Re: [asterisk-users] How to deal with error messages passed as Early Media
For calls that fail, even where early media is played, the call should terminate with a 4xx or 5xx SIP response which to a certain degree correlates to the nature of the actual failure. The SIP error code is delayed until the media playback completes, but should be no different whether or not early media is used (for the same actual failure). Early media is simply an audio stream for human consumption to explain the failure. There should be no need to attempt to recognize it, unless your ITSP is not terminating the call correctly. On Wed, Feb 3, 2016 at 8:41 AM, Olivier <oza.4...@gmail.com> wrote: > Hello, > > I'm trunking with an ITSP that, when treating an outbound to an unknown > destination, either: > - send a SIP error code (I can't be more explicit, at the moment), > - or cast a pre-recorded audio message using Early Media. > > At the same time, I'm also trunking with Contact Center solution which > doesn't support Early Media. > > > Beside asking my ITSP to treat calls consistently or ask Contact Centerto > support Early Media, is there a way to configure Asterisk to unify both > above error treaments into a single one ? > > How can I best deal with error messages passed as Early Media. > > Best regards > > -- > _ > -- 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 > -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11
What version of the ST2030 firmware are you using? On Thu, Jan 7, 2016 at 8:59 AM, Juergen Sauer <juergen.sa...@automatix.de> wrote: > Am 07.01.2016 um 10:55 schrieb Frank: > > On Wed, 2016-01-06 at 17:03 +0100, Juergen Sauer wrote: > Thx, 4answer. :) > > >> with in my sip.conf, I have got for this hardphone: > >> [...] > >> [hard1] > >> username=hard1 > >> secret=correct-and-three-times-checked-4-digit-pin > > > > In most cases, there is no need to set the "username=" option. The name > > of the device is the name within the square brackets above the > > configuration section. > > Delete the "username=hard1" and reload sip.conf. > > Should be so, agreed. But it worked quite a long time not this way. > :( > > Got now up. Why? I do not know. This Hard phone is really needing an > full expert". > > Now this piece of antique hardware it does recognize calls, which > asterisk sends. > Calling out, works, asterisk sees the device as "hard1". Calling "hard1" > shows up, "not avaible"... Same Setup on Snom 821 works perfectly. > > > mit freundlichen Grüßen > Jürgen Sauer > -- > Jürgen Sauer - automatiX GmbH, > +49-4209-4699, juergen.sa...@automatix.de > Geschäftsführer: Jürgen Sauer, > Gerichtstand: Amtsgericht Walsrode • HRB 120986 > Ust-Id: DE191468481 • St.Nr.: 36/211/08000 > GPG Public Key zur Signaturprüfung: > http://www.automatix.de/juergen_sauer_publickey.gpg > > -- > _ > -- 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 > -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] Custom PHP for Call Files
I happen to have some old crufty code in PHP that generates a call file to
trigger an AGI.
Look at function callagi() in
https://github.com/stgnet/stgagi/blob/master/stgagi.php
This works in a FreePBX environment where the Asterisk process is running
as user "asterisk". There are several other hard coded assumptions such as
paths, but the code should give you an idea how to make it work for you.
Note that Asterisk will normally delete the call file as soon as it sees it
and begins the call. There is an exception to this where the Archive flag
in the call file instructs Asterisk to move the file to another directory
and update it with the completion status.
For full details on the call file contents, see:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files
On Sun, Dec 27, 2015 at 9:14 PM, er ic <email.eherr9...@gmail.com> wrote:
> I am hoping to get some help here with building custom PHP to manage a
> 'wake up call' system.
>
> I have the script where the user can set the schedule for an extension
> wake up call.
>
> It appears to write to the /var/spool/asterisk/outgoing/ directory.
>
> My two issues:
>
> 1 - when the files do get moved over to outgoing/ directory via a cron
> job, the permissions show "-rw-r--r-- 1 apache apache 100 Jan 1 2016
> 5680a312a28b2.call" and the calls get sent when the date comes to pass. But
> my question is, if I mv 3 files from my php script, 'll
> /var/spool/asterisk/outgoing/' shows 'total 12' when there are only three
> files in the directory. What does total mean? Is my perl script doing
> something that I am not aware of and really there are 12 files overlapped
> or something funky?
>
> --- cron job perl script
> my @list = glob("/tmp/*.call");
> for( 0 .. $#list )
> {
> system "mv $list[$_] /var/spool/asterisk/outgoing/";
> }
> ---
>
> 2 - I would like to view and delete call files but as it currently stands,
> php gets a permission denied.
> obviously php is running as apache and the outgoing/ directory is
> asterisk:asterisk but the call files are apache:apache. My question is,
> what is the best way, without risking security, to allow php to list and
> delete the files? I know my scripts themselves work because when I chown
> apache:apache /var/spool/asterisk/outgoing the script works. I have seen
> front ends work with all the same permissions on outgoing/ and the files
> but I dont know how they are able to read/delete the files for monitor/
> which is the same as the outgoing/ directory.
>
> Thanks for your help in advance all!
> --Eric
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://asterisk.org
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Re: [asterisk-users] same sip username with realms and chan_sip
Just as a reminder: absolutely anytime that you succeed in crashing Asterisk (no matter the validity of your input), please make sure that either an issue covering the situation already exists, or please take the time to create a new one. When creating an issue (or if one is not already attached), please follow these [1] instructions for obtaining a backtrace and attach the file to the issue. Very often a backtrace on an issue is sufficient for us to identify and eliminate the bug that caused it. And if you can, please replicate using a currently supported version (11, 13, master) of Asterisk compiled from the latest git head -- this helps us to be confident that it's not something already fixed, and we can skip that step and get to fixing it faster. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace On Tue, Oct 13, 2015 at 5:22 AM, Ludovic Gasc <gml...@gmail.com> wrote: > pjsip crashes only with my realm experiments. > I'll test with the latest Asterisk 13 stable version to verify. > > However, even if I've found a solution for realm, I've the feeling that > realm in Asterisk isn't well tested/supported. > > For now, since September, I use a simpler solution in production: > integrate the account name as a prefix in the username: enough mainstream > to be sure is supported ;-) > > Ludovic Gasc (GMLudo) > http://www.gmludo.eu/ > On 11 Oct 2015 22:22, "Joshua Colp" <jc...@digium.com> wrote: > >> Ludovic Gasc wrote: >> >>> Hello, >>> >>> same sip username with realms is possible with Asterisk ? >>> I've tried to have this feature with Asterisk 13.3.2 and chan_pjsip, and >>> now, Asterisk crashes. >>> >> >> Did PJSIP crash in general (it's usually a build problem if that happens) >> or was it when you were experimenting with different realms and such? >> >> -- >> Joshua Colp >> Digium, Inc. | Senior Software Developer >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US >> 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 >> > > -- > _ > -- 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 > -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?
To add a header to the call leg that goes to the agent, try using a local
channel to activate dialplan on the outbound call:
Register Local/number@agent in the queue on behalf of the agent (replace
number with the agent's extension number)
In dialplan [agent], wild card match the number, add the header, and then
Dial(PJSIP/{$EXTEN}) to send the call to the agent.
On Thu, Aug 27, 2015 at 1:40 PM, Dan Cropp d...@amtelco.com wrote:
I have a call coming in.
I need to add a SIP Header to the channel.
Then, I need to send the call to the Queue so it is sent to the Agent.
The SIP header I added, I need to have appear in the INVITE sent to the
Agent.
It works in chan_sip. I send the call to a macro which does…
n,SIPAddHeader(My-Header-Name:${MY-HEADER-VALUE})
n,Queue(${ARG2})
In PJSIP , this doesn’t seem to work. Is there any way to add custom
PJSIP headers to be sent as part of the INVITE to the Agent?
When I look at the code, it seems as though the INVITE doesn’t look for
any custom headers to be included with the INVITE packet. Is this correct?
Have a great day!
Dan
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Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://asterisk.org
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_
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Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?
Are you using this method of setting headers on PJSIP?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_HEADER
On Thu, Aug 27, 2015 at 4:54 PM, Dan Cropp d...@amtelco.com wrote:
Thanks Scott.
I was able to get the basic concept to run.
However, it seems PJSIP INVITE for the Dial also does not support added
headers.
The Local channel dial plan did have the channel variable values. I added
them as SIP headers, then Dial(PJSIP/Agent).
The INVITE for the Dial on PJSIP continues to not include the SIP Headers
I added.
For chan_sip, I have no problem with this. Even the original Queue code I
had includes the added SIP headers with it’s INVITE to the Agent.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Scott Griepentrog
*Sent:* Thursday, August 27, 2015 4:28 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Is it possible to perform PJSIP Add
Header prior to calling Queue and have it part of the INVITE packet?
Local channels:
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeeperDialplan_id324598.html
This explains adding members to queues, although it doesn't specifically
provide an example of using local channels in a queue:
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289508.html
Basically, read that book, and if you get stuck ask for help.
On Thu, Aug 27, 2015 at 3:08 PM, Dan Cropp d...@amtelco.com wrote:
Thanks Scott.
I’m taking over for someone else’s code, so I must admit I’m still
learning the Agent and Queue concepts. Local channels are something I have
not used either. Would local channels essentially be an internal bridge?
How would I
“Register Local/number@agent in the queue on behalf of the agent (replace
number with the agent's extension number)”
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Scott Griepentrog
*Sent:* Thursday, August 27, 2015 1:57 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Is it possible to perform PJSIP Add
Header prior to calling Queue and have it part of the INVITE packet?
To add a header to the call leg that goes to the agent, try using a local
channel to activate dialplan on the outbound call:
Register Local/number@agent in the queue on behalf of the agent (replace
number with the agent's extension number)
In dialplan [agent], wild card match the number, add the header, and then
Dial(PJSIP/{$EXTEN}) to send the call to the agent.
On Thu, Aug 27, 2015 at 1:40 PM, Dan Cropp d...@amtelco.com wrote:
I have a call coming in.
I need to add a SIP Header to the channel.
Then, I need to send the call to the Queue so it is sent to the Agent.
The SIP header I added, I need to have appear in the INVITE sent to the
Agent.
It works in chan_sip. I send the call to a macro which does…
n,SIPAddHeader(My-Header-Name:${MY-HEADER-VALUE})
n,Queue(${ARG2})
In PJSIP , this doesn’t seem to work. Is there any way to add custom
PJSIP headers to be sent as part of the INVITE to the Agent?
When I look at the code, it seems as though the INVITE doesn’t look for
any custom headers to be included with the INVITE packet. Is this correct?
Have a great day!
Dan
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asterisk-users mailing list
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Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://asterisk.org
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://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
Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?
Local channels:
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeeperDialplan_id324598.html
This explains adding members to queues, although it doesn't specifically
provide an example of using local channels in a queue:
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289508.html
Basically, read that book, and if you get stuck ask for help.
On Thu, Aug 27, 2015 at 3:08 PM, Dan Cropp d...@amtelco.com wrote:
Thanks Scott.
I’m taking over for someone else’s code, so I must admit I’m still
learning the Agent and Queue concepts. Local channels are something I have
not used either. Would local channels essentially be an internal bridge?
How would I
“Register Local/number@agent in the queue on behalf of the agent (replace
number with the agent's extension number)”
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Scott Griepentrog
*Sent:* Thursday, August 27, 2015 1:57 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Is it possible to perform PJSIP Add
Header prior to calling Queue and have it part of the INVITE packet?
To add a header to the call leg that goes to the agent, try using a local
channel to activate dialplan on the outbound call:
Register Local/number@agent in the queue on behalf of the agent (replace
number with the agent's extension number)
In dialplan [agent], wild card match the number, add the header, and then
Dial(PJSIP/{$EXTEN}) to send the call to the agent.
On Thu, Aug 27, 2015 at 1:40 PM, Dan Cropp d...@amtelco.com wrote:
I have a call coming in.
I need to add a SIP Header to the channel.
Then, I need to send the call to the Queue so it is sent to the Agent.
The SIP header I added, I need to have appear in the INVITE sent to the
Agent.
It works in chan_sip. I send the call to a macro which does…
n,SIPAddHeader(My-Header-Name:${MY-HEADER-VALUE})
n,Queue(${ARG2})
In PJSIP , this doesn’t seem to work. Is there any way to add custom
PJSIP headers to be sent as part of the INVITE to the Agent?
When I look at the code, it seems as though the INVITE doesn’t look for
any custom headers to be included with the INVITE packet. Is this correct?
Have a great day!
Dan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://asterisk.org
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asterisk-users mailing list
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Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://asterisk.org
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Re: [asterisk-users] Windows Asterisk Help
On Wed, Jul 29, 2015 at 11:02 PM, Murthy Gandikota murth...@hotmail.com wrote: -- Date: Wed, 29 Jul 2015 11:47:19 -0500 From: sgriepent...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Windows Asterisk Help On Wed, Jul 29, 2015 at 10:16 AM, John Novack jnov...@stromberg-carlson.org wrote: Murthy Gandikota wrote: -- To: asterisk-users@lists.digium.com From: webaccounts...@jgoettgens.de Date: Wed, 29 Jul 2015 16:11:31 +0200 Subject: Re: [asterisk-users] Windows Asterisk Help Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7. Here is my sip.conf [general] context = demo ; Default context for incoming calls bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup = yes ; Enable DNS SRV lookups on outbound calls context=incoming disallow=all allow=ulaw allow=alaw allow=g729 allow=g723 externip=72.220.28.226 localnet=192.168.0.0 nat=yes maxexpiry=15 minexpiry=14 ;rtautoclear=no ;autofallthrough=yes register =16194077214:password@69.59.234.67:5060/202 [authentication] [3000] type = friend context = default username = 3000 host = dynamic mailbox = 3000 dtmfmode = rfc2833 [3001] type = friend context = default username = 3001 host = dynamic mailbox = 3001 dtmfmode = rfc2833 [3002] type = friend username = 3002 context = default host = dynamic mailbox = 3002 dtmfmode = rfc2833 [vonage-out] username=16194077214 type=friend secret=password port=5061 nat=yes host=69.59.234.67 fromuser=16194077214 fromdomain=69.59.234.67 dtmfmode=rfc2833 auth=md5 [vonage202] username=16194077214 ;type=friend type=peer ;type=user secret=password port=5061 nat=yes insecure=port,invite host=69.59.234.67 fromuser=16194077214 fromdomain=69.59.234.67 ;dtmfmode=inband context=from-pstn canreinvite=no ;auth=md5 disallow=all allow=ulaw ;allow=alaw ;allow=g729 ;allow=g723 Here is my extensions.conf [from-pstn] ;exten = 16194077214,1,verbose(0, hello) exten = 16194077214,1,Answer; exten = 16194077214,n,SayUnixTime() exten = 16194077214,n,Hangup I am able to connect with Asterisk on the first try after fresh load, but not on the subsequent tries. I have to re-reload sip.conf and extensions.conf to connect with Asterisk. Looking at the logs, it seems like a registration issue. So I set minexpirty and maxexpirty that seems to have no effect. can post the logs, if someone wants me to. Your kind help is appreciated. Best regards murthy www.asteriskwin32.com hosts only a very very old version of Asterisk (1.2.something). What speaks against setting up a small virtual machine to host a recent version of Asterisk? jg You have a point. My SIP provider at the moment is Vonage which I can't access from work (some security issue:) So I am confined to testing from home and I don't have any other machine to spare. If there is no other way to trouble-shoot the problem, I will have to do what you suggest. Thanks Regards murthy For very little $$$ you could obtain an HP thin client, load a modern version of Asterisk using AstLinux, and leave your Win 7 machine to do what it does best ( which is certainly NOT Asterisk ) Once installed, it can be completely controlled and configured remotely over your home LAN, consumes very little power, has a universal power supply, consumes little power and no noisy fans. HP5720 units can be had off eBay for $20-30 US. Even with shipping to your country, really low cost solution much more in the mainstream. AstLinux uses standard Asterisk confs. The GUI is used for management and editing, and doesn't use the difficult to troubleshoot and quirky overlays of a TrixBox or FreePBX Check out the astlinux website for more details John Novack -- Dog is my Co-pilot -- _ -- 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 Another option (assuming your computer has enough ram and disk space) is to run a copy of Linux in Vmware Player (which is available for free). It allows you to run the Linux environment in a virtual computer as if it was an application on windows. Then you can test the most recent release of Asterisk (version 13 at the moment). -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1
