@bashir123 wrote:
Dears,
I have added almost 80 outbound routes and they are working fine. When I add more outbound route and after testing this route I am getting the extension exceeds error and couldn’t find any way to resolve this issue. Can any one tell me that it is occurring due to the number of core processors used or due to any other issue? Or is it Hardware issue of any software?
Any support regarding this will be highly appreciated.
Best Regards,
Bashir Ahmad.
Posts: 4
Participants: 2
@adtopkek wrote:
We are currently using Chan_sip on 5060 and PJSIP on 5061 on one of our servers. We have a mix of PJSIP and chan_sip extensions now. We would like to convert most of our people to PJSIP and the easiest way of doing this is to put PJSIP on 5060 as we did on another server.
However we have people on 5061 that are using PJSIP. Most of them we can update easily enough but some like cell phone softphones are near impossible.
Can we set PJSIP to use port 5061 as well as 5060?
Posts: 1
Participants: 1
@adtopkek wrote:
Is there a way to convert extensions in bulk from chan_sip to PJSIP that isn’t opening every extension and pressing the PJSIP button?
Posts: 2
Participants: 2
@natnaree wrote:
Dear contributors,
I am Natnaree Asavaseri and currently undertaking a research internship at Nara Institute of Science and Technology, Japan. As a part of my research, I am analyzing the impact of Microsoft’s acquirement of GitHub.
I would like to conduct a survey to understand how developers perceive the Microsoft’s acquisition of GitHub, especially from contributors of Linux distributions and BSD families. So please consider voicing your opinion by allowing us up to 5 minutes to complete our short survey.
We would like to remind you that participation in this survey is completely voluntary and your identity is hidden for anonymity. Thank you in advance for your assistance.
Natnaree
NAIST, Japan
Posts: 1
Participants: 1
@adtopkek wrote:
When I call into my voicemail and it starts recording I can dial *1####0 then the voicemail application attempts a failed dial or something. What is this?
Does it happen to you when you try it?
Posts: 1
Participants: 1
@Hyperfocus wrote:
I installed SNG7 and I want to install XFCE on it, I ran the following commands:
yum groupinstall "XFCE" "Graphical Administration Tools" ln -sf /lib/systemd/system/runlevel5.target /etc/systemd/system/default.target rebootUnfortunately my system becomes unbootable and leaves me unable to inspect the issue.
I need it to use teamviewer since I will be managing the box remotely and I don’t want to expose the local network to the internet. Using teamviewer is the easiest solution for this situation.
Is there any recommended procedure on how to install a desktop GUI on SNG7?
If this is not possible I will install Debian instead, hopefully I will still be able to use commercial modules in the future with some tweaking.
Posts: 9
Participants: 5
@mhammett wrote:
To help people searching, you can use these commands to export your CDR and CEL information from your old install to put into your new. Older versions could have asteriskuser instead of freepbxuser
Old:
mysqldump --skip-add-drop-table --no-create-db --no-create-info --complete-insert -u freepbxuser -p asteriskcdrdb cdr > cdr.sql
mysqldump --skip-add-drop-table --no-create-db --no-create-info --complete-insert -u freepbxuser -p asteriskcdrdb cel > cel.sqlNew:
mysql -u root asteriskcdrdb < /home/mhammett/cdr2.sql
mysql -u root asteriskcdrdb < /home/mhammett/cel2.sql
Posts: 1
Participants: 1
@tonyg wrote:
hi, today i applied all os and modules updates. i have not applied updates in about 2 months. after the os and modules updates, i reboot the pbx and all seems to be working ok, but… i have noticed that i am getting critical errors on the dash. IVR seems to be working correctly but i get these invalid destination errors:
thanks in advance
Posts: 6
Participants: 3
@pbx wrote:
Distro Upgrade - Version 1807-2.sng7
Build Date: 2018-07-18Can no longer upgrade servers as resulting termination is because the boot does not have enough disk space. The boot section for the servers I used cannot be increased. Yet I have been able to upgrade servers over the past 3-4 weeks but from today we are blocked and the results report states the below. How can this be resolved?
Result for System requirements
Result: needs_actionRule ID: xccdf_preupg_rule_system_requirements_check
Time: 2018-07-19 12:03
The module checks if the system complies with the minimal requirements.
