FreePBX Setup for sip2sip.info

rchalk

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Can someone help me with the full trunk configuration details for both inbound and outbound settings, or point me to a thread with the information? Thanks. Richard
 

wardmundy

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I'm Not Very Good At This, But...

Here's what I'm using for the trunk settings...

Dial rules:
223NXXXXXX

In Outgoing Settings...

Trunk Name: sip2sip
username=223343xxxx
type=peer
canreinvite=no
secret=xxxx
qualify=yes
nat=yes
insecure=invite,port
host=proxy.sipthor.net
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=ulaw&gsm

Leave Incoming Settings Blank!

Register: 223343xxxx:[email protected]/223343xxxx

Next create an Inbound Route for your DID using the 223343xxxx number you put on the end of the register string. Point this DID to wherever you want to answer the calls. NOTE: GSM is required or you'll have problems.

Some have reported that the above didn't work for outbound calls so here's a workaround.

Create a custom trunk that looks like this:

Set your CallerID

Dial Rules:
223NXXXXXX
001NXXNXXXXXX <- only if you want to pay for calls

Custom DialString: SIP/$OUTNUM$@sip2sip.info

Then create an Outbound Route called sip2sip with...

Dial Rules:
same as above

Trunk: SIP/$OUTNUM$@sip2sip.info
 

rchalk

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Thanks, Ward. I appreciate the help!
Richard
 

warraich

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Didn't work outbound call for sip2sip

Didn't work outbound call for sip2sip. I did as ward said,
Create a custom trunk that looks like this:

Set your CallerID

Dial Rules:
223NXXXXXX
001NXXNXXXXXX <- only if you want to pay for calls

Custom DialString: SIP/$OUTNUM$@sip2sip.info

Then create an Outbound Route called sip2sip with...

Dial Rules:
same as above

Trunk: SIP/$OUTNUM$@sip2sip.info

but it will just dial out from GV.
can anyone help.
Thanks,
 

wardmundy

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Dialout is in Trunk order. If your new trunk is below your default (or some other) trunk, then it never gets used if there is a dial string match on another trunk.

So... move it on up.

128996156253503977.jpg
 

warraich

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Dialout is in Trunk order. If your new trunk is below your default (or some other) trunk, then it never gets used if there is a dial string match on another trunk.

So... move it on up.

128996156253503977.jpg
Thanks Ward. I tried moving the trunk up but it did not work. I will try again maybe I did some thing wrong.
 

warraich

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Thanks ward I moved it up above default and also changed 223NXXXXXX to 2233XXXXXX and now it is working.
 

atlpbxguy

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outgoing calls failing

Ok I have setup the system asterisk with two nortel phones. one is extension inside and the other is on sip2sip directly.

when I call in from sip2sip it works fine and the video/audio is perfect.

when I call out from the asterisk I see the call ring in on sip2sip phone but if I click to answer either by green button, pick up rec, or video button it still show call. the buttons at bottom are still ignore and decline.

when I do hang up it shows missed call.

all the time it is sending video and audio. the asterisk phone is seeing the video fine and audio is two ways. any ideas?

my setup is as follows

I have three trunks - one for each of the ips on the sip2sip proxy.
username=22334xxxxx
type=peer
canreinvite=yes
secret=xxx
qualify=yes
nat=yes
insecure=invite,port
host=81.23.228.150
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=ulaw&gsm&h263


then I added an outgoing as you stated above with custom trunk.

If I remove the custom outbound I end up with network congested error.

any thoughts would be appreciated as I am still fairly new.

thank you.
 

rossiv

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:( This isn't registering for me...
In sip show registry:
Code:
piaf-home*CLI> sip show registry
Host                            Username       Refresh State                Reg.Time                 
sipgate.com:5060                3932765e0          105 Registered           Thu, 04 Nov 2010 20:50:55
proxy.sipthor.net:5060          2233442659         120 Request Sent
 

warraich

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:( This isn't registering for me...
In sip show registry:
Code:
piaf-home*CLI> sip show registry
Host                            Username       Refresh State                Reg.Time                 
sipgate.com:5060                3932765e0          105 Registered           Thu, 04 Nov 2010 20:50:55
proxy.sipthor.net:5060          2233442659         120 Request Sent
[I changed host=proxy.sipthor.net to host=sip2sip.info and it worked for me.]
 

rossiv

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Doesn't work for me.:(
Originally Posted by rossiv :( This isn't registering for me...
In sip show registry:


Code:
piaf-home*CLI> sip show registry
Host Username Refresh State Reg.Time
sipgate.com:5060 3932765e0 105 Registered Thu, 04 Nov 2010 20:50:55
[
proxy.sipthor.net:5060 2233442659 120 Request Sent



[I changed host=proxy.sipthor.net to host=sip2sip.info and it worked for me.]
 
Last edited by a moderator:

warraich

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Doesn't work for me.:(
My trunk looks like this it is working with Video.

username=2233439xxx
type=peer
canreinvite=no
secret=xxxxxx
qualify=yes
nat=yes
insecure=invite,port
host=sip2sip.info
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=h263&ulaw&gsm

Sip registry

sip2sip.info:5060 223343xxx 105 Registered Sun, 07 Nov 2010 16:15:05
 

rossiv

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What domain do you have in your reg string?
 

jcusick

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I've tried every combination of:

host=proxy.sipthor.net and proxy.sipthor.net

register string with either sip2sip.info or proxy.sipthor.net

and no luck, can't register.

tcpdump gives me either udp port sip unreachable or other errors.

I have to assume the site is down this weekend or I'm doing something really wrong that I just cant see. I even did a nslookup and tried all the ip addesses, no luck.

Is this group reliable or am I just screwing up?
 

ojthecat

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I've tried every combination of:

host=proxy.sipthor.net and proxy.sipthor.net

register string with either sip2sip.info or proxy.sipthor.net

and no luck, can't register.

tcpdump gives me either udp port sip unreachable or other errors.

I have to assume the site is down this weekend or I'm doing something really wrong that I just cant see. I even did a nslookup and tried all the ip addesses, no luck.

Is this group reliable or am I just screwing up?

Post your trunk settings and register string for your sip2sip trunk you have something incorrect in there.
 

jcusick

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PEER Details:
username=223344xxxx
secret=xxxxxxxx
type=peer
canreinvite=no
qualify=yes
nat=yes
insecure=invite,port
host=sip2sip.info
dtmfmode=rfc2833
disallow=all
allow=h323&ulaw&gsm
context=from-trunk

I have also tried this with and without:
outboundproxy=proxy.sipthor.net

in the PEER settings

Register String:
223344xxxx:[email protected]/223344xxxx

Thanks in advance for the help :smile5:

JC
 

jcusick

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By the way, when I asked if this group was reliable, I meant sip2sip, not anyone here on this forum. I hope it wasn't taken the wrong way.

JC
 

ojthecat

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PEER Details:
username=223344xxxx
secret=xxxxxxxx
type=peer
canreinvite=no
qualify=yes
nat=yes
insecure=invite,port
host=sip2sip.info
dtmfmode=rfc2833
disallow=all
allow=h323&ulaw&gsm
context=from-trunk

I have also tried this with and without:
outboundproxy=proxy.sipthor.net

in the PEER settings

Register String:
223344xxxx:[email protected]/223344xxxx

Thanks in advance for the help :smile5:

JC

I dont use outboundproxy

i use
host=proxy.sipthor.net

everything else looks the same
 

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