Author Topic: 2.1 - Test Images, Progress, Feedback, Updates  (Read 20065 times)

dagisr

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Re: 2.1 - Test Images, Progress, Feedback, Updates
« Reply #30 on: February 28, 2011, 01:36:28 PM »
Hello,

I am running r1732 and it works fine as it do with 2.0.3.

As said in another spot here on this forum I am having problems with Phonzo.
We think they do not like 'asterisk' type of callers.
I have also seen when setting up my Nokia E51 to log in to my askozia PBX as an extension say 51 and pw 51 using WiFi that one must use  'asterisk' in the Realm field  with this exakt spelling,  or the phone will not register to the PBX.
Maybe this is the reason why we have this problem registering with Phonzo.


So, can you make Askozia even more flexible; maybe possibly in 'General Setup'
to program the Realm parameter?



Scapal

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Re: 2.1 - Test Images, Progress, Feedback, Updates
« Reply #31 on: March 03, 2011, 08:12:29 AM »
Hi Michael,

Are you still there? No SVN activity nor forum post for a while... is this project on pause?

Cheers,
Pascal

Michael

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Re: 2.1 - Test Images, Progress, Feedback, Updates
« Reply #32 on: March 04, 2011, 09:14:31 PM »
Oh no, no pause :)

Check out our newly launched http://www.askozia.com

CeBIT was a huge distraction the last few weeks and finalizing the website took a bit longer than expected. Other than that, we've been doing some internal hand-offs of different responsibilities so a TON of new commits from Benjamin are going to be showing up next week.

Lots of stuff going on, just hadn't had time to do much in the forums lately. All Askozia staff now have marked accounts and will be paying more attention here so that lonely feeling does not creep in again.

Scapal

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Re: 2.1 - Test Images, Progress, Feedback, Updates
« Reply #33 on: March 05, 2011, 01:18:52 PM »
Nice :-)

doucettom

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Re: 2.1 - Test Images, Progress, Feedback, Updates
« Reply #34 on: March 06, 2011, 06:47:20 PM »
Would be nice to have possibility to try fax2email from SIP providers, at least in the Integrator Panel. I know it's not going to be in the standard UI 'till T.38 gets fully supported but I'd try it anyway.
« Last Edit: March 06, 2011, 06:50:42 PM by doucettom »

Scapal

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Re: 2.1 - Test Images, Progress, Feedback, Updates
« Reply #35 on: March 29, 2011, 11:52:32 AM »
Bug on r1732

When multiple providers share the same hostname, you create a merged incoming context based on this hostname.
Ex. two lines on voip.belgacom.be will create a context SIP-PROVIDER-voip.belgacom.be-incoming.

The problem is that you don't merge the inbound routed numbers but redirect assume that those numbers are the usernames.
With voip.belgacom.be, the username is not the phone number, so your context looks like:

[ Context 'SIP-PROVIDER-voip.belgacom.be-incoming' created by 'pbx_config' ]
  'u234865xx' =>    1. Goto(SIP-PROVIDER-5028176174d9065435d38e-incoming,s,1) [pbx_config]
  'u288096xx' =>    1. Goto(SIP-PROVIDER-19976182854c9a299885c6b-incoming,s,1) [pbx_config]

Creating a new merged context is the way to go but you should really create a full context merging the two inbound rules instead, discarding, in this example, the SIP-PROVIDER-5028176174d9065435d38e-incoming and SIP-PROVIDER-19976182854c9a299885c6b-incoming.

As the same logic works even if you don't have multiple lines, why couldn't this be the general rule (ie. SIP-PROVIDER-fqdn-incoming) for SIP providers?

My current workaround is to create custom hostname aliases to avoid the merging and put all the incoming rules in the first profile, hence doing the merging manually.

Also, when corrected, it would be wise to do this based on the IP since, for example voip.nomado.eu and sip.3starsnet.com are in fact the same servers with the same IP. If you create 2 different contexts, only one will be used, the other being discarded.

Cheers,
Pascal

Dhaval12

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Re: 2.1 - Test Images, Progress, Feedback, Updates
« Reply #36 on: February 16, 2017, 11:54:36 AM »
Hi Michael,

Thank you for this wonderful thread. I like to test Version 2.1. i will to try this very soon.