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Getting help from Sangoma

January 4, 2019, 2:23 am
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I realize this thread is over a year old, but in my search for help with some problems I’m having with the Sangoma Partner Portal, I ran across this thread. When I log in to the portal, I don’t see anything about my account rep. I know it is Todd Bryant, however nothing on the portal indicates that. I went through the partner training up in Neenah with Tony and others, back in the Wayne Cook days, so I should be listed as a partner.

Also, who do I talk to about some problems I’m having using the partner portal?

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Asterisk Info, Peer report of SIP Peers indicating Host as PBX server IP

January 4, 2019, 2:37 am
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That’s a keen observation on extensions and NAT, Tom.

Let me share a brief background. We are making a wireless UC product that uses Asterisk/FreePBX for telephony. In order to popularize IP based telephony amongst small businesses, we have tried to simplify the telecom tech and made it plug-and-play for most common uses. Idea is to not require users to do complex configuration, irrespective of local or remote users. Hence, treating all users behind NAT by default, unless the network warrants a change due to layout or other use-case requirements. This config is working in over 25 locations but in this one location, I am noticing the morphed IP for remote users.

Like I mentioned, the system and calls are originating/terminating perfectly - just that this is causing brute-force protection to go kaput. I have manually compared all asterisk config files under /etc/asterisk between this system and other system where the IPs are recorded accurately as expected - and see no differences other than the environmental differences.

One key difference that is unique to this config is that it uses a private SIP trunk - hence, has two WAN eth ports active - with two isolated networks - one for internet and one for SIP trunk.

On turning on logs, I noticed that local address is reflecting as remote address too, just with a different port number.

I am enclosing the Asterisk sip settings screen below for your reference - but essentially, nat is set to route (so using rport and comedia) and there are four networks marked as local. Let me know if any other settings/log would be useful. Thanks.

Asterisk SIP Settings

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Voicemail Remote access

January 4, 2019, 3:41 am
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Hi Guys
I have managed to play around after reading some info and got this working

thanks
Billy G

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Fwconsole reload and limit of freepbx/asterisk

January 4, 2019, 4:04 am
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Hello dear community,

I would like to know if the reload time of the configuration is dependent on the number of extensions.
At the moment I only have 10 extensions and I already have 8 seconds of reload time.
I will have to install 1000 extensions and a dozen per day. My question is :
How can we not reload all the configuration during the reload but only that of the extension creates. Or if you know a realtime solution to keep it fluid. What’s the limit of Freepbx/asterisk ? I mean maximum supported of actives extensions with a lot of calls.

Config:
Cloud VPS
CPU: 4 vCore
RAM: 8192 MB
Storage: 100 GB SSD (voice messages sents to emails and not stored on the server)

Thank you in advance for your help.

Your friend freecall1

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Asterisk Info, Peer report of SIP Peers indicating Host as PBX server IP

January 4, 2019, 4:24 am
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I guess this thread comes closest to what we’re witnessing FreePBX can’t see extensions real IP

Unfortunately, no resolution, but the explanation hovers around doubting the router, which is a strong suspect in my opinion too. I did look at all possible configuration in this Aztech DSL8800GR router but to no avail. Frankly I don’t know what I should be looking for - I generally looked for NAT or SIP related settings, and there are none. Just the port forwarding rules that we’ve added.

But the fact that “sip show peer” is showing correct source IP in Reg. Contact field but replacing it with PBX IP in the Addr->IP field and using that to display as Host in Chan_sip Peers report is puzzling and causing me to also doubt some config in Asterisk playing up.

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Destination of No Answer Priority

January 4, 2019, 4:33 am
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Why would a ring group follow a Destination if no answer from a extension???

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FreePBX can't see extensions real IP

January 4, 2019, 4:36 am
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Hi Spaxton, sorry to be opening up this thread after 4 years. This is first well articulated thread about a problem we are facing too, on just one of the many boxes we have deployed. Were you able to find the root cause of the problem?

Captured in this ongoing thread… Asterisk Info, Peer report of SIP Peers indicating Host as PBX server IP

Hi Dicko, as you have explained, my hunch was also on the Router (Aztech DSL8800GR), but the fact that Asterisk “sip show peer” does show the correct source IP in “Reg. Contact” field but morphs it to PBX server IP in “Addr->IP” field, causes me to think it may be some interpretation of Asterisk as well.

Appreciate your analysis of our case. Thanks.