Re: [asterisk-users] Windows Asterisk Help
On Wed, Jul 29, 2015 at 10:16 AM, John Novack jnov...@stromberg-carlson.org wrote: Murthy Gandikota wrote: -- To: asterisk-users@lists.digium.com From: webaccounts...@jgoettgens.de Date: Wed, 29 Jul 2015 16:11:31 +0200 Subject: Re: [asterisk-users] Windows Asterisk Help Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7. Here is my sip.conf [general] context = demo ; Default context for incoming calls bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup = yes ; Enable DNS SRV lookups on outbound calls context=incoming disallow=all allow=ulaw allow=alaw allow=g729 allow=g723 externip=72.220.28.226 localnet=192.168.0.0 nat=yes maxexpiry=15 minexpiry=14 ;rtautoclear=no ;autofallthrough=yes register =16194077214:password@69.59.234.67:5060/202 [authentication] [3000] type = friend context = default username = 3000 host = dynamic mailbox = 3000 dtmfmode = rfc2833 [3001] type = friend context = default username = 3001 host = dynamic mailbox = 3001 dtmfmode = rfc2833 [3002] type = friend username = 3002 context = default host = dynamic mailbox = 3002 dtmfmode = rfc2833 [vonage-out] username=16194077214 type=friend secret=password port=5061 nat=yes host=69.59.234.67 fromuser=16194077214 fromdomain=69.59.234.67 dtmfmode=rfc2833 auth=md5 [vonage202] username=16194077214 ;type=friend type=peer ;type=user secret=password port=5061 nat=yes insecure=port,invite host=69.59.234.67 fromuser=16194077214 fromdomain=69.59.234.67 ;dtmfmode=inband context=from-pstn canreinvite=no ;auth=md5 disallow=all allow=ulaw ;allow=alaw ;allow=g729 ;allow=g723 Here is my extensions.conf [from-pstn] ;exten = 16194077214,1,verbose(0, hello) exten = 16194077214,1,Answer; exten = 16194077214,n,SayUnixTime() exten = 16194077214,n,Hangup I am able to connect with Asterisk on the first try after fresh load, but not on the subsequent tries. I have to re-reload sip.conf and extensions.conf to connect with Asterisk. Looking at the logs, it seems like a registration issue. So I set minexpirty and maxexpirty that seems to have no effect. can post the logs, if someone wants me to. Your kind help is appreciated. Best regards murthy www.asteriskwin32.com hosts only a very very old version of Asterisk (1.2.something). What speaks against setting up a small virtual machine to host a recent version of Asterisk? jg You have a point. My SIP provider at the moment is Vonage which I can't access from work (some security issue:) So I am confined to testing from home and I don't have any other machine to spare. If there is no other way to trouble-shoot the problem, I will have to do what you suggest. Thanks Regards murthy For very little $$$ you could obtain an HP thin client, load a modern version of Asterisk using AstLinux, and leave your Win 7 machine to do what it does best ( which is certainly NOT Asterisk ) Once installed, it can be completely controlled and configured remotely over your home LAN, consumes very little power, has a universal power supply, consumes little power and no noisy fans. HP5720 units can be had off eBay for $20-30 US. Even with shipping to your country, really low cost solution much more in the mainstream. AstLinux uses standard Asterisk confs. The GUI is used for management and editing, and doesn't use the difficult to troubleshoot and quirky overlays of a TrixBox or FreePBX Check out the astlinux website for more details John Novack -- Dog is my Co-pilot -- _ -- 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 Another option (assuming your computer has enough ram and disk space) is to run a copy of Linux in Vmware Player (which is available for free). It allows you to run the Linux environment in a virtual computer as if it was an application on windows. Then you can test the most recent release of Asterisk (version 13 at the moment). -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory
Re: [asterisk-users] asterisk segfault debian jessie asterisk 11.13
You'll want to follow these instructions to get a backtrace: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace And then create an issue here and attach the backtrace file: https://issues.asterisk.org This way the Asterisk team will have the best chance of being able to locate and resolve the problem, or at least advise you how to avoid it. On Tue, Jul 21, 2015 at 3:43 AM, Thomas thomasit...@gmail.com wrote: Hi, every two weeks the asterisk process has a segfault. Any idea whats reason or what I can do... thanks pc kernel: [1780743.239296] asterisk[11362]: segfault at 0 ip (null) sp 7f1e396b04a8 error 14 version is debian jessie Asterisk 11.13.1~dfsg-2+b1 built by buildd @ brahms on a x86_64 running Linux -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] Dell portability
Try turning off BUILD_NATIVE in menuselect. This will eliminate optimizations for the processor you last compiled on, which prevents crashes due to instructions not present on a different processor. This is frequently necessary when using in virtual environments. In cli form: # menuselect/menuselect --disable BUILD_NATIVE On Wed, Jul 1, 2015 at 1:36 PM, Jeff LaCoursiere j...@jeff.net wrote: Howdy, I built an LXC container with an image of asterisk 11.18 precompiled and installed. It runs fine on the dev platform, which is a Dell R320 running Ubuntu 14.04LTS. I shutdown the container, tarred it up, and untarred on a Dell PE1850, also running Ubuntu 14.04LTS. The container itself is Ubuntu 14.04LTS. Both platforms as far as I know are amd64. The container boots fine on the 1850, but trying to run asterisk segfaults. The source tree was still in the container, so I just did a make clean; make; make install. It now runs fine. Is there some compile flag I could use to make sure it is more compatible as I copy the container around? Can anyone suggest a debug sequence that would at least narrow down what is causing the fault? Cheers, j -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] chan_sip.c: Hanging up call
You mean sip set debug on ? Yes, that's correct for chan_sip. Sorry, I was vague -- there is now a different command for chan_pjsip, didn't know which you were using. On Thu, May 28, 2015 at 12:49 PM, Ethy H. Brito ethy.br...@inexo.com.br wrote: On Thu, 28 May 2015 11:15:45 -0500 Scott Griepentrog sgriepent...@digium.com wrote: The string 5a2600300339934f704528bb14ed05e9@MyAsterisk:5060 is the unique identifier for the call in SIP known as the Call-ID. If you have a packet capture of the port 5060 SIP traffic, that identifier will be in each SIP message related to the call, which also includes the full from and to details. That is the problem. Since the message occurs typically about 2~3 times a day (or even less), I will have tons of packets to sniff. But, I will give it a try. As an alternative to running a separate packet capture, you can enable SIP message logging in Asterisk, which puts the full SIP message into the same log file. You mean sip set debug on ? Be aware however that this can fill your hard drive quite rapidly, as well as put additional load on the disk storage system. I am pretty aware of that. Learn it the hard way. Cheers Ethy On Thu, May 28, 2015 at 11:03 AM, Ethy H. Brito ethy.br...@inexo.com.br wrote: Hi All I have a few lines like this at asterisk/messages. [May 25 15:22:42] WARNING[27725] chan_sip.c: Hanging up call 5a2600300339934f704528bb14ed05e9@MyAsterisk:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ). Since we have hundreds of clients with hundreds of simultaneous calls, how is it possible to know to which customer/IP those calls refer to? The above literature don't say much to help to narrow down the problem scope. Cheers Ethy -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org -- Ethy H. Brito /\ InterNexo Ltda. \ / CAMPANHA DA FITA ASCII - CONTRA MAIL HTML +55 (12) 3797-6860 X ASCII RIBBON CAMPAIGN - AGAINST HTML MAIL S.J.Campos - Brasil / \ PGP key: http://www.inexo.com.br/~ethy/0xC3F222A0.asc -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] chan_sip.c: Hanging up call
The string 5a2600300339934f704528bb14ed05e9@MyAsterisk:5060 is the unique identifier for the call in SIP known as the Call-ID. If you have a packet capture of the port 5060 SIP traffic, that identifier will be in each SIP message related to the call, which also includes the full from and to details. As an alternative to running a separate packet capture, you can enable SIP message logging in Asterisk, which puts the full SIP message into the same log file. Be aware however that this can fill your hard drive quite rapidly, as well as put additional load on the disk storage system. On Thu, May 28, 2015 at 11:03 AM, Ethy H. Brito ethy.br...@inexo.com.br wrote: Hi All I have a few lines like this at asterisk/messages. [May 25 15:22:42] WARNING[27725] chan_sip.c: Hanging up call 5a2600300339934f704528bb14ed05e9@MyAsterisk:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ). Since we have hundreds of clients with hundreds of simultaneous calls, how is it possible to know to which customer/IP those calls refer to? The above literature don't say much to help to narrow down the problem scope. Cheers Ethy -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] ARI echo test
I'm pretty sure there isn't a way to do that currently. My best guess would be that a new special type of bridge technology could be created that would implement the per-channel echo (no audio bridged between channels in the bridge). That would require new C code in Asterisk for the bridge, and then the usual methods of moving channels in to bridges with ARI could be used. On Sat, May 23, 2015 at 1:33 AM, Nick Awesome jl...@me.com wrote: recreate Echo, if that is possible. trying to recode all dialplan to stasis application On 22 May 2015, at 19:29, Scott Griepentrog sgriepent...@digium.com wrote: Nick- Are you wanting to recreate the dialplan Echo() application in stasis? Why not just send the call to Echo() instead of Stasis()? On Fri, May 22, 2015 at 11:25 AM, Matthew Jordan mjor...@digium.com wrote: On Fri, May 22, 2015 at 4:41 AM, Nick Awesome jl...@me.com wrote: Can anyone tell me how can I create echo test using ARI stasis application? I'm not sure an 'echo' test really makes much sense with ARI, but we do have some nice documentation on getting started with ARI on the wiki. The basic tutorial example should give you an ARI event over a WebSocket connection. https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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 -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] ARI echo test
Nick- Are you wanting to recreate the dialplan Echo() application in stasis? Why not just send the call to Echo() instead of Stasis()? On Fri, May 22, 2015 at 11:25 AM, Matthew Jordan mjor...@digium.com wrote: On Fri, May 22, 2015 at 4:41 AM, Nick Awesome jl...@me.com wrote: Can anyone tell me how can I create echo test using ARI stasis application? I'm not sure an 'echo' test really makes much sense with ARI, but we do have some nice documentation on getting started with ARI on the wiki. The basic tutorial example should give you an ARI event over a WebSocket connection. https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] Custom UUID in originate and AMI
As described in
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_Originate
:
In the AMI Originate request, if the channelId value is set, the new
channel originated will have that value as it's UUID or UniqueID.
On Sat, May 9, 2015 at 5:02 PM, Tiago Geada tiago.ge...@gmail.com wrote:
what do you mean by set
you can use like:
Variable: __CUSTOMID=UUID-string\r\n
to be able to read back ${CUSTOMID} back in the dialplan ... ?
On 8 May 2015 at 19:04, Mehdi Shirazi mahdi_shir...@yahoo.com wrote:
Hi
Could someone please help me how to set Custom generated UUID in
Originate action in AMI ?
Regards
Babak
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Scott Griepentrog
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Re: [asterisk-users] FXO advice
The Cisco/Linksys SPA devices are also able to be provisioned automatically. On Wed, Apr 15, 2015 at 3:20 PM, Bryant Zimmerman brya...@zktech.com wrote: Alejandro All of the Grandstream devices can be remote provisioned if you know what you are doing. Bryant -- *From*: Alejandro cdgr...@gmail.com *Sent*: Wednesday, April 15, 2015 4:17 PM *To*: asterisk-users@lists.digium.com *Subject*: [asterisk-users] FXO advice Hi All, I'll like to know if exist some Basic FXO that support some type of automatic provisioning of configuration. Our idea is avoid the users need to go into WebPage and setup our SIP gateway. Some advice or recommendation? Thanks Alejandro -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] Update peer IP address
That sounds like asterisk was working 100% correctly. If you receive an INVITE from an unknown IP address, then it should fail. Unless you want to allow anonymous, which is genearlly a very bad idea. If you are registering to IP X, but the provider may be transmitting invites from any number of other IP addresses, then you need a list of IP addresses, and have a trunk configuration set up for each one so that they are all recognized (with insecure=port,invite). If the provider is requiring you to accept invites from random IP addresses, get a new provider. On Thu, Apr 2, 2015 at 3:23 PM, Daniel Heckl daniel.he...@gmail.com wrote: Okay, Scott, I think we are on the wrong path. Maybe I'm wrong though. I will summarize again briefly the problems together: - The peer ip address could be another than the ip address of incoming invites - After an re-register the REGISTER is send to the new SIP server, answered with OK. But the peer ip address is still the old one (sip show peers). - If now is a INVITE, the request is answered with 401 Unauthorized. That’s why I would say, the problem is not the port or a needed authentication. My Asterisk works behind a NAT without port forwarding and nat=no, I have qualify=yes that it does not come to a NAT timeout. Here is an example. The peer ip address was at this time 217.0.23.100, the INVITE came from 217.0.23.68 an was rejected with 401 Unauthorized: INVITE sip:06123456789@80.000.111.222:45061 SIP/2.0 Max-Forwards: 58 Via: SIP/2.0/UDP 217.0.23.68:5060 ;branch=z9hG4bKg3Zqkv7ib7h2smv8whryjnos88srot1i7 To: sip:6123456...@telekom.de From: sip:+49123456...@tel.t-online.de;user=phone;tag=h7g4Esbg_44c62525 Call-ID: af71bbfbf269b895@62.155.0.75 CSeq: 3950540 INVITE Contact: sip:sgc_c@217.0.23.68;transport=udp Record-Route: sip:217.0.23.68;transport=udp;lr Min-Se: 900 P-Asserted-Identity: sip:+49123456...@tel.t-online.de;user=phone Session-Expires: 3600 Supported: histinfo Supported: timer Supported: norefersub Content-Type: application/sdp Content-Disposition: session Content-Length: 204 Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, UPDATE v=0 o=- 0 0 IN IP4 217.0.23.68 s=- c=IN IP4 217.0.4.134 t=0 0 m=audio 36480 RTP/AVP 9 8 102 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:102 telephone-event/8000 a=maxptime:20 a=ptime:20 Am 02.04.2015 um 22:00 schrieb Scott Griepentrog sgriepent...@digium.com : Actually, the IP address is still used to identify the incoming invite. With the insecure=port option set, Asterisk will presume the invite to still match the trunk account even if the NAT router has mangled (changed) the port number. My suspicion is that when the new register goes out, it's creating a new state in the firewall, resulting in a new port number, which is why you would have to allow anonymous calls to then accept it without insecure=port. The other possibility is that you have a port forward in the router set, which is similarly mangling the port number. With a valid registration being held, and assuming the router does not drop UDP states faster than 30 minutes, and also assuming that the provider is sending you invites on the registered port rather than always on 5060, there should not be a need for an inbound port forward to Asterisk, and you should not need insecure=port. The invite option disables authentication - which means only that Asterisk will not force a check of the password on the other end. Where the IP address is well known and trusted, the extra overhead and delay of authenticating incoming INVITEs is not needed. On Thu, Apr 2, 2015 at 2:28 PM, Daniel Heckl daniel.he...@gmail.com wrote: Scott, I have changed the configuration as said it and will test it. I’m curious. Can you briefly explain what insecure=invite,port does? ;insecure=port ; Allow matching of peer by IP address without ; matching port number ;insecure=invite ; Do not require authentication of incoming INVITEs ;insecure=port,invite ; (both) Do I understand correctly that in this mode the IP address is not checked and no authentication is required? Am 02.04.2015 um 20:11 schrieb Scott Griepentrog sgriepent...@digium.com : I'd be curious if setting insecure=invite,port makes any difference either (without alllowguest on). On Thu, Apr 2, 2015 at 9:03 AM, Daniel Heckl daniel.he...@gmail.com wrote: Ok, I have tested dnsmgr. This is not a solution, the situation has not changed. With dnsmgr I can not place outbound calls. I do not know why and what dnsmgr really do. My current solution is as follows: Say allowguest=yes, configure the default context that there can not be placed outbound calls. Use iptables to DROP all at your SIP port and allow only your local phones and the sip trunk ip range. I think srvlookup must be set to yes to place outbound calls if there is an ip address change. I think with the restriction
Re: [asterisk-users] Update peer IP address
