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netpro78

Demo of 5.2.6 (Win64)

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I was active back through the PBXnSIP 3.4 days, and was happy with the product back then, and lost interest when it started getting tailored for Snom only. Now that it is no longer a Snom product, I decided to revisit it to see what has been fixed, what is still an issue, and what is new. I did a fairly basic demo with the free version to evaluate if I should purchase or not. The following are some issues I see with the current product. I know that some are easy fixes, and some are far more complicated. I am hoping that some could be addressed to make it a worth while purchase. In the event that I did purchase and I wanted to migrate from PBXnSIP 3.4 is there still a migration path, since I see some posts saying the older versions have been removed?

 

Issues that still exist from 3.4

-The service is single threaded

-BLF's often show a ringing state when a call is answered with a pickup (*87)

-No support for BLA, or SCA

-Call sequence numbers get out of order when restarting the pbx for maintenance causing BLFs to stop functioning until the phones are rebooted (this is a killer when trying to run the switch for hosting)

 

New issues I ran across during the demo

-You can only add one extension at a time, unless you want to jump through the hoops of creating a CSV file

-Certain situations can get a call stuck in the system, and clicking the X next to the call will not remove it

-If the "Ignore Packets that do not match the Domain" is selected, then the phones can only place calls. When they attempt to receive calls, they continue to ring since the PBX ignores the SIP signaling from the phones saying that the phone is trying to answer the call.

-You cannot create trunks in Firefox

-No audio when calling from the web interface

-Include cell phone for web callback creates a call between the cellphone, and desk phone, not the cell phone, and the called number.

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Thanks for your feedback.

 

Yes we are working on some of these issues. Others are low priority, because we have to allocate resources and if some issues don't even register with most of our customers then they will get done last.

 

For bulk extensions CSV is still the way to go. We had added a table where multiple extensions could be created but had to remove it because customers didn't like it.

 

Call stuck has alluded us for a while. We are trying to recreate it consistently so that it can be fixed. Unfortunately it is very erratic and inconsistent and actually hardly ever happens in our test scenarios. We may have to build a bigger test scenario.

 

Hmmm, you can't create trunks in Firefox. That's new to me, we will test and fix.

 

No audio on calling from web interface is for WebRTC calls or the calls initiated by the phone through the web? Because both work perfectly here, I just tested it. Which version are you testing? Any logs?

 

There is a migration path but we do suggest to back up everything (in case you need to go back), and there may be minor corrections needed after migration because a lot has changed since version 3.

Other than that, it should not be a big problem. And our support will help you through the migration, once you decide to do so and call our support.

 

Thanks.

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It does generate a user disconnects call log whaen I attempt it. The interesting thing is it does not seem like it receives any packets:

 

Packets (Tx/Rx): 1429 0

Codec (Tx/Rx): PCMU/8000

Address: 79.146.173.76

SDP-State (Local/Remote): sendrecv sendrecv Sequence-Number (Tx/Rx): 47351 32768

User-Agent: Vodia WebRTC 1.0

Call-Quality:

VQSessionReport: CallTerm

LocalMetrics:

Timestamps:START=2015-08-26T22:20:01Z STOP=2015-08-26T22:20:25Z CallID:c1abf8a7@pbx

FromID:103@dev.testdomain.local;tag=gu8701

ToID:6443393@dev.testdomain.local;tag=6f7d20647a

SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=0 FPP=1 PPS=50 PLC=3

LocalAddr:IP=0.0.0.0 PORT=51788 SSRC=0x097b6f4a

RemoteAddr:IP=0.0.0.0 PORT=0 SSRC=0x

x-UserAgent:Vodia-PBX/5.2.6

x-SIPterm:SDC=OK

 

In the PBX log I am also seeing:

[6] 17:20:40.250 WEBS: Delete HTTP connection 8864 [3] 17:20:40.772 SIP: Could not send on socket 127.0.0.1:12345

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