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Alert-Info

January 4, 2019, 5:00 am
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Hi,
I’m migrating from FreePBX 12.0.76.6 to FreePBX 14.0.5.5. We like to have different ringer sounds for internal and external call.

For our Snom and Gigaset Phones, I need the Alert-Info like this

Alert-Info: <http://telefon.iris-spielwelten.ch/ringer239.wav>;x-line-id=0;info=iris;delay=0

This works well with my old PBX, I can see it in the log of a Snom Phone.

If I put the same line in Inbound-Routes in FreePBX 14.0.5.5.

inbound%20route
inbound route.png1123x916 100 KB

The header gets like this:

Alert-Info: &lt;http://telefon.iris-spielwelten.ch/ringer239.wav&gt;;x-line-id=0;info=iris;delay=0

So there is a problem with the encoding, FreePBX adds the &lt; (“lower than”) and &gt; (“greater than”) HTML-Tags, and those gets also in the Dialplan and SIP-Header. The phones don’t recognize this format.

How can I solve this problem?
Thank you for your answers.

Kind regards
pace44

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SIP Call without Authorization Header

January 4, 2019, 5:22 am
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If you use chan_sip extensions without a SIP secret, they should work for you.

Until now, I had never tested a pjsip extension without a SIP secret, and I see the same thing. FreePBX generates an auth line for the endpoint even tho the secret is null, which causes the call to fail. Removing the auth line from the endpoint definition resolves this, but it’s not easily done by the end user. Since there are a non-zero quantity of SIP endpoints that don’t support authorization, so you may want to file a feature request for pjsip to support this.

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Module Upgrade CLI - The "--onlystdout" option does not exist

January 4, 2019, 5:44 am
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This morning I was running some upgrades from the command line and after each upgrade (other than framework) I see the error below:

[root@Nimbus ~]# fwconsole ma upgradeall
No repos specified, using: [standard,extended,commercial,unsupported] from last GUI settings

Module(s) requiring upgrades: announcement, bulkhandler, calllimit, callrecording, cdr, conferencespro, contactmanager, core, dashboard, endpoint, fax, faxpro, findmefollow, framework, irc, ivr, languages, miscdests, music, paging, parking, parkpro, pm2, queueprio, queues, qxact_reports, restapps, sipsettings, sysadmin, ucp, userman, vega, voicemail
Upgrading module ‘framework’ from 14.0.5.5 to 14.0.5.20
Downloading module ‘framework’
Processing framework
Downloading…
12997025/12997025 [============================] 100%
Finished downloading
Extracting…Done
Download completed in 3 seconds
Updating tables admin, ampusers, cronmanager, featurecodes, freepbx_log, freepbx_settings, globals, module_xml, modules, notifications…Done
installing files to /var/www/html…done
installing files to /var/lib/asterisk/bin…done
installing files to /var/lib/asterisk/agi-bin…done
Checking for upgrades…
2 found
Upgrading to 14.0.5.9…
-> Running PHP script /var/www/html/admin/modules/framework/upgrades//14.0.5.9/upgrade.php
Upgrading to 14.0.5.9…OK
Upgrading to 14.0.5.12…
-> Running PHP script /var/www/html/admin/modules/framework/upgrades//14.0.5.12/upgrade.php
Upgrading to 14.0.5.12…OK
framework file install done, removing packages from module
file/directory: /var/www/html/admin/modules/framework/amp_conf removed successfully
file/directory: /var/www/html/admin/modules/framework/upgrades removed successfully
file/directory: /var/www/html/admin/modules/framework/start_asterisk removed successfully
file/directory: /var/www/html/admin/modules/framework/install removed successfully
file/directory: /var/www/html/admin/modules/framework/installlib removed successfully
Building Packaged Scripts…Done
Generating CSS…Done
Module framework successfully installed
Updating Hooks…Done
Upgrading module ‘core’ from 14.0.18.37 to 14.0.18.45
Downloading module ‘core’
Processing core
Downloading…
1108970/1108970 [============================] 100%
Finished downloading
Extracting…Done
Download completed in 1 seconds

The “–onlystdout” option does not exist.

ma [-f|–force] [-d|–debug] [–edge] [–stable] [–color] [–skipchown] [-e|–autoenable] [–skipdisabled] [–snapshot SNAPSHOT] [–format FORMAT] [-R|–repo REPO] [-t|–tag TAG] [–sendemail] [–] []…

Upgrading module ‘sipsettings’ from 14.0.27.5 to 14.0.27.7
Downloading module ‘sipsettings’
Processing sipsettings
Downloading…
259586/259586 [============================] 100%
Finished downloading
Extracting…Done
Download completed in 0 seconds

The “–onlystdout” option does not exist.

