Asterisk How to connect to IConnectHere

IConnectHere will sell you dialing plans that allow you to call out to the regualar phone system. You can then configure your SIP phones to register with them and call out thru them.

This page describes how to configure Asterisk so it acts as a SIP client, dialing out thru Asterisk, and then how to setup your dialplan so that your users calls are passed to Asterisk and then to IConnectHere.

First you need to configure something in sip.conf

[iconnect]
; for routing calls outbound to the PSTN numbers via iconnecthere
; (aka deltathree)
type=friend
secret=#SECRET#
username=#USERNAME#
host=natrelay.deltathree.com
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=G726

Then you'll need to configure something in your dialplan in extensions.conf (this might look like this)

[iconnecthere]
;; Including this into your context enables calls to the PSTN network
;; via the iconnect account SIP account configured sip.conf
exten => _91NXXNXXXXXX,1,SetCallerID(9195551212)
exten => _91NXXNXXXXXX,2,SetCIDName("John Smithe")
exten => _91NXXNXXXXXX,3,Dial(SIP/${EXTEN-1}@iconnect,60,r)
exten => _91NXXNXXXXXX,4,Congestion

You can then include that context into the dialplan for the users you wish allow to spend your iConnectHere minutes.

Inbound calls

If you have an iConnectHere number, you need to register it in sip.conf using the following format:

register => number:password:user@natrelay.deltathree.com

number: Your iConnectHere phone number. E.g. 12125551212
password: Same as #SECRET# above
user: same as #USERNAME# above

= Discusssion =

When I first set this up the calls would go thru but no audio; fixing that required experimenting with the codecs allowed, I have no idea if those are the optimal setup for those.

This works fine, but for me at least the caller ID doesn't get passed along and the ringing doesn't work well.

I had calls dropping after 5 seconds. Adding this to sip.conf fixed it:

canreinvite=no

Another "gotcha" is that iconnecthere doesn't work with qualify=yes or qualify=200 (or any given numeric value). I had added a qualify statement to all my SIP peers but hadn't realized the link between adding that and iconnecthere outbound stopping working. Having discussed it with someone else and checking out their config file, I noticed the difference. Removing the qualify line from my SIP peer solved the problem and I'm now able to call out successfully using iconnecthere again.

Support


See also, Asterisk sip canreinvite, the comments here, and at IconnectHere

IConnectHere will sell you dialing plans that allow you to call out to the regualar phone system. You can then configure your SIP phones to register with them and call out thru them.

This page describes how to configure Asterisk so it acts as a SIP client, dialing out thru Asterisk, and then how to setup your dialplan so that your users calls are passed to Asterisk and then to IConnectHere.

First you need to configure something in sip.conf

[iconnect]
; for routing calls outbound to the PSTN numbers via iconnecthere
; (aka deltathree)
type=friend
secret=#SECRET#
username=#USERNAME#
host=natrelay.deltathree.com
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=G726

Then you'll need to configure something in your dialplan in extensions.conf (this might look like this)

[iconnecthere]
;; Including this into your context enables calls to the PSTN network
;; via the iconnect account SIP account configured sip.conf
exten => _91NXXNXXXXXX,1,SetCallerID(9195551212)
exten => _91NXXNXXXXXX,2,SetCIDName("John Smithe")
exten => _91NXXNXXXXXX,3,Dial(SIP/${EXTEN-1}@iconnect,60,r)
exten => _91NXXNXXXXXX,4,Congestion

You can then include that context into the dialplan for the users you wish allow to spend your iConnectHere minutes.

Inbound calls

If you have an iConnectHere number, you need to register it in sip.conf using the following format:

register => number:password:user@natrelay.deltathree.com

number: Your iConnectHere phone number. E.g. 12125551212
password: Same as #SECRET# above
user: same as #USERNAME# above

= Discusssion =

When I first set this up the calls would go thru but no audio; fixing that required experimenting with the codecs allowed, I have no idea if those are the optimal setup for those.

This works fine, but for me at least the caller ID doesn't get passed along and the ringing doesn't work well.

I had calls dropping after 5 seconds. Adding this to sip.conf fixed it:

canreinvite=no

Another "gotcha" is that iconnecthere doesn't work with qualify=yes or qualify=200 (or any given numeric value). I had added a qualify statement to all my SIP peers but hadn't realized the link between adding that and iconnecthere outbound stopping working. Having discussed it with someone else and checking out their config file, I noticed the difference. Removing the qualify line from my SIP peer solved the problem and I'm now able to call out successfully using iconnecthere again.

Support


See also, Asterisk sip canreinvite, the comments here, and at IconnectHere

Created by: bhyde, Last modification: Tue 27 of Feb, 2007 (20:31 UTC) by echamberlain
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