This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
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Remote Access to freepbx without port forwarding
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Delay between pbx_variables.c: and netsock2.c - 10 seconds delay between dialing and establising connection
Hi,
I’m havig issues with dialing out. After dialing number # it takes 10 seconds to establish connection. As test I use *43. Anyone faced similar issue?
Asterisk version 15.7.2 - fresh install
log after pjsip set logger on
[2019-05-31 17:05:32] VERBOSE[19908] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘10.101.100.2’
[2019-05-31 17:05:32] VERBOSE[19908] res_pjsip_logger.c: <— Transmitting SIP response (358 bytes) to UDP:10.101.100.8:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.101.100.8:5060;rport=5060;received=10.101.100.8;branch=z9hG4bK79a3bc7e
Call-ID: 0011bb47-26f70004-67b15d99-118ee365@10.101.100.8
From: “297-297” <sip:297@10.101.100.2>;tag=0011bb4726f700043157518b-1d16ffbd
To: <sip:*43@10.101.100.2;user=phone>
CSeq: 102 INVITE
Server: FPBX-15.0.16(15.7.2)
Content-Length: 0
[2019-05-31 17:05:41] VERBOSE[19908] netsock2.c: Using SIP RTP Audio TOS bits 184
[2019-05-31 17:05:41] VERBOSE[19908] netsock2.c: Using SIP RTP Audio CoS mark 5
[2019-05-31 17:05:41] VERBOSE[22311][C-00000011] pbx.c: Executing [*43@from-internal:1] Set(“PJSIP/297-00000010”, “__COS_DEST=echotest”) in new stack
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Call Recording Download History
No, apache access log is your only resource for this.
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Call Recording Download History
Thank you for your help! Is there a way to keep the logs longer than 30 days, without having manually save them?
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Call Recording Download History
From bash
man logrotate
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Falla boot disco duro de estado solido [resuelto]
buen dia
si, muchas gracias los pasos funcionaron perfectamente, no se predio ni información, ni configuración y el sistema operativo inicia sin problemas. 
@gbarajas
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Phones unreachable from outside
A few days ago our Polycom phones stopped being able to receive calls from outside. We can call out and each other’s extensions just fine, but if anyone tries to call in it rings six times then hangs up.
The person who originally set up our PBX system and most of the phones left five months ago, but never told any of us how anything works. I’ve tried rebooting the phone system from the website, but that hasn’t fixed anything. I’ve updated a few of the modules that I thought might be the issue, but that didn’t help either.
We’re running |PBX Firmware: 2.210.62-3 and PBX Service Pack: 1.0.0.0.
Any help would be really appreciated.
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Programming the Message (Voice Mail) Key on Snom 821
I have configured some of my phones to share a common voicemail box, and this is working fine. When voicemail is received, the message light illuminates on all of the phones with a shared voicemail box.
I have programmed the message button on extension 234 to dial *98237 in an attempt to have it automatically dial the voicemail box for extension 237. However, when the button is pressed, the prompt asks for the mailbox number. So that it appears that the button is dialing *98 rather than *98237.
Do I need to insert any special sequence between *98 and 237 to have this function operate correctly on Snom 821 phones?
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Phones unreachable from outside
That version went end of life 7? 8? years ago and is long past due for an upgrade. Beyond your immediate need to get phones working, you need a plan get get on a supportable system. There is a tool for migrating settings to a new install: https://wiki.freepbx.org/display/PPS/Elastix+and+PBXinaFlash+to+FreePBX+Distro+Conversion+Tool
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Dial 3 digit extension immediately
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Weird one-way audio issue between FreePBX and Gateway
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Phones unreachable from outside
I’m not surprised it’s that old. The previous director wasn’t keen on updating/upgrading things. I do thank you for the link for upgrading. I suppose you don’t have any idea what’s wrong with our phones, do you?
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Digium D60 and FreePBX Distro VPN
This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
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Phones unreachable from outside
The log file is /var/log/asterisk/full and is kept for about 24 hours (before logrotate starts up a new file).
Look through the file and find an incoming call that isn’t being answered. If you need help interpreting the file, post the extract here and we can look at it and give you some ideas.
There are only 200 things it could be. The logs will narrow that down pretty quickly.
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System quit working
This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
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Help With E1 implementation and appliance selection
Hello All:
I have previously setup a few Asterisk systems but with plain POTS phone lines and a card or FXO device.
Now I need to do it for an E1 with 15 lines and I am not sure how to proceed.
These are the specs for the E1 line, translated form Spanish as best as possible:
E1 Type = PRI-ISDN
Number of Channels= 15 channels
Encoding Type= HDB3 (High Density Bipolar of order 3 code)
Framing: no-cr4
Channels= Ascending
Impendace = 75 Ohm
clock Type= Slave
dial type: Overlap
all 7 number digits delivered to the customers pbx
I need to select a commercial Appliance and anything else required as well as to know how to set it up for the E1 lines to be used by the ip phones and softphones.
We think we would have no more than 15 outside calls (obviously) simultaneously and about 10 internal calls at the same time.
Any help would be welcome.
Kind Regards
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Programming the Message (Voice Mail) Key on Snom 821
If I recall correctly, when you dial a mailbox that is not the one belonging to your extension, you always get the prompt to enter the password, but I might be wrong.
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Delay between pbx_variables.c: and netsock2.c - 10 seconds delay between dialing and establising connection
I’ve never seen anything like this and have no idea what may be wrong. However, a few comments and questions:
Asterisk 15 is no longer supported, near EOL and AFAIK not in any official distros. See
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
On the chance that it may fix your problem and you’ll want to do it anyhow, get or make a system with Asterisk 16. If you build it yourself, use --with-pjproject-bundled rather than building pjproject separately.
Also, I’m guessing that __COS_DEST is from the Class of Service module. Test before installing commercial modules, to see whether they are related to your issue.
See if a higher verbosity log shows anything interesting.
See if any processes are running continuously, you’re out of memory, etc.
Start a tcpdump capture (of everything), make test call, stop capture, copy to your PC and look at it with Wireshark. Possibly, there is a failing network operation between the incoming INVITE and the dial plan running, e.g. a DNS lookup, external logging attempt, etc.
If still no luck: FreePBX and Asterisk versions? Install script used? Cloud or on-site? If cloud, whose? If on-site, describe physical platform, virtualization used if any, etc.
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Callerid_entries: Integrity constraint violation: 1062 Duplicate entry '0' for key 'PRIMARY'
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Delay between pbx_variables.c: and netsock2.c - 10 seconds delay between dialing and establising connection
I’ve seen symptoms like this if you have a flaky STUN server configured.
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