Actually, the IP address is still used to identify the incoming invite. With the insecure=port option set, Asterisk will presume the invite to still match the trunk account even if the NAT router has mangled (changed) the port number. My suspicion is that when the new register goes out, it's creating a new state in the firewall, resulting in a new port number, which is why you would have to allow anonymous calls to then accept it without insecure=port. The other possibility is that you have a port forward in the router set, which is similarly mangling the port number. With a valid registration being held, and assuming the router does not drop UDP states faster than 30 minutes, and also assuming that the provider is sending you invites on the registered port rather than always on 5060, there should not be a need for an inbound port forward to Asterisk, and you should not need insecure=port. The invite option disables authentication - which means only that Asterisk will not force a check of the password on the other end. Where the IP address is well known and trusted, the extra overhead and delay of authenticating incoming INVITEs is not needed. On Thu, Apr 2, 2015 at 2:28 PM, Daniel Heckl daniel.he...@gmail.com wrote: Scott, I have changed the configuration as said it and will test it. I’m curious. Can you briefly explain what insecure=invite,port does? ;insecure=port ; Allow matching of peer by IP address without ; matching port number ;insecure=invite ; Do not require authentication of incoming INVITEs ;insecure=port,invite ; (both) Do I understand correctly that in this mode the IP address is not checked and no authentication is required? Am 02.04.2015 um 20:11 schrieb Scott Griepentrog sgriepent...@digium.com : I'd be curious if setting insecure=invite,port makes any difference either (without alllowguest on). On Thu, Apr 2, 2015 at 9:03 AM, Daniel Heckl daniel.he...@gmail.com wrote: Ok, I have tested dnsmgr. This is not a solution, the situation has not changed. With dnsmgr I can not place outbound calls. I do not know why and what dnsmgr really do. My current solution is as follows: Say allowguest=yes, configure the default context that there can not be placed outbound calls. Use iptables to DROP all at your SIP port and allow only your local phones and the sip trunk ip range. I think srvlookup must be set to yes to place outbound calls if there is an ip address change. I think with the restriction of the firewall that should be a secure solution. Am 01.04.2015 um 19:23 schrieb Sebastian Kemper sebastian...@gmx.net: On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote: On 4/1/15 10:48 AM, Daniel Heckl wrote: John, thank you four your answer. I think you have misunderstood the problem. It’s about a ip address change of the sip trunk, not of my asterisk server. You would probably benefit by enabling the DNS Manager to allow for dynamic IP changes: # cat dnsmgr.conf [general] enable=yes ; enable creation of managed DNS lookups ; default is 'no' refreshinterval=180 ; refresh managed DNS lookups every n seconds ; default is 300 (5 minutes) Hello Andres, I read that same suggestion elsewhere in connection with Deutsche Telekom, so it seems there's some benefit in it. Daniel, did you try it out already? Kind regards, Sebastian -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com
Re: [asterisk-users] Update peer IP address
I'd be curious if setting insecure=invite,port makes any difference either (without alllowguest on). On Thu, Apr 2, 2015 at 9:03 AM, Daniel Heckl daniel.he...@gmail.com wrote: Ok, I have tested dnsmgr. This is not a solution, the situation has not changed. With dnsmgr I can not place outbound calls. I do not know why and what dnsmgr really do. My current solution is as follows: Say allowguest=yes, configure the default context that there can not be placed outbound calls. Use iptables to DROP all at your SIP port and allow only your local phones and the sip trunk ip range. I think srvlookup must be set to yes to place outbound calls if there is an ip address change. I think with the restriction of the firewall that should be a secure solution. Am 01.04.2015 um 19:23 schrieb Sebastian Kemper sebastian...@gmx.net: On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote: On 4/1/15 10:48 AM, Daniel Heckl wrote: John, thank you four your answer. I think you have misunderstood the problem. It’s about a ip address change of the sip trunk, not of my asterisk server. You would probably benefit by enabling the DNS Manager to allow for dynamic IP changes: # cat dnsmgr.conf [general] enable=yes ; enable creation of managed DNS lookups ; default is 'no' refreshinterval=180 ; refresh managed DNS lookups every n seconds ; default is 300 (5 minutes) Hello Andres, I read that same suggestion elsewhere in connection with Deutsche Telekom, so it seems there's some benefit in it. Daniel, did you try it out already? Kind regards, Sebastian -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] Update peer IP address
or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] PJSIP configuration for AWS/EC2 based Asterisk 13.1.0
BTW, the allow=!all is equivalent to disallow=all, so you can drop the disallow line. On Thu, Mar 5, 2015 at 7:26 PM, Sonny Rajagopalan sonny.rajagopa...@gmail.com wrote: OK. I think I found the issue. The key is to add rtp_symmetric=yes Here's what my final configuration looks like: [transport-udp] type=transport protocol=udp bind=0.0.0.0 ;; for within EC2 local_net=172.31.32.0/20 ;; For softphones within EC2 local_net=192.168.1.0/24 external_media_address=publicIPOfEC2Instance external_signaling_address=publicIPOfEC2Instance ;Templates for the necessary config sections [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=!all,ulaw direct_media=no rtp_symmetric=yes On Thu, Mar 5, 2015 at 5:52 PM, Sonny Rajagopalan sonny.rajagopa...@gmail.com wrote: Hello All, I have an Asterisk server v13.1.0 running on EC2 and I am able to connect and register SIP devices and see them on the asterisk CLI. I am also able to place calls, but I am not able to hear any audio on either end after the call is picked up. I was wondering if you can tell me what a minimal configuration for Asterisk on EC2 looks like. My current pjsip.conf configuration looks like this: type=transport protocol=udp bind=0.0.0.0 local_net=172.31.32.0/20 ; In the following two lines, replace publicIP with the output of ; curl -s http://169.254.169.254/latest/meta-data/public-ipv4 external_media_address=publicIP external_signaling_address=publicIP [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=ulaw direct_media=no [auth_userpass](!) type=auth auth_type=userpass [aor_dynamic](!) type=aor max_contacts=1 remove_existing=yes ;Definitions for our phones, using the templates above ;; usernames and passwords etc. below My security group configuration allows TCP, UDP posrt 5060 inbound, outbound from/to 0.0.0.0/0 and TCP, UDP ports 1-2 from/to 0.0.0.0/0. Should I turn on STUN for my zoiper softphones? Any specific flavor? What am I doing wrong? Any help appreciated. -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] connect call to queue to specified agent
When the call comes in, before sending it into the queue, you could consult a database of last agent who helped the user, then check availability of that agent, and send the call directly to the agent instead of putting it into the queue. You can use QueueLog to record that action so that any queue monitoring data is not unaware of it, but otherwise you would need to understand it won't show up in your queue metrics. On Fri, Feb 13, 2015 at 8:49 AM, Marek Cervenka cerv...@fpf.slu.cz wrote: hi, is it possible connect call to queue to specified agent? like Mr. Neo called helpdesk queue, call picked by agent Smith Mr. Neo is calling again and i want connect him with agent Smith -- --- Marek Cervenka === -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] asterisk -r spammy
Use the -m option to mute console logging. On Fri, Feb 13, 2015 at 12:47 PM, thufir hawat.thu...@gmail.com wrote: when running asterisk -r, is there a way to turn off the messages? I didn't find the answer in the man page. thanks, Thufir -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] Dial Plan Issue
I am trying to transition an application over from a FreePbx box to a Standard build Asterisk 11.6 box. I have a job that creates a call file and plays a sound file. If it detects a voicemail, then it plays it, waits 1 second and replays it. The FreePbx box works fine but the Standard Asterisk build is dropping the call during the first Voicemail playback and it does not leave the voicemail. Here is the printout of the log file for both boxes. Free PBX: [2015-02-10 12:12:34] VERBOSE[10502] pbx.c: -- Executing [XX@subMachine:4] Playback(SIP/trunk503out-9728, temp/0250002) in new stack [2015-02-10 12:13:29] VERBOSE[10502] pbx.c: -- Executing [XX @subMachine:5] Wait(SIP/trunk503out-9728, 1) in new stack [2015-02-10 12:13:30] VERBOSE[10502] pbx.c: -- Executing [XX @subMachine:6] Playback(SIP/trunk503out-9728, temp/0250002) in new stack Standard Asterisk Build: [2015-02-10 15:01:12] VERBOSE[32567][C-000f] pbx.c: -- Executing [xx @subMachine:1] SendDTMF(SIP/SMtrunk1-000f, w1w) in new stack [2015-02-10 15:01:16] VERBOSE[32567][C-000f] pbx.c: -- Executing [xx @subMachine:2] Set(SIP/SMtrunk1-000f, IVR_MSG=temp/0250002) in new stack [2015-02-10 15:01:16] VERBOSE[32567][C-000f] pbx.c: -- Executing [xx@subMachine:3] System(SIP/SMtrunk1-000f, /bin/echo -e xx,RUN2,i9,02102015145822,MACHINE,SIP/SMtrunk1-000f,02.10.2015 15.01log/outbound.txt) in new stack [2015-02-10 15:01:16] VERBOSE[32567][C-000f] pbx.c: -- Executing [xx @subMachine:4] Playback(SIP/SMtrunk1-000f, temp/0250002) in new stack [2015-02-10 15:01:16] VERBOSE[32567][C-000f] file.c: -- SIP/SMtrunk1-000f Playing 'temp/0250002.slin' (language 'en') [2015-02-10 15:01:50] VERBOSE[32567][C-000f] pbx.c: == Spawn extension (subMachine, xx, 4) exited non-zero on 'SIP/SMtrunk1-000f' I copied the context from the FreePbx box over to the new box so the code should be the same. Any help would be appreciated. Thanks, Scott If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosurehttp://www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -- _ -- 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] Dial Plan Issue
One follow-up. At the end of the call, after it dis-connects I get the following error: [2015-02-10 15:33:42] NOTICE[4524]: pbx_spool.c:402 attempt_thread: Call completed to SIP/SMtrunk1/xx Thanks, Scott Haley 5-2244 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A Sent: Tuesday, February 10, 2015 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dial Plan Issue I am trying to transition an application over from a FreePbx box to a Standard build Asterisk 11.6 box. I have a job that creates a call file and plays a sound file. If it detects a voicemail, then it plays it, waits 1 second and replays it. The FreePbx box works fine but the Standard Asterisk build is dropping the call during the first Voicemail playback and it does not leave the voicemail. Here is the printout of the log file for both boxes. Free PBX: [2015-02-10 12:12:34] VERBOSE[10502] pbx.c: -- Executing [XX@subMachine:4] Playback(SIP/trunk503out-9728, temp/0250002) in new stack [2015-02-10 12:13:29] VERBOSE[10502] pbx.c: -- Executing [XX @subMachine:5] Wait(SIP/trunk503out-9728, 1) in new stack [2015-02-10 12:13:30] VERBOSE[10502] pbx.c: -- Executing [XX @subMachine:6] Playback(SIP/trunk503out-9728, temp/0250002) in new stack Standard Asterisk Build: [2015-02-10 15:01:12] VERBOSE[32567][C-000f] pbx.c: -- Executing [xx @subMachine:1] SendDTMF(SIP/SMtrunk1-000f, w1w) in new stack [2015-02-10 15:01:16] VERBOSE[32567][C-000f] pbx.c: -- Executing [xx @subMachine:2] Set(SIP/SMtrunk1-000f, IVR_MSG=temp/0250002) in new stack [2015-02-10 15:01:16] VERBOSE[32567][C-000f] pbx.c: -- Executing [xx@subMachine:3] System(SIP/SMtrunk1-000f, /bin/echo -e xx,RUN2,i9,02102015145822,MACHINE,SIP/SMtrunk1-000f,02.10.2015 15.01log/outbound.txt) in new stack [2015-02-10 15:01:16] VERBOSE[32567][C-000f] pbx.c: -- Executing [xx @subMachine:4] Playback(SIP/SMtrunk1-000f, temp/0250002) in new stack [2015-02-10 15:01:16] VERBOSE[32567][C-000f] file.c: -- SIP/SMtrunk1-000f Playing 'temp/0250002.slin' (language 'en') [2015-02-10 15:01:50] VERBOSE[32567][C-000f] pbx.c: == Spawn extension (subMachine, xx, 4) exited non-zero on 'SIP/SMtrunk1-000f' I copied the context from the FreePbx box over to the new box so the code should be the same. Any help would be appreciated. Thanks, Scott If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.commailto:messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosurehttp://www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -- _ -- 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] constantly increasing load in Asterisk 11.14
Can you tell me if the memory usage by Asterisk is also increasing with load over time? On Thu, Feb 5, 2015 at 4:53 AM, Sebastian Damm d...@sipgate.de wrote: Hi, we have quite a few Asterisk machines running and try to keep them on a current version of the Asterisk 11 branch. But since we upgraded to 11.14.0 a couple weeks ago, we have to restart the Asterisk process every week because the load gets too high and our monitoring complains. Those machines are doing only SIP-to-SIP call relay, the dialplan is quite complex, transcoding is done only on a few percent of the calls processed. During the daytime, there are at max around 200 SIP channels (100 calls) running at the same time. After one week, one machine has processed about 170k calls. I have uploaded a comparison of cacti load graphs for one week of a machine running with 11.14.0 and one running with 11.6.0: http://pbrd.co/1v0SO3R As you can see, after a restart, both machines have about the same load. But after the really quiet weekend, the 11.14 Asterisk starts the new week with a much higer load than the 11.6 Asterisk, where it stays constant. We've had an 11.5.1 machine running for about half a year without the need of restarting, but right now, this is not possible. Has anyone seen this before? Or does anyone know a reason, what change somewhere between 11.6 and 11.14 could cause this behaviour? It looks like we have to go back to 11.6. Best Regards, Sebastian -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] Problem with Cisco Phones
If I remember correctly, 9.x firmware dropped UDP support altogether. On Thu, Jan 22, 2015 at 4:31 AM, Jordan Cook - Gyron Networks jordan.c...@gyron.net wrote: Apparently this is a known problem past v8 firmware: http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update- version-9/ I've done some more playing about and what I've noticed is that even when using TCP SIP on the 8.x Firmware conferencing doesn’t work - making it use UDP fixes this. So has anyone managed to get the 9.x firmware working with UDP? Possibly worth a try to see if this resolves the issue? This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] MWI issue
I have a situation that I need help with. I have 2 phone systems, 1 Asterisk and 1 Avaya. All voicemail is kept on the Avaya system. Whenever a call comes into an extension that the Asterisk server owns, I re-direct it to a different number that is owned by the Avaya System. If that Avaya extension does not answer it, I send it to the voicemail on the Avaya Messaging system for the extension that it came in on the Asterisk box. Once that happens, I need to send a MWI indicator to an application on the desktop of the Avaya User that there is a voicemail for that mailbox. I see the SIP Notify come in from Avaya for the extension (I did this with a tcpdump). My question is how do I configure Asterisk to act on that request and call an agi program to do what I want. Any help would be appreciated. Thanks, Scott Haley If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosurehttp://www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -- _ -- 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] Problem with Cisco Phones
Next step is packet capture to see if there is a clue as to the cause of the failure in the SIP signalling. On Tue, Jan 20, 2015 at 10:41 AM, Jordan Cook - Gyron Networks jordan.c...@gyron.net wrote: We were using G722 - I thought similarly and tried a call with alaw. Same problem occurred, any other ideas? I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can only do a single G729 channel, and if you require G729 for the second leg of a conference, it will fail. This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] Problem with Cisco Phones
Apparently this is a known problem past v8 firmware: http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update-version-9/ On Tue, Jan 20, 2015 at 11:16 AM, Jordan Cook - Gyron Networks jordan.c...@gyron.net wrote: Next step is packet capture to see if there is a clue as to the cause of the failure in the SIP signalling. Right, I see the following when running SIP Debug. Looks to me like the phones are expecting the server to do the conference mixing, which I guess it would do in CallManager? --- SIP read from TCP:xxx.xxx.xxx.xxx:50604 --- REFER sip:xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/TCP xxx.xxx.xxx.xxx:50604;branch=z9hG4bK48c7492c From: 4005 sip:4...@xxx.xxx.xxx.xxx ;tag=203a07fceb4b00eff1377deb-da93e2ee To: sip:4...@xxx.xxx.xxx.xxx Call-ID: outofdialog--001e-67a906f5-5333c...@xxx.xxx.xxx.xxx Max-Forwards: 70 Date: Tue, 20 Jan 2015 17:10:19 GMT CSeq: 101 REFER User-Agent: Cisco-CP7945G/9.4.2 Contact: sip:4...@xxx.xxx.xxx.xxx:50604;transport=tcp Referred-By: 4005 sip:4...@xxx.xxx.xxx.xxx Refer-To: cid:9a2a9191@xxx.xxx.xxx.xxx Content-Length: 963 Content-Type: application/x-cisco-remotecc-request+xml Content-Disposition: session;handling=required Content-Id: 9a2a9...@xxx.xxx.xxx.xxx ?xml version=1.0 encoding=UTF-8? x-cisco-remotecc-request softkeyeventmsg softkeyeventConference/softkeyevent dialogid callid203a07fc-eb4b001c-1bf7ad61-614d3...@xxx.xxx.xxx.xxx/callid localtag203a07fceb4b00ed3e4e2321-d9cb1581/localtag remotetagas4a087ee2/remotetag /dialogid linenumber0/linenumber participantnum0/participantnum consultdialogid callid203a07fc-eb4b001d-14750420-d3d10...@xxx.xxx.xxx.xxx/callid localtag203a07fceb4b00ee46f74fd6-4ed3acbd/localtag remotetagas18747c6d/remotetag /consultdialogid statefalse/state joindialogid callid/callid localtag/localtag remotetag/remotetag /joindialogid eventdata invocationtypeexplicit/invocationtype /eventdata userdata/userdata softkeyid0/softkeyid applicationid0/applicationid /softkeyeventmsg /x-cisco-remotecc-request - --- (16 headers 3 lines) --- Sending to xxx.xxx.xxx.xxx:50604 (no NAT) Call outofdialog--001e-67a906f5-5333c...@xxx.xxx.xxx.xxx got a SIP call transfer from caller: (REFER)! --- Transmitting (no NAT) to xxx.xxx.xxx.xxx:50604 --- SIP/2.0 603 Declined (No dialog) Via: SIP/2.0/TCP xxx.xxx.xxx.xxx:50604;branch=z9hG4bK48c7492c;received=xxx.xxx.xxx.xxx From: 4005 sip:4...@xxx.xxx.xxx.xxx ;tag=203a07fceb4b00eff1377deb-da93e2ee To: sip:4...@xxx.xxx.xxx.xxx;tag=as141fffdd Call-ID: outofdialog--001e-67a906f5-5333c...@xxx.xxx.xxx.xxx CSeq: 101 REFER Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] Problem with Cisco Phones
I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can only do a single G729 channel, and if you require G729 for the second leg of a conference, it will fail. On Tue, Jan 20, 2015 at 10:03 AM, Jordan Cook - Gyron Networks jordan.c...@gyron.net wrote: Possibly slightly off topic, has anyone ever had Cisco 79xx Series phones come up with “cannot complete conference” errors when trying to conference two calls together? This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] sip show channelstats reliable?
