Re: [asterisk-users] Asterisk 13 - sorcery realtime for pjsip publish objects

2015-02-19 Thread Matt Hoskins
Good Morning,

After further investigation, I found that the res_pjsip_publish_asterisk
module does not use the realtime sorcery wizard, but instead only reads
from the configuration files.  I've been able to patch the module, using
the logic from the other modules to learn how to make the sorery
configuration read from the other sorcery wizards and it's now working for
the asterisk-publication.  This thread probably belongs on the
asterisk-dev list.  I've never submitted a patch to the asterisk project,
and I doubt my code follows your standards.  I'm wondering - is there
planned support for asterisk-publication to be realtime-aware?  If so, I
can continue to do my development on 13.2.0 and simply wait for it to get
fixed in a future point release.  Or, if submitting the patch is the best
course of action - I can do that too.

Suggestions?



 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hoskins
Sent: Wednesday, February 18, 2015 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13 - sorcery realtime for pjsip
publish objects

Excellent.  I was using ast-13.1.0 with no luck.  I upgraded to 13.2.0 and
have made it further, but am having a little difficulty.  The
outbound-publish object types seems to be working in realtime now.  But
the asterisk-publication object is only reading from sorcery.conf.  I know
you said that it *should* work, with no guarantee, which I'm fine with.  I
just want to make sure I don't have a possible misconfiguration issue.

Here is my sorcery.conf and extconfig file:

Sorcery.conf

[res_pjsip]
endpoint=realtime,ps_endpoints
auth=realtime,ps_auths
aor=realtime,ps_aors
domain_alias=realtime,ps_domain_aliases

[res_pjsip_endpoint_identifier_ip]
identify=realtime,ps_endpoint_id_ips

[res_pjsip_outbound_publish]
outbound-publish=realtime,ps_outbound_publish

[res_pjsip_pubsub]
inbound-publication=realtime,ps_inbound_publication

[res_pjsip_publish_asterisk]
asterisk-publication=realtime,ps_asterisk_publication

extconfig.conf
---
ps_endpoints = odbc,asterisk-realtime
ps_auths = odbc,asterisk-realtime
ps_aors = odbc,asterisk-realtime
ps_domain_aliases = odbc,asterisk-realtime ps_endpoint_id_ips =
odbc,asterisk-realtime ps_outbound_publish = odbc,asterisk-realtime
ps_inbound_publication = odbc,asterisk-realtime ps_asterisk_publication
= odbc,asterisk-realtime




 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Wednesday, February 18, 2015 10:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13 - sorcery realtime for pjsip
publish objects

Matt Hoskins wrote:
 Hello,

Kia ora,

 I am currently trying to set up pjsip realtime and would like to have 
 outbound-publish, inbound-publication, and asterisk-publication 
 sorcery object types in ODBC realtime. Is that currently supported? I 
 know that some object types are known working and others are not. I 
 was curious what the status of those objects are.

It should be possible with recent changes that have been done. I
personally have not tried it, and it would still require a reload to pick
up changes regardless.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
http://spamaway.npgco.com/canit/urlproxy.php?_q=aHR0cDovL3d3dy5kaWdpdW0uY2
9t_r=YmFzZQ%3D%3D 
http://spamaway.npgco.com/canit/urlproxy.php?_q=aHR0cDovL3d3dy5hc3Rlcmlzay
5vcmc%3D_r=YmFzZQ%3D%3D



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Re: [asterisk-users] Asterisk 13 - sorcery realtime for pjsip publish objects

2015-02-19 Thread Matt Hoskins
Awesome!



https://issues.asterisk.org/jira/browse/ASTERISK-24811







Matt Hoskins | NPG Corp | Systems Architect

816.749.2815 (Internal: ext. 10015)







From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of George Joseph
Sent: Thursday, February 19, 2015 10:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13 - sorcery realtime for pjsip 
publish objects



On Thu, Feb 19, 2015 at 9:15 AM, Joshua Colp jc...@digium.com wrote:

Matt Hoskins wrote:

Good Morning,

After further investigation, I found that the res_pjsip_publish_asterisk
module does not use the realtime sorcery wizard, but instead only reads
from the configuration files.  I've been able to patch the module, using
the logic from the other modules to learn how to make the sorery
configuration read from the other sorcery wizards and it's now working for
the asterisk-publication.  This thread probably belongs on the
asterisk-dev list.  I've never submitted a patch to the asterisk project,
and I doubt my code follows your standards.  I'm wondering - is there
planned support for asterisk-publication to be realtime-aware?  If so, I
can continue to do my development on 13.2.0 and simply wait for it to get
fixed in a future point release.  Or, if submitting the patch is the best
course of action - I can do that too.



