For anyone having problems configuring w/provider

wiseoldowl

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#1
A few quick suggestions to start this off:

See if your provider is listed here:
Howto: Setting up VOIP Provider Trunks
Note that the pages in that section were contributed by many people and some may be more accurate than others (I'm particularly suspicious that the one about Broadvoice isn't right, for example).

Make sure that there is a context=from-trunk line in your trunk settings. Often they tell you that this goes in the USER details, but many providers treat you as an extension, not a peer, and in that case the USER context and details are totally ignored! So, if you can't get incoming calls to work, try moving the context=from-trunk to the PEER details and see if that makes any difference (if it does, you should probably clean out the USER context and details).

And, make sure you have created an incoming route and that the DID matches the number that follows the forward slash in your trunk registration. In other words, your registration string should take the form accountid:password@your.provider/yourDIDnumber, and whatever you have in the yourDIDnumber position is the DID you should be using for your inbound route. If that doesn't help, see How to get the DID of a SIP trunk when the provider doesn't send it (and why some incoming SIP calls fail) (the stinking smiley is hiding a colon and a lowercase "p" - please, someone, for the love of God, configure this forum software properly, or put it out of its misery!).
 

rafael

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#2
Nice +1 for your karma :)

Rafael
 

blakefrance

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#3
Hi, I'm new to elastix but have been using FreePBX for a bit using Broadvoice. I found I had to setup my trunks a little differently than both examples on the FreePBX webpage. I was wondering if you could guide me in posting my information so that people can find it, and hopefully it will help someone?
 

wiseoldowl

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#4
What did you have to do that was different that in Broadvoice (alternative/corrected)? Probably the best thing to do would be to post a comment on that page (you will need to be logged in).
 

haramarcuse

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#5
This might be an old thread but I just want to say that the source provided by wiseoldowl is very informative. Thanks for sharing.
 

scenarist

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#6
http://www.callwithus.com provider

I have a problem with configuring my Elastix server on http://www.callwithus.com provider.
I followed this officialy link for configuration
http://www.callwithus.com/configs/trixbox.html
but nothing.
When I dial via Xlite any number in e.164 format which is CountryCode_AreaCode_Phone_Number (example: 004917674187137 - mobile number for Germany) I got a voice message: "The person you are calling is unavailable, please try again".

In attachments I put my screenshoots of SIP Trunk and Outbound Route. Also I configured registration string on SIP trunk as username:password@sip.callwithus.com

What could be the problem?
Please help! http://forum.elastix.org/old_files/SIPTrunkandOutboundRoute.rar
 

nanoc84

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#7
hello u can help you , send me a e - mail
 

ccccp

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#8
dialling rule on trunk must be empty
also on outbound route dot is enough but looks fine to me.
try to make this route first as well
 

scenarist

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#9
ok.
My dialing rule on sip trunk is now empty and dialing rule on outbound rute contained only dot "."
But when I call e.g mobile of germany 4917674187137 , I got very low ringing tones voice quality and the call can not be established. But on my call history on web account I got report of this call. See attachment
I dont know what could be the problem. I need to call only Germany(fixed and mobile phones) but it seems with callwithus that is impossible.

I also tried that dialing phone number directly via X-lite but I got the same problem.
Callwithus voip provider is not enough good for calls to European countries! http://forum.elastix.org/old_files/Image_024.rar
 

scenarist

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#10
this is my asterisk cli output when I dialing mobile for germany

Code:
== Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called callwithus/4917674187137
    -- SIP/callwithus-0000070c is making progress passing it to SIP/201-0000070b
    -- SIP/callwithus-0000070c answered SIP/201-0000070b
    -- Packet2Packet bridging SIP/201-0000070b and SIP/callwithus-0000070c
    -- Executing [h@macro-dialout-trunk:1] Macro("SIP/201-0000070b", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/201-0000070b", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] NoOp("SIP/201-0000070b", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/201-0000070b", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/201-0000070b", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,10)
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/201-0000070b", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,12)
    -- Executing [s@macro-hangupcall:12] Hangup("SIP/201-0000070b", "") in new stack
  == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/201-0000070b' in macro 'hangupcall'
  == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/201-0000070b' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 4917674187137, 4) exited non-zero on 'SIP/201-0000070b'
       > doing dnsmgr_lookup for 'sip.callwithus.com'
       > ast_get_srv: SRV lookup for '_sip._udp.sip.callwithus.com' mapped to host sip.callwithus.com, port 5060
 

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