okay so the date/time issue eventually resolved itself but I still can’t get click to dial to work at all. Right now I have two different php files on the server that click to dial can point to. For pjsip users I have a pjsipcall.php and for normal chan sip extensions I have the default call.php. In the beginning only the polycom would ring but after toggling it back and forth now neither devices ring. Back to more troubleshooting.
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One extension - two phones. Both work but not showing registered in endpoint
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S505 not showing in models
Your EPM module is way out of date.
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Cisco outgoing call display name not working
We have a mixture of Cisco 7942 and 7962s, and just recently our phones have stopped displaying the Name of the extension being dialed. For example if I dial 66022 my phone displays 66022 (66022), instead of Nav-Test (66022). So with the help of a colleague, we figured out that enabling Find Me/Follow Me on the extension (66022) fixed the display name issue. With FM/FM turned on, when I dial 66022, my phone displays Nav-Test (66022) correctly.
Did a packet capture and noticed some differences with FM/FM both disabled and enabled.
FM/FM Disabled
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.21.18.67:5060;branch=z9hG4bK3fbc08f2;received=172.21.18.67
From: “66008” sip:66008@10.50.160.81;tag=00260bd7db3400244680b498-148ee722
To: sip:66022@10.50.160.81;tag=as1eac74ec
Call-ID: 00260bd7-db34000d-c6fa6b10-014178ca@172.21.18.67
CSeq: 102 INVITE
Server: FPBX-14.0.13.23(13.29.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:66022@10.50.160.81:5160
Remote-Party-ID: “Nav-Test” sip:66022@10.50.160.81;party=called;privacy=off;screen=no
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.21.18.67:5060;branch=z9hG4bK3fbc08f2;received=172.21.18.67
From: “66008” sip:66008@10.50.160.81;tag=00260bd7db3400244680b498-148ee722
To: sip:66022@10.50.160.81;tag=as1eac74ec
Call-ID: 00260bd7-db34000d-c6fa6b10-014178ca@172.21.18.67
CSeq: 102 INVITE
Server: FPBX-14.0.13.23(13.29.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:66022@10.50.160.81:5160
Remote-Party-ID: “66022” sip:66022@10.50.160.81;party=called;privacy=off;screen=yes
Content-Length: 0
So in this capture the first “Status 180 Ringing” shows Remote-Party-ID: “Nav-Test” which is correct. But then for some reason the second “Status 180 Ringing” changes to Remote-Party-ID: “66022”. Also noticed “screen”=no in the first 180 Ringing packet, then changes to “screen”=yes in the second 180. This traffic is being sent from FreePBX (10.50.160.81) to my desk phone (172.21.18.67). The server (10.50.160.81) for some reason is altering the RPID.
FM/FM Enabled
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.21.18.67:5060;branch=z9hG4bK0b95eb9f;received=172.21.18.67
From: “66008” sip:66008@10.50.160.81;tag=00260bd7db34049c70533a7e-66827c33
To: sip:66022@10.50.160.81;tag=as258bc2f5
Call-ID: 00260bd7-db34001e-1d7da114-5f5d4145@172.21.18.67
CSeq: 102 INVITE
Server: FPBX-14.0.13.23(13.29.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:66022@10.50.160.81:5160
Remote-Party-ID: “Nav-Test” sip:66022@10.50.160.81;party=called;privacy=off;screen=no
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.21.18.67:5060;branch=z9hG4bK0b95eb9f;received=172.21.18.67
From: “66008” sip:66008@10.50.160.81;tag=00260bd7db34049c70533a7e-66827c33
To: sip:66022@10.50.160.81;tag=as258bc2f5
Call-ID: 00260bd7-db34001e-1d7da114-5f5d4145@172.21.18.67
CSeq: 102 INVITE
Server: FPBX-14.0.13.23(13.29.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:66022@10.50.160.81:5160
Content-Length: 0
In this capture the first “Status 180 Ringing” packet has the Remote-Party-ID: “Nav-Test”, but the second “Status 180 Ringing” doesn’t even include the Remote-Party-ID header at all. This traffic is also coming from FreePBX (10.50.160.81) to my phone (172.21.16.67).