Remediation instructions
Free space in the /boot/ directory is smaller then the estimated free space required
for the secure upgrade. You need to release more space on the partition./usr:
(only for migration process)
Estimated needed free space: 2689038 kB
Estimated safe free space: 3092388 kB
Available free space: 43363004 kB/var:
(only for migration process)
Estimated needed free space: 2689038 kB
Estimated safe free space: 3092388 kB
Available free space: 43363004 kB/boot
Estimated required free space: 112640 kB
Available free space: 87853 kBAdditional output:
preupg.risk.HIGH: Not enough free space on the /boot. Release more space for upgrade.
Posts: 1
Participants: 1
@adtopkek wrote:
I just updated to 14.0.2.97 and got this error:
To fix it clear your cache. That is all.
Posts: 1
Participants: 1
@artello73 wrote:
Hello,
Long story short - I’m running FreePBX 13 in a virtual machine in oVirt cluster. The FreePBX setup is a bit unusual - we have a 4 sip trunks, and each trunk connected to own Ethernet interface (so we are using PJSIP due to this). And I there are a number of commercial modules running on this setup.
As this is a live system, I’d like to clone this VM and try to do an upgrade to FreePBX 14. The question is how to not mess up with system activation and commercial modules? How to do this in a proper way?
One option is to test upgrade on the clone and if successful start doing upgrade of main VM.
Second option - if clone upgrade successful - somehow move activation from old VM to the new one. Is this possible?Regards,
Artem
Posts: 1
Participants: 1
@adtopkek wrote:
I grepped the etc/asterisk/* directory but couldn’t find these settings. Where can I view their current settings and change them at?
https://wiki.asterisk.org/wiki/display/AST/Performance+Tuning
Posts: 1
Participants: 1
@Basildane wrote:
FreePBX distro 14.0.3.10.
[2018-07-29 08:24:56] WARNING[3102] db.c: Error executing SQL (COMMIT): disk I/O error
[2018-07-29 08:24:56] WARNING[3102] db.c: Error executing SQL (ROLLBACK): cannot rollback - no transaction is activeImmediately after that the call drops.
The system is on a RAID server, and I checked everything I could think of. Raid is green. The host server checks out, disk system says no errors. I can’t find anything in the host hardware logs. I also ran checkdb on the MariaDB databases. No errors found.
I would like to do something like fsck on the partition, but I have not been able to figure out what to do, because FreePBX installs a LVM volume and I’m not familiar with the commands.
This is what my drive looks like.
lsblk -o NAME,FSTYPE,SIZE,MOUNTPOINT,LABEL
NAME FSTYPE SIZE MOUNTPOINT LABEL
sda 128G
├─sda1 ext4 2G /boot bootvol
└─sda2 LVM2_member 126G
├─SangomaVG-root xfs 122.1G /
└─SangomaVG-swaplv1 swap 3.9G [SWAP]I was able to run fsck on /dev/sda1, but that doesn’t tell me what I need to know.
Any suggestion on next steps?By the way, this system has run flawlessly for years. The problem just cropped up recently.
Posts: 1
Participants: 1
@adtopkek wrote:
I checked out htop today and there is a thread
php /var/html/html/admin/modules/firewall/bin/getservices
That is using 100% cpu 100% of the time. I turned off the firewall and it is still stuck like this. Anyone else had this recently?
I am full stable on modules as of the time of this post. Asterisk 13 Distro 7.
Posts: 1
Participants: 1
@munozj wrote:
I’ve been able to get system messages such as /var/log/messages to forward to my syslog server but is it possible to also forward some of the Freepbx and Asterisk logs such as:
/var/log/asterisk/full
/var/log/asterisk/fail2ban
/var/log/asterisk/freepbx.log
/var/log/asterisk/freepbx_security.logIt’s just easier for me to parse through them in syslog rather than the flat log files.
(FreePBX Distro 13)
Posts: 3
Participants: 2
@chemnic wrote:
I’m trying to upgrade my system from 10.13.66 to SNG7, but it fails shortly after executing “distro-upgrade”.
I’m following the documentation at https://wiki.freepbx.org/display/PPS/Upgrading+from+FreePBX+10.13.66+to+SNG7.
