I am going to give it a go myself; I already have a couple of basic VOIP phones delivered and I also have some new computer hardware arriving for another project; but I will attempt an install on this at first ... having watched the crosstalk videos it looks relatively straight forward .. famous last words ...
Now I need to review/decide upon a SIP trunk to test with here in the UK?
It is in our edge repo because QA is going through it. I don't know how the licensing etc works on it. That is why I directed to sales because they would have those answers
Hi!
I added a comment to the ticket I had opened for this problem for the FreePBX distro.
re: Latest system updates (October 2017) breaks Wanpipe drivers.
I am not sure the FreePBX devs can take care of this but they will surely be able to forward this information to the appropriate people.
Have a nice day!
Nick
my ISP provided me with a DHCP address I have to be manually updating the address each time it changes. Can this be automated. I will also like to know if it will be possible to use a domain name their line a no-ip domain that points to my modem ISP IP address
Hi James!
Good point, I am indeed running Edge...
Maybe I should install it, I am curious to see if Franck (which I believe developed it) remembered what we talked about about using regionalisms... **
Have a nice day!
Nicolas
** Some words do not have the same meaning depending on where you are in the world and these are best avoided and replaced by words or combination of words which do mean the same thing the world over.
Hi all!
I have realized that sometimes, many calls are not execute because zulu sends phone number with '+' symbol.
I wonder if it's possible to process the number before dial, to send the correct number to the sip provider. ¿How could I do it?
Regards
The issue that we have is related to inbound call that are direct from the PSTN to the PBX. It has nothing to do with the trunk between our UCM and the PBX. Also, there is no firewall between our voice gateway and our PBX. It is direct. I have logs from our SIP provider showing that the PBX is dropping the call at the one hour mark. I just tested again this morning and the same issue. This is the PBX log:
[2017-10-10 06:14:00] VERBOSE[28780][C-00000514] file.c: -- Playing 'digits/7.ulaw' (language 'en')
[2017-10-10 06:14:01] VERBOSE[28780][C-00000514] file.c: -- Playing 'conf-otherinparty.ulaw' (language 'en')
[2017-10-10 06:14:02] WARNING[2063] chan_sip.c: Retransmission timeout reached on transmission B95ADB55-ACF911E7-8152A57C-5A2F1691@10.50.107.2 for seqno 104 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[2017-10-10 06:14:02] WARNING[2063] chan_sip.c: Hanging up call B95ADB55-ACF911E7-8152A57C-5A2F1691@10.50.107.2 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[2017-10-10 06:14:02] VERBOSE[26166][C-0000050b] res_musiconhold.c: -- Stopped music on hold on SIP/10.50.107.2-000004fe
[2017-10-10 06:14:02] VERBOSE[26166][C-0000050b] file.c: -- Playing 'confbridge-leave.gsm' (language '')
[2017-10-10 06:14:03] VERBOSE[26166][C-0000050b] pbx.c: -- Executing [h@ext-meetme:1] Hangup("SIP/10.50.107.2-000004fe", "") in new stack
Here are the logs from our SIP provider from our previous instance:
2017-10-02 11:57:30.214 Wightman-ISC-1 Outgoing SIP message sent : INVITE (SDP) From: anonymous@10.10.106.20:5060 To: 519*******@10.1.0.18 519*******@10.1.0.18:5060
2017-10-02 11:57:30.218 Wightman-ISC-1 Incoming SIP message received : 100 Trying From: "Anonymous" To: <sip:519*******@10.1.0.18>
2017-10-02 11:57:30.221 Wightman-ISC-1 Incoming SIP message received : BYE 519*******@10.1.0.18->anonymous@10.10.106.20:5060
2017-10-02 11:57:30.221 Wightman-ISC-1 A SIP message was received with authentication information.
2017-10-02 11:57:30.221 Wightman-ISC-1 519*******is disconnecting from the call.
There are a few instances earlier in the call where we send the INVITE, receive a TRYING followed by a 491 Request Pending message before we get the expected ACK message.
