I notice today that Call Data Record report does not show detail (blank) when I click individual UniqueID link.
UniqueID link shows call detail for the timestamp before the module update, I am wondering whether it is related to the module update or not. (CDR module update was performed on 1/11, I can see UniqueID info prior to the date and all calls shows blank after module update)
Just wondering anyone experiencing this. Is there a way to roll back the module version so that I can test and see whether the issue is related to module update?
Any feedback would be appreciated.
Best regards,
Environment:
Asterisk version: 13.24.1
FreePBX version: 14.0.5.25
CDR module version: upgraded from 14.0.5.14 to 14.0.5.16
hello I used freepbx sangoma with this personalization of file extension.custom.conf , then i changed “from-internal” with “from-internal-custom” in the 211 internal advanced , but did this configuration the internal stops working . In the past i used this configuration in pika asterisk appliance without any problem.
Could someone help me ? Thank you
edit==> extension-custom.conf
…
include => from-internal-custom
Hi,
I want to move to self hosted FreePBX. For my extension, I have an endpoint in the office and one at home. For that reason, I want to use PJSIP.
I created the extension and configured my audiocodes 440HD phone. Phone registers and makes calls, but there is no audio at all.
I have the UDP port 5060 and UDP Range 10000 to 20000 forwarded to the FreePBX.
For testing purposes, I set up CHANSIP and configured the endpoint at home and it worked fine.
I also used a softphone at home using PJSIP and it worked fine.
Lastly, DMZ’d my audiocodes endpoint at home and then I had one way audio working using PJSIP.
From my reading, PJSIP requires NAT to be enabled on the endpoint, but I see nothing in the configuration of the Audiocodes 440hd to change NAT settings.
Any ideas what I should do at this point to get my phone working with PJSIP?
I have to say that it has been forever since this system was initially set up. The system has been moved onto new hardware and everything is working fine other than the analog channels. My thought was to remove and re-add the analog trunk but that has turned out to be a mistake, now all the analog channels are not working.
For the time being I have dug up some ATA FXO/FXS boxes and have things sorta working but faxing, ya, old-school faxing with fax machines, is not working well as you might expect. So I’m keen to get the TDM400 back cookin’ again.
DAHDI, Asterisk and FreePBX all seem to start clean and the TDM400 appears to be there as lspci shows “Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface” as expected. No apparent interrupt conflicts and the ports are lighting up as configured.
I have 3 site’s in each site has different phone system 1 avaya office, 1 avaya bcm450 and freepbx 14. from avaya office they can call to freepbx. But bcm450 cant call freepbx when I dial extension from bcm450 I receive a voice tone extension;phone-context=subscriber.private
Sometimes call audio drops in the middle of a call so that the other end can’t hear me. When that happens, the only thing I can do is hang up and call them back. Where would I begin troubleshooting something like this?
We have Grandstream gxp2130 phones managed with the commercial endpoint manager with a fresh install of FreeBPX 14.0.5.25
All has been going according to plan (we have a similar setup with an older version of FreePBX at a different location) until we attempt to transfer calls.
The blind transfer seems to try and transfer, according to the phone’s display - but nothing appears in FreePBX. But that’s just the related issue.
When I hang up the phone nothing happens in FreePBX. It doesn’t recognize the phone hung up - ever.
This same issue happens when I perform a Page. I can make the page but when I hang up all of the paged phones remain off-hook, because the phone didn’t tell FreePBX that it hung up.
But when I dial an extension directly, talk and hang up, the phone appears to send the hangup signal properly and it appears on the Asterisk Console.
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We have a Polycom IP 6000 that is registered using the commercial endpoint manager. It can make internal calls to other extensions just fine.
It cannot receive internal calls - all calls go to the unavailable voicemailbox option.
I’ve checked the DND in FreeBPX for this extension, it’s disabled
I’ve checked the DND on the Polycom, it’s disabled
I’ve rebooted the FreePBX server
I’ve rebooted the Polycom
Viewing the FreePBX logs shows the phone registers itself and immediately becomes unavailable:
[2019-02-04 15:16:47] VERBOSE[8181] res_pjsip_registrar.c: Added contact 'sip:1622@10.8.2.106:5060' to AOR '1622' with expiration of 120 seconds
[2019-02-04 15:16:50] VERBOSE[8181] res_pjsip/pjsip_options.c: Contact 1622/sip:1622@10.8.2.106:5060 is now Unreachable. RTT: 0.000 msec
Any thoughts as to why?
Is the Polycom IP6000 broken with the current pjsip driver software?