I've got some free time and can take a look.  Give me the issue number after 
you've created it.




The guidelines for filing an issue are here[1] and include links for 
submitting a patch. I know of noone currently working on it.







  _


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Re: [asterisk-users] Asterisk 13 - sorcery realtime for pjsip publish objects

2015-02-18 Thread Matt Hoskins
Excellent.  I was using ast-13.1.0 with no luck.  I upgraded to 13.2.0 and
have made it further, but am having a little difficulty.  The
outbound-publish object types seems to be working in realtime now.  But
the asterisk-publication object is only reading from sorcery.conf.  I know
you said that it *should* work, with no guarantee, which I'm fine with.  I
just want to make sure I don't have a possible misconfiguration issue.

Here is my sorcery.conf and extconfig file:

Sorcery.conf

[res_pjsip]
endpoint=realtime,ps_endpoints
auth=realtime,ps_auths
aor=realtime,ps_aors
domain_alias=realtime,ps_domain_aliases

[res_pjsip_endpoint_identifier_ip]
identify=realtime,ps_endpoint_id_ips

[res_pjsip_outbound_publish]
outbound-publish=realtime,ps_outbound_publish

[res_pjsip_pubsub]
inbound-publication=realtime,ps_inbound_publication

[res_pjsip_publish_asterisk]
asterisk-publication=realtime,ps_asterisk_publication

extconfig.conf
---
ps_endpoints = odbc,asterisk-realtime
ps_auths = odbc,asterisk-realtime
ps_aors = odbc,asterisk-realtime
ps_domain_aliases = odbc,asterisk-realtime
ps_endpoint_id_ips = odbc,asterisk-realtime
ps_outbound_publish = odbc,asterisk-realtime
ps_inbound_publication = odbc,asterisk-realtime
ps_asterisk_publication = odbc,asterisk-realtime


Matt Hoskins | NPG Corp | Systems Architect

816.749.2815 (Internal: ext. 10015)




 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Wednesday, February 18, 2015 10:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13 - sorcery realtime for pjsip
publish objects

Matt Hoskins wrote:
 Hello,

Kia ora,

 I am currently trying to set up pjsip realtime and would like to have 
 outbound-publish, inbound-publication, and asterisk-publication 
 sorcery object types in ODBC realtime. Is that currently supported? I 
 know that some object types are known working and others are not. I 
 was curious what the status of those objects are.

It should be possible with recent changes that have been done. I
personally have not tried it, and it would still require a reload to pick
up changes regardless.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
http://spamaway.npgco.com/canit/urlproxy.php?_q=aHR0cDovL3d3dy5kaWdpdW0uY2
9t_r=YmFzZQ%3D%3D 
http://spamaway.npgco.com/canit/urlproxy.php?_q=aHR0cDovL3d3dy5hc3Rlcmlzay
5vcmc%3D_r=YmFzZQ%3D%3D

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[asterisk-users] Asterisk 13 - sorcery realtime for pjsip publish objects

2015-02-18 Thread Matt Hoskins
Hello,

 

I am currently trying to set up pjsip realtime and would like to have
outbound-publish, inbound-publication, and asterisk-publication sorcery
object types in ODBC realtime.  Is that currently supported?  I know that
some object types are known working and others are not.  I was curious
what the status of those objects are.

 

Thanks!

 

Matt Hoskins | NPG Corp | Systems Architect

816.749.2815 (Internal: ext. 10015)





 

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[asterisk-users] PJSIP vs SIP channeltype

2015-01-26 Thread Matt Hoskins
Hello,

 

I'm currently evaluating asterisk 13 (Currently on 11).  We're testing the
migration from SIP to PJSIP.  Is there a way to alias the SIP channeltype
to PJSIP when exlusively using pjsip?

 

Matt Hoskins | NPG Corp | Systems Architect

816.749.2815 (Internal: ext. 10015)





 

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Re: [asterisk-users] Wiki (pjsip+realtime) says don't put the transports into realtime. Still true?

2015-01-25 Thread Matt Hoskins
Joshua,

Regarding Outbound Registrations in realtime and a reload.  Does it require 
a pjsip reload or full asterisk restart?

