i wasnt able to make a ticket in the commercial module category but it let me in the freepbx section so i did that. someone already logged into my system and fixed it yesterday. thanks!
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Broke after zulu 3.0 install
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ZULU 3 Fax & SMS
They do and will do.
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Could not reload FOP server after v14 upgrade
This happened a few days ago.
An hour or so after posting my question I returned to the dashboard to find that the FOP server was no longer complaining. Further checks determined that FOP was indeed running.
Sorry, forgot to post back here that all is well.
I consider this issue closed.
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Zulu 2 and Zulu 3
I’ve been slowly exploring my way around Zulu 3 and while there is some learning curve, it is very nice. One major change that has impacted us greatly is popups.
In Zulu 2 we had FM/FM turned on for many of our extensions. These would forward to agents on an Avaya system; however, these agents would also have the Zulu 2 client open with the softphone disabled. They would use this for chat, and more importantly when a call hit their Asterisk extension it would trigger the popup before ringing their Avaya phone.
In my testing it appears that the popup will not popup unless the call is answered on the Zulu 3 client. Is that correct? Is there anyway to circumvent this if it is? We are big advocates for Zulu and just purchased 1,000 licenses based on the belief that if the product keeps moving in this direction it could be a full replacement for us, in time.
Today it is not due to the lack of call center agent call control features. We have submitted a feature request to have these added, hopefully it will be achievable. In the interim what can I do to get back on track with popups (popup still pops up even tough the call is answered on an external number)?
I noticed in the logs that it is storing the URL as a variable, is there a way for me to invoke the popup if I add a custom context to one of the macros maybe?
Thanks for your help/insight!
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PJSIP Error with NO PJSIP configured
Hello,
The negative affect is the phone doesn’t ring and calls for the phone go directly to voicemail 
.
When first installed the PBX about 8 months ago, default was PJSIP and I flipped it to CHANSIP, everyting was then created as CHANSIP ( Trunks & Extensions ) CHANSIP was changed to 5060 and PJSIP to 5160.
Never had this issue until yesterday.
I didn’t see anywhere to disable PJSIP so if it is sending both, not sure how to stop that.
Thanks for the responses, and looking forward to the magic answer .
***** after revisting the advanced menu, I saw SIP CHANNEL DRIVER option and there I did see BOTH, so I changed it to CHANSIP, I’ll post back shortly.
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Restore backup to newer hardware fails
I built a new FreePBX box to upgrade the hardware of an existing (functional) system.
Then, my older functional FreePBX system I upgrade to the latest version (v14).
Both machines have the same software versions.
The backup from the older hardware will not restore to the newer hardware. “Restore” finds the *.tgz file but, “clicking” the ‘restore’ button the page flashes, the *.tgz drops off and nothing appears to happen. Further checking and no, the backup did not restore.
Now, the old hardware is an Intel box running a dual core Pentium with Raid1 mirrored hard drives. The newer hardware is a quad core box with 8Gb RAM and a single 180Gb msata ssd. (If the hardware is the issue.)
Forum searches of this issue yield nothing on the latest FreePBX version 14.
I thought it was this simple but perhaps not. What am I doing wrong?
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PJSIP Error with NO PJSIP configured
Pjsip can be disabled in advanced settings, but your problem doesn’t seem to be related to it.
Post logs of a failing call.
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PJSIP Error with NO PJSIP configured
I am moving everything away from Chan_SIP to PJSIP because of major vulnerabilities in it that allow people sending random packets to our server to make international calls, among many other reasons.
It might be best long term to figure out what is going on with PJSIP.
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Disable Specific CIC
Hi All,
We are using such device Sangoma Netborder SS7 Gateway for interconnecting with our ISP through SS7.
The device is configured and we can receive calls.
Is there any way to block some channels (for ex. cic 10) and set it down?
I tried to use something like s1,2-9,g10,11-31 - but on the other side - the ISP anyway sees it as idle, but they expect to see it down.
Thank You.
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Steps for downgrade Zulu from 3 to 2?
I looked through the bug reports and can find no report of this.
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Zulu 2 and Zulu 3
Correct
Unfortunately not at this time. Please report your issue so we can add a setting for you to decide.
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Disable Specific CIC
it’s very simple actually,
ftdm ss7 blo span 1 chan 4 - did the job.
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PJSIP Error with NO PJSIP configured
Since the log is a moving target with lots of calls, I cant get a clean call log for just that extension.
If there is a way from within asterisk to do that I would appreciate the info.
Basically this is the error:
[2018-07-06 09:58:07] ERROR[9281][C-00000048] pbx_functions.c: Function PJSIP_HEADER not registered
[2018-07-06 09:58:07] ERROR[9281][C-00000048] pbx_functions.c: Function PJSIP_HEADER not registered
Which makes the extension look busy and goes to voicemail.
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PJSIP Error with NO PJSIP configured
You’re moving away from chansip despite your problems with cpu spikes in pjsip, or has this since been resolved?
I assume the adtoptek in the Asterisk forum is you:
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PJSIP Error with NO PJSIP configured
Do a sip debug and paste the output.
sip set debug peer xxxx
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PJSIP Error with NO PJSIP configured
We have narrowed down things and it doesn’t seem to be PJSIP but asterisk (MySQL might be involved too). PJSIP actually uses a bit less CPU when compared to Chan_SIP.
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PJSIP Error with NO PJSIP configured
Ok thanks.
If you wouldn’t mind, please keep updating that thread with your findings.
We are concerned about moving larger installs to pjsip due to that and other things, and also concerned about the performance of Asterisk/Freepbx on larger 300+ installs in general.
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Steps for downgrade Zulu from 3 to 2?
Not in bug reports, it’s on commercial support.
Regards
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PJSIP Error with NO PJSIP configured
I was talking about asterisk and not specifically PJSIP in that post. The server was originally a chan_sip that we are slowly converting to PJSIP.
We have under 1000 extensions on one server and the BLF reloads after freepbx loads the dial plan into asterisk cause problems on both chan_sip/pjsip.
After a lot of testing PJSIP seems to have about a 5-10% performance increase on our server compared to Chan_SIP (Have both enabled so might save more if Chan_sip is off). Not to mention it fixes 2 exploits we have been dealing with when using chan_sip. The added features are nice too. However it seems like “MWI Subscription Type” (Advanced ext settings) doesn’t like being on auto for our server so we set it to Unsolicited, saves some more CPU cycles so we are ok with that.
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Implementing and Testing Naf's GVsip Google Voice Sip Ubuntu 18.04 Freepbx
@tm1000 without make samples the /etc/asterisk dir is empty on a fresh build. Thus the step where you start asterisk with ./start_asterisk start before running the installer will fail. How does this work without at least /etc/asterisk/asterisk.conf in place?
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