I would recommend capturing traffic outside your Asterisk server with Wireshark, then running the Telephony/Rtp/Analysize Streams option to determine if you have packet loss at that point in the network. On Mon, Jan 19, 2015 at 1:00 PM, Todd R. tjrl...@live.com wrote: Thanks but no Adtran here. I do think these stats are indicating an issue, I just don't know how to prove it outside Asterisk. -- From: ewiel...@nyigc.com To: tjrl...@live.com; asterisk-users@lists.digium.com Date: Mon, 19 Jan 2015 13:55:33 -0500 Subject: RE: [asterisk-users] sip show channelstats reliable? I’ve seen something similar with Adtran SIP gateways.When a re-invite happens the Adtran gets all confused about call stats and marks the pre-reinvite leg of the call as losing large numbers of packets.BTW, IIRC reinvites happen when a codec changes or the channel switches to T.38. Also Adtran SIP gateways appear not to support OPTIONS packets when running in SIP proxy mode, which is very annoying. At some point I’ll try and arrange a slugfest between Digium and Adtran and they can figure out why it doesn’t work. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd R. *Sent:* Monday, January 19, 2015 1:45 PM *To:* Asterisk-Users List *Subject:* Re: [asterisk-users] sip show channelstats reliable? Additional info: At the moment I am running 1.8.x but the other day I was getting the same results on 11.x Here is a sample from show channelstats. I do think this command is showing that there is trouble between specific IP's and my Asterisk box but I don't know if the numbers are accurate and reliable. Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter x.x.x.x 5531341d06b 00:07:42 023123 063836 (73.41%) 0. 023102 00 ( 0.00%) 0.0007 Peer IP changed to protect the innocent :-) -- From: tjrl...@live.com To: asterisk-users@lists.digium.com Date: Mon, 19 Jan 2015 12:17:25 -0600 Subject: [asterisk-users] sip show channelstats reliable? I am seeing lots of lost packets when running the command sip show channelstats at the CLI. There are issues across multiple Asterisk servers I am trying to diagnose but everything I read seems to point to this command being pretty unreliable. Can I trust the info this command shows? I am showing lots of lost packets in sip show channelstats but I can't see any packet loss when pinging the same IP's to/from. Since I don't 100% control the network my gear is on, I need something outside of Asterisk to show the network engineer to convince here and myself that there are network issues. All I have is the loss that's shown from this command with no real network stats to back it up. Is there a magic command in CentOS anyone can recommend to diagnose and match up the issues shown in Asterisk using this command? Moving gear around on the network changes the info Asterisk shows a LOT. For example, if I point traffic to the main physical gateway I get loss to a particular customer's IP (their PBX), if I move it to another place on the network (as a VM) their IP is good and other customers IP's start showing loss using the channelstats info. Driving me freakin' crazy. It does appear there are network issues causing my troubles but I can't get help if I can't point to some hard and fast issues outside of Asterisk. The only thing I have right now is collissions showing on one of a few of our pfSense devices but they are virtual running on XenServer, still this would indicate a problem in my opinion. Thanks in advance for any assistance on this issue. Stepping back from the ledge now LOL -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org -- _ -- Bandwidth
Re: [asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue
To sort out RTP problems, I would recommend: 1) on all endpoints use codec of allow=!all,ulaw -- this is or should be supported by all endpoints and eliminates any issues of mismatch, translation, etc., and can be adjusted later once everything is working 2) add an Echo() application to your dialplan so you can call it and check RTP to and from Asterisk 3) start with direct_media=no to run all the RTP through Asterisk first 4) packet capture at/on the asterisk server, as well as at endpoints if need be, to identfy if and where RTP streams are being sent and received. The goal being to get two way audio calls up through Asterisk, and then change one thing at a time towards your desired configuration and retest. On Thu, Jan 8, 2015 at 7:03 PM, Sonny Rajagopalan sonny.rajagopa...@gmail.com wrote: Well, I thought it worked, but it actually doesn't--I am able to get the caller pick up the phone, but for some reason, I cannot hear anything on either side no matter who does the calling. Again, my two SIP phones are on the local 192.168.1.0/24 network (do not go over the Internet) and the Asterisk server is located in the same network (not accessed over the Internet). Any help is appreciated. Does the fact that Asterisk is running on a VirtualBox VM on the same machine as one of the SIP phones matter? I am able to access the ARI REST interface of the Asterisk server quite fine on the host machine. I suspect it has to do with RTP not being set up, but all the codec support is there. Here's a log for the SIP request from 192.168.1.50: --- Received SIP request (1229 bytes) from UDP:192.168.1.50:64009 --- INVITE sip:6002@192.168.1.139;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 146.115.163.234:64009 ;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z- Max-Forwards: 70 Contact: sip:demo-alice@146.115.163.234:64009;transport=UDP To: sip:6002@192.168.1.139;transport=UDP From: sip:demo-alice@192.168.1.139;transport=UDP;tag=b661670b Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.21933 r21903 Authorization: Digest username=demo-alice,realm=asterisk,nonce=[removed],uri= sip:6002@192.168.1.139 ;transport=UDP,response=[removed],cnonce=[removed],nc=0001,qop=auth,algorithm=md5,opaque=[removed] Allow-Events: presence, kpml Content-Length: 245 v=0 o=Z 0 0 IN IP4 146.115.163.234 s=Z c=IN IP4 146.115.163.234 t=0 0 m=audio 8000 RTP/AVP 0 3 110 8 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- Transmitting SIP response (319 bytes) to UDP:192.168.1.50:64009 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 146.115.163.234:64009 ;rport=64009;received=192.168.1.50;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z- Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE. From: sip:demo-alice@192.168.1.139;tag=b661670b To: sip:6002@192.168.1.139 CSeq: 2 INVITE Content-Length: 0 Any help is appreciated. A topology is shown below in ASCII. ( Big bad Internet ) GW/NAPT/Router | -- / \ || Host A Host B - - | Alice | | Bob | | 192.168.1.50 | | 192.168.1.149 | |---| |---| | Asterisk sr | |(VM) | | 192.168.1.239 | |---| On Thu, Jan 8, 2015 at 2:32 PM, Sonny Rajagopalan sonny.rajagopa...@gmail.com wrote: Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is now [transport-udp] type=transport protocol=udp bind=0.0.0.0 local_net=192.168.1.0/24 ;Templates for the necessary config sections [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=ulaw [auth_userpass](!) type=auth auth_type=userpass [aor_dynamic](!) type=aor max_contacts=1 remove_existing=yes ;Definitions for our phones, using the templates above [demo-alice](endpoint_internal) auth=demo-alice aors=demo-alice mailboxes=box_a rewrite_contact=yes [demo-alice](auth_userpass) password=demo
Re: [asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue
It would appear that you have the Asterisk server on a public IP address, your two endpoints are behind a NAT, and you have rewrite_contact enabled in pjsip.conf. In which case, what you are seeing is correct. In order to be able to send a call to an extension where it is behind NAT, Asterisk must update the contact to have the current IP and port that the phone registered via (i.e. the WAN IP of the NAT, and the WAN port that it is retaining state for). On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan sonny.rajagopa...@gmail.com wrote: I am following the instructions in https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I am trying to make a call from extension Alice (6001) to extension for Bob (6002). When I make the call, I can hear the ringing on Alice's phone (caller), but Bob's phone (callee) doesn't ring, or show a call coming in from Alice. My setup and environment is as follows: Alice, Bob and Asterisk all in the same 192.168.1.0/24 network, and they are able to register to the Asterisk server running 13.1.0/PJSIP. The rest of the configuration is the same as the aforementioned wiki page, but is shown here for clarity: root@asterisk13FFP:/var/log/asterisk# more /etc/asterisk/extensions.conf [from-internal] exten=6001,1,Dial(PJSIP/demo-alice) exten=6002,1,Dial(PJSIP/demo-bob) exten=6003,1,Answer() same =6003,n,Playback(hello-world) same =6003,n,Hangup() What I do observe is that I when I request the output of pjsip show endpoints, I get Contact information for the two SIP peers that have registered different from their actual IP addresses. I suspect this has something to do with their calls being routed elsewhere. If my assumption is correct--how do I fix this? Alice should be at 192.168.1.50 and Bob should be at 192.168.1.149, instead, they (both) show IP address 146.115.163.234. Any help is deeply appreciated. Thanks. asterisk13FFP*CLI pjsip show endpoints Endpoint: Endpoint/CID. State. Channels. I/OAuth: AuthId/UserName... Aor: Aor MaxContact Contact: Aor/ContactUri... Status RTT(ms).. Transport: TransportId Type cos tos BindAddress.. Identify: Identify/Endpoint. Match: ip/cidr. Channel: ChannelId.. State. Time(sec) Exten: DialedExten... CLCID: ConnectedLineCID... = Endpoint: demo-alice Unavailable 0 of inf InAuth: demo-alice/demo-alice Aor: demo-alice 1 Contact: demo-alice/sip:demo-alice@*146.115.163.234*:38519 Unknown nan Endpoint: demo-bob Not in use0 of inf InAuth: demo-bob/demo-bob Aor: demo-bob 1 Contact: demo-bob/sip:demo-bob@*146.115.163.234*:38321;tra Unknown nan -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] Smartphone Mobility App?
The main problem we are trying to solve is when our staff forward to their cell phones they cant distinguish if the call was directed at their cell phone or the business DID. The easiest way to solve that is to have an audio prompt announce the calls that were passed through from the business DID before connecting the call through. That does require using a follow-me approach instead of forwarding, but is easily done by just changing the confimration prompt. On Fri, Dec 19, 2014 at 8:29 AM, chris tknch...@gmail.com wrote: Anyone found any good smartphone apps that connect with their asterisk boxes that provides basic mobility features? The main problem we are trying to solve is when our staff forward to their cell phones they cant distinguish if the call was directed at their cell phone or the business DID. We also would like to give user ability to control DND and forwarding of their extension from the smartphone. I know there are many cloud service providers with a offering like this but we are not looking to change our service infrastructure but rather looking for just a software product that connects to our existing asterisk systems and provides this functionality. We would ideally like something for both iphone and android but the immediate need is for iPhone Curious to hear what people have tried, their experiences, etc. We are open to both free/open source as well as commercial software as long as it is multitenant or scalable beyond single server. TIA, chris -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] Register multiple phones to a single AOR with PJSIP
You need to change your dialplan to use the PJSIP_DIAL_CONTACTS function
like this:
exten = _X.,1,Dial(${PJSIP_DIAL_CONTACTS(200)},30)
It expands to the list of contacts, separated by , so that the contacts
are dialed at the same time.
The documentation page you reference should be updated to include that
detail.
On Thu, Oct 30, 2014 at 2:18 PM, Carlos Chavez cur...@telecomabmex.com
wrote:
I just finished installing Asterisk 13 on our test server and I can
now use PJSIP to register phones and make and receive calls. The only
problem I am having is that when I register multiple phones to a single
account only one of them rings. The AOR for the account has maxcontacts at
3.
If I do a pjsip show endpoints I can see two Contact entries which I
take to mean that both phones have registered:
Endpoint: 101 Not in
use0 of inf
InAuth: 101/101
Aor: 1013
Contact: 101/sip:101@192.168.2.193:5063 Avail 178.681
Contact: 101/sip:101@192.168.2.197:58086;transport=UDP;r Avail
4.198
Transport: transport-udp udp 0 0 0.0.0.0:5060
I have tried with several phones and have rebooted the Asterisk server
and phones several times just to make sure configs are loaded properly but
I cannot get Asterisk to ring multiple phones at once. I used
https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime to
configure this instance of Asterisk. Am I missing some setting to allow
Asterisk to ring all phones registered to a single AOR?
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Re: [asterisk-users] Call forwarding from Phones and getting the referrer IP
After a quick perusal of the chan_sip.c code (from svn trunk), I'm not seeing where the address (p-sa) logged in that message is passed to the redirecting functions handling the 302, thus it is unlikely there is a way to obtain it other than reading the log. It wouldn't be hard to set a channel variable with that value however, should you want to patch the code, possibly even submit that. On Tue, Oct 28, 2014 at 7:05 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On 24 October 2014 16:51, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm using asterisk 1.8 but I'm sure this applies to other versions. If someone puts a call divert on a handset such as a Snom phone I get this type of SIP message on receipt of an inbound call: Got SIP response 302 Moved Temporarily back from xxx.xxx.xxx.xxx:x Which then triggers a local channel to make the call. Is there any way I can access that IP address inside my dialplan? I've done a ChanDump and there's no sign of it. Regards Ish Bumping this as I originally sent it late on Friday. If anyone has any idea, please let me know. Thanks in Advance Ish -- Ishfaq Malik 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, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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 reload context
Using current svn trunk, that option isn't available. It would appear that the patch from that issue did not get into the code. On Tue, Oct 28, 2014 at 10:22 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, is it possible to reload just a context in stead of the whole dialplan ? I see this on the tracker : https://issues.asterisk.org/jira/browse/ASTERISK-19934 But is it possible in some Asterisk version ? Kind regards, Jonas. -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] [asterisk-dev] AstriDevCon 2014: Agenda itemDeprecate AMI/AGI(Ben Klang)
is asterisk abandoning the dial plan? It's clear that there is a desire to have a way of running Asterisk with little or no dialplan. While currently there is no way to abandon the dialplan as you point out, that could actually happen, someday, many years and versions from now. But even then I would expect there could be a loadable module to add dialplan support for those who still need it, where the dependencies on dialplan have since been removed from the core. So, to answer your question, yes, and no. On Wed, Oct 22, 2014 at 1:55 PM, Paul Albrecht palbre...@glccom.com wrote: On Oct 22, 2014, at 11:31 AM, Matthew Jordan mjor...@digium.com wrote: On Wed, Oct 22, 2014 at 11:14 AM, Paul Albrecht palbre...@glccom.com wrote: On Oct 22, 2014, at 10:33 AM, Joshua Colp jc...@digium.com wrote: Paul Albrecht wrote: Really? Shouldn’t something this major affecting the entire Asterisk community get discussed on the lists? Any idea what Leif is talking about when he says the community is in transition, moving from dial plan model to external control. It was something Ben Klang brought up and wanted to talk about - it's not something that has been decided 'nor does anyone know what the future entails. Any further discussions will naturally occur on the mailing list and in fact some things have explicit action items to bring them up on here. The suggestion that Asterisk should consider deprecating AMI/AGI is “crazy talk.” It doesn’t merit discussion and shouldn’t be on the agenda in the first place. It’s completely impractical and can never happen. Moreover, Leif seems to think we (the asterisk community) are in transition. What does that mean? Are we abandoning the dial plan? Seriously? That’s never gonna happen either. ARI isn’t easier to use than dial plan scripting. I guess one could hope that what happens in Vegas stays in Vegas”, but I don’t think the Asterisk community has that kind of luck. Just because someone decided to bring up a radical idea does not mean we refuse to discuss it. So you agree that deprecating AMI/AGI is “crazy talk” but you’ll discuss it because of your open-mindedness? This is an open source project. Communication is done in an open, transparent manner. People should feel like they can bring up interesting, radical, and yes - even crazy - ideas. By the same token, when you propose ideas, you must be prepared for honest criticism and accept it in graciously rather than simply resorting to argument ad hominem. If you don't like that, you don't have to participate in the discussion. You haven’t really responded to the substance of my post, that is, is asterisk abandoning the dial plan? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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 -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] PJSIP and NAT behind a dynamic IP address
If you review the current asterisk 12 sample pjsip config for extension 6002 (viewable here: http://svnview.digium.com/svn/asterisk/branches/12/configs/pjsip.conf.sample), you will find it contains the correct settings for an endpoint behind NAT. Specifically note that you need rewrite_contact enabled so that the contact address is rewritten to match the inbound SIP registration, and also with rtp_symmetric enabled to do the same thing for RTP. Also be aware that you will have less problems by omitting the transport= line from the endpoint configuration altogether. It's generally not required to define that the endpoint is restricted to using a specific transport, and doing so interferes with the automatic transport selection, possibly including the symmetric SIP operation. On Wed, Oct 22, 2014 at 9:13 PM, Jeffrey Ollie j...@ocjtech.us wrote: What should the PJSIP configuration be if your external IP address is dynamic, as is common with most home networks, and probably a lot of small business networks as well? The external_media_address and external_signaling_address transport settings are static. It would be possible to write a script that would detect the external IP address and rewrite the pjsip configuration file, but since you can't change transports without a full restart of the server that doesn't seem very friendly. Is the only alternative to rely on your firewall/router to fix up the address in the SDP? -- Jeff Ollie -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] Issue playing high quality white noise
The limitation of 8khz sample rate (ulaw or alaw on pstn) should only affect the audio spectrum - for example there will be a loss of frequencies above 3.3-4khz if the band pass filter is done correctly, or an overly loud static sound where higher frequencies were in the original if not. If by 'broken up' you mean to say that there are periods of no audio, then there is a separate issue affecting the audio stream such as packet loss or problems getting the audio file to stream reliably. On Tue, Oct 14, 2014 at 10:47 AM, asteriskus...@dovid.net wrote: Hi, I have a client that wants a phone system that will play sounds from a sleep machine. I tried using all different formats (GSM, WAV, WV49, MP3 etc.). Over SIP it was OK however with the PSTN it broke up from time to time. I assume this has to do with the fact that the PSTN is limited to 8khz. Is there something I am missing here or is this simply a limitation of the PSTN? Regards, Dovid -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://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] Sent ami event from AGI?
You can use the AGI command EXEC to execute a dialplan application, and the application UserEvent can be used to generate custom events that AMI clients can receive. https://wiki.asterisk.org/wiki/display/AST/AGICommand_exec https://wiki.asterisk.org/wiki/display/AST/Application_UserEvent On Thu, Oct 2, 2014 at 4:02 AM, Ilya Awesome jl...@me.com wrote: hello, is there way to send event to all ami clients from AGI script? Sent from my iPhone -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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 Questions
You can create an extension 456, but change the DIAL string to be Local/$97@from-internal The extension can be any type really, but normally in this case you would use Custom rather than SIP to avoid creating an actual extension. On Thu, Oct 2, 2014 at 12:32 PM, Phil Ledon ple...@lodgetech.com wrote: We are trying to add voice mail to our hotel rooms. Our current phone instruction cards say 'to reach voice mail dial ext 456. Replacing those instructions is not feasible at the moment. We have Feature Code *97 that takes them directly to their voice mail box. Question - What is an easy way to have exten 456 dial *97. We are using AsteriskNow distro, version11. *Phil Ledon* -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] how can queue agents choose which call to answer?
