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Choppy audio
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Got RTP packet from , Sent RTP packet to on console
Thank you sir, that was it.
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Outbound CID how to add +country code
I agree with the concept but not the specifics. Your internal format should match what is used on UK landlines and mobiles.
On an incoming call from London, you would change +442072345678 to 02072345678. On a call from Paris, you would change +33123456789 to 0033123456789. If the called number comes in as +44113334455 it would be changed to 0113334455. These changes would be done with a custom context similar to from-pstn-e164-us but modified for UK.
On outgoing calls, you would define [macro-dialout-trunk-predial-hook] to rewrite caller ID. See
Rewriting the outbound number can be done in the Dialed Number Manipulation Rules for the trunk.
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ARI Password keeps changing and I don't know why
Hello, so occasionally I will see “Apply Settings” pop up after I haven’t made any changes, and so sometimes I will backup /etc/asterisk and then Apply, and compare the diff to see what changed.
It always seems to be the ARI Password, (or maybe just the salt?). The only file that is changed, is ari_additional.conf, and it’s the secret line that is changed. I don’t think I’m doing anything to cause the password to be changed, and I’m definitely not doing it intentionally, and no one else has authorization for the server, and it’s behind a whitelist only firewall. (Although it wasn’t always)
Is this normal behavior? I’ve looked at the cron logs and don’t see anything that sticks out, but I haven’t yet been able to nail down the exact time this ‘change’ is made (that makes the Apply Changes button appear)
Let me know if you have any questions, thanks 
FreePBX 14.0.5.2
Asterisk 13.22.0
PS. ARI Is disabled in Advanced Settings, however in ari_additional.conf there is still the ‘freepbxuser’ user, (and it’s this user’s password that keeps changing).
tldr; Is is normal for the ARI password for user ‘freepbxuser’ to change on it’s own?
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Location of EPM config files
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Inbound Calls from Sprint Poor Quality
You betcha. There could have been an internal port. Look up the problematic DID at https://apeiron.io/lrn . Who is the carrier? Does is show a ported_on_date much later than when you ported it in?
Look up both the LRN and the DID at https://www.telcodata.us/search-area-code-exchange-detail . What rate center(s) are shown? If they don’t match, that could be an issue.
Do you have Google Voice Enterprise? If so, you should get awesome support; call and tell them that you have trouble reaching a number that works fine from a landline.
Otherwise, do you have access to a phone on Sprint (or a Sprint MVNO) for testing?
Can you see the trouble calling the DID from your own Flowroute trunk?
Were the multiple invites from Flowroute SIP level retries (same Call-ID), or route level (INVITE was CANCELed and resubmitted with a different Call-ID header)?
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Upgrade HW from Raspberry Pi to something a bit more durable
IF you wanted to stay with a Pi, I believe that the Pi3 and some Pi2’s have the ability to boot and run from a USB attached hard drive. No need to worry about killing an SD card over time.
https://www.raspberrypi.org/documentation/hardware/raspberrypi/bootmodes/msd.md
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Upgrade HW from Raspberry Pi to something a bit more durable
Instead of booting from USB (which requires blowing a fuse, so that’s permanent), you can simply leave the boot partition on the SD card and put everything else on a USB SSD or hard drive. Unless you are updating the kernel, the boot partition never gets written, so it shouldn’t wear out.
For example:
~# cat /etc/fstab
/dev/mmcblk0p1 /boot vfat defaults 0 2
/dev/sda1 / ext4 errors=remount-ro,noatime,nodiratime,commit=120 0 1
tmpfs /tmp tmpfs defaults,nodev,nosuid 0 0
/dev/sda2 /dvr ext4 noatime,commit=120 0 0
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Upgrade HW from Raspberry Pi to something a bit more durable
Just as a point of information, cheap ssd usb disks and sd cards use the the same underlying technology, so wear-levelling is not necessarily as better as you would like.
I would definitely get an atomic pi as has been suggested for $40 (they are flying off the shelf so be quick) the emmc drives are well thought out.
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IVR - Invalid Retry Recording language
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Callback app, Agent First
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Inbound Calls from Sprint Poor Quality
First of all, it seems like the duplicate origination symptom is no longer appearing. I’m still having the poor quality audio issue, and longer (although not as long as before) ring time before hitting my system. Also, I think that the multiple invites may have stopped happened before and then come back later, so this seems like more of a problem than a good thing, because Flowroute is coming back to me saying they can’t reproduce the multiple invites anymore, but the other problems are still happening, but slightly less intense. Ugh.
To answer you questions:
The carrier seems to be “Flowroute LLC”
The ported_on_date matches the date I would expect, December of 2018
Rate centers do seem to be different. When I input the area code and prefix of my DID into telcodata.us it lists the ratecenter as the city I would expect, while the LRN rate center is different.