Our SIP trunk provider is Twilio. We have configured in Twilio the trunk “Termination Sip Uri” as “alliancemaintenance.pstn.twilio. com”, and that is what is entered in the FreePBX -> Trunk -> sip Settings -> Outgoing -> PEER Details:
host=alliancemaintenance.pstn.twilio. com
type=peer
dtmfmode=rfc2833
disallow=all
allow=ulaw
As far as I can tell, the URL resolves to Twilio’s IPs properly in DNS. I check it using MXToolBox DNS Lookup.
We have periods where we can’t call out, and I’m thinking it might be related to these messages in the Asterisk Log File:
[2019-01-30 15:02:08] ERROR[11964] netsock2.c: getaddrinfo("alliancemaintenance.pstn.twilio. com", "(null)", ...): Name or service not known
[2019-01-30 15:02:08] WARNING[11964] acl.c: Unable to lookup 'alliancemaintenance.pstn.twilio. com'
[2019-01-30 15:07:08] ERROR[11964] netsock2.c: getaddrinfo("alliancemaintenance.pstn.twilio. com", "(null)", ...): Name or service not known
[2019-01-30 15:07:08] WARNING[11964] acl.c: Unable to lookup 'alliancemaintenance.pstn.twilio. com'
[2019-01-30 15:12:08] ERROR[11964] netsock2.c: getaddrinfo("alliancemaintenance.pstn.twilio. com", "(null)", ...): Name or service not known
[2019-01-30 15:12:08] WARNING[11964] acl.c: Unable to lookup 'alliancemaintenance.pstn.twilio. com'
These two messages will repeat every five minutes, and continue for 20 or so minutes, sometimes up to a few hours (with no other log activity mixed in).
Note: since I’m a new user, the Forum doesn’t allow me to post “links”. So each time dot-com is used in this message, I have put a space in front of the dot-com so the Forum doesn’t consider it a link.
Does anyone have any insight into this issue? Thanks.
In case it is helpful, here is a larger section of the log with the log lines before and after the period in question:
[2019-01-30 13:59:00] VERBOSE[25490][C-000001c5] pbx.c: Executing [s@crm-hangup:7] AGI("SIP/Twilio1-000003ba", "sangomacrm.agi") in new stack
[2019-01-30 13:59:00] VERBOSE[25490][C-000001c5] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2019-01-30 13:59:01] VERBOSE[25490][C-000001c5] res_agi.c: <SIP/Twilio1-000003ba>AGI Script sangomacrm.agi completed, returning 0
[2019-01-30 13:59:01] VERBOSE[25490][C-000001c5] pbx.c: Executing [s@crm-hangup:8] Return("SIP/Twilio1-000003ba", "") in new stack
[2019-01-30 13:59:01] VERBOSE[25490][C-000001c5] app_stack.c: Spawn extension (ext-group, h, 1) exited non-zero on 'SIP/Twilio1-000003ba'
[2019-01-30 13:59:01] VERBOSE[25490][C-000001c5] app_stack.c: SIP/Twilio1-000003ba Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
[2019-01-30 15:02:08] ERROR[11964] netsock2.c: getaddrinfo("alliancemaintenance.pstn.twilio. com", "(null)", ...): Name or service not known
[2019-01-30 15:02:08] WARNING[11964] acl.c: Unable to lookup 'alliancemaintenance.pstn.twilio. com'
[2019-01-30 15:07:08] ERROR[11964] netsock2.c: getaddrinfo("alliancemaintenance.pstn.twilio. com", "(null)", ...): Name or service not known
[2019-01-30 15:07:08] WARNING[11964] acl.c: Unable to lookup 'alliancemaintenance.pstn.twilio. com'
[2019-01-30 15:12:08] ERROR[11964] netsock2.c: getaddrinfo("alliancemaintenance.pstn.twilio. com", "(null)", ...): Name or service not known
[2019-01-30 15:12:08] WARNING[11964] acl.c: Unable to lookup 'alliancemaintenance.pstn.twilio. com'
[2019-01-30 15:17:08] ERROR[11964] netsock2.c: getaddrinfo("alliancemaintenance.pstn.twilio. com", "(null)", ...): Name or service not known
[2019-01-30 15:17:08] WARNING[11964] acl.c: Unable to lookup 'alliancemaintenance.pstn.twilio. com'
[2019-01-30 15:22:08] ERROR[11964] netsock2.c: getaddrinfo("alliancemaintenance.pstn.twilio. com", "(null)", ...): Name or service not known
[2019-01-30 15:22:08] WARNING[11964] acl.c: Unable to lookup 'alliancemaintenance.pstn.twilio. com'
[2019-01-30 15:27:08] ERROR[11964] netsock2.c: getaddrinfo("alliancemaintenance.pstn.twilio. com", "(null)", ...): Name or service not known
[2019-01-30 15:27:08] WARNING[11964] acl.c: Unable to lookup 'alliancemaintenance.pstn.twilio. com'
[2019-01-30 15:29:09] VERBOSE[12031][C-000001c6] netsock2.c: Using SIP RTP TOS bits 184
