB), I need to check how much space voice messages take to save and play back. The appliance is connected to a landline in Europe (in case that makes a difference as far as codecs are concerned). Is there a document that shows the different options, with/without compression, so I can make an informed choice? Thank you. -- _ -- 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] Up-to-date list of Asterisk appliances?
Hello Googling for this type of non-PC hardware returns products that could be missing in action for years. www.google.com/search?q=asterisk+appliance Is there an up-to-date list of Asterisk appliances, ideally broken down by price (ie. not just entreprise stuff, but also SOHO)? Thank you. -- _ -- 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] Big time system
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stuart Elvish Sent: Saturday, June 26, 2010 6:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Big time system Hi All, I am not sure if my comments will be helpful but here goes. Let me know if you need anymore pointers. Also happy to consult but you would need to contact me off list for that... Stuart Elvish === Thanks, your info is most helpful, as is the other info I have received, some of it in private messages. Your snapshot description of a ~4000 user system and architecture is a good starting point for our planning. Cary Fitch -- _ -- 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 -- _ -- 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] [CRON] Right way to restart Asterisk and Zaptel?
On Fri, 25 Jun 2010 08:59:32 +0200, Gilles wrote: >Before I go ahead, I'd like to know if I can just send the following >commands, or if there are issues I should know about: To avoid issues about the host hanging after a reboot due to upgrades... I think I'll just run a CRON job to stop/start Zaptel and see if it solves the issue I'm having about Asterisk no longer answering calls after X weeks of running fine. Can someone confirm that Zaptel can't be unloaded with Asterisk is still running (this is on a FreeBSD 6.3 host)? = # /usr/local/etc/rc.d/asterisk stop Stopping asterisk. kill: 735: No such process # ps aux | grep asterisk | grep -v grep # /usr/local/etc/rc.d/zaptel stop zaptel # kldstat Id Refs AddressSize Name 17 0xc040 7a05b0 kernel 21 0xc0ba1000 5c304acpi.ko 121 0xc2d69000 19000linux.ko # /usr/local/etc/rc.d/zaptel start zaptel Keyword: [fxsks], Value: [1] Keyword: [loadzone], Value: [fr] Keyword: [defaultzone], Value: [fr] # kldstat Id Refs AddressSize Name 1 17 0xc040 7a05b0 kernel 21 0xc0ba1000 5c304acpi.ko 121 0xc2d69000 19000linux.ko 139 0xc2ca7000 32000zaptel.ko 141 0xc2cdd000 7000 qozap.ko 151 0xc2ce7000 2tau32pci.ko 161 0xc2d09000 5000 wcfxo.ko 171 0xc2d0f000 a000 wcfxs.ko 181 0xc2d1f000 6000 wct1xxp.ko 191 0xc2d25000 c000 wct4xxp.ko 201 0xc2d31000 a000 wcte11xp.ko 211 0xc2d3b000 d000 wcte12xp.ko # /usr/local/etc/rc.d/asterisk start Starting asterisk. # ps aux | grep asterisk | grep -v grep root 4555 0,0 4,3 21896 16444 ?? Ss 13:22 0:00,67 /usr/local/sbin/asterisk # /usr/local/etc/rc.d/zaptel stop zaptelkldunload: can't unload file: Device busy # /usr/local/etc/rc.d/asterisk stop Stopping asterisk. kill: 4555: No such process # /usr/local/etc/rc.d/zaptel stop zaptelkldunload: can't find file wcte12xp.ko: No such file or directory kldunload: can't find file wcte11xp.ko: No such file or directory kldunload: can't find file wct4xxp.ko: No such file or directory kldunload: can't find file wct1xxp.ko: No such file or directory kldunload: can't find file wcfxo.ko: No such file or directory kldunload: can't find file tau32pci.ko: No such file or directory kldunload: can't find file qozap.ko: No such file or directory kldunload: can't find file zaptel.ko: No such file or directory # /usr/local/etc/rc.d/zaptel start zaptel Keyword: [fxsks], Value: [1] Keyword: [loadzone], Value: [fr] Keyword: [defaultzone], Value: [fr] # /usr/local/etc/rc.d/asterisk start Starting asterisk. # ps aux | grep asterisk | grep -v grep root 4629 0,0 4,3 21896 16444 ?? Ss 13:27 0:00,65 /usr/local/sbin/asterisk = => I guess it's OK to ignore Zaptel's "kldunload: can't find file X.ko: No such file or directory", and just go ahead and stop/start Asterisk and start Zaptel. -- _ -- 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] Detecting hook flash in asterisk
Hello, Is it possible to detect a hook flash in asterisk. I want to be able to perform some functions an hook flash. I have the following entry in features.conf which executes a Macro on detecting key press '**'. [applicationmap] test => **,caller,Macro,testflash Is it possible to do this action on hook flash? -- _ -- 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] [CRON] Right way to restart Asterisk and Zaptel?
