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Software and hardware performance for freepbx

April 4, 2018, 2:55 pm
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The resources required per call vary by more than an order of magnitude, depending on whether the system is transcoding, recording or playing media, listening for DTMF and/or performing functions such as jitter buffer and echo cancellation. IMO automated evaluation is not feasible. OTOH, both free and commercial load emulation software is readily available, so you can easily determine how many calls can be handled at a given quality level.

For example, see http://sipp.sourceforge.net/ . IMO you should have one or two real calls in concurrent with the artificial load, to subjectively assess the severity of any quality degradation.

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Set variable by phone and change recording depending on it

April 4, 2018, 3:00 pm
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Freepbx in HyperV

April 4, 2018, 3:08 pm
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Interesting. We have had issues with Conferences for years on Hyper-V hosted FreePBX systems. Audio sync/lag and participants getting “automuted” that we eventually opened a support case for and were ultimately told “Hyper-V is not a supported platform.”

These FreePBX VMs have not been live-migrated ever, but I’m wondering if the issue could be related.

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Sipp

April 4, 2018, 3:30 pm
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i have a freepbx server i want to have how to use sipp to have total number of simultaneous calls can support my pbx server.

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Sipp

April 4, 2018, 3:31 pm
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Dimensioning an Asterisk system (benchmarking)

April 4, 2018, 3:34 pm
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Freepbx in HyperV

April 4, 2018, 3:40 pm
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The time sync issue on my HyperV deployment never suffered (knowingly) from any issue. I rebooted the guest OS and the message stopped. Again, for me it was the same issue, but a completely different deployment of an PBiaF from probably 3 years ago. I’ll have to watch to see if it comes back. Not sure when this problem started, but would have been relatively recently. Admittedly haven’t looked at the console of the box in several month (which is where the message was showing for me) vs a remote SSH session.

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Chan_dongle

April 4, 2018, 4:19 pm
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Trying to connect on /dev/ttyUSB2…
[2018-04-04 23:20:48] ERROR[15173]: chan_dongle.c:137 lock_create: open(’/var/lock/LCK…ttyUSB2’) failed: Permission denied
[2018-04-04 23:20:48] ERROR[15173]: chan_dongle.c:137 lock_create: open(’/var/lock/LCK…ttyUSB1’) failed: Permission denied

i still getting these even i did

usermod -a -G dialout asterisk

chmod 777 /dev/ttyUSB*

nothing happen

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How to use Sipp

April 4, 2018, 4:25 pm
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i have a freepbx server i want to have how much simultaneous call can support with tools sipp is there anybody who can help me how to use this tools ??

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Software and hardware performance for freepbx

April 4, 2018, 4:28 pm
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thanks, sipp is that it can give me number of simultaneous call in my ipbx server ???

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Presence Watcher BLF key? (DND vs Busy?)

April 4, 2018, 4:32 pm
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I’m currently using pjsip rather than chan_sip and there are a lot of states that seem to be available:
UNKNOWN | NOT_INUSE | INUSE | BUSY | INVALID | UNAVAILABLE | RINGING | RINGINUSE | ONHOLD

I believe it’s up to the phone as to how it chooses to display each one. (Some will show green, some red, some flashing, some off, etc.) I’ve successfully added the script I linked above to the extensions_custom.conf file and it works – it does change the hint to “HOLD” or “RINGING” as specified, but unfortunately Asterisk’s DEVICE_STATE has to be set to a custom name. Some of the statuses are then pushed down to the hints, but others are not. I suspect it’s doing some magic in the background to decide that if my custom status is “UNAVAILABLE” but my actual phone reports “NOT_INUSE” then the hint becomes “NOT_INUSE” – however, if my custom status is “BUSY” and my phone is idle, the higher priority leads it to report the hint as “BUSY.” At least that’s my guess without having dug into the source code yet.

In any event, I think I’ll be able to solve my issue with a Yealink T46S rather than the Grandstream 2170. It looks like it will allow some more customization as to how it displays the various hints. I still wish there was a better way to show presence on a BLF key, though – something similar to the Grandstream UCM. I’ll keep digging. :slight_smile:

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Presence Watcher BLF key? (DND vs Busy?)