[2019-01-30 15:29:09] VERBOSE[12031][C-000001c6] netsock2.c: Using SIP RTP CoS mark 5
[2019-01-30 15:29:09] VERBOSE[40851][C-000001c6] pbx.c: Executing [+18008508610@from-trunk:1] NoOp("SIP/Twilio0-000003bc", "Catch-All DID Match - Found +18008508610 - You probably want a DID for this.") in new stack
[2019-01-30 15:29:09] VERBOSE[40851][C-000001c6] pbx.c: Executing [+18008508610@from-trunk:2] Log("SIP/Twilio0-000003bc", "WARNING,Friendly Scanner from 54.172.60.0") in new stack
[2019-01-30 15:29:09] WARNING[40851][C-000001c6] Ext. +18008508610: Friendly Scanner from 54.172.60.0
[2019-01-30 15:29:09] VERBOSE[40851][C-000001c6] pbx.c: Executing [+18008508610@from-trunk:3] Set("SIP/Twilio0-000003bc", "__FROM_DID=+18008508610") in new stack
[2019-01-30 15:29:09] VERBOSE[40851][C-000001c6] pbx.c: Executing [+18008508610@from-trunk:4] Goto("SIP/Twilio0-000003bc", "ext-did,s,1") in new stack
[2019-01-30 15:29:09] VERBOSE[40851][C-000001c6] pbx_builtins.c: Goto (ext-did,s,1)
[2019-01-30 15:29:09] VERBOSE[40851][C-000001c6] pbx.c: Executing [s@ext-did:1] Set("SIP/Twilio0-000003bc", "__DIRECTION=INBOUND") in new stack
When my clients receive an inbound call to their phone system, they pick up and the caller states they need to be transferred to a different extension. Once that call is transferred to the next endpoint, when the user picks up their handset or hits the answer button, they hear a “busy signal” and the caller gets their voicemail.
Anyone else experience this and find a solution? My clients are using Grandstream GXP 2170 phones.
I’m in the middle of an interesting puzzle of transferring active calls to specific extension’s voicemail with a press of a button.
One of our client users has a sidecart for their Grandstream GXP2170, solely for transferring calls to specific extension’s voicemails as needed.
The call comes in, the front desk takes the call and routes it as needed. If the extension to be transferred to is unavailable for calls (yet their DND is not turned on…), the call is to be sent straight to voicemail, without ringing the phone. The front desk would press the transfer-speeddial button to transfer the call to that extension’s voicemail.
I have the sidecart’s buttons set as Transfer > *103 for an example.
I’ve thumbed down the issue to be using buttons, as manually hitting transfer, then *, followed by the extension. I currently have featurecodes such as *1, *10, *11, and *12 disabled, as they were conflicting, and now transfer to voicemail work.
Other than *1, the others are not used. I’ve changed *1 to **1 for continued use. I’ve tried changing the transfer to voicemail feature code to use **, however the Grandstreams displays on screen, no audio recording, I don’t have the exact wording off hand, not matching dialplan.
I’m looking to see if there are better solutions to this issue. As far as I’ve seen, disabling *10, *11, and *12, and changing *1 to **1 has been the best alternative. As I write this, I realized I didn’t test and confirm **1 is a conflict while in an active call, though it’ll be manually pressed when needed.
I’m migrating my small office (10 phones) off an old Cisco POTS IP-PBX to FreePBX and trying out Nexmo elastic trunking.
I’m trying to configure an Edgemarc 2900 SBC, but I’m finding the interface a little esoteric. Things like blacklisting/whitelisting, firewall rules, DMZ, aren’t configured in any way I’m familiar with. As of now, I have the ALG turned off. I can register with the provider and make outgoing calls with two way audio. I cannot call in. When I call in I just get silence for a while before the call terminates. Looking at the asterisk console (asterisk -vvvvvr), nothing is coming into the PBX which makes sense since I haven’t told the router to send it anything.
When I try to configure the ALG using Edgewater’s guide which seems to be their recommended way, I lose registration. I’ve been tinkering for a while, but can’t figure it out. Am I correct that my three options are, setting up the ALG, putting the PBX in a DMZ, or port forwarding?
