i have a spa3102 ata, most of the programming is straight forward and it partially works. I can make outbound calls ,which i dont need to make calls, i need inbound calls for the ringer. We have a s8300 server and g450 gateways,i have the license and everything is setup thru session manager but when i use putty and traceSM i see 408 timeout,503 service unavailable and publish 404 unauthorized. My thoughts are Response Status Code Handling under the sip tab,or dial plan might affect inbound traffic, but again im not sure and cant find any documentation.I dont understand alot of the ip side,im learning it slowly but surely. Any help or direction would be great.
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Help with spa3102 ata
- Thread starter Cityman13
- Start date
At the very least you would need a user profile setup in System Manager for the sip line/account. That profile would have a sip communication address - 1234@yourcompany.com, then it would have a session manager profile, and originating and terminating application sequence, and be associated to a CM endpoint.
All that is on the user profile page in user management. Basically, the application sequence is what Session Manager proxies you to (CM). CM has to process calls for your gateway for things like if you're call forwarded.
The link below outlines all the steps to do it with an Audiocodes. Its a decent doc as it covers everything Avaya SM/SMGR/CM involved as well as the Audiocodes box - which would be the only place you differ from that reference config.
Good luck!
All that is on the user profile page in user management. Basically, the application sequence is what Session Manager proxies you to (CM). CM has to process calls for your gateway for things like if you're call forwarded.
The link below outlines all the steps to do it with an Audiocodes. Its a decent doc as it covers everything Avaya SM/SMGR/CM involved as well as the Audiocodes box - which would be the only place you differ from that reference config.
Good luck!
- Thread starter
- #3
Thanks kyle555 for the help,i went thru all the settings and I should have been clearer on the setup. Our avaya dealer setup all the sip routing and everything needed for our ivr system to go sip,we had them setup what was needed on the switch side for the ata to work. what seems to be the issue is the ata isnt responding to the invite from a incoming call. Thats where i cant find any documentation or setup and linksys said call who we purchased it from (amazon),cisco said buy a service agreement,and avaya said its not been approved and offer no help.
And therein lies the problem of cobbling together a mishmash of things that isn't explicitly supported. Have you tried an Audiocodes MP112? They're relatively cheap to get your hands on one and test with. But more specifically, what is your call flow? I would expect something like: dial an extension of a "SIP" phone in CM, that SIP phone routes via AAR to Session Manager, and Session Manager would proxy that invite to the SPA that is registered to SM as a station.
I wouldn't think you're doing this as "trunking" to the SPA as a SIP Entity with an Entity Link in Session Manager, but more as if the SPA were registering to SM with the profile of a SIP phone in CM - like the Audiocodes doc would read, or like you would register any old SIP phone.
Also, stick with TCP!
I understand the SPA isn't responding to an "invite" message - but what does its registration from the get go look like?
I wouldn't think you're doing this as "trunking" to the SPA as a SIP Entity with an Entity Link in Session Manager, but more as if the SPA were registering to SM with the profile of a SIP phone in CM - like the Audiocodes doc would read, or like you would register any old SIP phone.
Also, stick with TCP!
I understand the SPA isn't responding to an "invite" message - but what does its registration from the get go look like?
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