Antonio Gómez Soto wrote:
 Hi,

 The asterisk wiki page says:

 Sorcery.conf allows you to try to configure other PJSIP objects such
 as transport using realtime and it currently won't stop you from doing so.
 However, some of these object types should not be used with realtime
 and this can lead to errant behavior.

 Which objects and is this still true in 1.13.1 ?

Yes, still true for transports. Outbound registrations, though, have been 
tweaked so that they can exist in realtime but require a reload to be picked 
up. Those are the two that come to mind.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: 
http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5kaWdpdW0uY29tr=YmFzZQ%3D%3D
  
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[asterisk-users] MWI publish VIA pjsip for non sip channels

2014-10-30 Thread Matt Hoskins
Before I go down a rabbit hole, does the mwi publish/subscription work for
non SIP phones?

For instance, I have a single voicemail server, connected to multiple
asterisk boxes via SIP.  On each of those servers, there are a mix of SIP
and SCCP phones attached.  Currently, I'm using res_xmpp to distribute mwi
from the voicemail server to the endpoint servers.  Would this type of
setup work with PJSIP?  The net effect here is that I want to get away
from res_xmpp, if possible.

Matt Hoskins | NPG Corp | Systems Architect

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Re: [asterisk-users] MWI publish VIA pjsip for non sip channels

2014-10-30 Thread Matt Hoskins
Of course, I left out a detail that may (or may not change) the answer.
I'm using the external chan-sccp-b sccp module, not the chan_skinny
bundled with asterisk.

Matt Hoskins | NPG Corp | Systems Architect

816.749.2815 (Internal: ext. 10015)




 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Thursday, October 30, 2014 5:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MWI publish VIA pjsip for non sip channels

Matt Hoskins wrote:
 Before I go down a rabbit hole, does the mwi publish/subscription work 
 for non SIP phones?

Yes. SIP is simply used as the transport mechanism. It works pretty much
the same as res_xmpp except without needing an XMPP server.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5kaWdpdW0uY29
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Re: [asterisk-users] MWI publish VIA pjsip for non sip channels

2014-10-30 Thread Matt Hoskins
Awesome - Thanks for the quick replies.  I'm sure I could have
tried-and-see but with going from Asterisk 11 to 13, there'd be so many
things changing - it helps to know from the outset.

Thanks again.

Matt Hoskins | NPG Corp | Systems Architect



 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Thursday, October 30, 2014 5:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MWI publish VIA pjsip for non sip channels

Matt Hoskins wrote:
 Of course, I left out a detail that may (or may not change) the answer.
 I'm using the external chan-sccp-b sccp module, not the chan_skinny 
 bundled with asterisk.

Still doesn't matter. Provided it works with res_xmpp it'll work with the
new SIP method.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5kaWdpdW0uY29
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Re: [asterisk-users] FYI: Block Comments

2014-08-25 Thread Matt Hoskins
I love the templates and used them extensively in the beginning of my
asterisk journey.  But abandoned them once I went realtime.  Moving them
to realtime - WOULD BE AWESOME!

Matt Hoskins | NPG Corp | Systems Architect

816.749.2815 (Internal: ext. 10015)




 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Monday, August 25, 2014 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FYI: Block Comments

On 8/25/2014 2:36 AM, Brian LaVallee wrote:
 Hello,

 Here's a fun issue that recently caused me some serious heartache.
 Hope this helps others from making the same mistake.

 Did you know that the configuration parser supports block-comments.
 Like an idiot, I've been highlighting text between dashes.

The configuration parser can do a lot of things. Out of curiosity amongst
those reading this - how many of you know about templates?

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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
www.digium.com  www.asterisk.org

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Re: [asterisk-users] FYI: Block Comments

2014-08-25 Thread Matt Hoskins
Holy damn - I didn't know you could use templates in extensions!  Mind =
Blown.

Matt Hoskins | NPG Corp | Systems Architect

816.749.2815 (Internal: ext. 10015)




 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
Sent: Monday, August 25, 2014 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FYI: Block Comments

On Mon, 25 Aug 2014, Joshua Colp wrote:

 how many of you know about templates?

(You may get more replies with a more 'on-target' subject. I lost interest
in 'block comments' but was curious why the thread was still getting
replies.)

Love templates. Use them in extensions.conf, sip.conf, and iax.conf every
day.

Here's an example from extensions.conf:

[party-line](digit-timeout,h,i,max-timeout,pound-main,s)
 same = n,   agi(write-cdr)
 same = n,   background(${PROMPTS-PATH}/0116)
...