Any guidance would be greatly appreciated. Thank you.
Is it possible to integrate freepbx with a Sip Trunk provider and Skype for Business at the same time?
The reason why I am wondering because SIP trunk provider uses 5060, Chansip using 5160
Skype for Business using 5060 but TCP and Chansip has 5160 tcp.
Would this work? wouldn’t there be any port conflict in this case? I tried to integrate it but I have problem with calling from Skype for Business to FreePBX after I integrated my Sip Trunk provider.
I’m having problems using endpoint manager to configure some DB20/D10 handsets. I had a configuration that was working with 2 handsets and we’ve decided to scale up and add eight more. It has been quite a while since I first configured the original two handsets and now I’m having problems. I added a third and a forth, but something went strange when I added the 5th.
The attached screens shots show the situation with 4 and 5 handsets configured. You’ll notice that the IPEIs have shifted around, and none are for the new added extension (6001) which should have an IPEI of 02EB6C51DC. Also one of the IPEIs is repeated (02EB687E5E) which is clearly wrong. Deleting the added extension (6001) restored the original situation.
I have tried various things to try to clear this problem. Deleted and re-adding extensions; Removing and reinstalling Endpoint Manager in a bid to delete any ‘corrupted’ database state. None of these has worked and IPEI 02EB687E5E seems to be very sicky and accrues to every handset configure when viewed in the DB20 web UI. The correct IPEI are showing in the EPM config pages.
So I’m stuck and seeking inspiration and help.
Another strangeness in the config view from the DB20 Web UI is the server configuration which includes two servers configured with just port numbers. That may be normal, but seems strange and may be a clue.
No issues here I just wanted to share with you how I expanded the drive of my freepbx import in aws.
So I followed the advice on this topic How to install Freepbx Distro on Amazon AWS EC2 machine and created a virtual machine which I then imported to aws and my instance was running with almost no issues however as noted in the aforementioned topic even if when you launch the instance you assign it a bigger drive then what you originally had on the virtual machine the system will still run on the original size, to overcome this @yitzyf recommended to use growpart to grow the partition but it seems that there are more steps involved so I did some digging and figured it out, I should have posted this on that topic but it’s already closed so I’m posting it here so that if anyone else has this issue I hope I can save them some time.
My virtual machine originally had a 20GB drive and I launched it with a 200GB drive your situation may vary but the same instructions should apply, also since we are gonna play with partitions make sure to create a backup image in AWS before continuing.
So after you launch the instance with a bigger drive log in with ssh.
Run lsblk to list disks, partitions, lvms and structure:
[root@freepbx ~]# lsblk
NAME MAJ:MIN RM SIZE RO TYPE MOUNTPOINT
xvda 202:0 0 200G 0 disk
├─xvda1 202:1 0 2G 0 part /boot
└─xvda2 202:2 0 18G 0 part
├─SangomaVG-root 253:0 0 15.2G 0 lvm /
└─SangomaVG-swaplv1 253:1 0 2G 0 lvm [SWAP]
What this tells you is that you have one disk xvda with 200GB, the disk has 2 partitions xvda1 which is the boot partition with 2GB and xvda2 with 18GB, on the xvda2 partition there’s a volume group named SangomaVG which contains 2 logical volumes (LVM) SangomaVG-root which is mounted at / with 15.2GB this is the one you wanna expand and then a swap volume SangomaVG-swaplv1 with 2GB.
If you login to the web gui and check the storage in system admin you will indeed see Drive Usage 4,851.02M / 15,594.00 M (32%)
This shows you the full names of the the partitions and LVM’s along with some other useful info, Take note of the partition and LVM names.
If you run vgdisplay it will display Volume group information:
[root@freepbx ~]# vgdisplay
— Volume group —
VG Name SangomaVG
System ID
Format lvm2
Metadata Areas 1
Metadata Sequence No 4
VG Access read/write
VG Status resizable
MAX LV 0
Cur LV 2
Open LV 2
Max PV 0
Cur PV 1
Act PV 1
VG Size 18.04 GiB
PE Size 4.00 MiB
Total PE 50699
Alloc PE / Size 4413 / <17.24 GiB
Free PE / Size 46286 / 180.80 GiB
VG UUID sE36sO-ye0A-FeW9-AM4F-a0kv-nysi-MvIfGc
All good so far now to the actual editing you will need to follow these 4 steps:
expand partition.
expand volume group to take advantage of new partition space.
expand logical volume SangomaVG-root within the volume group.
expand underlying file system after logical volume expansion.
