Yes, that’s what I was thinking.
Is it a process to setup SIP URI calling? Is there a tutorial out there that you know of that explains how to set this up? I’d ultimately like to setup a specific SIP URI destination in our client PBXs and allow incoming SIP URI calls in our PBX?
You are always down to the point. I just reviewed the guide and can’t wait to act on it when I get to work. Thank you.
Check the last lines of
/var/log/asterisk/full
are they quite recently timestamped? If not check in /var/log/messages at about that time. Also check for coredumps in /tmp
Asterisk version? Does this apply to you: Removal of Asterisk 13.26.0-1 and Asterisk 15.7.2-1
So a SIP URI is easy enough to set up on the sending PBX, but the receiving PBX needs to be able to distinguish the incoming INVITE from all of the spurious internet SIP traffic that it’s hard at work keeping out. This is done by setting up a SIP trunk, the thing you are trying to avoid. There is nothing special about a trunk, in it’s most basic form it’s identifying a peer to Asterisk saying “calls from this IP are trusted and go to xxx context”. That’s what you need on the receiving PBX.
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I’ve got a client we deployed a couple dozen S705’s. The client is a warehouse with most of the phones in their office portion, and a few in the warehouse. They heavily use walkie talkies to communicate.
I’ve observed their radios frequently interfere with the Sangoma Phones and cause all kinds of strange behavior. It frequently causes false button presses, like placing calls on hold, or enabling DND among other unwanted behavior.
I’m 100% certain the radios are the cause. Phones are powered by POE. Obviously asking the client not to use their radios isn’t going to fly. Tomorrow I’m going to put some Digium phones out there to see if the behavior persists. It did not happen with their old Merlin system FWIW.
I thought about adding ferrite cores around the patch cables, but am unclear if the patch cable itself is acting as the antenna here or if the phone itself is just accepting the interference (as FCC mandates they do) because the radios are frequently only a few feet away from the phones.
Anyone run into this, or have ideas on how to address it short of replacing the phones with a different brand?
Understood.
I don’t mind setting up a single SIP trunk on the receiving side. What I really want to avoid is setting up DEDICATED IAX trunks for each client PBX that may call into our PBX. The only tutorials I’ve read describe how to configure a trunk directly between two PBXs. This would be a nightmare if I had to manually setup a trunk between us and each of our 40 clients.
But a single trunk for all URI traffic would be fine for me. Couldn’t I program a Shared Secret that our client PBXs would need to allow a call into our PBX?
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I need help understanding what this dial plan does. I would like users to be able to add a “1” when making a call with the area code and number(10 digits) or a toll free number. Now if they proceed a number with a “1” they get “number not in service” recording.
Admittedly, this is a newbie question. Guilty as charged.
You are prepending +1 you said you wanted to prepend 1 for 10 digit dialing, you should also add 1NxxNxxxxxx for folks who have already dialed an eleven digit call if the + belongs anywhere is in the trunk dialing ruled that require it.
Thanks. I understand the need to add 1NxxNxxxxxx. But, please explain what prepending +1 in the current dialplan does. I “inherited” this system from someone else and I’m trying to make sense of what was intended here to see if I still want/need it.
Some trunks like what is called e164 dialling, whereby the + is a metacharacter for international access , i.e. 011 for NANP land and 00 for MOTROTW . most phones dont have that key but all cell phones do (by long pressing the 0) so if your carrier needs it on their trunks then add that to that particular trunks dial manipulation rules
Thanks for that explanation. But just to be sure I understand … if I just want my users to have the option of calling a number “long distance” or toll-free (10 numbers) and adding a 1 before the number, I need to add 1NxxNxxxxxx to the existing dial plan. Correct?
If they dial 10 digits but the trunk needs 11 prepend 1 to NXXNXXXXXX
if they dial 11 digits ------------------------- then also have 1NXXNXXXXXX
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.( its all in the wiki linked at the top of this page)
If you really want to troubleshoot the ‘line’ feature, at the Asterisk command prompt type
pjsip set logger on
which will include the SIP traffic in the normal Asterisk log. Look at whether an incoming INVITE includes the line tag in the URI; if not check the Contact header of an outgoing REGISTER request. Or, post the relevant log segments.
However, I know nothing (so far) about your system that would require the line feature. In pjsip Settings -> Advanced for your trunks, try setting Contact User to the DID for that trunk (in the same format that you have in your Inbound Routes).
Alternatively, In pjsip Settings -> General for your trunks, try changing Context from from-pstn to from-pstn-toheader, which should cause the DID (or perhaps the account number, whatever the provider sends) for the trunk to be used for routing incoming. Temporarily set up an Inbound Route with DID Number and CallerID Number left blank, pointing to a working extension. Call in and see what appears in the DID field of your CDR. Change your production Inbound Routes accordingly, then delete the temporary route.
As it turns out my SIP trunking provider (Bandwidth.com) supports only 2 number formats - E.164 and 10-digit. Even if I add 1NxxNxxxxxx to my dial plan outbound calls fail. Is there a way to create an entry in my dial plan to accommodate a user that adds a “1” before a 10-digit number as this is the way a lot of numbers are posted for contact information?
Create outbound routes
11 digit = 1NXXNXXXXXX
10 digit = NXXNXXXXXX
in the trunk manipulation rules prepend + to 1NXXNXXXXXX and either accept NXXNXXXXXX as-is or prepend +1 to it.
If you only have that one trunk , do it in or both places but just understand what the prefix and the prepend does to the base number, again all in the wiki and most in the hover over help.