You can use any number of methods for redirecting a call from the queue to a specific agent. These include off the shelf products such as FOP or iSymphony, or even something custom built that can display calls and direct Asterisk (usually through AMI) to transfer the call to a new destination. However, you will need to be aware that your queue metrics may not count it as a normally handled call, since the call is yanked out of the queue to transfer directly to an agent via a separate tool. You may also want to look into building a custom queue-like solution through ARI, using a Stasis application to manage callers on hold in waiting bridges, and then delivering them to agents completely under control of your application. In this case you would need to create your own queue logging data to your metrics solution, which would allow you to record calls correctly even when transferred early. On Tue, Sep 23, 2014 at 1:41 PM, Michael Keuter li...@mksolutions.info wrote: Am 23.09.2014 um 19:49 schrieb Marie Fischer ma...@vtl.ee: Hi everybody, I'm looking for a solution for the following scenario: • Asterisk queue • At peak hours, there will be more callers then queue members/agents, so some callers will spend some time on hold • Agents should be able to choose which of the on hold calls to answer instead of answering the next one in queue We already have a web interface where agents can see the callers on hold, so the best solution would be if they could just click a callers number to get his call. But I have not found a way to tell Asterisk to do something to a call on hold in a queue. Priority queues are not really an option, as the agents will be deciding on the fly which caller is more important. I am not really sure if queues are the correct solution for this problem. However, we have existing statistics built for queue logs, so it would be really nice if the solution was queue-based. Thanks for any thoughts, -- marie Hello Marie, maybe FOP2 [1] is an option for you. There you can visually pick up a call from a queue. It's not open source though. [1] http://www.fop2.com Michael http://www.mksolutions.info -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] compiling Asterisk
I am trying to compile the certified-asterisk-11.6-cert5 code and when I try to start it and then go into the console I am getting the error message asterisk dead but subsys locked. Can anyone help with why this is happening? I have never seen this before. This is a fresh install on a new server CentOS 6.5. Thanks, Scott Haley IS Voice Projects Team Edward Jones Investments Phone: 314-515-2244 Email: scott.ha...@edwardjones.commailto:scott.ha...@edwardjones.com If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosurehttp://www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -- _ -- 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] compiling Asterisk
Had to re-install and change selinux to disable. Works now. Thanks, Scott Haley 5-2244 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A Sent: Friday, September 12, 2014 1:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] compiling Asterisk I am trying to compile the certified-asterisk-11.6-cert5 code and when I try to start it and then go into the console I am getting the error message asterisk dead but subsys locked. Can anyone help with why this is happening? I have never seen this before. This is a fresh install on a new server CentOS 6.5. Thanks, Scott Haley IS Voice Projects Team Edward Jones Investments Phone: 314-515-2244 Email: scott.ha...@edwardjones.commailto:scott.ha...@edwardjones.com If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.commailto:messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosurehttp://www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -- _ -- 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] Call Transfer Fails - Not a Valid Extension
The file /var/log/asterisk/full will contain helpful log messages that show how Asterisk is internally handling the call. It may be necessary to increase the verbosity of the log to get more details however. From the linux command line, you can follow these steps to get a copy of the relevant messages: # asterisk -rx core set verbose 5 # cat /var/log/asterisk/full mylogfile (perform a transfer that fails with the message now, then press CTRL-C to cancel the above command) The mylogfile will have the log entries necessary to understand what happened, although it may also require an understanding of the FreePBX dialplan to interpret it. If you can post your log file (recommend using a pastebin rather than emailing the whole thing) it should be fairly easy to spot the problem and advise you how to fix it. On Sun, Sep 7, 2014 at 10:55 PM, Phil Ledon ple...@lodgetech.com wrote: We have a plain vanilla installation of AsteriskNOW using Digium D40/50 phones. All transfers are failing from any source to any extension with the message “that is not a valid extension”. Does anyone have any ideas about where to begin looking for the source of that error? *Phil Ledon* -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] asterisk SugarCrm integration
Unfortunately, my knowledge of SugarCRM is also a little dated. I checked on SugarForge ( http://www.sugarforge.org/softwaremap/trove_list.php?form_cat=407) and there doesn't appear to be an Asterisk integration listed, although there are some tapi dialers (which may allow routing to asterisk via another app). I would recommend filing an issue on the yaai project for 7 support. There may also be some other resources I've missed. On Thu, Aug 28, 2014 at 3:18 PM, Marek Cervenka cerv...@fpf.slu.cz wrote: it's old. sugarcrm v7 is not supported Dne 28.8.2014 v 14:54 Scott Griepentrog napsal(a): I've used this before, and it appears to still be an active project. https://github.com/blak3r/yaai On Thu, Aug 28, 2014 at 7:29 AM, Marek Cervenka cerv...@fpf.slu.cz wrote: hello, can you recommend good asterisk-SugarCrm integration plugin? i googled a lot, but i want something what is used on daily basis 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] asterisk SugarCrm integration
I've used this before, and it appears to still be an active project. https://github.com/blak3r/yaai On Thu, Aug 28, 2014 at 7:29 AM, Marek Cervenka cerv...@fpf.slu.cz wrote: hello, can you recommend good asterisk-SugarCrm integration plugin? i googled a lot, but i want something what is used on daily basis thank you -- --- Marek Cervenka === -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] PRI timing settings
On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.netmailto:j...@jeff.net wrote: I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as busy. I don’t know what expectations for CLID your carrier might have, or for that matter the upstream carrier, however, we found through our CLEC here in the US that while the CLEC was happy to take e.164 formatted numbers from us as CLID, Global Crossing would reject them further upstream resulting in our calls to many toll frees being rejected. Switching to 10 digit CLID on all outbound calls through that PRI solved the problem. I don’t know if this is your problem but be sure your CLID is in the most simple format possible for your region to help rule it out. sl -- _ -- 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] Asterisk peer definition registration
Registering on a configuration reload (or startup) is written into the code of chan_sip. There isn't a way to defeat that using configuration. Since you presumably are not attempting to register with invalid credentials, the fact that you sometimes have a higher frequency of successful registrations should not be a trigger for being blocked. I would work with them to identify precisely why they are blocking you and if you are not doing anything wrong suggest they review their policy. On Sat, Aug 16, 2014 at 10:21 AM, Steve Ng steveng.1...@gmail.com wrote: Hi, I am using Asterisk currently as a SIP proxy to a SIP provider. Inside my real-time, I would set the SIP credential based on what the user has provided. For example [name] type=peer defaultuser=USER_PROVIDED secret=USER_PROVIDED host=USER_PROVIDED When I reset Asterisk, Asterisk will attempt to register with the sip provider. And if there are sufficiently amount of records with invalid credentials, I'll get blocked by the SIP provider as they might think that I'm brute forcing. Just a question to check if there's any chance I could ask Asterisk not to register when I reset. Or is there any other possible solution for this? -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] [OT] Split a recording based on a presence of beep sound
You would probably have better results from using a specific frequency tone (or dual tones) as the beep and then using a tone detection algorithm to locate it, in the same way that DTMF works. On Tue, Aug 12, 2014 at 2:25 AM, Satish Barot satish4aster...@gmail.com wrote: Hi All, I have been working on a project where I need to record a call in Asterisk and then split the recording into multiple audio files based on a presence of particular sound (i.e. beep) in a recording. I know this is out of scope for Asterisk but I wanted to benefit from someone else's experience if it has been done earlier. I have googled a bit and seems that Audio fingerprint( http://en.wikipedia.org/wiki/Acoustic_fingerprint) is something I should concentrate on. Your views are highly appreciated. Thanks, --Satish -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] Asterisk seding 2 INVITEs all of a sudden
There is right at 500 ms between the two invites. You are seeing a retransmission due to a lack of response to the first INVITE in time. This is normal, correct, and expected behavior. The retransmission can occur even sooner in the case where QUALIFY is used to determine that the endpoint usually responds faster. On Tue, Aug 12, 2014 at 6:49 AM, Nick Cameo sym...@gmail.com wrote: Hello Everyone, Today we observed asterisk sending two invites for the initial call before the call was established (ie, not re-invites). There were no changes made to the configuration for a very long time, and was kind of confused when seeing this action. Can someone please suggest where to look to remove this behaviour? U 2014/08/12 07:34:20.405029 192.168.2.10:5060 - 192.168.2.20:5080 INVITE sip:873359633037@192.168.2.20:5080 SIP/2.0. Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK541d5594;rport. Max-Forwards: 70. From: 555955599 sip:555955...@victoria.example.com;tag=as285d2896. To: sip:873359633037@192.168.2.20:5080. Contact: sip:555955599@192.168.2.10:5060. Call-ID: 5a51eef8064a0d360009f64e34c70...@victoria.example.com. CSeq: 102 INVITE. User-Agent: EXAMPLE Systems. Date: Tue, 12 Aug 2014 11:34:20 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 279. . v=0. o=root 1631923320 1631923320 IN IP4 192.168.2.10. s=EXAMPLE Systems. c=IN IP4 192.168.2.10. t=0 0. m=audio 52034 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2014/08/12 07:34:20.903830 192.168.2.10:5060 - 192.168.2.20:5080 INVITE sip:873359633037@192.168.2.20:5080 SIP/2.0. Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK541d5594;rport. Max-Forwards: 70. From: 555955599 sip:555955...@victoria.example.com;tag=as285d2896. To: sip:873359633037@192.168.2.20:5080. Contact: sip:555955599@192.168.2.10:5060. Call-ID: 5a51eef8064a0d360009f64e34c70...@victoria.example.com. CSeq: 102 INVITE. User-Agent: EXAMPLE Systems. Date: Tue, 12 Aug 2014 11:34:20 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 279. . v=0. o=root 1631923320 1631923320 IN IP4 192.168.2.10. s=EXAMPLE Systems. c=IN IP4 192.168.2.10. t=0 0. m=audio 52034 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. Thanks in Advance, Nick -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] enable features
To enable transfers using in-call DTMF sequences, you'll need to use the t and/or T options in the Dial() command that initiates the call. For details see: https://wiki.asterisk.org/wiki/display/AST/Application_Dial On Thu, Aug 7, 2014 at 2:29 AM, Aristeidis Tsitras tsit...@hotmail.com wrote: i do have asterisk 1.8 (no gui, no distro based) and i would like to enable some features: -call forward (conditional, unconditional,...) -DND -call waiting -attended transfer -follow me all the features i would like to enable/disable them through digit codes such #45# and *45. all these fetures should apply to asterisk only and not use the features from the service provider. i have edited the /etc/asterisk/features.conf file and uncommented the option for attended transfer (*2). the thing is that it did not work. is there something else that i have to write to sip/extensions.conf? -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] Internal timing under load is critical ?
If you were running on Asterisk 1.4, a Zaptel or Dahdi timing source (including the Sangoma USB device) was necessary to avoid sometimes unreliable timing from the dummy interface. For modern releases (1.6, 1.8, 11, 12, etc) this isn't necessary for most systems. However, you may have better results with such a large number of calls by using a hardware timing source. The difference will vary between different systems and loads -- I recommend testing it on your own platform. Note that changing to a different model with a different motherboard or even just a different chipset can result in a difference in timing accuracy. -- so your best option is to try it both ways under load to see if you see a benefit, and re-test should you change the platform, such as using a different motherboard. On Wed, Jul 30, 2014 at 4:08 AM, babak bk1...@yahoo.com wrote: Hi I am evaluating some voice broadcasting solutions based on Asterisks for more than 1000 simultaneous calls. Connection to Asterisk all are based on SIP and SIP Trunks so no DAHDI hardware is required. According to some recommendations like http://osdial.org/howto/ Internal timing is very critical with Asterisk when it is under load and we must use DAHDI hardware or USB Voice Synch Tool http://www.sangoma.com/accessories/specialty-tools/ But according to my understanding of wiki https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces It seems it is not necessary now. Please tell me your opinions. Regards -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] Directory app not working with realtime
The last time I looked at the directory application, it was hard coded to read the voicemail.conf file directly. Unless there is a newer version that can be configured to read the database, it would have to be modified. On Wed, Jul 30, 2014 at 8:55 AM, Tech Support aster...@voipbusiness.us wrote: All; I’m currently running Asterisk 1.8.15-cert7 and am using realtime to store my voicemail configuration. The voicemail application works fine, but the problem I have is that the ‘Directory’ app cannot find any entries because there are no entries in the voicemail.conf file. When I add a context and an extension entry in voicemail.conf, it works the way it should. Is there something that I’m missing here? Any insight at all would be greatly appreciated. Thanks; John *Tech Support* Tech Support VoIP Business Solutions 240-215-3479 (Work/Fax) supp...@voipbusiness.us f...@voipbusiness.us -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] Directory app not working with realtime
For clarification: I was speaking of the directory.php which didn't support realtime last I looked at the code. The app_directory built in to Asterisk should support realtime. Can you determine which one you're using? On Wed, Jul 30, 2014 at 9:46 AM, Scott Griepentrog sgriepent...@digium.com wrote: The last time I looked at the directory application, it was hard coded to read the voicemail.conf file directly. Unless there is a newer version that can be configured to read the database, it would have to be modified. On Wed, Jul 30, 2014 at 8:55 AM, Tech Support aster...@voipbusiness.us wrote: All; I’m currently running Asterisk 1.8.15-cert7 and am using realtime to store my voicemail configuration. The voicemail application works fine, but the problem I have is that the ‘Directory’ app cannot find any entries because there are no entries in the voicemail.conf file. When I add a context and an extension entry in voicemail.conf, it works the way it should. Is there something that I’m missing here? Any insight at all would be greatly appreciated. Thanks; John *Tech Support* Tech Support VoIP Business Solutions 240-215-3479 (Work/Fax) supp...@voipbusiness.us f...@voipbusiness.us -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] Directory app not working with realtime
I just took a peak at that version of app_voicemail and the code definitely reads from realtime. I would suggest: 1) Posting your (password sanitized) configs to see if someone can spot a problem 2) Running with debug and verbose messages enabled and checking the log for helpful diagnostics describing why it isn't working. On Wed, Jul 30, 2014 at 10:32 AM, Tech Support aster...@voipbusiness.us wrote: Scott; I’m using Asterisk’s built-in application “Directory”, not the php script. Thanks; John *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Scott Griepentrog *Sent:* Wednesday, July 30, 2014 10:59 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Directory app not working with realtime For clarification: I was speaking of the directory.php which didn't support realtime last I looked at the code. The app_directory built in to Asterisk should support realtime. Can you determine which one you're using? On Wed, Jul 30, 2014 at 9:46 AM, Scott Griepentrog sgriepent...@digium.com wrote: The last time I looked at the directory application, it was hard coded to read the voicemail.conf file directly. Unless there is a newer version that can be configured to read the database, it would have to be modified. On Wed, Jul 30, 2014 at 8:55 AM, Tech Support aster...@voipbusiness.us wrote: All; I’m currently running Asterisk 1.8.15-cert7 and am using realtime to store my voicemail configuration. The voicemail application works fine, but the problem I have is that the ‘Directory’ app cannot find any entries because there are no entries in the voicemail.conf file. When I add a context and an extension entry in voicemail.conf, it works the way it should. Is there something that I’m missing here? Any insight at all would be greatly appreciated. Thanks; John *Tech Support* Tech Support VoIP Business Solutions 240-215-3479 (Work/Fax) supp...@voipbusiness.us f...@voipbusiness.us -- _ -- 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 -- [image: Image removed by sender. Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org -- [image: Image removed by sender. Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] Limit Asterisk
Whether SSD drives allow you to add any additional calls depends entirely on whether or not they can be written to faster than the SAS drives you have. My experience shows SSD's can be twice as fast as run-of-the-mill SATA, but the performance difference compared to SAS is likely not as great, and could even be worse. You'll need to test two drives to find out. I recommend mounting both to test them and copying a very large ISO file using dd which will give you the transfer rate when finished. Then you should have your answer. On Wed, Jul 23, 2014 at 4:03 PM, Eduardo Leones edua...@ypytecnologia.com.br wrote: Thanks for the feedback. In this case SSD disks you think it solves? Eduardo 2014-07-23 18:01 GMT-03:00 Ron Wheeler rwhee...@artifact-software.com: I would also do some math on the bandwidth requirement. If you divide your disk bandwidth by your recording bit rate what is the theoretical maximum number of calls that you can record at once? Assumes that you have infinite CPU and memory and that you can actually drive the disks at their maximum. If this comes out to 300, you are already there. If it comes out to 3000, you have something wrong in your setup or your assumptions and a target to work towards. What quality are you using in the recording? 44k per second(CD quality sound) uses a lot more bandwidth than 3K (telephone quality) What encoding are you using? How low a bit rate can you use and still have usable recordings? If they are for legal or audit use, you can go pretty low. If you are recording soundtracks for reuse in training or publication, you may require higher bit rates. If you disable recording, how many simultaneous calls can you support? Just to be sure that recording is the issue. Ron On 23/07/2014 4:29 PM, Scott Griepentrog wrote: Your bottleneck is most likely your drive bandwidth. Even with SAS drives, you'll need to move to a raid 5+ solution with 6+ drives to continue to increase the concurrent calls, or use a storage appliance. To confirm this, install the tool nmon and use the v and d options to bring up the resource usage indicators and drive busy/throughput statistics. On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones edua...@ypytecnologia.com.br wrote: people I have a running Asterisk 1.8.28 in great Dell server with two xeon processors and 16gb of ram and HD SAS 15k (Raid 1). This server is recording all calls (placed to record the audio in a ram disk), the entire CDR goes straight to MySQL by cdr_mysql.so. Each call runs some validation and AGI's have an auto dialer system that generates calls over the manager. Calls originate and terminate via SIP (no transcode). With this structure, even being a great server, we can not spend 150 simultaneous calls. When it reaches 140, the load average goes up a lot and the calls start to get very bad audio, tear, etc.. Using the top we see that all the processing is for asterisk. In this scenario, I think there is some limitation in Asterisk, or even the manager due to the auto dialer. Can anyone give me any tips where I can look where is the bottleneck? I need to get at least 250 calls that server quality. tks -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029
Re: [asterisk-users] Asterisk 12.4.0 not able to install pjsip
1) What platform are you on (i.e. Ubuntu/Centos/etc) 2) What steps did you take to install the PJSIP libraries? On Wed, Jul 23, 2014 at 7:30 AM, Sameer Rathod sam...@hostnsoft.com wrote: Hi, I had tried all the steps which I used to inatall Asterisk 12.3.2 Pjsip in Asterisk 12.3.2 is working but in new release Asterisk 12.4.0 it is not working I am getting XXX in make menuselect resource_module. I tried all trouble shooting steps along with ldconfig etc. I think its a bug can any one help me on this ? -- Regards Sameer Rathod 8109413462 -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] Limit Asterisk
Your bottleneck is most likely your drive bandwidth. Even with SAS drives, you'll need to move to a raid 5+ solution with 6+ drives to continue to increase the concurrent calls, or use a storage appliance. To confirm this, install the tool nmon and use the v and d options to bring up the resource usage indicators and drive busy/throughput statistics. On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones edua...@ypytecnologia.com.br wrote: people I have a running Asterisk 1.8.28 in great Dell server with two xeon processors and 16gb of ram and HD SAS 15k (Raid 1). This server is recording all calls (placed to record the audio in a ram disk), the entire CDR goes straight to MySQL by cdr_mysql.so. Each call runs some validation and AGI's have an auto dialer system that generates calls over the manager. Calls originate and terminate via SIP (no transcode). With this structure, even being a great server, we can not spend 150 simultaneous calls. When it reaches 140, the load average goes up a lot and the calls start to get very bad audio, tear, etc.. Using the top we see that all the processing is for asterisk. In this scenario, I think there is some limitation in Asterisk, or even the manager due to the auto dialer. Can anyone give me any tips where I can look where is the bottleneck? I need to get at least 250 calls that server quality. tks -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] Simultaneous Ring
I have this working but I have one problem. I need to grab values from
variables that I have set in the calling context to dial. How would I do that.
[tbs-utils]
exten = s,1,NoOp(Entering tbs-utils for extension ${ARG1})
;Set local variables to be used in the call
same = n,Set(NUMBER=${ARG1})
same = n,Set(GLOBAL(DIALGROUP1)=)
same = n,Set(GLOBAL(DIALGROUP2)=)
same = n,Set(_VM=)
same = n,Set(_TIMER1=)
same = n,Set(_TIMER2=)
same = n,Set(BRANCH=)
same = n,Set(_TO_VM=0)
;Check to see if the Primary SIP trunk is up
same = n,Set(NETWORKSTATUS=${SIPPEER(${GLOBAL(TRUNK1)},status)})
;Setting the TRUNK variable based upon the status of whether Trunk1 is reachable
same =
n,Set(TRUNK=${IF($[$[NETWORKSTATUS=UNREACHABLE]]?${GLOBAL(TRUNK2)}:${GLOBAL(TRUNK1)})})
;Calling the agi script
same = n,AGI(agi://localhost/tbs.agi)
;Displaying the values of the variables set in the agi script
same = n,NoOp(Branch number is: ${BRANCH})
same = n,NoOp(DIALGROUP1 is: ${DIALGROUP1})
same = n,NoOp(DIALGROUP2 is: ${DIALGROUP2})
same = n,NoOp(TIMER1 is: ${TIMER1})
same = n,NoOp(TIMER2 is: ${TIMER2})
same = n,NoOp(VM is: ${VM})
same = n,NoOp(TO_VM is: ${TO_VM})
;Check to see if we should go straight to VM
same = n,Gotoif($[${TO_VM} = 1]?200:)
;Dial the primary number and to to the return status
same = n,Dial(Local/Group1-101@DelayLocal/Group2-101@Delay,30)
same = n,Hangup();
[Delay]
;Dial Group 1
exten = Group1-101,1,Verbose(2,Dialing Group 1 set of phones
${GLOBAL(DIALGROUP1)})
same = n,Dial(${DIALGROUP1},20,t)
;Dial Group 2
exten = Group2-101,1,Verbose(2,Dialing Group2 set of phones)
same = n,Verbose(2, Waiting 10 seconds before dialing)
same = n,Wait(10)
same = n,Dial(${DIALGROUP2},${TIMER2},t)
Thanks,
Scott Haley
5-2244
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Thursday, July 17, 2014 6:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Simultaneous Ring
Thanks AJ, this sounds like what I need.