I don’t see any relevant information in the output, or in the log files:
[root@pbx-test ~]# distro-upgrade ┏━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━┓ ┃ ┃ ┃ Sangoma 6 to 7 Upgrade Tool ┃ ┃ ┃ ┃ Distro Upgrade - Version 1807-2.sng7 ┃ ┃ Build Date: 2018-07-18 ┃ ┃ ┃ ┗━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━┛ Checking prerequsites... Checking FreePBX Version [ ✔ ] - 13.0.195.4 Checking bitsize of machine [ ✔ ] - x86_64 Checking for Hyper-V [ ✔ ] Checking for 32 bit packages [ ✔ ] - No i686 rpms found Checking available disk space [ ✔ ] - 341G Available Checking for IonCube Loader [ ✔ ] - IonCube Loader not detected Checking for outdated system [ ✔ ] - No pending updates. Checking for HA Setup [ ✔ ] - Not a HA system All prerequsites passed! Are you ready to upgrade your machine to SNG7? This process requires two reboots, and will download approximately 400mb of files before starting. There will be no interruption to service until this machine is rebooted. Download files required for upgrade [Yn]? y ######### Starting setup upgrade on jue ago 2 11:58:27 CST 2018 ######### ######### Creating upgrade repofile ######### ######### Installing needed packages ######### Loaded plugins: downloadonly, fastestmirror, kmod Setting up Install Process Loading mirror speeds from cached hostfile upgrade | 2.9 kB 00:00 Package openscap-1.0.8-1.0.1.el6.centos.1.x86_64 already installed and latest version Nothing to do Loaded plugins: downloadonly, fastestmirror, kmod Setting up Install Process Loading mirror speeds from cached hostfile Package preupgrade-assistant-2.6.0-2.el6.sangoma.x86_64 already installed and latest version Package preupgrade-assistant-el6toel7-0.7.1-2.el6.sangoma.noarch already installed and latest version Package 1:redhat-upgrade-tool-0.7.52-1.0.1.el6.sangoma.noarch already installed and latest version Package sangoma-pbx-1-1.sng7.noarch already installed and latest version Nothing to do ######### Running preupgrade to inventory running system ######### ######### An error occured, please check /var/log/sngupdate ######### Error! setup_upgrade did not exit cleanly. Not enabling upgrade serviceThe file /var/log/sngupdate is here: https://pastebin.com/PkzdFh9J
Posts: 4
Participants: 2
@Krunal wrote:
Hi, I am Using FreePBX Distro version 14.0.3.6.
We are having 2 trunks one is PRI Gateway and other one is GSM Gateway.
We have no issue with PRI Gateway. But When user makes an outbound call from GSM Gateway they heard multiple ring back tone.After subsequent troubleshooting I have found that my GSM Gateway also have Ringback tone feature. So most likely Ringback tone is generated by GSM Gateway and FreePBX both at the same time and that may cause the issue. I have disable ringback tone feature in GSM Gateway and found that there is no multiple ringback tone. but at the same time I found some other issues with ringback tone. I.e. If remote user not answering or cut the call then, user behind freepbx heard ringing instead of playing User busy tone.
I have remove “r” from trunk dial option from advance setting but no success. Is there any way such that pbx dont create ring back tone itself?
Any other solution from i have stated is highly appreciated.
Thanks In Advance.
Here I am Attaching a log for your reference.
