Juan Manuel Acevedo
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Posts posted by Juan Manuel Acevedo
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Hi:
For PSTN I use Quintum gateways wich I can configure to drive one incoming call to one account, (extension,group, AA) based on called party or calling party. (DNIS or ANI)
I need to use a SIP registration Trunk to one ITSP wich give me several numbers. I need to drive the call based on the called party as well based on calling party, but I don´t see at trunk definition any parameter to do that. It is possible or not?, If yes how can I to configure it?
Thanks and best regards-
Juan Acevedo
consultorit@umi.com.co
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Do you ever want the call to go to vmail? What about call forward no answer or set a cell phone number to an extension so it will fork the call.
Thanks for the answer, in fact I need is that when extension is not registered, the calls be forwarded to one specific extension, but the default behavior of PBxNSIP is that when the extension is not registered and a call is incoming it goes to the vmail.
The point is that my csatomer uses Softphone and when he turn off the PC, the extension loses the register, and he want that teh call goes to another extension.
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Hi:
Does exist some way to redirect the extension when it is not registered but dedfined at PBXnSIP?
A lot of my customers use Softphone and when they power off the PC, the extension get not register, and they want that when it happens the call goes to another extension.
Of course exists the option for forward alls calls and the customer has the option to do that before power off, but.. you now : Alzaimer exists.
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STUN works only in < 80 % of the cases, leaving the remaining > 20 % as a support nightmare. We gave up troubleshooting all kinds of one-way audio problems and explicity removed the support for STUN-allocated identities. Our life and the life of our customers became much easier after that. And we saved a lot of money buying all these different DSL and cable routers to find out what the problem was with this and that installation.
Pulver must get a session border controller. That is what practically all SIP service providers are doing.
Or they should start supporting IPv6 addresses (no more NAT the way it was done in IPv4). If you don't have a routable address, don't expect that someone calls you . At least not in a reliable way.
Thanks for your answer, but how can I do that Pulver get a session border controller?
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Hi:
I have downloaded Version 2.1.6.2450, but I don't see room to configure STUN Server, and my provider, Pulver rejects the registration becuase non-routable addresss.
Any body knows how to configure it?
Thanks And Best Regards
Juan Acevedo
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Hi:
I see a lot of expired files (18.000 per day) recorded at PBX folder, I dont know why these files are produced niether how to avoid his recording.
The fact is that in a PBXnSIP with a large number of calls the existence of these files make waste a lot of processor resources making inestable the system.
If anybody knows how to avoid them please tell me.
Best regards and thanks
Juan Manuel Acevedo
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Hi:
Does anybody know SIP phones with two ports autosensing 100/1000?
Better if are tested with PBXnSIP.
Thanks
Juan Acevedo
acevedo1@une.net.co
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Juan,
I just tried a key mapped to a wireless MAC and the system did not like the key. I pasted it in a second time, hit SAVE, and it seemed to like it the second time.
I am running 2.1.6.2446 on my laptop. I did not restart the pbxctrl service...
Paul:
Can you test with 2.1.6.2447? I have pasted twice and the license is not recognize
Thanks
Juan Acevedo
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I have installed the 2.1.6.2447, but the license was not activated.
I use a Wireless NIC in order to provide the server with the MAC address, but the license was not activated.
The 2.0.3.1715 version, the last I am using activates the license with the wireless NIC.
What can I do? I want to use the 2.1.6.2447 because the new features it has.
Anybody can give some idea?
Thanks
Juan Acevedo
acevedo1@une.net.co
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Did you use a higher priority for the next entry in the dial plan?
http://wiki.pbxnsip.com/index.php/Trunk_Se...tbound_Settings was not clear on this, has just been updated.
Thanks, using a higher priority for for the next entry failover it works fine, it can not be the same or less
Juan Acevedo
acevedo1@une.net.co
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It is seconds. A value of ten means that after ten seconds not receiving anything from the ISP, the PBX would continue processing the dial plan and possibly choose another trunk for the call. I would say values between 5 and 10 seconds are a good balance between reasonable response times from the ISP and user patience until they hear a ringback tone.
Hi:
It seems trunk
I have the following settings:
Dial plan:
9* 9* to trunk119, failover on 5xx codes, and request timeout 5 secs
9* 9* to trunk120
Trunks:
trunk119: SIP Gateway outbound: 192.168.2.119
trunk120: SIP Gateway outbound: 192.168.2.120
I have populate all the channels of the gateway 119, and make a new call starting by 9, at ethereal I can see that PBXnSIP returns 503 error. but dont search the trunk 120, and returns bussy tone at phone originates the call.
I will appreciate any help about.
Thanks and best regards
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There were a couple of important fixes around the daily sending of emails that caused problems in several places. I think everyone running 2.1 versions should move to that version. The release notes are available at http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.5.
Hi:
I use one wireless NIC in order to provide the MAC address to the system, this way I can mover to one server to another por backup purposes.
I see at V2.1.5 that the system does not recognize the wireless NIC MAC , It recognizes only the MAC of the wired NIC included at server.
Thanks
Juan Acevedo
email: acevedo1@une.net.co
Called and Calling Party on inbound calls on trunks
in Trunk Setup
Posted
Thanks Maribel, when can I to find the way to use it?
Juan