Seems like my solution to this problem is enabling FM/FM on every single extension, but I thought I would post this on the forums to see if anyone has seen this before. This is my first time posting on these forums, if more information is needed, please let me know.
Thanks!
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BLF light green even if device is offline
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Zero Touch Provisioning not working over HTTPS
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Latest Update Breaks Trunk
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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A Community Update
The AGI stuff was mostly what I was referring to w/r/t your implementing NodeJS for performance reasons. It’s not optional, as one can’t do the install without node, and with the majority of our PBXs being still on EL6, this was a challenging proposition to say the least! Trying to get it installed – and stable – was just too much of a hassle, with too many changes needed. (EL7 is no problem of course, so newer installs are running 15 without issue, and I’m running tests on EL8 now, with everything looking good.) It wasn’t the different language that was a problem, but the requirement for those additional daemons. I realize that most users are happy to let FreePBX control their entire system and just install the distro, so I know I’m a minority voice; I just like to grumble about these things every now and then!
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S505 not showing in models
Ah! I always forget that the modules can have different expiration dates! And indeed even though this install in brand new we used the old deployment ID.
Thanks for the check!
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BLF light green even if device is offline
The question is does “Offline, Unknown” = State:Unavailable
Then also there is the question of wtf FreePBX is doing. Likely I changed something. I test things out on this PBX, but still.
My test extension is 121. There is no device currently registered. Yet the hints show State:Idle
[jbusch@pbx ~]$ rasterisk -x 'core show hints' | grep 121
121@ext-local : PJSIP/121&Custom:DND State:Idle Presence:available Watchers 1
auto_hint_121@from-i: PJSIP/121,CustomPres State:Unavailable Presence:available Watchers 0
[jbusch@pbx ~]$ rasterisk -x 'pjsip show contacts' | grep 121
[jbusch@pbx ~]$
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A Community Update
Thanks, I don’t want to just grumble about things!
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P-preferred-identity
There’s a few old threads on this here have you had a look at them?
This one for example: Freepbx sip custom header P-Preferred-Identity
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Zulu erroring out
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Llamadas se cortan inmediatamente
Gracias dolesec, los codecs están bien a mi parecer, en ambos lados están registrados los correctos.-
veracruz-atp*CLI> iax2 show peer iaxaereo
- Name : iaxaereo
Secret :
Context : default
Parking lot :
Mailbox :
Dynamic : Yes
Callnum limit: 0
Calltoken req: No
Trunk : Yes
Encryption : (aes128,keyrotate)
Callerid : “” <>
Expire : -1
ACL : No
Addr->IP : 192.168.14.253 Port 4569
Defaddr->IP : 0.0.0.0 Port 4569
Username :
Codecs : 0x703 (g723|gsm|g729|speex|ilbc)
Codec Order : (none)
Status : OK (124 ms)
Qualify : every 60000ms when OK, every 10000ms when UNREACHABLE (sample smoothing Off)
Sistemas 2.-
freepbx-aereo*CLI> iax2 show peer iaxveracruz
- Name : iaxveracruz
Description :
Secret :
Context : from-internal
Parking lot :
Mailbox :
Dynamic : Yes
Callnum limit: 0
Calltoken req: No
Trunk : No
Encryption : No
Callerid : “” <>
Expire : 13
ACL : No
Addr->IP : 192.168.8.254 Port 1045
Defaddr->IP : (null) Port (null)
Username :
Codecs : (g723|gsm|ulaw|alaw|g726aal2|adpcm|slin|g729|speex|g726|g722|siren7|siren14|slin16|g719|speex16)
Codec Order : (ulaw|alaw|gsm|g726|g729|g723|speex|speex16|g722|siren7|adpcm|g719|slin|slin16|g726aal2|siren14)
Status : OK (155 ms)
Qualify : every 60000ms when OK, every 10000ms when UNREACHABLE (sample smoothing Off)
Resultado de tcpdump.