I hadn’t heard of Google Voice Enterprise, but I do have a gsuite account, so I think I do. I’ll check out that possibility.
Unfortunately I don’t have access to a Sprint phone. But I can still test with GV and Flowroute numbers.
I do see the problem when I call from another flowroute trunk/DID associated with my account. Is that what you were asking?
Like I mentioned, the invite problem doesn’t seem to be happening anymore, but I opened up a packet capture I have, and it seems like the problem is route level. Kinda. So each of the invites has a different Call-ID, but I don’t see the CANCEL part. I’m not great at interpreting packet captures, so I may just be missing the CANCELs.
Also, I forgot to mention this initially, but the Invites come in multiple seconds apart. In the packet capture I have, the second came 14 seconds after the first, and the third came a little less than 4 seconds after the second.
Do these “puzzle pieces” point to anything?
Should I bring up the rate center potential problem with Flowroute, or the fact that this seems to be a route level problem?
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Outbound Caller ID Name
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Synchronize Ringing on Grandstream Phones
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Anonymous:phone-context=unknown
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Problems recognizing distinctive ring using Sangoma B600E
I am not having much success using the Sangoma B600E to recognize distinctive ring patterns.
I have four ring patterns (the maximum permitted in the USA) on my POTS line:
- very long (standard ring pattern)
- long long
- short long short
- short short long
I have added the following to the chan_dahdi_channels_custom_conf file:
usedistinctiveringdetection=yes
distinctiveringaftercid=no
I have configured the Asterisk Logfile Settings so that everything is turned on in my full log.
The Detected ring pattern messages in my full log show that two of my ring patterns (very long and short short long) are always detected as 0,0,0. One of my ring patterns (long long) is fairly consistently detected as 332,0,0 to 335,0,0. The fourth ring pattern is often detected as 0,0,0, but has also been detected as 369,0,0 and 307,0,0. All four ring patterns sound correct on an analog phone connected to the same POTS line. I have tried two different FXO ports on the B600E with similar results.
My questions are:
- Are there any DAHDI settings that I need to adjust to achieve consistent distinctive ring detection?
- The second and third value of the detected ring pattern is always 0. Is this normal? What are the meanings each of the three numbers?
- It appears as if RING_PATTERNS is hardcoded to 3 in sig_analog.h. Although I believe I can make this work (once the rings are correctly detected), it is not optimal, since I would need to make one of the ring patterns use my default context, and would thus not have any way to direct errors to a different extension. Is there any chance that RING_PATTERNS could be increased to 4 in a subsequent update? I suppose I could build this module myself, but then I would need to deal with code signing.
- What is the proper way to define the dring1context, dring2context, dring3context, and default context? I have tried making this the phone number (e.g. 5035551212 – not my real phone number) and then using the matching value as the DID number on my inbound routes, but the calls (even for the long long ring) don’t get routed to the desired extension. Once the rings are properly detected, what is the correct way to route them? (See the error messages below.)
chan_dahdi.c: Matched Distinctive Ring context 5035551212
pbx.c: Channel ‘DAHDI/1-1’ sent to invalid extension but no invalid handler: context,exten,priority= 5035551212,s,1
sig_analog.c: Hanging up on ‘DAHDI/1-1’
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Trunk stops working after PBX IP address change
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How to configure the new Property Manager app
Where is the HPMS Directory?
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Issues With System Firewall
Yes. I had turn on/off firewall, restart asterisk, fwconsle stop start but it’s not run. But bewteen 2hours it’s auto run again
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Issues With System Firewall
Yes. I had turn on/off firewall, restart asterisk, fwconsle stop start but it’s not run. But bewteen 2hours it’s auto run again.
It’s error at monday each off week. That is show error:
Unparseable output from getservices - [“Exception: Asterisk is not connected in file /var/www/html/admin/libraries/php-asmanager.php on line 236”,“Stack trace:”," 1. Exception->() /var/www/html/admin/libraries/php-asmanager.php:236"," 2. AGI_AsteriskManager->send_request() /var/www/html/admin/modules/firewall/Smart.class.php:447"," 3. FreePBX\modules\Firewall\Smart->getPjsipContacts() /var/www/html/admin/modules/firewall/Smart.class.php:437"," 4. FreePBX\modules\Firewall\Smart->getRegistrations() /var/www/html/admin/modules/firewall/Smart.class.php:69"," 5. FreePBX\modules\Firewall\Smart->getAllPorts() /var/www/html/admin/modules/firewall/Firewall.class.php:1026"," 6. FreePBX\modules\Firewall->getSmartPorts() /var/www/html/admin/modules/firewall/bin/getservices:22"] - returned .
I Have restart asterisk, fwconsle stop start but it’s not run.
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