Where the templates look like:

[digit-timeout](!)
 exten = t,1,goto(${CONTEXT},s,1)
[h](!)
 exten = h,1,goto(finish-call,h,1)
[i](!)
exten = i,1,goto(${CONTEXT},s,1)
[max-timeout](!)
 exten = T,1,goto(max-time,s,1)
[pound-main](!)
 exten = #,1,goto(main-menu,s,1)
[s](!)
 exten = s,1,
verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])

Note that the 's' template has to be the last template specified in the
template list. Also, that '${EXTEN}@${CONTEXT}' makes for a quick
cut-n-paste into the 'dialplan show' CLI command.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Replying to Posts

2014-03-13 Thread Matt Hoskins
 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry
Brummell
Sent: Thursday, March 13, 2014 1:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
rwhee...@artifact-software.com
Subject: Re: [asterisk-users] Replying to Posts

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] on behalf of Eric Wieling
[ewiel...@nyigc.com]
Sent: Thursday, March 13, 2014 2:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
rwhee...@artifact-software.com
Subject: Re: [asterisk-users] Replying to Posts

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
Sent: Thursday, March 13, 2014 1:39 PM
To: rwhee...@artifact-software.com; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] Replying to Posts

On Thu, 13 Mar 2014, Ron Wheeler wrote:

 -1
 Prefer top posting.

Your preferences are in conflict with the mailing list rules
(http://www.asterisk.org/community/discuss), specifically #5.

It has to be all one way or the other. This is an English language list. 
Thus, the natural expectation is top to bottom, left to right, answers
follow questions.

If Digium does not like my top posting then they can remove me from the
mailing list.  Your battle is already lost unless Outlook is banned from
the mailing list.

This is an example of why I top post.   Who wrote what?



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Ditto, bottom posting is from the 90's.  We've passed that era.

 






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[asterisk-users] multicastRTP source interface

2014-02-06 Thread Matt Hoskins
I have an asterisk 11.4.0 server with two interfaces, eth0 and eth1.

 

Eth0 has a default gateway on it, eth1 is connected the subnet that has my
phones registered.

 

I'd like to use the multicastRTP driver to do paging.  However, when a
phone dials an extension with multicastRTP, the multicast stream goes to
the primary interface (eth0) and it really needs to go to eth1.  

 

Is there a way to specify which interface the rtp is sourced from?

 

Matt Hoskins | NPG Corp | Systems Architect

816.749.2815

 

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[Asterisk-Users] Cisco 7960 Multiple Line Appearance

2005-11-15 Thread Matt Hoskins

Hello:

I'm currently using asterisk 1.0.7 with cisco 7960 SIP 7.5 phones.  I'd 
like to use this phone for a receptionist so that she can take calls for 
 4 other people.  Is this possible?


I tried setting buttons 2 - 5 to register with the extensions of the 
people she'd be taking calls from, but then the other people couldn't 
receive calls.  I'm assuming this is because only one phone can register 
with the same extension.


I also tried ringing multiple channels with the extension, ie:

1234 = 1,Dial(SIP/1234SIP/4321)

This led to the intended result (both phones rang), but it did not show 
which extension was being called.  (IE: on the phone display, it should 
that I was getting a call from 5551212, but did not say who 5551212 was 
calling)


Is there any way to do this with SIP and the 7960?  I've seen the 7914 
but then I'd have to use SCCP and I'm not sure if it is stable enough 
for production use.


Thank you for your time,

Matt Hoskins

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Re: [Asterisk-Users] Cisco 7960 Multiple Line Appearance

2005-11-15 Thread Matt Hoskins
Alright, I'm inspired.  I'll give it a shot.  Should I use the asterisk 
hint system or is line appearance done in the sccp config file seperately?


Do you have a configuration example?

Thanks!

Matt

Sergio Chersovani wrote:

Matt Hoskins ha scritto:

I'm currently using asterisk 1.0.7 with cisco 7960 SIP 7.5 phones.  
I'd like to use this phone for a receptionist so that she can take 
calls for  4 other people.  Is this possible?



The SIP firmware does not support it.
You have to use SCCP to do that

Is there any way to do this with SIP and the 7960?  I've seen the 7914 
but then I'd have to use SCCP and I'm not sure if it is stable enough 
for production use.



Well give it a chance :-)

http://chan-sccp.berlios.de

Sergio
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