Extend partition 2 in /dev/xvda to fill empty space until end of disk or next partition:
[root@freepbx ~]# growpart /dev/xvda 2
CHANGED: partition=2 start=4098048 old: size=37844992 end=41943040 new: size=415332319,end=419430367
Run lsblk again to verify that partition xvda2 was expanded:
[root@freepbx ~]# lsblk
NAME MAJ:MIN RM SIZE RO TYPE MOUNTPOINT
xvda 202:0 0 200G 0 disk
├─xvda1 202:1 0 2G 0 part /boot
└─xvda2 202:2 0 198G 0 part
├─SangomaVG-root 253:0 0 15.2G 0 lvm /
└─SangomaVG-swaplv1 253:1 0 2G 0 lvm [SWAP]
It now shows xvda2 has 198GB great!
2. expand volume group to take advantage of new partition space.
This command resizes the amount of space that a LVM volume group can use on a partition and needs to be run if you resize its partition:
[root@freepbx ~]# pvresize /dev/xvda2
Physical volume “/dev/xvda2” changed
1 physical volume(s) resized / 0 physical volume(s) not resized
Run vgdisplay to confirm volume group resize:
[root@freepbx ~]# vgdisplay
— Volume group —
VG Name SangomaVG
System ID
Format lvm2
Metadata Areas 1
Metadata Sequence No 4
VG Access read/write
VG Status resizable
MAX LV 0
Cur LV 2
Open LV 2
Max PV 0
Cur PV 1
Act PV 1
VG Size 198.04 GiB
PE Size 4.00 MiB
Total PE 50699
Alloc PE / Size 4413 / <17.24 GiB
Free PE / Size 46286 / 180.80 GiB
VG UUID sE36sO-ye0A-FeW9-AM4F-a0kv-nysi-MvIfGc
It now shows VG size as 198.04 GB great!
3. expand logical volume SangomaVG-root within the volume group.
Now we need to resize the logical volume, SangomaVG-root, remember the full name from earlier whae we ran fdisk -l ? good, We use lvresize for this. There is a funny looking argument that is passed to resize. We don’t say that we want to fill the rest of the volume, we say that we want to add 100% of the free space to the volume.
[root@freepbx ~]# lvresize -l +100%FREE /dev/mapper/SangomaVG-root
Size of logical volume SangomaVG/root changed from <15.24 GiB (3901 extents) to 196.04 GiB (50187 extents).
Logical volume SangomaVG/root successfully resized.
Now re-run lsblk to make sure the LVM was expanded:
[root@freepbx ~]# lsblk
NAME MAJ:MIN RM SIZE RO TYPE MOUNTPOINT
xvda 202:0 0 200G 0 disk
├─xvda1 202:1 0 2G 0 part /boot
└─xvda2 202:2 0 198G 0 part
├─SangomaVG-root 253:0 0 196G 0 lvm /
└─SangomaVG-swaplv1 253:1 0 2G 0 lvm [SWAP]
It shows us that LVM SangomaVG-root now has 196GB great!
4. expand underlying file system after logical volume expansion.
Finally we want to resize the actual underlying file system on the logical volume. Since we’re using centos and xfs file system we will run xfs_growfs:
[root@freepbx ~]# xfs_growfs /dev/mapper/SangomaVG-root
meta-data=/dev/mapper/SangomaVG-root isize=512 agcount=4, agsize=998656 blks
= sectsz=512 attr=2, projid32bit=1
= crc=1 finobt=0 spinodes=0
data = bsize=4096 blocks=3994624, imaxpct=25
= sunit=0 swidth=0 blks
naming =version 2 bsize=4096 ascii-ci=0 ftype=1
log =internal bsize=4096 blocks=2560, version=2
= sectsz=512 sunit=0 blks, lazy-count=1
realtime =none extsz=4096 blocks=0, rtextents=0
data blocks changed from 3994624 to 51391488
Now go back to the web gui and refresh the system admin storage page:
I’m having a crazy issue with VLANs in the latest version of FreePBX.
We’re using Grandstream GXP 2130 phones with FreePBX 14. It’s running on a Dell R710.
When the phones are on the VLAN (eth0.20) they don’t hang up after a call completes. There are also strange issues like some calls going straight to voicemail instead of ringing the destination phone.
When the phones are NOT on the VLAN they hang up properly - all is well.
Could these issues be because I’m running on old hardware with a newer CentOS release?
The Dell has Broadcom NICs and the kernel module being used is BNX2.
I’m going back to FreePBX 13 and testing the exact same setup to confirm my suspicions but I have a strong feeling there’s something wrong with VLANs in the newest kernel with the BNX2 driver. I’ll report back with some findings tomorrow.