tcpdump host 192.168.14.253
tcpdump: verbose output suppressed, use -v or -vv for full protocol decode
listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes
18:49:35.363745 IP pc253.veracruz.mex.iax > 192.168.14.253.iax: UDP, length 31
18:50:03.364693 IP pc253.veracruz.mex.iax > 192.168.14.253.iax: UDP, length 31
18:50:05.364253 IP pc253.veracruz.mex.iax > 192.168.14.253.iax: UDP, length 31
18:50:15.364111 IP pc253.veracruz.mex.iax > 192.168.14.253.iax: UDP, length 31
No sé si tenga que ver con el problema, pero se comunican a travez de una vpn “172.18.8.1” no de manera directa…
Gracias de antemano
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Outbound CID not passing but shows in CDR report
The Asterisk log appears to show the correct 416 number set as caller ID, but your tcpdump doesn’t show any of the communication with the trunking provider, so we don’t know whether it’s a trunk configuration issue or a problem at the provider.
For this, there is no need to use tcpdump; at the Asterisk command prompt type
sip set debug on
then make a test call. The SIP traces will appear interspersed in the regular Asterisk log.
Paste that at https://pastebin.freepbx.org and post the link here.
Also, post your outbound trunk settings.
BTW, according to https://apeiron.io/lrn , the 416 number ported to Iristel only yesterday at about 09:42 Toronto time, so it’s possible that your provider hasn’t set everything up properly.
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FreePBX design discussion
It’s kind of crazy how people nitpick AdHominem while the main message is being missed. The product is not intuitive. Documentation is not very strong and in no way seems to enable a new user. I find myself having to go into the forums a lot. I actually had to use YouTube videos made by CrossTalk Solutions to figure out how just to get started. FusionPBX seems to be a step backwards in what AdHominem is talking about…had you said VitalPBX that would had made more sense. As new user (as I’m always trying to learn) I’m with AdHominem. BTW weeks back I complained that the release notes for FreePBX SNG7-PBX-64bit-1910 was actually for SNG7-PBX-64bit-1904 and still today they are wrong.
AdHominem - I get it, you’re wanting to help. I would visit other forums that would welcome ideas and synergize…this thread seems to be more trolling then wanting to build upon a solution. I get your message and its not lost on me.
Thanks,
. .
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FreePBX design discussion
Could be that the tone of the help/suggestions had something to do with the defensive response posture. Also that many simply disagree.
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FreePBX design discussion
Gordon,
Thank you very much for your support.
Although FreePBX and Asterisk are both ostensibly open-source projects, the reality is that neither would exist without the commercial support from Digium, Tony Lewis, and now Sangoma. The continued existence of FreePBX depends upon people choosing FreePBX over other options (3CX, Cisco UCM, etc.) and buying either commercial modules, Sangoma hardware, SIPStation, or commercial support.
Like many technology products (e.g., Apple), FreePBX has a group of blindly loyal supporters (“fanboys”). They know the product. They love the product. And for same reason, they take criticism of the product as a personal attack on their decision to do so. To them, anyone who criticizes FreePBX should immediately quit using it and use something else. The problem with that approach is that if new users don’t keep choosing FreePBX, the corporate support that Sangoma provides will dry up and there won’t be a product for them to love down the road.
In some ways, I’m a FreePBX fanboy too. Aside from air conditioning and mobile phones, I cannot think of another product that has changed the way that I live and do business more than FreePBX. But, my love for the product has not blinded me to FreePBX’s faults. I’ve tried hard to get the developers to ensure that FreePBX stays easy to adopt and use so that new users will keep choosing it.
Now, I’m sure that the devs will reply to this post by pointing out that x,000 new systems were installed last month/quarter/year. Whatever that statistic is, however, it ignores the fact that there could have been exponentially more installations had the product been more user friendly and better documented. In the long term, if 3CX steals enough of FreePBX’s business, there won’t be a FreePBX left to love.
Seriously, FreePBX will let anyone set up a corporate grade PBX for FREE. 3CX costs money, and UCM is absurdly expensive. Financially speaking, there is no reason for anyone to ever choose any of the FreePBX alternatives. They choose them because they’re easier. Time is money.