On Fri, 25 Jun 2010 12:32:03 -0400, Barry Miller wrote: >Hi Gilles. You appear to be both posting to newsgroup >gmane.comp.telephony.pbx.asterisk.user AND sending the same message >directly to the asterisk-users list. This means that we list subscribers >see two copies of all your messages: one from gmane, one from you. (They >don't show up that way on gmane because it suppresses duplicates.) Sorry guys :-/ I think I fixed it in my NNTP reader. -- _ -- 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] [CRON] Right way to restart Asterisk and Zaptel?
On Fri, 25 Jun 2010 06:45:29 -0700 (PDT), Steve Edwards wrote: >However, the "philosophy" of regularly rebooting the box ensures that you >will never find the source of the problem. I think I'll just stop/start Zaptel and see if this is the cause for Asterisk no longer taking call after X weeks. >Hey Gilles, any chance of you fixing whatever it is that you are doing >that causes you to double-post EVERYTHING? Sorry about that. I think I fixed it. It looks like a wrong setting in the NTTP reader I'm using to access the mailing list through Gmane. -- _ -- 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] [CRON] Right way to restart Asterisk and Zaptel?
On Fri, 25 Jun 2010 13:25:18 +0300, Tzafrir Cohen wrote: >Asterisk 1.4.x works with Zaptel as well. > >But yes, this means an upgrade of Asterisk, and maybe you'd like to >avoid that. Yup. I'll just stop/start Zaptel every night and see if that fixes the problem. Thank you. -- _ -- 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] Big time system
Hi All, I am not sure if my comments will be helpful but here goes. Recently completed a roll out of Asterisk to provide a FTTH (fibre to the home) based telephone exchange. It uses Asterisk real time and a MySQL database. The design is for a somewhat smaller audience (~4000 extensions) but should provide some insight into a successful larger build which is basically a server scaling exercise. The setup works like this: 2x MySQL servers, one primary and one backup with a shared IP address. 2x Asterisk "gateway" servers, one primary and one secondary, a shared IP address. 2x Asterisk "SIP" servers, one primary and one secondary with a shared IP address. The Asterisk servers are joined by IAX which allows the SIP server (only serving internal connections) to connect to ISDN or via SIP through one of the two gateway servers whilst providing redundancy and isolation of the different parts of the network. Each house has a termination box that supplies cable TV, an ATA and internet access via a single QoS managed fibre link. The equipment at the end of each fibre is proprietary but is the same as an ATA for the purposes of voice (therefore each house is presented with a two-wire "exchange" connection). Each termination box provides two copper lines. Use of MySQL for provisioning has been fantastic - it has been stable and allows us to provide third parties with access to provision every aspect of the server (voicemail, DID's, extensions for the ATA's, call type restrictions and control line level settings such as voicemail and ring delay). We have some code in the dial plan which checks these settings using ODBC integration. For reliability we have a local copy of extensions.conf for each server. In terms of system design, there have been three things that we needed to know inside out: * How to make MySQL bullet proof. In your case you are probably considering a cluster with multiple physical locations for redundancy. * What changes to the default kernel settings need to be made to facilitate large network groups and large buffers. This really depends on how many individual registrations and calls each server is going to handle. * Setup of Asterisk to work well for high availability and ease of configuration. One interesting design project I have also worked on (which may help here) is for emergency telephones on a motorway. Every telephone is fed from an alternating supply cabinet (phone 1 - cabinet 1, phone 2 - cabinet 2, phone 3 - cabinet 1, phone 4 - cabinet 2) so if one cabinet fails (due to a switch, an ATA or power supply failure) only every second phone is "taken out". There are other issues - faxing and data calls don't work well in most setups. On