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No, it is not even getting to asterisk, that is why I am pursuing the settings if anyone knows. Cheers

I an currently trying to connect a SIP Trunk from Vonex (Australian VoIP Provider) to my Asterisk box. It appears to be Registering Name/username Host Dyn Forcerport ACL Port Status 075413XXXX/XXX 175.XXX.XXX.XX...

Including answering an "on hold" call of another phone in the system not just providing the indications? Garry

Spa 942 Are you sure that the SPA 942 can show other phones the Hold indication and allow it to be recovered on another phone?

Hi, My sip_general_custom is as follows ; Anything that is not to be overwritten during ; updates or upgrades is to be put in this file ; otherwise FreePBX replaces sip.conf with a standard file. srvlookup=no...

Hi, Only if there is at least one SIP Trunk provider. I fixed mine with svrlookup=no and all addresses hard coded e.g. 192.168.0.3 Nothing else ever fixed the problem for me. Cheers, Garry

Thanks for that Cheers Garry

Use a different Announcement per IVR determined by time Cheers Garry

Hi drvcrash, Don't suppose you would like to share the setup information? Cheers Garry

Hi Shane, Send me the file and I will try it on my system as the file I posted worked on my system. garry at palamo dot com dot au I usually test it from cron within Webmin Cheers Garry

Hi Shane, Unless the the actual issue is in the following line:- open(SIPTRUNKS,"</etc/asterisk/trunkalerts_sip.conf"); The "<" should not be there. If that doesn't fix it, try commenting out the line with a # Failing that you could pm me the...

OK My file shows 202.169.178.10:5060 but that is only because I use hard addresses OK Mine are owned by asterisk but that should not be the issue. Can you post the two lines you have with IAXTRUNKS and SIPTRUNKS Please. The only issues I ever had was in the email area. Cheers Garry

Contents Could you please post the contents of the trunkalerts_sip.conf and trunkalerts_iax.conf in /etc/asterisk My Line 18 is open(SIPTRUNKS,"/etc/asterisk/siptrunks"); Have you added another line ? Who owns the iax and sip files? Cheers Garry

Hi, This fixed my same problem Check the ownership and the permissions of the file in /var/lib/asterisk/bin: module_admin -rwxrwxr-- found to be set set for 755, changed it to 777 Regards Garry

Rhino Rhino Cards work well and IMO are value for money Garry

Hi, Yes Hookflash was used when we needed to transfer to another PSAP Cheers Garry

Hi John, Darn good question - to be perfectly honest I don't recall and don't have access anymore to the Audiocodes documentation and GUI. I do recall that different Ring Groups for the Consoles were used depending on Time of Day and Geographical location of the Calling Number as well as...

This may also help but was from a couple of years ago:- Do NOT set Enable911PSAP, but use the following settings (which can be used with the current 4.8 release installed on the M1k). To enable MF detection instead of DTMF detection configure: DTMFDetectorEnable = 0 MFR1DetectorEnable =...

It was tested with Analogue Modules See Below The Network looks like an FXO and the Mediant 1000 looks like FXS port(s). When there is an Incoming 911 Call, the Network applies a loop to the FXS port on the Mediant 1000. When the FXS is ready to receive the ANI, it applies a "wink" aka...

The Mediant 1000 Cheers Garry