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [29724963997@from-internal:1] Macro(“SIP/540-00000658”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/540-00000658”, “TOUCH_MONITOR=1533701946.1656”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/540-00000658”, “AMPUSER=540”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/540-00000658”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/540-00000658”, “1?Set(REALCALLERIDNUM=540)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/540-00000658”, “AMPUSER=540”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/540-00000658”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/540-00000658”, “AMPUSERCIDNAME=Krunal Thakar”) in new stack
– Executing [s@macro-user-callerid:8] ExecIf(“SIP/540-00000658”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:9] GotoIf(“SIP/540-00000658”, “0?report”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/540-00000658”, “AMPUSERCID=540”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/540-00000658”, “__DIAL_OPTIONS=Tt”) in new stack
– Executing [s@macro-user-callerid:12] Set(“SIP/540-00000658”, “CALLERID(all)=“Krunal Thakar” <540>”) in new stack
– Executing [s@macro-user-callerid:13] GotoIf(“SIP/540-00000658”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“SIP/540-00000658”, “1?Set(GROUP(concurrency_limit)=540)”) in new stack
– Executing [s@macro-user-callerid:15] ExecIf(“SIP/540-00000658”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:16] NoOp(“SIP/540-00000658”, “Macro Depth is 1”) in new stack
– Executing [s@macro-user-callerid:17] GotoIf(“SIP/540-00000658”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,18)
– Executing [s@macro-user-callerid:18] GotoIf(“SIP/540-00000658”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,37)
– Executing [s@macro-user-callerid:37] Set(“SIP/540-00000658”, “CALLERID(number)=540”) in new stack
– Executing [s@macro-user-callerid:38] Set(“SIP/540-00000658”, “CALLERID(name)=Krunal Thakar”) in new stack
– Executing [s@macro-user-callerid:39] GotoIf(“SIP/540-00000658”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:40] Set(“SIP/540-00000658”, “CDR(cnam)=Krunal Thakar”) in new stack
– Executing [s@macro-user-callerid:41] Set(“SIP/540-00000658”, “CDR(cnum)=540”) in new stack
– Executing [s@macro-user-callerid:42] Set(“SIP/540-00000658”, “CHANNEL(language)=en”) in new stack
– Executing [29724963997@from-internal:2] Gosub(“SIP/540-00000658”, “sub-record-check,s,1(out,29724963997,force)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“SIP/540-00000658”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“SIP/540-00000658”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“SIP/540-00000658”, “NOW=1533701946”) in new stack
– Executing [s@sub-record-check:4] Set(“SIP/540-00000658”, “__DAY=08”) in new stack
– Executing [s@sub-record-check:5] Set(“SIP/540-00000658”, “__MONTH=08”) in new stack
– Executing [s@sub-record-check:6] Set(“SIP/540-00000658”, “__YEAR=2018”) in new stack
– Executing [s@sub-record-check:7] Set(“SIP/540-00000658”, “__TIMESTR=20180808-094906”) in new stack
– Executing [s@sub-record-check:8] Set(“SIP/540-00000658”, “__FROMEXTEN=540”) in new stack
– Executing [s@sub-record-check:9] Set(“SIP/540-00000658”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“SIP/540-00000658”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“SIP/540-00000658”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“SIP/540-00000658”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“SIP/540-00000658”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“SIP/540-00000658”, “3?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“SIP/540-00000658”, “1?sub-record-check,out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [out@sub-record-check:1] NoOp(“SIP/540-00000658”, “Outbound Recording Check from 540 to 29724963997”) in new stack
– Executing [out@sub-record-check:2] Set(“SIP/540-00000658”, “RECMODE=no”) in new stack
– Executing [out@sub-record-check:3] ExecIf(“SIP/540-00000658”, “0?Goto(routewins)”) in new stack
– Executing [out@sub-record-check:4] ExecIf(“SIP/540-00000658”, “1?Goto(routewins)”) in new stack
– Goto (sub-record-check,out,7)