I may well be the last in the line of long-terms users who worked to try to keep FreePBX easy to use. I’m often reminded of WiseOldOwl and MichiganTelephoneBlog, both of whom used write extensively in an effort to help people understand how to make FreePBX work, but both eventually just gave up. Ward Mundy’s love/hate relationship with FreePBX and its developers is well documented at his NerdVittles Blog. Now that I’ve invoked Ward’s name, I suspect he’ll be along shortly to comment… 
I used to post here and wrote on the Wiki pretty regularly, but I gave up as decisions by the devs to change the UI rendered much of what I wrote obsolete. Much of it was eventually deleted by the CM. Notwithstanding the claims made by FreePBX’s blindly loyal fanbase, FreePBX’s documentation is now abysmal. If, like me, you’ve been using it for ten years, its easy. However, if I had to start using the product today, I’m certain that its complexity would have caused me to give up.
So, again, I really do appreciate your support. Thank you.
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Net2Phone SIP Trunk-
This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.
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FreePBX design discussion
No, it does not. Installed the Distro straight up with no commercial modules and even with Sangoma phones does not give you a corporate grade PBX. It lacks numerous things a commercial (as in paid for) PBX has. In order to start be comparable, side by side, you need to start adding in commercial and third party modules.
Let’s clear the air on something as well, while FreePBX does not have any upfront licensing costs it is never free to setup or install. I know how much I’m worth per hour and because I can setup the PBX faster and more in depth that the other guys out there I know my worth is even more.
One of the biggest problems with FreePBX is the installers who undercut and devalue FreePBX (and themselves) because of a mind set of “corporate PBX for FREE”. So while the Mitel/Avaya/Nortel/et al guys are dropping down quotes around $10K someone that uses FreePBX comes in and figures cost of server + a few hours time = waaaay cheaper. That becomes the other problem. Because FreePBX is “free” and the person is doing it on the cheap, the server is cheap, the phones are cheap (generally EOL and out of date models).
Just remember just because you can do the project for $1500 while everyone else is quoting $5K-$7K for a project doesn’t mean you should.
Look, this is from experience and just in 2019. I worked with a company and did numerous installs/migrations of FreePBX 14. They had the same mind set about “corporate PBX for free” and in basically 90% of the cases they lost money. That also meant I lost money. In fact there are a couple in limbo because the project was under priced and now they have to deal with it out of pocket or convince the end user to put up more money. There were a couple that pretty much died because no one wanted to put up the money due to it being under priced.
Now for me on average the installs alone are about $2K-$3K if it’s hosted and can get over double that for 100% on-prem. installs. That includes setting up the voice services (or PBX), endpoints, equipment, wiring, punch downs and testing internal, external and 9-1-1 from all the endpoints. It averages out to about 90 endpoints per install give or take. Usually about 6-8 hours between setup before install and install/turnup onsite.
Again, just because FreePBX is $0 upfront to license and use to get you out the gate and running. It’s never $0 to setup/install and your time should never be treated like a $0 cost.
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Outbound CID not passing but shows in CDR report
Resolved:
So it looks like there has been some change with the latest Distro FreePBX as this issue is not happening to my other pbx’s.
So the resolution was to modify the SIP Trunk Outbound registration string in the following way
Original:
canreinvite=nonat
nat=yes
context=from-trunk
host=voip.itsvoiptime.com
secret=xxxxxxx
type=peer
username=xxxxxxxx
disallow=all
allow=ulaw&g729
fromuser=xxxxxxxxxxxxx
trustrpid=yes
sendrpid=yes
insecure=invite
qualify=yes
Modified: resolved the issue
username=xxxxxxx
type=peer
trustrpid=yes
sendrpid=yes
secret=xxxxxxx
qualify=yes
nat=yes
insecure=invite
host=voip.itsvoiptime.com
disallow=all
context=from-trunk
canreinvite=nonat
allow=ulaw&g729
removedfromuser=xxxxxxxxxxxremoved
It seems that the “fromuser=” entry was overriding any configured CID
I was able to see through my Telcom partner reseller portal what CID was showing and the CID matched the “fromuser=” entry. I figured since “username=” was already commented I didn’t need the “fromuser=” and I was right.
Hope this assists anyone.
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