Thanks,
Scott Haley
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rights reserved.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles
Sent: Thursday, July 17, 2014 2:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Simultaneous Ring
On Wednesday 16 Jul 2014, Haley,Scott A wrote:
I have a need to issue a dial command to a number:
same = n,Dial(${DIALGROUP1},${TIMER1},t)
After a number of seconds, let's say 10 seconds. I want to dial
another set of numbers while continuing to ring, or interrupting the
first group of numbers.
same = n,Dial(${DIALGROUP2},${TIMER1},t)
Is there a way to do this without interrupting the first call?
This sounds exactly like the sort of situation for which local channels were
invented .
Dial(${DIALGROUP1}LOCAL/foo@bar) with a longer timeout than 10 seconds. Then
in your local channel, wait 10 and Dial(${DIALGROUP2}). The first Dial() will
be satisfied when someone answers either a phone in dial group 1, or a phone in
dial group 2 set ringing by the Dial() in the local channel.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
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Re: [asterisk-users] Simultaneous Ring
That worked. I had to use the *two* underscores in the agi script where I was setting the values. Thanks. Thanks, Scott Haley If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Friday, July 18, 2014 9:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Simultaneous Ring On Friday 18 Jul 2014, Haley,Scott A wrote: I have this working but I have one problem. I need to grab values from variables that I have set in the calling context to dial. How would I do that. I think you need to prefix your variable names with *two* underscores, to make them indefinitely heritable down the succession of channels. If they are prefixed with a single underscore, then they only get inherited *once*; so if the child channel spawns a grandchild, then any _VARS it inherited from the parent channel won't exist in the grandchild, but any __VARS will. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] Simultaneous Ring
Thanks AJ, this sounds like what I need.
Thanks,
Scott Haley
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or if you have received this message in error, immediately notify us and
delete it and any attachments.
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D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO
63131 © Edward Jones. All rights reserved.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles
Sent: Thursday, July 17, 2014 2:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Simultaneous Ring
On Wednesday 16 Jul 2014, Haley,Scott A wrote:
I have a need to issue a dial command to a number:
same = n,Dial(${DIALGROUP1},${TIMER1},t)
After a number of seconds, let's say 10 seconds. I want to dial
another set of numbers while continuing to ring, or interrupting the
first group of numbers.
same = n,Dial(${DIALGROUP2},${TIMER1},t)
Is there a way to do this without interrupting the first call?
This sounds exactly like the sort of situation for which local channels were
invented .
Dial(${DIALGROUP1}LOCAL/foo@bar) with a longer timeout than 10 seconds. Then
in your local channel, wait 10 and Dial(${DIALGROUP2}). The first Dial() will
be satisfied when someone answers either a phone in dial group 1, or a phone in
dial group 2 set ringing by the Dial() in the local channel.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
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[asterisk-users] Simultaneous Ring
I have a need to issue a dial command to a number:
same = n,Dial(${DIALGROUP1},${TIMER1},t)
After a number of seconds, let's say 10 seconds. I want to dial another set of
numbers while continuing to ring, or interrupting the first group of numbers.
same = n,Dial(${DIALGROUP2},${TIMER1},t)
Is there a way to do this without interrupting the first call?
Thanks,
Scott Haley
If you are not the intended recipient of this message (including attachments),
or if you have received this message in error, immediately notify us and delete
it and any attachments.
If you do not wish to receive any email messages from us, excluding
administrative communications, please email this request to
messa...@edwardjones.com along with the email address you wish to unsubscribe.
For important additional information related to this email, visit
www.edwardjones.com/US_email_disclosurehttp://www.edwardjones.com/US_email_disclosure.
Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St.
Louis, MO 63131 © Edward Jones. All rights reserved.
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Re: [asterisk-users] recording in mp3
You will not be able to able to save much space if any by using MP3 instead of ulaw or wav -- at least not without expending a lot of CPU time to encode the file at a very low bitrate which sounds pretty bad even with just speech. One of the better space savings options for recordings or voicemail is gsm. Of course, using an MP3 format just because you prefer that is understandable. Additionally, I'm nearly 100% certain that Asterisk does not support encoding and directly writing MP3 files. On Mon, Jun 30, 2014 at 3:11 PM, andrew Colin and...@vsave.co.za wrote: Hey guys Is it possible to record with mixmonitor straight into mp3. I am trying to reduce disk space and want my calls to be recorded in mp3 Instead of wav. Sent from Samsung Mobile Original message From: Sameer Rathod Date:30/06/2014 9:23 PM (GMT+02:00) To: asterisk-users@lists.digium.com Subject: [asterisk-users] Fwd: Regarding packet2packet bridging Dear concern, I want to configure packet2packet bridging in asterisk. How could I do this any of the tutorial or instructions will help ? I found the setting the canreinvite=yes will do the stuff but it is not working I am using asterisk 12.3 version I am very new to asterisk please help me in doing the same. Thanks in advance. -- Regards Sameer Rathod 8109413462 -- Regards Sameer Rathod 8109413462 -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] wct4xxp Excessive Interrupts Resulting in Unusable System or Card
reveals that the card generated 100,000 interrupts without being serviced and the kernel disabled it (and also reveals that the card is apparently on its own IRQ): maintenance@sip:~$ cat /proc/interrupts CPU0 CPU1 0: 46 0 IO-APIC-edge timer 1: 10 0 IO-APIC-edge i8042 7: 1 0 IO-APIC-edge 8: 0 0 IO-APIC-edge rtc0 9: 0 0 IO-APIC-fasteoi acpi 12: 4 0 IO-APIC-edge i8042 14: 0 0 IO-APIC-edge pata_amd 15: 0 0 IO-APIC-edge pata_amd 16:304 0 IO-APIC-fasteoi nouveau 19: 1221 0 IO-APIC-fasteoi eth1 21: 8681 0 IO-APIC-fasteoi sata_nv 22: 0 0 IO-APIC-fasteoi ehci_hcd:usb1 23: 0 0 IO-APIC-fasteoi ohci_hcd:usb2 25: 10 1 IO-APIC-fasteoi wct4xxp NMI: 1 1 Non-maskable interrupts LOC: 17884 19728 Local timer interrupts SPU: 0 0 Spurious interrupts PMI: 1 1 Performance monitoring interrupts IWI: 1554815 IRQ work interrupts RTR: 0 0 APIC ICR read retries RES: 6566 8577 Rescheduling interrupts CAL:220 4521 Function call interrupts TLB:638504 TLB shootdowns TRM: 0 0 Thermal event interrupts THR: 0 0 Threshold APIC interrupts MCE: 0 0 Machine check exceptions MCP: 1 1 Machine check polls ERR: 1 MIS: 0 Any ideas on how I can further diagnose and pursue this? Google does not reveal much related to this issue that is useful. Thank you! -- Scott L. Lykens Keystone Medical Management Solutions, Inc. +1 814 325-7500 x501 -- www.kmmsinc.comhttp://www.kmmsinc.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] wct4xxp Excessive Interrupts Resulting in Unusable System or Card
Just to be sure, what's the output of vmstat 10 10? From within a minute or so of the system starting, keep in mind that the TE410P’s IRQ is disabled so the sys value is not representative of actual use had it been. maintenance@sip:~$ vmstat 10 10 procs ---memory-- ---swap-- -io -system-- --cpu- r b swpd free buff cache si sobibo in cs us sy id wa st 0 0 0 7714300 42712 17629200 45837 369 369 1 3 91 4 0 0 0 0 7714336 42720 17632400 0 4 194 396 0 0 99 0 0 0 0 0 7714676 42720 17632400 0 5 197 397 0 0 100 0 0 0 0 0 7714732 42736 17632400 0 8 216 443 0 0 99 0 0 0 0 0 7714736 42744 17632400 0 2 195 395 0 0 99 0 0 0 0 0 7714736 42744 17632400 0 0 200 420 0 0 99 0 0 0 0 0 7714712 42752 17632400 0 4 205 414 0 0 99 0 0 0 0 0 7714760 42804 17632400 023 216 430 0 0 98 2 0 0 0 0 7714756 42812 17632400 0 4 201 409 0 0 99 0 0 Thank you. -- Scott L. Lykens Keystone Medical Management Solutions, Inc. +1 814 325-7500 x501 -- www.kmmsinc.comhttp://www.kmmsinc.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] wct4xxp Excessive Interrupts Resulting in Unusable System or Card
On Jun 1, 2014, at 11:01 AM, jg webaccounts...@jgoettgens.de wrote: Yes, I can see this. Another thing to check would be to start from a different OS (eg from a USB stick) and see how the card behaves on the otherwise same hardware. Since your ProLiant G2 server is almost 10 years old, and the TE410P works with 3.3V only (http://www.digium.com/en/products/telephony-cards/digital/quad-span), it might be worth to check this. The server is equipped with a 3.3v PCI-X slot. (https://h10057.www1.hp.com/ecomcat/hpcatalog/specs/provisioner/05/411095-421.htm). It is an old server but it has worked just fine for the task of hosting Asterisk for some time and I prefer not to spend $2,000+ to replace both the server and the PCI card with more modern hardware. Admittedly, the TE410P is new to the equation in the last several months but only in the last few weeks has this really become a problem to the point of affecting use. In fact, I was on a call Thursday morning for about an hour that was entirely SIP but during that time the system started blocking and other users could no longer make calls - even though my call was unaffected. The server is equipped with an AMD 8132 PCI-X bridge which apparently is known for being difficult in regards to interrupts. Google reveals that a few drivers have workarounds related to this chipset and to a range of revisions that mine happens to fall into. I will build a live-cd based usb key later on today and test the hardware independent of its present OS. Thank you. Scott -- _ -- 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] Trunk issue
I am trying to run an agi script and asterisk is not finding it. The output of the cli is as follows: -- Executing [s@tbs-utils:7] AGI(SIP/7002-001a, tbsdial.agi) in new stack [Apr 28 12:00:05] WARNING[21812][C-000f]: res_agi.c:1681 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File does not exist. The file is in that directory and is owned by the user asterisk. Why does it say the file does not exist? Thanks, Scott Haley 5-2244 If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -- _ -- 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] Trunk issue
It runs but hangs with the output of: perl tbsdial.agi 81101 GET VARIABLE astexten Right now, it is a simple perl script. Here is the entire script. #!/usr/bin/perl use Asterisk::AGI; my $agi = new Asterisk::AGI; my $dialgroup1 = DIALGROUP1; my $dialgroup2 = DIALGROUP2; my $vmvariable = VM; my $timer = TIMER; my $branch = BRANCH; my $input; my $dg1value; my $dg2value; my $vmvalue; my $branchvalue; $input = $agi-get_variable(astexten); #$agi-answer(); #$agi-stream_file(welcome); $agi-set_variable($dialgroup1, $dg1value); $agi-set_variable($dialgroup2, $dg2value); $agi-set_variable($vmvariable, $vmvalue); $agi-set_variable($timer, $timervalue); $agi-set_variable($branch, $branchvalue); Thanks, Scott Haley 5-2244 If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Monday, April 28, 2014 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk issue Does the script generate an error when run outside of Asterisk? An AGI should simply wait for input when run outside of Asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A Sent: Monday, April 28, 2014 1:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk issue I am trying to run an agi script and asterisk is not finding it. The output of the cli is as follows: -- Executing [s@tbs-utils:7] AGI(SIP/7002-001a, tbsdial.agi) in new stack [Apr 28 12:00:05] WARNING[21812][C-000f]: res_agi.c:1681 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File does not exist. The file is in that directory and is owned by the user asterisk. Why does it say the file does not exist? -- _ -- 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] Trunk issue
Here is the directory listing: [root@nxdasterisk-3 agi-bin]# ls -al total 12 drwxr-xr-x. 2 asterisk asterisk 4096 Apr 28 12:11 . drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 .. -rwxrwxr-x. 1 asterisk asterisk 590 Apr 28 11:55 tbsdial.agi Thanks, Scott Haley 5-2244 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Monday, April 28, 2014 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk issue Odd. AGI scripts should hang waiting for input when run from the CLI. They should not output anything. If the script is not set as executable you'd get an error. If you were not running it as the same user as asterisk runs as you should still get a different error. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A Sent: Monday, April 28, 2014 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk issue It runs but hangs with the output of: perl tbsdial.agi 81101 GET VARIABLE astexten Right now, it is a simple perl script. Here is the entire script. #!/usr/bin/perl use Asterisk::AGI; my $agi = new Asterisk::AGI; my $dialgroup1 = DIALGROUP1; my $dialgroup2 = DIALGROUP2; my $vmvariable = VM; my $timer = TIMER; my $branch = BRANCH; my $input; my $dg1value; my $dg2value; my $vmvalue; my $branchvalue; $input = $agi-get_variable(astexten); #$agi-answer(); #$agi-stream_file(welcome); $agi-set_variable($dialgroup1, $dg1value); $agi-set_variable($dialgroup2, $dg2value); $agi-set_variable($vmvariable, $vmvalue); $agi-set_variable($timer, $timervalue); $agi-set_variable($branch, $branchvalue); Thanks, Scott Haley 5-2244 If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Monday, April 28, 2014 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk issue Does the script generate an error when run outside of Asterisk? An AGI should simply wait for input when run outside of Asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A Sent: Monday, April 28, 2014 1:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk issue I am trying to run an agi script and asterisk is not finding it. The output of the cli is as follows: -- Executing [s@tbs-utils:7] AGI(SIP/7002-001a, tbsdial.agi) in new stack [Apr 28 12:00:05] WARNING[21812][C-000f]: res_agi.c:1681 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File does not exist. The file is in that directory and is owned by the user asterisk. Why does it say the file does not exist? -- _ -- 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
Re: [asterisk-users] Trunk issue
One more thing. I have this exact same script working on an Asterisk 1.8 box. This is a new Asterisk 11.7 box. Thanks, Scott Haley 5-2244 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A Sent: Monday, April 28, 2014 12:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk issue Here is the directory listing: [root@nxdasterisk-3 agi-bin]# ls -al total 12 drwxr-xr-x. 2 asterisk asterisk 4096 Apr 28 12:11 . drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 .. -rwxrwxr-x. 1 asterisk asterisk 590 Apr 28 11:55 tbsdial.agi Thanks, Scott Haley 5-2244 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Monday, April 28, 2014 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk issue Odd. AGI scripts should hang waiting for input when run from the CLI. They should not output anything. If the script is not set as executable you'd get an error. If you were not running it as the same user as asterisk runs as you should still get a different error. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A Sent: Monday, April 28, 2014 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk issue It runs but hangs with the output of: perl tbsdial.agi 81101 GET VARIABLE astexten Right now, it is a simple perl script. Here is the entire script. #!/usr/bin/perl use Asterisk::AGI; my $agi = new Asterisk::AGI; my $dialgroup1 = DIALGROUP1; my $dialgroup2 = DIALGROUP2; my $vmvariable = VM; my $timer = TIMER; my $branch = BRANCH; my $input; my $dg1value; my $dg2value; my $vmvalue; my $branchvalue; $input = $agi-get_variable(astexten); #$agi-answer(); #$agi-stream_file(welcome); $agi-set_variable($dialgroup1, $dg1value); $agi-set_variable($dialgroup2, $dg2value); $agi-set_variable($vmvariable, $vmvalue); $agi-set_variable($timer, $timervalue); $agi-set_variable($branch, $branchvalue); Thanks, Scott Haley 5-2244 If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Monday, April 28, 2014 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk issue Does the script generate an error when run outside of Asterisk? An AGI should simply wait for input when run outside of Asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A Sent: Monday, April 28, 2014 1:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk issue I am trying to run an agi script and asterisk is not finding it. The output of the cli is as follows: -- Executing [s@tbs-utils:7] AGI(SIP/7002-001a, tbsdial.agi) in new stack [Apr 28 12:00:05] WARNING[21812][C-000f]: res_agi.c:1681 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File does not exist. The file is in that directory and is owned by the user asterisk. Why does it say the file does not exist? -- _ -- 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
Re: [asterisk-users] Trunk issue
Now I am getting Permission denied. -- Executing [4000@phones:1] NoOp(SIP/7001-003a, Starting TBS Dailer App) in new stack -- Executing [4000@phones:2] NoOp(SIP/7001-003a, 4000) in new stack -- Executing [4000@phones:3] Gosub(SIP/7001-003a, tbs-utils,s,1,(4000)) in new stack -- Executing [s@tbs-utils:1] NoOp(SIP/7001-003a, Entering tbs-utils for 4000) in new stack -- Executing [s@tbs-utils:2] Set(SIP/7001-003a, DIALGROUP1=) in new stack -- Executing [s@tbs-utils:3] Set(SIP/7001-003a, DIALGROUP2=) in new stack -- Executing [s@tbs-utils:4] Set(SIP/7001-003a, VM=) in new stack -- Executing [s@tbs-utils:5] Set(SIP/7001-003a, TIMER=) in new stack -- Executing [s@tbs-utils:6] Set(SIP/7001-003a, BRANCH=) in new stack -- Executing [s@tbs-utils:7] AGI(SIP/7001-003a, tbsdial.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/tbsdial.agi tbsdial.agi: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': Permission denied Thanks, Scott Haley 5-2244 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asghar Mohammad Sent: Monday, April 28, 2014 12:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk issue if that is the case then check again Perl Asterisk AGI. On Mon, Apr 28, 2014 at 7:33 PM, Haley,Scott A scott.ha...@edwardjones.commailto:scott.ha...@edwardjones.com wrote: One more thing. I have this exact same script working on an Asterisk 1.8 box. This is a new Asterisk 11.7 box. Thanks, Scott Haley 5-2244 -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A Sent: Monday, April 28, 2014 12:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk issue Here is the directory listing: [root@nxdasterisk-3 agi-bin]# ls -al total 12 drwxr-xr-x. 2 asterisk asterisk 4096 Apr 28 12:11 . drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 .. -rwxrwxr-x. 1 asterisk asterisk 590 Apr 28 11:55 tbsdial.agi Thanks, Scott Haley 5-2244 -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Monday, April 28, 2014 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk issue Odd. AGI scripts should hang waiting for input when run from the CLI. They should not output anything. If the script is not set as executable you'd get an error. If you were not running it as the same user as asterisk runs as you should still get a different error. -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A Sent: Monday, April 28, 2014 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk issue It runs but hangs with the output of: perl tbsdial.agi 81101 GET VARIABLE astexten Right now, it is a simple perl script. Here is the entire script. #!/usr/bin/perl use Asterisk::AGI; my $agi = new Asterisk::AGI; my $dialgroup1 = DIALGROUP1; my $dialgroup2 = DIALGROUP2; my $vmvariable = VM; my $timer = TIMER; my $branch = BRANCH; my $input; my $dg1value; my $dg2value; my $vmvalue; my $branchvalue; $input = $agi-get_variable(astexten); #$agi-answer(); #$agi-stream_file(welcome); $agi-set_variable($dialgroup1, $dg1value); $agi-set_variable($dialgroup2, $dg2value); $agi-set_variable($vmvariable, $vmvalue); $agi-set_variable($timer, $timervalue); $agi-set_variable($branch, $branchvalue); Thanks, Scott Haley 5-2244 If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.commailto:messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosurehttp://www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun
Re: [asterisk-users] Trunk issue
That seemed to fix it. Thanks to everyone. Thanks, Scott Haley 5-2244 If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Monday, April 28, 2014 12:58 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Trunk issue On 28-04-14 19:49, Haley,Scott A wrote: Now I am getting Permission denied. Have you checked if SELinux is blocking the app? Any blockage should show up as an 'AVC' in /var/log/audit/audit.log You can temporarily set SELinux to permissive with 'setenforce 0' and check if the problem goes away. HTH, Patrick -- _ -- 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] Trunk issue
It is just plain Asterisk. I solved the original problem of it not being in the from-pstn context, now I am getting a rejected error I believe from the CM. Thanks, Scott Haley 5-2244 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of richard.seg...@marisec.ca Sent: Wednesday, April 23, 2014 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Trunk issue Are you using freeswitch, or just plain asterisk? I just setup a trunk between Asterisk and CM this morning, and it works great providing that you allow for anonymous calls. -Original Message- From: Haley,Scott A scott.ha...@edwardjones.com Sent: Wednesday, April 23, 2014 9:36am To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Subject: [asterisk-users] Trunk issue -- _ -- 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-usersI have setup a trunk on Asterisk 11.7 to an Avaya Session Manager. Every time I try to send a call over it, the call gets rejected. Here is the sip debug trace. Could anyone tell me what may be going wrong? nxdasterisk-2*CLI [Apr 23 08:20:59] WARNING[19047]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/scott.call: Operation not permitted Audio is at 18380 Adding codec 14 (alaw) to SDP Adding codec 100012 (g722) to SDP Adding codec 13 (ulaw) to SDP Reliably Transmitting (no NAT) to 192.168.175.135:5060: INVITE sip:913145152244@192.168.175.135 SIP/2.0 Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632 Max-Forwards: 70 From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f To: sip:913145152244@192.168.175.135 Contact: sip:3145152000@192.168.122.57:5060 Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.7.0 Date: Wed, 23 Apr 2014 13:20:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 229 v=0 o=root 424150695 424150695 IN IP4 192.168.122.57 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.122.57 t=0 0 m=audio 18380 RTP/AVP 8 9 0 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- --- SIP read from UDP:192.168.175.135:5060 --- SIP/2.0 100 Trying Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060 CSeq: 102 INVITE From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f To: sip:913145152244@192.168.175.135 Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632 Content-Length: 0 - --- (7 headers 0 lines) --- --- SIP read from UDP:192.168.175.135:5060 --- INVITE sip:913145152...@devjones.com SIP/2.0 P-AV-Message-Id: 1_1 Route: sip:192.168.122.57;lr;phase=terminating Supported: replaces, timer Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Date: Wed, 23 Apr 2014 13:20:59 GMT Contact: sip:3145152000@192.168.122.57:5060;gsid=d13ae820-caef-11e3-9b9c-6c3be5a59e68 Via: SIP/2.0/UDP 192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4 Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967 Via: SIP/2.0/UDP 192.168.175.130:15060;rport;ibmsid=local.1389145532068_1778704_1816625;branch=z9hG4bK605195140054947 Via: SIP/2.0/UDP 192.168.175.135;rport=5060;branch=z9hG4bK6add1632-AP;ft=192.168.175.135~13c4;received=192.168.175.135 Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632 Record-Route: sip:2ca13a6d@192.168.175.135;transport=udp;lr Record-Route: sip:192.168.175.130:15060;transport=udp;ibmsid=local.1389145532068_1778704_1816625;lr Record-Route: sip:2ca13a6d@192.168.175.135;transport=udp;lr P-Charging-Vector: icid-value=d13ae820-caef-11e3-9b9c-6c3be5a59e68 User-Agent: Asterisk PBX 11.7.0 AVAYA-SM-6.3.1.0.631004 P-Asserted-Identity: Edward Jones sip:3145152...@devjones.com From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f To: sip:913145152244@192.168.175.135 Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060 Max-Forwards: 66 CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 229 Av-Global-Session-ID: d13ae820-caef-11e3-9b9c-6c3be5a59e68 P-Location: SM;origlocname=Asterisk-2;origsiglocname=Asterisk-2;origmedialocname=Asterisk-2;termlocname=Asterisk-2;termsiglocname=Asterisk-2;smaccounting=true v=0 o=root 424150695 424150695 IN IP4 192.168.122.57 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.122.57 t=0 0 m=audio 18380 RTP/AVP 8 9 0 a=rtpmap:8 PCMA/8000
Re: [asterisk-users] Asterisk -rx, how expensive is it? Should you avoid spamming it?