– Executing [out@sub-record-check:7] Gosub(“SIP/540-00000658”, “recordcheck,1(force,out,29724963997)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“SIP/540-00000658”, “Starting recording check against force”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“SIP/540-00000658”, “force”) in new stack
– Goto (sub-record-check,recordcheck,5)
– Executing [recordcheck@sub-record-check:5] Set(“SIP/540-00000658”, “__REC_POLICY_MODE=FORCE”) in new stack
– Executing [recordcheck@sub-record-check:6] GotoIf(“SIP/540-00000658”, “1?startrec”) in new stack
– Goto (sub-record-check,recordcheck,16)
– Executing [recordcheck@sub-record-check:16] NoOp(“SIP/540-00000658”, “Starting recording: out, 29724963997”) in new stack
– Executing [recordcheck@sub-record-check:17] Set(“SIP/540-00000658”, “AUDIOHOOK_INHERIT(MixMonitor)=yes”) in new stack
[2018-08-08 09:49:06] ERROR[6946][C-0000032e]: pbx_functions.c:701 ast_func_write: Function AUDIOHOOK_INHERIT not registered
– Executing [recordcheck@sub-record-check:18] Set(“SIP/540-00000658”, “__CALLFILENAME=out-29724963997-540-20180808-094906-1533701946.1656”) in new stack
– Executing [recordcheck@sub-record-check:19] MixMonitor(“SIP/540-00000658”, “/var/spool/asterisk/monitor/2018/08/08/out-29724963997-540-20180808-094906-1533701946.1656.wav,abi(LOCAL_MIXMON_ID),”) in new stack
– Executing [recordcheck@sub-record-check:20] Set(“SIP/540-00000658”, “__MIXMON_ID=0x7fa4d009af40”) in new stack
== Begin MixMonitor Recording SIP/540-00000658
– Executing [recordcheck@sub-record-check:21] Set(“SIP/540-00000658”, “__RECORD_ID=SIP/540-00000658”) in new stack
– Executing [recordcheck@sub-record-check:22] Set(“SIP/540-00000658”, “__REC_STATUS=RECORDING”) in new stack
– Executing [recordcheck@sub-record-check:23] Set(“SIP/540-00000658”, “CDR(recordingfile)=out-29724963997-540-20180808-094906-1533701946.1656.wav”) in new stack
– Executing [recordcheck@sub-record-check:24] Return(“SIP/540-00000658”, “”) in new stack
– Executing [out@sub-record-check:8] Return(“SIP/540-00000658”, “”) in new stack
– Executing [29724963997@from-internal:3] ExecIf(“SIP/540-00000658”, “0 ?Set(CDR(accountcode)=)”) in new stack
– Executing [29724963997@from-internal:4] Set(“SIP/540-00000658”, “MOHCLASS=default”) in new stack
– Executing [29724963997@from-internal:5] ExecIf(“SIP/540-00000658”, “1?Set(TRUNKCIDOVERRIDE=<7800>)”) in new stack
– Executing [29724963997@from-internal:6] Set(“SIP/540-00000658”, “_NODEST=”) in new stack
– Executing [29724963997@from-internal:7] Macro(“SIP/540-00000658”, “dialout-trunk,3,9724963997,off”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/540-00000658”, “DIAL_TRUNK=3”) in new stack
– Executing [s@macro-dialout-trunk:2] ExecIf(“SIP/540-00000658”, “0?Set(DIAL_OPTIONS=t)”) in new stack
– Executing [s@macro-dialout-trunk:3] GosubIf(“SIP/540-00000658”, “0?sub-pincheck,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:4] ExecIf(“SIP/540-00000658”, “0?Set(CALLERID(num)=540)”) in new stack
– Executing [s@macro-dialout-trunk:5] GotoIf(“SIP/540-00000658”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/540-00000658”, “DIAL_NUMBER=9724963997”) in new stack
– Executing [s@macro-dialout-trunk:7] Set(“SIP/540-00000658”, “DIAL_TRUNK_OPTIONS=Tt”) in new stack
– Executing [s@macro-dialout-trunk:8] Set(“SIP/540-00000658”, “OUTBOUND_GROUP=OUT_3”) in new stack
– Executing [s@macro-dialout-trunk:9] Set(“SIP/540-00000658”, “DIAL_TRUNK_OPTIONS=Tt”) in new stack
– Executing [s@macro-dialout-trunk:10] GotoIf(“SIP/540-00000658”, “0?nomax”) in new stack
– Executing [s@macro-dialout-trunk:11] GotoIf(“SIP/540-00000658”, “0?chanfull”) in new stack
– Executing [s@macro-dialout-trunk:12] GotoIf(“SIP/540-00000658”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:13] Macro(“SIP/540-00000658”, “outbound-callerid,3”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/540-00000658”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/540-00000658”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:3] ExecIf(“SIP/540-00000658”, “0?Set(REALCALLERIDNUM=540)”) in new stack
– Executing [s@macro-outbound-callerid:4] GotoIf(“SIP/540-00000658”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,7)
