I recently redid my asterisk system and added FreePBX, i was able to get everything setup correctly for receiving and making calls to and from my Cisco SPA8800 (which required editing extensions_custom.conf)
Now when i receive a call from the SPA8800 the phones ring, but it displays the SPA’s extension, not the CID it received, i know the SPA is receiving CID because of the Last call has the cid listed.
When i didn’t have freepbx it would just automatically translate the cid.
I appreciate any help in getting freepbx to translate the caller id properly to the end phones
Does anybody have any suggestions, this issue is pretty frustrating and any help is really appreciated.
Are you having the FXO trunks actually route into the Asterisk system or just using the internal switching ability of the SPA8800 to ring the phones attached on the FXS ports? You have the option of either method on the SPA8800.
On the SPA8800, the FXO ports should work like the SPA3000 and the setup on Asterisk is a little complicated. If you Google for SPA3000 and Asterisk setup, you may get more insight. As I recall, you have to set delay timers which wait long enough to receive the CID from the CO trunk before ringing the call through.
Thanks for the reply
The SPA8800 is just used to dial the ring group in asterisk, the calling part works fine it is just the passing of the CID
the 8800 waits the proper amount of time to receive the CID, and they CID worked fine with just asterisk and no freePBX installed.
On the 8800 it is configured to send the CID as voip cid, but i think it is as simple as freePBX redefines the CID as the extension that the 8800 is using.
I hope that helps make more sense.
I feel that this is something simple that i am missing.
So i guess really my question is: Is there a way to allow what the CID was set on the extension to be passed to the ring group?
This should not be related to a FreePBX issue as people use SPA3000’s and SPA8800’s without issue. My only advice is to research the setup further. If the incoming trunk group and trunk extension is setup correctly, the incoming caller-id will be delivered to the ring group.
On the SPA8800, the FXO ports work just like the one on the SPA3000/3100 series and the FXS ports work just like the ones on the SPA2000 or PAP2T.
Manual Asterisk conf files are quite different from those generated by FreePBX so you need to do your due diligence to find the correct way to program it. That is specifically why I mentioned the similarity to the SPA3000. A quick Google search yields this link: http://www.freepbx.org/support/documentation/howtos/howto-linksys-spa-3102-sipura-spa-3000-freepbx
alright, well it sounds like my problem may be that i kind of did a workaround that setup the 8800 as a extension and not as a truck of any kind and did a custom dialplan to send out the calls, i will look into this article
I followed the guide provided by kenn10, and i followed the instructions the best i could, despite being for a SPA3300 instead of the 8800. I can call out but it won’t answer incoming calls. using (S0<:1234567890>) as the dialplan (1234567890 being the num i put in instead of my phone number), or using the way i did it b4 where it called the ring group directly, either way it never rings in, with asterisk cli verbosity set up to like 16 all i get on a call in is:
“Using SIP RTP TOS bits 184
Using SIP RTP CoS mark 5”
and nothing more, all i changed on the spa was the dialplan and the userid/ password. But i configured the spa as a trunk, configured incoming and outgoing.
I am pretty sure that it is getting past the SPA and asterisk isn’t picking it up, but i am not positive.
I could really use some help here and i need to get the system back up and running before Monday
Any help is appreciated.
I ran a syslog server on my computer and set the SPA8800 to report to that so i could try to get a more detailed error report, when i make a external call into our system it goes:
M0: CC:Failed w/ Calling
M0: [1:0]AUD Rel Call
M0: FM Alert Stop RxTx (c=002b6da0;a=0)
M0: [1:0]RTP Rx Up
M0: [1:0]AUD ALLOC CALL (port=16459)
M0: Calling:[email protected]:0
bottom most being first in time
any help is really appreciated.
alright well i feel stupid, this morning i get in, fight with it for a while, then decide to do a reboot of the server, and low and behold that fixes everything.
This has been strange as a reboot fixed 2 other issues when i first set this up, but it doesn’t quite make sense as to why.
Regardless it is working, CID is passing and all is good.
Except that my music on hold just happened to completely stop working now. On to my next challenge…
My next recommendation is that you back up all your config files to an external media and install a fully functional package like PBX In A Flash from the ISO. Or you can keep chasing little things forever…
Hi. Need your help to configure the ability FXO lines to route directly to internal FXS ports if SIP registration fail.