How much Asterisk is affected depends on both how often you run a command, and even more significantly, what command you run (and which version of Asterisk). Commands that display information about every active channel, for example sip show peers, may slow other processing significantly because they have to briefly lock the data structures to insure valid information. There have been improvements in more recent versions of Asterisk that reduce the negative affects of this by looking at cached information instead of locking everything. On the other hand, requesting specific information (sip show peer X) or more generic information (sip show inuse) will have much less affect on other activity in Asterisk. On Thu, Apr 24, 2014 at 5:20 AM, Mikael Fredin mik...@wiraya.com wrote: Just like the subject sais - how expensive is it to execute a lot of these commands to keep track of different things in asterisk? I have avoided doing this because it feels a bit like a risk to spam the asterisk CLI this way, but is it really? CPU-wise it doesn't seem very expensive to do it 100 times a second (from a simple test I did), but is it possible it will affect the asterisk service in any other negative way? Regards, Mikael -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] Asterisk -rx, how expensive is it? Should you avoid spamming it?
That's a good point also - if you're doing something automated, AMI is likely a better option. The connection to Asterisk is persistent, and information output is structured and we take pains not to break the API definition, which is not true of CLI output. On Thu, Apr 24, 2014 at 12:47 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Thu, Apr 24, 2014 at 12:20:37PM +0200, Mikael Fredin wrote: Just like the subject sais - how expensive is it to execute a lot of these commands to keep track of different things in asterisk? I have avoided doing this because it feels a bit like a risk to spam the asterisk CLI this way, but is it really? CPU-wise it doesn't seem very expensive to do it 100 times a second (from a simple test I did), but is it possible it will affect the asterisk service in any other negative way? It feels very expensive. Part of it is because of starting a new instance of Asterisk. It will not load any module and such, but if you care about speed, you can use netcat (it takes some care). You'll also encounter some artificial delays in the response which make it feel more expensive. The main reason to avoid it is because its output is not intended for automated parsing. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] Asterisk -rx, how expensive is it? Should you avoid spamming it?
The Stasis message bus and caching is introduced in Asterisk 12. https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+API+Improvements Note that as it's fairly new, in some cases older code may still lock data structures during operations rather than read the cache. You will also want to see if ARI ( https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI) can provide what you need. This is newer code and more likely to use the cache and be efficient. On Thu, Apr 24, 2014 at 2:12 PM, Mikael Fredin mik...@wiraya.com wrote: Thank you, that's very useful information! Does the same go for issuing a sip show peers through the AMI? And do you know where I could find information of what asterisk versions may use cached information instead? What would you suggest be better ways to monitor asterisk information? On 24 April 2014 17:58, Scott Griepentrog sgriepent...@digium.com wrote: How much Asterisk is affected depends on both how often you run a command, and even more significantly, what command you run (and which version of Asterisk). Commands that display information about every active channel, for example sip show peers, may slow other processing significantly because they have to briefly lock the data structures to insure valid information. There have been improvements in more recent versions of Asterisk that reduce the negative affects of this by looking at cached information instead of locking everything. On the other hand, requesting specific information (sip show peer X) or more generic information (sip show inuse) will have much less affect on other activity in Asterisk. On Thu, Apr 24, 2014 at 5:20 AM, Mikael Fredin mik...@wiraya.com wrote: Just like the subject sais - how expensive is it to execute a lot of these commands to keep track of different things in asterisk? I have avoided doing this because it feels a bit like a risk to spam the asterisk CLI this way, but is it really? CPU-wise it doesn't seem very expensive to do it 100 times a second (from a simple test I did), but is it possible it will affect the asterisk service in any other negative way? Regards, Mikael -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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 -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] Trunk issue
=z9hG4bK605195140054947 Via: SIP/2.0/UDP 192.168.175.135;rport=5060;branch=z9hG4bK6add1632-AP;ft=192.168.175.135~13c4;received=192.168.175.135 Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632 From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f To: sip:913145152244@192.168.175.135;tag=as119fde8b Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060 CSeq: 102 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 [Apr 23 08:20:59] NOTICE[19026][C-0003]: chan_sip.c:25450 handle_request_invite: Call from 'SMtrunk' (192.168.175.135:5060) to extension '913145152244' rejected because extension not found in context 'from-pstn'. Scheduling destruction of SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060' in 32000 ms (Method: INVITE) --- SIP read from UDP:192.168.175.135:5060 --- ACK sip:913145152...@devjones.com SIP/2.0 Route: sip:192.168.122.57;lr;phase=terminating Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060 From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f To: sip:913145152244@192.168.175.135;tag=as119fde8b Via: SIP/2.0/UDP 192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4 Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967 CSeq: 102 ACK Max-Forwards: 66 Content-Length: 0 - --- (10 headers 0 lines) --- Really destroying SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060' Method: ACK --- SIP read from UDP:192.168.175.135:5060 --- SIP/2.0 403 Forbidden (Denial 1732) Av-Global-Session-ID: d13ae820-caef-11e3-9b9c-6c3be5a59e68 Server: Avaya CM/R016x.02.0.823.0 AVAYA-SM-6.3.1.0.631004 Warning: 399 192.168.175.252 Restricted Access To: sip:913145152244@192.168.175.135;tag=8072a3b71bcde31d444535cfeab00 From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632 Content-Length: 0 - --- (10 headers 0 lines) --- Transmitting (no NAT) to 192.168.175.135:5060: ACK sip:913145152244@192.168.175.135 SIP/2.0 Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632 Max-Forwards: 70 From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f To: sip:913145152244@192.168.175.135;tag=8072a3b71bcde31d444535cfeab00 Contact: sip:3145152000@192.168.122.57:5060 Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 11.7.0 Content-Length: 0 --- [Apr 23 08:20:59] WARNING[19026][C-0002]: chan_sip.c:22991 handle_response_invite: Received response: Forbidden from 'Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f' Scheduling destruction of SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060' in 32000 ms (Method: INVITE) [Apr 23 08:20:59] NOTICE[19157]: pbx_spool.c:389 attempt_thread: Call failed to go through, reason (1) Hangup [Apr 23 08:20:59] NOTICE[19157]: pbx_spool.c:392 attempt_thread: Queued call to SIP/SMtrunk/913145152244 expired without completion after 0 attempts Thanks, Scott Haley IS Voice Projects Team Edward Jones Investments Phone: 314-515-2244 Email: scott.ha...@edwardjones.commailto:scott.ha...@edwardjones.com If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosurehttp://www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -- _ -- 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 dropped calls
I would suggest starting with a packet capture of the SIP messages that will include both call legs (i.e. capture at the Asterisk box). This should tell you who initiated the hangup - the carrier side, the phone side, or Asterisk. On Wed, Mar 26, 2014 at 11:46 AM, Mike Diehl mdiehlena...@gmail.com wrote: Hi all, I have a user who is reporting dropped calls at his site. We don't have any other users complaining of this. So far, this is what we know: 1. The manager bought all new Polycom phones. (POE) 2. They replaced the network switch with a POE version. 3. It's not just one or two of the phones that have problems. 4. It doesn't matter if they use the headset or the cordless set. 5. The ISP reports a very clean circuit. (Ethernet from the CLEC.) 6. We don't see their phones become unavailable very often. 7. They are the only site that seems to be having trouble. So, where else can/should I look? Mike. -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://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] SIP 603 Declined error message
I have a SIP trunk from my Asterisk server to an Avaya CM server. If I place calls inbound, everything works fine. If I place calls outbound, originating from the Asterisk box, everything works fine (I have done this with the use of the .call files). If I setup an extension with the findme-followme feature and have it try to hair-pin a call back out the same trunk to the Avaya, I get a SIP/2.0 603 Declined message. Here is the output. Any reason that this might be happening? It has been working up until now this week. I rebooted the machine on Tuesday. --- SIP read from TCP:172.17.184.46:31285 --- INVITE sip:51...@edj.devjones.com SIP/2.0 From: Haley, Scott sip:3145152...@edwardjones.com;tag=8066eb6f589ce3124b652973b4b00 To: sip:51...@edj.devjones.com Call-ID: 8066eb6f589ce3125b652973b4b00 CSeq: 1 INVITE Max-Forwards: 71 Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00 Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00 Supported: 100rel,histinfo,join,replaces,sdp-anat,timer Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,INFO,PRACK,PUBLISH,UPDATE User-Agent: Avaya CM/R016x.02.0.823.0 Contact: Haley, Scott sip:3145152244@172.17.184.46;transport=tcp Route: sip:192.168.122.51;transport=tcp;lr;phase=terminating Accept-Language: en;q=1 Alert-Info: cid:internal@edj.devjones.com;avaya-cm-alert-type=internal History-Info: sip:51...@edj.devjones.com;index=1 History-Info: 51104 sip:51...@edj.devjones.com;index=1.1 Min-SE: 1200 P-Asserted-Identity: Haley, Scott sip:3145152...@edwardjones.com Record-Route: sip:172.17.184.46;transport=tcp;lr Session-Expires: 1200;refresher=uac Content-Type: application/sdp Content-Length: 257 v=0 o=- 1393419743 1 IN IP4 172.17.184.46 s=- c=IN IP4 172.17.184.93 b=AS:64 t=0 0 a=avf:avc=n prio=n a=csup:avf-v0 m=audio 28196 RTP/AVP 0 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 - --- (23 headers 13 lines) --- Sending to 172.17.184.46:31285 (NAT) Using INVITE request as basis request - 8066eb6f589ce3125b652973b4b00 Found peer 'trunk503in' for '3145152244' from 172.17.184.46:31285 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 127 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 127 Capabilities: us - 0x100c (ulaw|alaw|g722), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) Peer audio RTP is at port 172.17.184.93:28196 Looking for 51104 in from-trunk-sip-trunk503out (domain edj.devjones.com) list_route: hop: sip:172.17.184.46;transport=tcp;lr --- Transmitting (NAT) to 172.17.184.46:31285 --- SIP/2.0 100 Trying Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285 Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00 Record-Route: sip:172.17.184.46;transport=tcp;lr From: Haley, Scott sip:3145152...@edwardjones.com;tag=8066eb6f589ce3124b652973b4b00 To: sip:51...@edj.devjones.com Call-ID: 8066eb6f589ce3125b652973b4b00 CSeq: 1 INVITE Server: FPBX-2.8.1(1.8.13.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1200;refresher=uac Contact: sip:51104@192.168.122.51:5060;transport=TCP Content-Length: 0 -- Executing [51104@from-trunk-sip-trunk503out:1] Set(SIP/trunk503in-010b, GROUP()=OUT_1) in new stack -- Executing [51104@from-trunk-sip-trunk503out:2] Goto(SIP/trunk503in-010b, from-trunk,51104,1) in new stack -- Goto (from-trunk,51104,1) -- Executing [51104@from-trunk:1] Set(SIP/trunk503in-010b, __FROM_DID=51104) in new stack -- Executing [51104@from-trunk:2] Gosub(SIP/trunk503in-010b, app-blacklist-check,s,1) in new stack -- Executing [s@app-blacklist-check:1] GotoIf(SIP/trunk503in-010b, 0?blacklisted) in new stack -- Executing [s@app-blacklist-check:2] Set(SIP/trunk503in-010b, CALLED_BLACKLIST=1) in new stack -- Executing [s@app-blacklist-check:3] Return(SIP/trunk503in-010b, ) in new stack -- Executing [51104@from-trunk:3] Gosub(SIP/trunk503in-010b, cidlookup,cidlookup_1,1) in new stack -- Executing [cidlookup_1@cidlookup:1] GotoIf(SIP/trunk503in-010b, 1?cidlookup,cidlookup_return,1) in new stack -- Goto (cidlookup,cidlookup_return,1) -- Executing [cidlookup_return@cidlookup:1] ExecIf(SIP/trunk503in-010b, 0?Set(CALLERID(name)=)) in new stack -- Executing [cidlookup_return@cidlookup:2] Return(SIP/trunk503in-010b, ) in new stack -- Executing [51104@from-trunk:4] ExecIf(SIP/trunk503in-010b, 0 ?Set(CALLERID(name)=3145152244)) in new stack
[asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Unknown)
2012/12/19 Scott Huang gyration.hu...@gmail.com Hi I've saw some similar case in the mail list, but seems no standard answers, so I decide ask here again. Is there anyone see the message below ? I use asterisk(1.8.11-cert 9) in my openbts2.8, and when I made a phone call, the Asterisk CLI poppd the following messages. = *CLI == Using SIP RTP CoS mark 5 -- Executing [8690@phones:1] Dial(SIP/IMSI466974600011287-, SIP/IMSI466974104638690) in new stack [Dec 18 16:00:49] WARNING[2934]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/IMSI466974600011287-' status is 'CHANUNAVAIL' == The attached files are the sip.conf and extension.conf and wireshark trace log. The part of my setting in sip.conf is: [IMSI466974104638690]; callerid=8690 8690 ; regexten=8690; canreinvite=no type=friend allow=gsm context=phones host=dynamic registertrying=yes [IMSI466974102820333]; callerid=0333 0333 ; regexten=0333; canreinvite=no type=friend allow=gsm context=phones host=dynamic registertrying=yes [IMSI466974600011287]; callerid=1287 1287 ; regexten=1287; canreinvite=no type=friend allow=gsm context=phones host=dynamic registertrying=yes The part of my setting in extensions.conf is: [phones] exten = 8690,1,Dial(SIP/IMSI466974104638690) exten = 0333,1,Dial(SIP/IMSI466974102820333) exten = 1287,1,Dial(SIP/IMSI466974600011287) How to exactly configure asterisk for a sip call ? Thanks very much ! BR/Scott -- _ -- 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] No rtpmap codec info in 200 OK
Ok. Asterisk sends the rtpmap info for the codec. Is it possible to remove this from the 200 OK sent by Asterisk? Possible direction I should look. -- _ -- 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] No rtpmap codec info in 200 OK
It seems quite unlikely that the presence of an 'a=rtpmap' line in the SDP for G.729 is going to cause a call to have any problems. Thanks for the reply. I'll expand on the scenario... This particular ATA does not send 'a=rtpmap' for any codec. When talking to a Asterisk PBX everything works fine. When talking to a VSP that sends an INVITE with User-Agent: Sippy the call is setup then drops after 32 seconds. Packet captures shows that no ACK is received after the ATA sends the 200 OK (missing rtpmap). After sending 200 OK about 6 times it then sends BYE and the call disconnects. Every other ATA I have sends rtpmap and works fine. The idea was to manipulate Asterisk into not sending rtpmap for the codec to confirm what happens. I'll now look for another solution. -- _ -- 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] No rtpmap codec info in 200 OK
On 20 December 2011 12:51, Bruce B bruceb...@gmail.com wrote: I could be wrong but this sounds like a NAT issue rather SIP related packet issue. I looked at this to start with. Spent sometime comparing addresses and ports between successful and failure packets. Couldn't see any ports that weren't opened on the way out or the use of private ip addresses. I cleared the nat translation table between tests. This ATA works fine with Asterisk based VSPs. I'm just going to have to get more methodical. FYI, the ATA is a GW211 (mass produced OEM device, this one labelled Cormain) and the VSP is Pennytel here in Australia. -- _ -- 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] No rtpmap codec info in 200 OK
On 20 December 2011 14:22, Bruce B bruceb...@gmail.com wrote: Can you register with Eyebeam to VSP and have it work? Make sure you are on the exact same network as the ATA when making this test. This should isolate the NAT issue. Great tip. Eyebeam dosen't send a rtpmap for known codecs unless you select the option too. Well, without it Eyebeam works fine so I better start looking at the firewall. Strange that this particular ATA fails with this particulat VSP only with three different firewalls... vyatta, microtik and a Billion modem/router. Thanks again. -- _ -- 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] No rtpmap codec info in 200 OK
Hi, My VSP uses Asterisk to which I'm connected with an ATA. When I receive an inbound call the invite includes the following... v=0 o=root 32218 32218 IN IP4 202.52.129.50 s=session c=IN IP4 202.52.129.50 t=0 0 m=audio 16864 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off – - – - a=ptime:20 a=sendrecv My ATA's 200 OK reply after call setup has the following... v=0 o=CMI-SIPUA 13369 0 IN IP4 211.30.XXX.XXX s=SIP CALL c=IN IP4 211.30.XXX.XXX t=0 0 m=audio 20216 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=rtcp:20217 a=silenceSupp:off – - – - a=sendrecv Notice there is no rtpmap:18 G729/8000 in the reply. The call continues fine. Is it right that there is no codec info in the reply and the call continues? -- _ -- 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] About X100P and TDM400P analog card in China