the FTTH project we use an override code for faxes and send them via ISDN rather than VoIP. Unfortunately one of the contractors chose VoIP as the backbone of the network rather than ISDN despite our advice to the contrary. Your choice of ATA will also be important to faxing. >> >> Asterisk may carry you a way down this road, but in the end, it's not, >> and was never designed to be a class 5 telecom switch. There are people >> working on a carrier grade implementation that may or may not be fully >> class 5, but I don't know what the status is on that. I haven't gotten >> an answer from Digium on that lately. >> I disagree with this statement for a purely voice network. A few years ago this statement would be true. With the correct hardware and engineering, Asterisk can competently handle class 5 and higher (for example class 4) switching. They key however is good design and separating servers to run core functions as the network gets larger. A large network like this means that you would most likely have specific servers running highly specialized routing rather than having one server route all sorts of different calls. >> >> You'll want some type of Multiservice Access Platform (MSAP). Zhone >> makes the MALC and their newer MXK box. Adtran has the TA-5000 shelf. >> Neither are what you'd call cheap. Both will provide T1 access, DSL, >> SDSL, VDSL, bonded, and even ethernet access to the customer over a >> variety of transport options, including copper pairs. >> There are a few other vendors that have this, CoreCess is one and I believe Motorola also have been looking at manufacturing this type of unit. (The Motorola conversation wasn't exactly specific, it was mentioned they were interested in a job which was similar in requirements to what the Zhone equipment was recommended for.) But, I think you were looking for exchange equipment. Your suggestion of using a server at each local exchange seems to be most logical. I am assuming you will be using existing copper which you can then put into a channel bank. I unfortunately don't have too many brand recommendations but this is similar in style to what I have done on some closed networks. One thing that I can't emphasize enough is to test the system thoroughly. You will need to make sure you comp
Re: [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?
On Fri, 25 Jun 2010 11:19:25 +0100, Gareth Blades wrote: >If you are going to reboot the server regularly then make sure and >system updates are set to not automatically install new kernel versions. >Otherwise if you get a kernel update and reboot zaptel/dahdi wont load >until you recompile it. Mmm... Great tip. I hadn't thought of this. Thanks -- _ -- 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] Error - Failed to extend from xxx to xxx
Hi List, I have a problem with Asterisk 1.6.1.6 realtime (MySQL databases hosted on a separate machine). When Asterisk is in verbose mode, it prints messages saying "failed to extend from 512 to 664" (quite a few lines in a block) and then the last message is mostly "failed to extend from 512 to 663". The number of lines varies unpredictably. The full message (in the logs) is: [Jun 26 04:02:30] VERBOSE[3257] utils.c: failed to extend from 512 to 664 I haven't been able to track down much information about this using web searches but it appears (based on what I have read) that perhaps it is a database connection issue. There doesn't appear to be any problem in terms of network connectivity or database load on the database server which would lead to this situation. The Asterisk server is also well resourced generally running at <10% load. The system is designed for high availability so the extensions re-register quite frequently (which is no problem as the extensions are on an internal network) and there are approximately 1,570 extensions with more being added each week. We use cached RT peers. The database is used for CDR's, sip.conf and voicemail.conf but extensions.conf is static. So with all the above information, I am leaning towards the error being related to the database connection for real time and it occurring when an extension re-registers. Any thoughts? Thanks in advance. Stuart Elvish -- _ -- 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