– Executing [s@macro-outbound-callerid:7] Set(“SIP/540-00000658”, “USEROUTCID=“Krunal Thakar”<7540>”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/540-00000658”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:9] Set(“SIP/540-00000658”, “TRUNKOUTCID=12345678”) in new stack
– Executing [s@macro-outbound-callerid:10] GotoIf(“SIP/540-00000658”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,15)
– Executing [s@macro-outbound-callerid:15] ExecIf(“SIP/540-00000658”, “1?Set(CALLERID(all)=12345678)”) in new stack
– Executing [s@macro-outbound-callerid:16] ExecIf(“SIP/540-00000658”, “1?Set(CALLERID(all)=“Krunal Thakar”<7540>)”) in new stack
– Executing [s@macro-outbound-callerid:17] ExecIf(“SIP/540-00000658”, “1?Set(CALLERID(all)=<7800>)”) in new stack
– Executing [s@macro-outbound-callerid:18] ExecIf(“SIP/540-00000658”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:19] ExecIf(“SIP/540-00000658”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:20] Set(“SIP/540-00000658”, “CDR(outbound_cnum)=7800”) in new stack
– Executing [s@macro-outbound-callerid:21] Set(“SIP/540-00000658”, “CDR(outbound_cnam)=”) in new stack
– Executing [s@macro-outbound-callerid:22] ExecIf(“SIP/540-00000658”, “1?Set(CALLERID(all)=<7800>)”) in new stack
– Executing [s@macro-outbound-callerid:23] ExecIf(“SIP/540-00000658”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:24] ExecIf(“SIP/540-00000658”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:25] Set(“SIP/540-00000658”, “CDR(outbound_cnum)=7800”) in new stack
– Executing [s@macro-outbound-callerid:26] Set(“SIP/540-00000658”, “CDR(outbound_cnam)=”) in new stack
– Executing [s@macro-dialout-trunk:14] GosubIf(“SIP/540-00000658”, “0?sub-flp-3,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:15] Set(“SIP/540-00000658”, “OUTNUM=9724963997”) in new stack
– Executing [s@macro-dialout-trunk:16] Set(“SIP/540-00000658”, “custom=SIP/Dinestar_out”) in new stack
– Executing [s@macro-dialout-trunk:17] ExecIf(“SIP/540-00000658”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)”) in new stack
– Executing [s@macro-dialout-trunk:18] ExecIf(“SIP/540-00000658”, “0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:19] Macro(“SIP/540-00000658”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/540-00000658”, “”) in new stack
– Executing [s@macro-dialout-trunk:20] GotoIf(“SIP/540-00000658”, “0?skipcrm”) in new stack
– Executing [s@macro-dialout-trunk:21] Set(“SIP/540-00000658”, “__CRM_DIRECTION=OUTBOUND”) in new stack
– Executing [s@macro-dialout-trunk:22] Set(“SIP/540-00000658”, “__CRM_DESTINATION=9724963997”) in new stack
– Executing [s@macro-dialout-trunk:23] Set(“SIP/540-00000658”, “__CRM_SOURCE=540”) in new stack
– Executing [s@macro-dialout-trunk:24] AGI(“SIP/540-00000658”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <SIP/540-00000658>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@macro-dialout-trunk:25] Set(“SIP/540-00000658”, “CHANNEL(hangup_handler_push)=crm-hangup,s,1”) in new stack
– Executing [s@macro-dialout-trunk:26] NoOp(“SIP/540-00000658”, “CRM Finished”) in new stack
– Executing [s@macro-dialout-trunk:27] GotoIf(“SIP/540-00000658”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:28] ExecIf(“SIP/540-00000658”, “1?Set(CONNECTEDLINE(num,i)=9724963997)”) in new stack
– Executing [s@macro-dialout-trunk:29] ExecIf(“SIP/540-00000658”, “1?Set(CONNECTEDLINE(name,i)=CID:7800)”) in new stack
– Executing [s@macro-dialout-trunk:30] ExecIf(“SIP/540-00000658”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)7800)”) in new stack
– Executing [s@macro-dialout-trunk:31] GotoIf(“SIP/540-00000658”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:32] Dial(“SIP/540-00000658”, “SIP/Dinestar_out/9724963997,300,Ttb(func-apply-sipheaders^s^1)”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– SIP/Dinestar_out-00000659 Internal Gosub(func-apply-sipheaders,s,1) start
– Executing [s@func-apply-sipheaders:1] ExecIf(“SIP/Dinestar_out-00000659”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
– Executing [s@func-apply-sipheaders:2] NoOp(“SIP/Dinestar_out-00000659”, “Applying SIP Headers to channel”) in new stack