Hello. All. I am a bit new to asterisk, started from half a month ago. I am setting up a home asterisk server with analog card. I am using asterisk 1.4.27. At the moment, I bought a X100P card and installed it on my computer. I used it to connect my home phone line. For the moment, it works fine when dial in. Soon I noticed when I dial out through it to my mobile, it can't hang up automatically after I hang up my mobile. After googled, I found the reason as described as below link and some solutions. http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html For me, none of solutions works. So I am rethinking should I buy another TDM400P card. But I am wondering because in China. The phone system looks different so I don't know if TDM400P will work or not. Here is the flow when I am using X100P to dial out. 1. Pick up phone I hear tone. DA~~~ 2. press the number tone: DA~~~ 3. dialing No more tone. Music playing~(lalala, I love lalal) At the same time, on asterisk console, it prints out. The call has been answered. Actually it is still dialing and my mobile is ringing because I didn't answer the call.. The music was played by ISP 4. whether I answered the call or refuse the call. No more prints on asterisk console. But on phone end, when I refuse the call, instead of busytone, I hear the voice The phone you're dialing is busy now. Please try again later.. So the whole thing is, during the whole call process, only before dialing, we can hear the phone tone, for all other time, Dialing, refused, the ISP will play music/voice instead of providing the tone. I don't understand how x100p identify the status, I guess should be on the tone. 5. I wait asterisk/x100p to hangup the call and after 5 minutes, I have to cut the phone line to force it hang up. So can TDM400X work with such a system without tone only with music and voice? Thanks. Regards. Scott -- _ -- 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] About X100P and TDM400P analog card in China
So does this mean no solution when used ZAP/DAHDI with PSTN line? If I installed an E1, will that work? Thanks. Regards. On Wed, May 11, 2011 at 12:57 AM, John Novack jnov...@stromberg-carlson.org wrote: Remember that ZAP/DAHDI channels don't receive ( because most PSTN/POTS lines don't provide ) answer supervision. This will certainly complicate what you want do do. John Novack Scott Zhang wrote: Hello. All. I am a bit new to asterisk, started from half a month ago. I am setting up a home asterisk server with analog card. I am using asterisk 1.4.27. At the moment, I bought a X100P card and installed it on my computer. I used it to connect my home phone line. For the moment, it works fine when dial in. Soon I noticed when I dial out through it to my mobile, it can't hang up automatically after I hang up my mobile. After googled, I found the reason as described as below link and some solutions. http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html For me, none of solutions works. So I am rethinking should I buy another TDM400P card. But I am wondering because in China. The phone system looks different so I don't know if TDM400P will work or not. Here is the flow when I am using X100P to dial out. 1. Pick up phone I hear tone. DA~~~ 2. press the number tone: DA~~~ 3. dialing No more tone. Music playing~(lalala, I love lalal) At the same time, on asterisk console, it prints out. The call has been answered. Actually it is still dialing and my mobile is ringing because I didn't answer the call.. The music was played by ISP 4. whether I answered the call or refuse the call. No more prints on asterisk console. But on phone end, when I refuse the call, instead of busytone, I hear the voice The phone you're dialing is busy now. Please try again later.. So the whole thing is, during the whole call process, only before dialing, we can hear the phone tone, for all other time, Dialing, refused, the ISP will play music/voice instead of providing the tone. I don't understand how x100p identify the status, I guess should be on the tone. 5. I wait asterisk/x100p to hangup the call and after 5 minutes, I have to cut the phone line to force it hang up. So can TDM400X work with such a system without tone only with music and voice? Thanks. Regards. Scott -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] About X100P and TDM400P analog card in China
Thanks. I see. Regards. Scott On Wed, May 11, 2011 at 3:43 AM, John Novack jnov...@stromberg-carlson.orgwrote: Assuming you have read the link you provided, and understand most of what it said, the link really doesn't address calling out over a POTS (copper) line. When Asterisk dials out and finishes the dial string, it considers it answered. IF your POTS provider doesn't provide any clue, other than audio, that the line is answered, not answered, or the call terminates, then you will have to do some coding. You could set an absolute limit, or IF the call will always go to you, you could listen for some DTMF and hang up then. OR, if there is an option, you could use some sort of digital trunk, SIP or what have you, where there is more complete communication. SIP isn't the most desirable, IMO, as some of your countrymen ( and other counties s well ) seem to have nothing better to do than to attempt to break in to VOIP systems and steal telephone time. T1/E1 will certainly provide much better communication, as will ISDN. Remember the POTS analog technology was built and constantly modernized over the last 130 years, but was never designed for anything other than human communication. Once stupid machinery became involved, the problems became larger and larger. John Novack Scott Zhang wrote: So does this mean no solution when used ZAP/DAHDI with PSTN line? If I installed an E1, will that work? Thanks. Regards. On Wed, May 11, 2011 at 12:57 AM, John Novack jnov...@stromberg-carlson.org wrote: Remember that ZAP/DAHDI channels don't receive ( because most PSTN/POTS lines don't provide ) answer supervision. This will certainly complicate what you want do do. John Novack Scott Zhang wrote: Hello. All. I am a bit new to asterisk, started from half a month ago. I am setting up a home asterisk server with analog card. I am using asterisk 1.4.27. At the moment, I bought a X100P card and installed it on my computer. I used it to connect my home phone line. For the moment, it works fine when dial in. Soon I noticed when I dial out through it to my mobile, it can't hang up automatically after I hang up my mobile. After googled, I found the reason as described as below link and some solutions. http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html For me, none of solutions works. So I am rethinking should I buy another TDM400P card. But I am wondering because in China. The phone system looks different so I don't know if TDM400P will work or not. Here is the flow when I am using X100P to dial out. 1. Pick up phone I hear tone. DA~~~ 2. press the number tone: DA~~~ 3. dialing No more tone. Music playing~(lalala, I love lalal) At the same time, on asterisk console, it prints out. The call has been answered. Actually it is still dialing and my mobile is ringing because I didn't answer the call.. The music was played by ISP 4. whether I answered the call or refuse the call. No more prints on asterisk console. But on phone end, when I refuse the call, instead of busytone, I hear the voice The phone you're dialing is busy now. Please try again later.. So the whole thing is, during the whole call process, only before dialing, we can hear the phone tone, for all other time, Dialing, refused, the ISP will play music/voice instead of providing the tone. I don't understand how x100p identify the status, I guess should be on the tone. 5. I wait asterisk/x100p to hangup the call and after 5 minutes, I have to cut the phone line to force it hang up. So can TDM400X work with such a system without tone only with music and voice? Thanks. Regards. Scott -- _ -- 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 -- Dog is my Co-pilot -- Dog is my Co-pilot -- _ -- 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] Registration problems - Vitelity
Hi All- I have successfully routed calls into our asterisk system from several DID providers in the USA, but for some reason I'm having a problem getting Vitelity to work. We are using the IAX protocol, and the symptom is that only about 50% of the calls terminate properly into my asterisk system - the rest get a busy signal. The ones that do not come in don't show up at all on the asterisk console, even with IAX2 debug enabled. I know that the uncompleted calls are getting into the Vitelity server because I get an uncompleted call email every time one fails. It seems to be acting like the registration is falling off somehow. I have confirmed that I am registering successfully with the Vitelity server every 50 seconds, ie the Vitelity server is acknowledging every registration. So far, after a couple weeks of calls back and forth with Vitelity customer service, no progress has been made, however the Vitelity tech reps did make some vague references to other IAX users occasionally having registration issues. Anyone else having similar issues? 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
Re: [asterisk-users] [Dahdi 2.3.0.1] Does it need all those modules?
I just went through a Dahdi rebuild, and I seem to recall a message that all modules will be loaded until you set up the dahdi configuration files. regards Scott On 7/9/2010 11:41 AM, Gilles wrote: Hello To use Dahdi + Asterisk with a PCI card with a single FXO port, I just... 1. compiled and installed Dahdi 2. edited /etc/modprobe.d/dahdi.blacklist.conf to blacklist netjet and unblacklist wctdm: == # cat /etc/modprobe.d/dahdi.blacklist.conf blacklist wct4xxp blacklist wcte12xp blacklist wct1xxp blacklist wcte11xp blacklist wctdm24xxp blacklist wcfxo #blacklist wctdm blacklist wctc4xxp blacklist wcb4xxp blacklist netjet == 3. rebooted, and checked that netjet was gone and wctdm was in: == # lsmod | grep -i wc wctc4xxp 32414 0 dahdi_transcode 5751 1 wctc4xxp wcb4xxp33905 0 wcfxo 8968 0 wctdm24xxp116684 0 wcte11xp 22995 0 wct1xxp12971 0 wcte12xp 26308 0 dahdi_voicebus 39947 2 wctdm24xxp,wcte12xp wct4xxp 230713 0 wctdm 35677 0 dahdi 197809 11 xpp,dahdi_transcode,wcb4xxp,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp,wctdm crc_ccitt 1339 3 wctdm24xxp,dahdi,hisax == Does Dahdi really need all those modules, or is there another configuration file that I missed to disable unneeded modules? 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] Conditional includes in iax.conf
Hello- For maintenance purposes, if possible I'd like to use the same iax.conf file in several different asterisk systems. However, on one of the systems only, I would like to include an IAX register command to another external system. Within iax.conf or other configuration files (other than extensions.conf), is there a way of determining what system I'm running on, and include a particular configuration item conditionally? I guess what I'm asking is there a way to conditionally include lines in a configuration depending on the value of some linux environment variable? 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
Re: [asterisk-users] Conditional includes in iax.conf
On 7/7/2010 11:25 AM, Danny Nicholas wrote: -- Rather than trying to determine what system you are on, just make the included file be empty on all except the desired server. OK, thanks. I thought I might have to do it that way, which is slightly less desirable, as it makes the systems different from each other. cheers Scott -- _ -- 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] Conditional includes in iax.conf
On 7/7/2010 11:52 AM, Kevin P. Fleming wrote: On 07/07/2010 01:46 PM, Scott Stingel wrote: On 7/7/2010 11:25 AM, Danny Nicholas wrote: -- Rather than trying to determine what system you are on, just make the included file be empty on all except the desired server. OK, thanks. I thought I might have to do it that way, which is slightly less desirable, as it makes the systems different from each other. You could also enable 'execincludes' in asterisk.conf, then use #exec to execute a small script (even just a shell script) that outputs the desired iax.conf content for the server it is running on. That's much easier and more effective than trying to put conditional logic and other programming constructs into the configuration file reader. Ok, thanks Kevin. Something I haven't used before but will look into! -- _ -- 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] Warning spamming for any unsynchronized ISDN port with dahdi-2.3.0.1
On 6/30/2010 3:56 PM, Alex Villacís Lasso wrote: whenever an ISDN port is in RED alarm (unsynchronized), we get a stream of warnings in /var/log/asterisk/full that look like this: [Jun 30 17:38:41] WARNING[9637] chan_dahdi.c: No D-channels available! Using Primary channel 78 as D-channel anyway! [Jun 30 17:38:41] WARNING[9638] chan_dahdi.c: No D-channels available! .. question I have is this: is this warning message something to be expected from ports with RED alarms? Or is this message a symptom of a deeper misconfiguration? Alex- On my system (D410P) the above message appears when EITHER: (a) A span is configured in dahdi-channels.conf (or chan_dahdi.conf), but nothing is plugged into it OR (b) A span is configured in dahdi-channels.conf (or chan_dahdi.conf), an E1 is plugged in, BUT signalling type is incorrectly configured (pri_cpe vs. pri_net) I agree with the other person, that a single Red Alarm message would be preferable rather than have the above message repeat forever if nothing is plugged in. You can disable it if the lines are inactive by commenting out the configuration information in dahdi-channels.conf (or chan_dahdi.conf depending on your setup) Scott -- _ -- 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] Anybody using TE410P on BT ISDN with DAHDI?
On 6/22/2010 4:26 AM, A J Stiles wrote: Is anybody else using the following combination: * a TE410P card (wct4xxp driver) * a BT ISDN connection * DAHDI 2.3.0.1 * Asterisk 1.6.2.9 I'm trying to configure a new box to replace a legacy system (same hardware; some old version of Asterisk with Zaptel; works lovely but hopelessly out-of-date) and not having much joy. Specifically, I couldn't get it to see a D-channel on channel 16 of span 1. And without a D-channel, there is no way I'm going to be able to get a call in or out. This could well be because the syntax of modern /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf is slightly incompatible with the old zapata.conf and zaptel.conf files. So I guess the first question should be, has anybody else managed to make this combination work? (I'm new here and I may have missed some important information, so please ask.) Hi- I've been going through the same upgrade process recently, and had the same error (shown in your other message). I had forgotten that the equipment I was plugged in to was CPE, so I had to change my new setting for that span to NET rather than CPE. I notice in your old zapata files that you had CPE for two spans and NET for the other two, and your dahdi_chan setup is set up the same. But I'm thinking perhaps during testing you plugged a CPE on your new setup to a CPE on the other, which would produce the symptoms you see. -Scott -- _ -- 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] DAHDI: Inbound BRI call, DDI not presented
On 6/22/2010 2:03 AM, Tzafrir Cohen wrote: On Mon, Jun 21, 2010 at 09:08:02AM -0700, Scott Stingel wrote: Hello- I have a system with one D410P and one B200P (both OpenVox). All is well with the D410P, inbound and outbound, and I can initiate calls on the B200P BRI span, but there may be something wrong with my inbound BRI setup: there is no indication of an inbound call when I dial in to it from the PSTN. When I run pri intense debug and make a call to the BRI span, I can see a message containing the DDI that I'm dialing, in this case 336027 (BT supplies only the last 6 digits of a delivered number). See debug output below... Is there anything you see in the dialplan trace itself? Also, 'intense debug' shows a lot of noise of ISDN layer 2 (Q.921). But that is normally not interesting. Do you see anything on a simple 'pri debug span 1' (only layer 3 debug)? Have I neglected to set up some needed parameter? This all worked on older boards when using bristuff, but now I want to use dahdi. My client is in the UK, connected to BT, and I have specified euroisdn as the switch type. many thanks - (snippet during inbound call to 336027) Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 008 P/F: 1 0 bytes of data -- ACKing all packets from 8 to (but not including) 8 -- Stopping T200 timer -- Starting T203 timer Shouldn't an RR be sent back? Handling message for SAPI/TEI=0/0 TEI: 0 State 7 V(S) 8 V(A) 8 V(R) 8 K 1, RC 0, l3initiated 0, reject_except 0 ack_pend 0 T200 0, N200 3, T203 1 [ 02 ff 03 08 01 01 05 a1 04 03 80 90 a3 18 01 89 1e 02 84 83 6c 02 21 a3 70 07 81 33 33 36 30 32 37 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 30 bytes of data Handling message for SAPI/TEI=0/127 TEI: 0 State 7 V(S) 8 V(A) 8 V(R) 8 K 1, RC 0, l3initiated 0, reject_except 0 ack_pend 0 T200 0, N200 3, T203 1 [ 02 ff 03 08 01 01 05 a1 04 03 80 90 a3 18 01 89 1e 02 84 83 6c 02 21 a3 70 07 81 33 33 36 30 32 37 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 30 bytes of data Handling message for SAPI/TEI=0/127 - Thanks, will try the less intense debug. I thought it was interesting however that the incoming DDI was in the message, but not showing up in the dialplan trace.. -Scott -- _ -- 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