April 4, 2018, 5:05 pm
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You are correct. I used the word state instead of hint. You only get 3. At least it was last time we messed with this. It may of changed but you could only get those 3 low level hints of in use, not in use or flashing.

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Extension IAX2 remota se desconecta despues de un tiempo cuando bloqueo el iphone

April 4, 2018, 6:32 pm
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extension IAX2 remota se desconecta despues de un tiempo cuando bloqueo el iphone

chan_iax2.c:12384 __iax2_poke_noanswer: Peer ‘101’ is now UNREACHABLE!

que es?

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Cisco 7940 connection issue

April 4, 2018, 7:52 pm
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Hi everyone!

I recently purchased 2 Cisco 7940’s with the latest SIP firmware already pre-installed. One of these phones in now located in Minnesota and one in Colorado. The one in Minnesota is working fine with Freepbx and was a breeze to setup. The freepbx machine is hosted on a raspberry pi 3 with ethernet in Colorado. The phone in Colorado will not send a ping to the freepbx server located on the local network. The log files show nothing about this phone trying to ping, the other phone is pinging via the external IP address. The local phone just will not connect. Any advice or help is appreciated. I have read a-lot about TFTP being required however I have no idea on how to do this.

I am new to Freepbx as well as Asterisk. Thank you all!

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Cisco 7940 connection issue

April 4, 2018, 7:57 pm
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configure manual not tftp check you network with other phone or softphone and try

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Cisco 7940 connection issue

April 4, 2018, 7:58 pm
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I have been configuring manually and can confirm that the softphone using x-lite does work.

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AT&T/Cricket Blocking Traffic

April 4, 2018, 9:49 pm
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Thanks @dicko and @GSnover. I changed the extension to iax2 and configured such in Zoiper. That fixed the problem. I’m sure it’s a 4G/LTE SIP blocking, as I’ve changed no other settings other than changing this one extension to iax2.

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AT&T/Cricket Blocking Traffic

April 4, 2018, 11:22 pm
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I’m glad that you got it working, though I’d really like to know what the SIP issue was.

IMHO it’s unlikely they would be actively blocking. I hope that a member with a phone on Cricket or AT&T can take a look at what goes wrong.

AT&T doesn’t block VoIP in general; Skype, FaceTime, Messenger, WhatsApp, Viber, Hangouts, SideLine, Line2, … all work fine.

I just did two tests with CSipSimple <-> FreePBX; both had good bidirectional audio. I admit that they weren’t definitive, limited by my present resources. The first used a Samsung tablet with an AirVoice Wireless SIM (AT&T MVNO). The other used a Motorola phone with an SFR SIM, roaming on AT&T.

I don’t know whether it mattered in this case, but my PBX listens on a non-standard port rather than 5060.

Some reasons why IAX2 is IMO not a good general workaround:

  1. Won’t work with the native VoIP function built into most Android dialers.
  2. Won’t work with other fine softphones (CSipSimple, Bria, Grandstream).
  3. Won’t work with other PBXes (FreeSWITCH, 3CX, Vodia), though some offer an alternative tunnel mechanism.
  4. For those who don’t need a PBX or are satisfied with the PBX features of their provider, many don’t support IAX (even some that are Asterisk based).

Your observed symptoms seem strange for blocking. The signaling was apparently normal – did they tweak the SDP so RTP went to the wrong address or port? It certainly would be easier to just block the SIP ports.

Since it’s easy to capture a SIP trace with both FreePBX and CSipSimple, it should be easy to see what is going wrong.

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No such context 'macro-outisbusy' for macro 'outisbusy'. Was called by 100@from-internal

April 4, 2018, 11:35 pm
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No such context 'macro-outisbusy' for macro 'outisbusy'. Was called by 100@from-internal

April 4, 2018, 11:42 pm
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image
image.jpg1366x768 100 KB

The account(asterisk extension) is registered in softphone
When I call someone, first it asks the outbound route password and says thank you for entering the correct password, later it gets disconnected.
Then the above logs appear.

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