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IAX / SIP from Asterisk

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IAX / SIP from Asterisk

Will F9 be producing a list of settings for those of us in the Asterisk realm who'd like to connect to Plustalk?
6 REPLIES
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IAX / SIP from Asterisk

Considering it looks like F9 _is_ using Asterisk to provide the PlusTalk SIP proxy (judging by the default voicemail app (chameleon mail!) - they didn't even change the default install voce prompts!) - I would guess it would be pretty easy to peer your Asterisx server to theirs - although the headaches for F9 to try to support all these additional peers would take some managing. Just look at the sticky for all the **nnn shortcodes for well established VoiP providers (glad to see FWD in there!), this list could grow to be very large indeed. _BUT_, what a great idea - you buy an Asterisx PBX solution, get F9 to add your server to their peer list - if a standard way of defining these short codes could be agreed (like the UK 01nnn/02nn dial prefixes), then making VoIP calls would be a _lot_ simpler!
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IAX / SIP from Asterisk

I'm not even looking for IAX2 peering - merely the correct configuration for SIP peering so I can route calls through them as needed Smiley
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IAX / SIP from Asterisk

I don't have Asterisk running in front of me right now, but I think you should be able to add F9 as a SIP peer to your Asterisx config. If I remember, it's all setup in the sip.peers config file. Create a dialplan entry (e.g. all numbers beginning with **6) and then make an entry in sip.peers with all your SIP account details in the relevant places. Then, when one of the users on your Asterisk PBX wants to dial another PlusTalk user, they could dial **6nnnnnn and directly dial PlusTalk user nnnnnn. Without this, your Asterisk users would have to enter the full SIP address of the party they want to call.

I'll have a tinker with Asterisk and see if I can get it to work...
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IAX / SIP from Asterisk

Yeah, I've tried a sip.conf entry, but getting the username etc parameters in the right order has been a bit non-easy. Oh well, I'll poke it some more tonight and if I get it working, post to the voip-info wiki.
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IAX / SIP from Asterisk

Having applied the SPA instructions instead of XLite, I believe I have a working config.

sip.conf:

[general]
register => SIP_ID:F9_PASSWORD@sip.force9.net

[plustalk-out]
type=peer
username=SIP_ID
host=sip.force9.net
secret=F9_PASSWORD
authuser=SIP_ID
fromdomain=username.force9.co.uk
context=home
dtmfmode=rfc2833
insecure=very
allow=all



extensions.conf:

[plustalk]
exten => _758.,1,SetCallerId(My Name <SIP_ID>)
exten => _758.,2,Dial(SIP/**393${EXTEN:3}@plustalk-out,60,tr)
exten => _758.,3,Hangup()



My plustalk extension is set up for calling FWD explicitly right now, just so I could test with 613. I was able to connect, and Asterisk registered for inbound, so I'd guess it's working fine.


-- Executing SetCallerID("SIP/2000-2097", "Me <4>") in new stack
-- Executing Dial("SIP/2000-2097", "SIP/**393613@plustalk-out|60|tr") in new stack
-- Called **393613@plustalk-out
-- SIP/plustalk-out-e9d3 is ringing
-- SIP/plustalk-out-e9d3 answered SIP/2000-2097
-- Attempting native bridge of SIP/2000-2097 and SIP/plustalk-out-e9d3
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IAX / SIP from Asterisk

Nice one! Cheesy
Just compiled Asterisk on my machine. Entered my account details etc. into sip.conf, started up Asterisk, and it all "just worked":

Quote

*CLI>sip show peers
Name/username Host Dyn Nat ACL Port Status
plustalk-out/***760 84.92.*.*** 5060 Unmonitored
1 sip peers [1 online , 0 offline]
*CLI> sip show peer plustalk-out


* Name : plustalk-out
Secret : <Set>
MD5Secret : <Not set>
Context : home
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
CallingPres : Presentation Allowed, Not Screened
FromDomain : *********.force9.co.uk
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : -1
Call limit : 0
Dynamic : No
Callerid : "" <>
Expire : -1
Insecure : port,invite
Nat : RFC3581
ACL : No
CanReinvite : Yes
PromiscRedir : No
User=Phone : No
Trust RPID : No
Send RPID : No
DTMFmode : rfc2833
LastMsg : 0
ToHost : sip.force9.net
Addr->IP : 84.92.*.*** Port 5060
Defaddr->IP : 0.0.0.0 Port 0
Def. Username: ***760
SIP Options : (none)
Codecs : 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p)
Codec Order : (none)
Status : Unmonitored
Useragent :
Reg. Contact :


Interesting set of codecs negotiated - including some video too...I wonder how long 'til F9 start offering "videoconferencing". Need to set up a few more extentions, but this is now starting to get very interesting...Hmmm.

Cheers!