Figured it out
asterisk -vvvvvr
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View "Trunk" traffic?
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Multiple GVGW Trunks
Trunk name must be unique; but will not validate if trunk name is not gvgw.simonics.com
This is an issue as I have 3 GVGW trunks I would like to set up using one system.
Please let me know if I can provide any other info to help. Basically, I can be logged into one trunk or the other, but not two or more at once, since it will not validate if the trunk name is anything other than gvgw.simonics.com, and each trunk name must be unique. I've tried using the IP, and I've also tried inserting credentials for both trunks in one file. Perhaps I have to edit the Asterisk .conf files directly? I already had a peek inside sip_registrations.conf which has the login credentials for all accounts already, but I'm not sure where the host would go.
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Multiple GVGW Trunks
Change the trunk name. It doesn’t have to be the host name.
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Our bound rout from CUCM to PSTN trough Asterisk
I have an outbound route on my CUCM so I can call out side, when I call the range of ext for asterisk they work correctly how ever if I try to dial an external number form cucm it is ringing my ring group that I have configure for inbound. how can I configure a route so when a call comes from ext 1xxx with a dial number xxxxxxxxx and routes it to the pstn
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Twilio Inbound route calles get “the number you have dialed is not in service”
I enable debug mode. I have not been able to get a log because my server is not accessible right now and is on a remote location. but I did had twilio look at a log snippet I sent them and this is what they told me:
Twilio is sending the initial INVITE to 184.98.217.155 however it is sending the ACK to a 184.98.160.208 address, and it appears your equipment is not getting the ACK, as it is resending the 200 OK, until it eventually times out and disconnects the call after 32 seconds.
The reason we are sending the ACK to a 184.98.160.208 is due to the fact that this is the address which is in the Contact header returned in the 200 OK from your equipment (184.98.217.155). I am thinking that .155 never sees the ACK and times out the call (sends a BYE) after 32 seconds.
So to rectify there are 2 options:
(a) Have .155 insert itself into a Record-Route header in the 200 OK
(b) Figure out why .208 is being inserted into the contact header rather than .155
My question is why will it be seeing 2 ip address. It is a simple dsl modem with dhcp address
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Twilio Inbound route calles get “the number you have dialed is not in service”
Check the external IP configured in Asterisk SIP Settings.
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Multiple GVGW Trunks
I have changed it. If it is not the host name, it will not connect.
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Trying to configure incoming fax-to-email
Nope. Nothing in the log for the time the Fax was sent.
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Multiple GVGW Trunks
Please follow our freepbx configuration guide found on the support site. I recommend configuring as type friend and using auth name matching as described in the guide. This requires that you set the peer name the same as the username, which would be unique for each trunk.
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Multiple GVGW Trunks
I will try it first thing in the morning. Pretty sure I followed Option one to the tee, but it still did not work. I will let you guys know how it goes. Thank you
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Eth ports on Sangoma UC 40
The reason we add 4 is most people want 2 for LAN and WAN and 1 spare. Its cheaper to do 4 ports than 4 with the switch we are using on the MB.
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Intrusion detection blocks by home ip
These are the messages I'm getting from fail2ban
The IP ...... has just been banned by Fail2Ban after
20 attempts against apache-auth on .
The IP ..... has just been banned by Fail2Ban after
34 attempts against recidive on .
both for the same ip address.
Why is there so many attempts on apache-auth? Can somebody help figure this out? My home IP constantly gets banned by fail2ban. It's crazy!
Why is there 34 attempts
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Eth ports on Sangoma UC 40
Thank you!
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Eth ports on Sangoma UC 40
Thank you for the update. When should we expect to see the new and improved UC 40 with the WAP features added?
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Which is more efficient, ChanSIP or PJSIP?
chan_sip is no longer maintained and hasn't been in 4ish years
As of the 25th of this month it will receive no more security patches.
Performance depends on what you are doing. 1 call chan_sip will win performance wise, under a normal to high call load pjsip will win.
Most of the "issues" people have are likely configuration issues. This is not super suprising with ~200 config options in PJSIP it is pretty daunting from the outside. You don't need then all. It is especially more difficult than pasting a blob from some cha_sip example and pressing go.
There are resources like https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip#Migratingfromchan_siptores_pjsip-SidebySideExamplesofsip.confandpjsip.confConfiguration but this doesn't mean much when configuring from the gui. We are going to work on documenting "chan_sip blob" to "pjsip GUI settings"
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Inbound call terminates at 32 second
This has been fixed. I restarted my modem and everything the IP address updated
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Multiple random incoming calls
After resetting my modem and updating the sip address this is no longer happening
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Eth ports on Sangoma UC 40
No clue what you mean by WAP being added.
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Fxotune doesn't put best settings into fxotune.conf
When FreePBX reboots, I have horrible echo. I can run fxotune and then it is great until next boot.
When I run fxotune, I get lines like this in the output
Done!
Found best echo coefficients: 1=9,0,251,252,2,255,0,0,0
/etc/fxotune.conf shows a new modification time, but does NOT have the above settings in the file, it has
1=6,0,0,0,0,0,0,0,0
instead. I have manually put in the data from the fxotune run, but I wonder why fxotune does not write these numbers into the /etc/fxotune.conf file?
Jon
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Fxotune doesn't put best settings into fxotune.conf
It should so write
man fxotune
youi need to fxotune -s on startup to get them restored
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