– Executing [s@func-apply-sipheaders:3] Set(“SIP/Dinestar_out-00000659”, “SIPHEADERKEYS=”) in new stack
– Executing [s@func-apply-sipheaders:4] ExecIf(“SIP/Dinestar_out-00000659”, “0?Set(Rheader=1)”) in new stack
– Executing [s@func-apply-sipheaders:5] While(“SIP/Dinestar_out-00000659”, “0”) in new stack
– Jumping to priority 9
– Executing [s@func-apply-sipheaders:10] ExecIf(“SIP/Dinestar_out-00000659”, “0?SIPRemoveHeader(Alert-Info:)”) in new stack
– Executing [s@func-apply-sipheaders:11] ExecIf(“SIP/Dinestar_out-00000659”, “0?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
– Executing [s@func-apply-sipheaders:12] Return(“SIP/Dinestar_out-00000659”, “”) in new stack
== Spawn extension (from-pstn, 29724963997, 1) exited non-zero on ‘SIP/Dinestar_out-00000659’
– SIP/Dinestar_out-00000659 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
– Called SIP/Dinestar_out/9724963997
– SIP/Dinestar_out-00000659 is making progress passing it to SIP/540-00000658
Posts: 1
Participants: 1
@adtopkek wrote:
We are having a problem with BLF lights freezing when using Yealink’s 83.0.X firmware. We caught it acting up and sent Yealink a bug report. Their response is that the BLF’s need to use a different RFC standard.
Dear Customer,
This is East from Yealink Technical Support team, nice to meet you.
Thank you for the detailed information.
After I checked the syslog that you provided, I can confirm the reason. Please see the detail bellow:
When the monitored status change, the server will send NOTIFY to the phone, there is a BLF version on the NOTIFY.
In our V81 version, the phone will not check the BLF version in the NOTIFY. But in the V83 (actually updated in V82), the phone will check the BLF version.
We modified it because it is compliant with standard RFC4235, it is a better update. Please see description of BLF version below:
Version: This attribute allows the recipient of dialog information documents to properly order them. Versions start at 0, and increment by one for each new document sent to a subscriber . Versions are scoped within a subscription. Versions MUST be representable using a non-negative 32 bit integer.
But according to the syslog, after received NOTIFY from server, the phone found that the version is smaller than the previous one, it is a substandard parameter. So the phone will not update the status of the BLF, and prompt error like: BLF bad version param. Cur ver=71, new ver=26
As our phones refer to the standard RFC4235, please contact your service provider to modify the NOTIFY message, and let it also refer to the standard. Then it will work well in BLF.
Where can I set the blf’s to use RFC4235 as they suggest?
Freepbx 14 asterisk 13.22
Asterisk Forum post: https://community.asterisk.org/t/blf-s-not-working-right-with-yealink-83-0-x-firmware-response-from-yealink/75708
Posts: 9
Participants: 4
@cbelcher wrote:
Hey guys running latest stable build, SNG7-PBX-64bit-1805-1 with Asterisk 13.
Simple pjsip trunk to TSP and my pcaps are showing the From field has the private IP address. At that point the TSP gives me a 403 forbidden.
A working call via another server has the correct FROM with the public ip and all works fine.
Where the heck to make the changes to get this to show the correct IP.
Nothing to do with my Firewall these are captures before they hit the firewall.
I think I’ve got all my NAT Settings in General SIP Settings tab correct. It picks up the correct External Address when I press the Detect Netowork Settings. The local networks are correct, just 172.16.0.0/16,
The from field looks like this on the non working side.
From: sip:ESI_SIP_PSTN@172.16.10.32; tag asdfasdf blah blah blahThe via field, contact and SDP address show the correct public IP, what am I doing wrong!
Thanks I know it’s got to be something super easy I’m just missing. Let me know if you need any further details! Screenshot ect
Using
Posts: 1
Participants: 1
@GSnover wrote:
When did this change? I have successfully upgraded MANY Hyper-V machines using this method - Where can I get the old Upgrade Package so I can continue moving all my machines off of 10?
Sad in Albuquerque…
Posts: 6
Participants: 4
