TIPS New problem with Anveo (first time)

sirdotcom

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Hi,
I just installed the RaspberryPi IPBX 13-13 system. Things are working great, until, after Anveo has worked on "regular" PC or VPS installs, I get this error on an incoming call:

Code:
 chan_sip.c:26472 handle_request_invite: Call from 'anveo-1' (50.22.101.14:1054) to extension '15050000000' rejected because extension not found in context 'from-anveo'

I don't get it ... what is the context looking for? It's sending it a number which it should be happy about.

Thanks,
Steve

EDIT: Well I figured it out finally. In the peer section it says context=from-anveo-trunk. But there isn't one. So I put it to from-trunk and it works great. Only problem now is I get NO audio and I bet the cursed firewall is blocking RTP ...
 
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jreming

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Anveo is a tricky beast, you can actually set it up as 1 pjsip trunk and get in/out in 1 swoop.

But the issue I got and recently solved was 1 way audio for outbound calls.

With how Anveo works, you do not have a list of known ip addresses, instead Anveo's signaling server will use ice to tell you what server to communicate with to receive audio.

If you created a pjsip trunk for Anveo, you can enable the trunk to use ice but you can only do it from editing the config.

nano /etc/asterisk/pjsip.endpoint.conf

find your anveo pjsip trunk and append

icesupport=yes

to it
 

proftech

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Anveo is a tricky beast, you can actually set it up as 1 pjsip trunk and get in/out in 1 swoop.

But the issue I got and recently solved was 1 way audio for outbound calls.

With how Anveo works, you do not have a list of known ip addresses, instead Anveo's signaling server will use ice to tell you what server to communicate with to receive audio.

If you created a pjsip trunk for Anveo, you can enable the trunk to use ice but you can only do it from editing the config.

nano /etc/asterisk/pjsip.endpoint.conf

find your anveo pjsip trunk and append

icesupport=yes

to it
Hi jreming
Yours is the closest thing I've seen to add anveo as a pjsip endpoint. Question; Are you using Anveo direct or retail? I'm wishing to use Anveo direct, but only for outbound. I use Vanilla command line Asterisk 13.6 w/ pjsip and haven't seen any samples of pjsip endpoint, etc settings. Should be simple for outbound only but my endpoint shows unavailable. The ad for PBXinaflash said the are a lot of Asterisk people on this forum. I thought surely somebody is using Anveo Direct. I know I will have to experiment with NAT settings but I need to get the endpoint working first. Thanks
 

proftech

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Yes. I was looking for the link in my history just now and couldn't find it. Thanks for sending, however as far as I can tell still no joy. :(
 

wardmundy

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Yes. I was looking for the link in my history just now and couldn't find it. Thanks for sending, however as far as I can tell still no joy. :(

Then perhaps you missed a step. We use Anveo Direct all the time without any problems. If you had context=from-anveo-trunk, then you weren't following our tutorial.

Have you set up your public and private IP addresses in Settings -> SIP Settings -> NAT Settings?
 

proftech

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With a little experimentation I think I have it working. Will try it out on a long call on Monday. Thanks for the help.
 

jreming

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Hi jreming
Yours is the closest thing I've seen to add anveo as a pjsip endpoint. Question; Are you using Anveo direct or retail? I'm wishing to use Anveo direct, but only for outbound. I use Vanilla command line Asterisk 13.6 w/ pjsip and haven't seen any samples of pjsip endpoint, etc settings. Should be simple for outbound only but my endpoint shows unavailable. The ad for PBXinaflash said the are a lot of Asterisk people on this forum. I thought surely somebody is using Anveo Direct. I know I will have to experiment with NAT settings but I need to get the endpoint working first. Thanks


I use Anveo Direct and I have found that pjsip will pretty much work on any sip system, you just need to adapt it correctly.

If you plan on outgoing only make sure that you can reach the endpoint sbc.anveo.com:5060 (i think, port is important). Also make sure you enable icesupport on the trunk as well as open up your rtp ports as we have no idea what the ip address of the final destination will be (present in the ice headers)

I personally have never really dealt with just asterisks alone and always used some form of a gui to assist so I am afraid I cant really help you with the setup other then confirming some settings.

What do the logs say when trying to make a call? (also it should never register, that should be disabled)


EDIT:

Looks like I was late to the party, sorry crazy day at the office.

Glad you got it working and let me know if you need any further help.
 

proftech

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Replying to jreming
EDIT:

Looks like I was late to the party, sorry crazy day at the office.

Glad you got it working and let me know if you need any further help.

After a bit of tinkering, it seems to work perfectly and not a really massive configuration at all [for outbound only]. And as a side benefit I was able to carry the setup over to my Flowroute trunk, which also offers direct audio. And with the correct pjsip option settings I don't seem to need to mess with router / port settings at all! Will be doing further testing ...
 

jreming

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Yea in general pjsip doesnt require much outside of the basic, server, port, user/pass pertty much everything else can be left alone and it will auto determine the match ips.

Highly recommend switching everything over to it sooner rather then later.
 

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