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Asterisk Info, Peer report of SIP Peers indicating Host as PBX server IP

January 4, 2019, 5:52 am
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OK so there are a few issues with this. First, pointing to a 4 year old thread isn’t going to help. Things have changed in the last four years and they could pretty much be irrelevant. Second, you’ve made custom modifications and well that means you’ve changed how FreePBX normally works. That can be an issue. Third, you’re not understanding how this is working in regards to the Addr->IP and the Reg. Contact.

The Reg. Contact is actually what is in the Contact Header sent by the device. The Addr->IP is the Received Address i.e. the public WAN the request came from. So like I said, if you are seeing the Asterisk IP there instead, you have NAT/network configuration issues.

Without seeing what “custom” work you’ve done, no clue as to what the actual issue is. Is there anything different about this box then the others you’ve done? Are they in the same locations? Do they go on-premises?

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Commercial Voicemail Notification doesn't work

January 4, 2019, 6:05 am
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Hi Dave, sorry for my late reply

I’ll check the logs and attach them to you.

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Asterisk13-odbc code change

January 4, 2019, 6:23 am
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This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.

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Any physical phones handle more than 50 BLFs?

January 4, 2019, 6:33 am
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While I second the opinion of using FOP2 (that’s what we use for our receptionists), I have used a Yealink T46G with 3 EXP40 side cars (for 120 BLF’s) without issue.

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This issue appeared when upgrade framework

January 4, 2019, 6:37 am
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SIP Call without Authorization Header

January 4, 2019, 6:38 am
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I managed to do it by deleting the auth-part in the pjsip.endpoint.conf file.

I think this is really bad practice because it gets overwritten whenever I press the “Reload Config” button on the web interface.
I hope somebody knows a better way. the CHAN_SIP possibility did not work.

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Software to send fax

January 4, 2019, 6:40 am
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I know you mentioned that Hylafax isn’t what you’re looking for… but I would urge you to take another look.

  1. I have Hylafax server set up on my PBX, with 8 IAX Virtual modems.
  2. My PC’s use the Winprint Hylafax Gui, which works as a virtual printer. Anything that can be printed, can be faxed. They literally hit “Print” like they normally would, and choose Hylafax… they are then prompted for the fax number and off it goes.
  3. The users can chain multiple pages together into a single fax, there is an “address book”, they get confirmation emails if it did or didn’t send.

It’s realy an amazing solution.

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Commercial Voicemail Notification doesn't work

January 4, 2019, 6:49 am
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I checked the file myself and it’s saying that the length of the voice mail that was left was not enough. tried to call and leave a voice mail with 10 seconds or longer. it worked fine and I got a voice mail notification email however it’s not coming from the Voice Mail notification commercial module.

Attaching log, please change the extension of full.tgz to full because it’s not compressed file.
Thanks

full.tgz (189.4 KB)

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FaxPro module operation

January 4, 2019, 6:50 am
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@tm1000, I followed your advise to post the question in a new topic and it seems like it has lost as no one bother to even looking at it!

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Asterisk Info, Peer report of SIP Peers indicating Host as PBX server IP

January 4, 2019, 6:53 am
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Tom, you’re right - the other thread symptomatically looks similar but could be different underneath. The thread doesn’t document Asterisk versions in play, so can’t be sure.

Thanks for explaining the interpretation of the two fields. I was under a wrong impression that probably Addr->IP is derived from Reg. Contact in someway, but probably per your explanation, they are two independent fields populated independently. (An aside, can you please point me to any document that explains all these fields and their source of information?). Is there any line of investigation you can suggest to carried out on the Router? What typically should be looked at? I could not find a explicit NAT/SIP related settings on it. I could enable logging on it and see if it throws anything.

All boxes are in different locations; they deploy on-premise and sit in client’s existing networks, over wire or wirelessly. Our box is also an integrated wireless router in itself, in addition to UC/PBX. However, asterisk/freepbx is configured under prescribed norms. As I stated earlier, on networking front, there are four local networks, NAT is set to route for SIP peers as well as trunks (in Advanced as well as SIP and Extensions settings for all users).

Sorry if this information is insufficient to draw some ideas or suggest an investigative path. Please suggest whatever best you can and feel free to ask for any specific data/log that can help assess. Thanks.

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