2017-10-02 11:42:10.835 Wightman-ISC-1 Outgoing SIP message sent : INVITE (SDP) From: anonymous@10.10.106.20:5060 To: 519*******@10.1.0.18 519*******@10.1.0.18:5060
2017-10-02 11:42:10.840 Wightman-ISC-1 Incoming SIP message received : 100 Trying From: "Anonymous" To: <sip:519*******@10.1.0.18>
2017-10-02 11:42:10.841 Wightman-ISC-1 Incoming SIP message received : 491 Request Pending From: To: <sip:519*******@10.1.0.18>
2017-10-02 11:42:10.842 Wightman-ISC-1 Outgoing SIP message sent : ACK From: anonymous@10.10.106.20:5060 To: 519*******@10.1.0.18 519*******@10.1.0.18:5060
Still struggling to figure this one out. I read that updating the rtpkeepalive helps. Default is 0 and I set it to 30. Any feedback?
Also, don't know if this is related? I see this quite often in the logs:
[2017-10-10 06:14:32] WARNING[2063][C-00000517] channel.c: Unable to find a codec translation path from (ulaw) to (g729)
[2017-10-10 06:14:32] WARNING[2063][C-00000517] channel.c: Unable to find a codec translation path from (ulaw) to (g729)
[2017-10-10 06:14:33] ERROR[28825][C-00000517] channel.c: Could not return write format to its original state
Any assistance or guidance is appreciated.
Is member timeout enabled on the conference? This was affecting us at the default timeout period of 6 hours.
This specifies the number of seconds that the participant may stay in the conference before being automatically ejected. 0 = disabled, default is 21600 (6 hours)
Where is this setting in the GUI?
Looks like the remote end is using a re-INVITE as a keepalive mechanism and for some reason Asterisk isn't able to respond to it properly. You might want to configure your trunk with "session-timers=refuse" (see https://community.freepbx.org/t/asterisk-session-timers/41371) or just get more thorough SIP traces to see why that transaction is failing.
Well, /etc/fxotune.conf is owned by root, and shows rw permissions, and the modified date shows it WAS written by fxotune. But, the values in the file are NOT the same as were written out when I ran fxotune. I have now manually edited the settings into the file, and whenever we reboot FreePBX, I'll find out if that fixed it. Probably will never have to touch it again, IF that does fix the problem.
Thanks,
Jon
I have Reinvite Behaviour set to "NO" Does that matter?
Logs. The /var/log/asterisk/full log should be tracking when the phones become active and when they die.
Hi,
I know there was a firmware update - 2.0.4.32 - that addressed this specifically but we continue to have this issue on several of our S300's. Has anyone else had this continue and if so have you found any resolution for it? Thanks
can you file a ticket with support
https://support.sangoma.com
"never" is a long time. If you have bad copper to the dmarc in either direction or degrading lines this will be a continued issue. I would evaluate your wiring. It is unlikely your provider will care as it will work fine for their standards. If you have above ground wiring outside it is likely a new drop would help if you can make it happen.
Hi,
First issue:
We need to execute zulu-uc on a single system so that several agents can execute zulu-uc on that system (not simultaneously). We don't want to install multiple versions or unpack the zulu-uc client in each users home directory or change ownership and group of that directory so it corresponds to that particular user.
Second issue
I installed zulu-uc version 2.1.11/2.1.14 When executing zulu out of the installation path, it does not kill the process and we have to kill -9 pid manually.
Any suggestions?
Please check Your NAT settings in SIP extension settings. Also post the output of sip set debug on .
Hello @fetoa ,
Add this to your extensions_custom.conf file:
[macro-dialout-trunk-predial-hook]
;Remove spaces or dashes or + from outbound number when calling
exten => s,1,Set(OUTNUM=${FILTER(0123456789*#,${OUTNUM})})
exten => s,n,MacroExit()After saving it in your extensions_custom.conf file, run the following command from the linux console:
rasterisk -x'dialplan reload'
Thank you,
Daniel Friedman
Trixton LTD.
Depends a lot on the phone. I know that the Cisco phone ring tones can be replaced and that the Polycom phones have different ring tones that can be selected.