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Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

This application note describes how to configure and deploy a SPA8800 in an Asterisk environment. It incluldes SPA8800, Asterisk sip.conf, and Asterisk extensions.conf files.

Also included is a troubleshooting section with sample traces showing registration and call flows.

Comments
Cisco Employee

Hi SPA8800 owners and potential owners,

How can I help you with the SPA8800?

Do you need more information or examples? Let me know and I'll do my best to provide the content / help.

Regards,

Patrick

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New Member

How about a SPA8800 to UC540 configuration example- that is to add an additional 4 FXO ports to UC5x0 total capacity?

New Member

In the works. Check back in a couple of weeks.


Thanks,

Marcos

New Member

Also maybe interesting might be to add a section to the document on how to integrate the spa8800 in asterisk using the web-gui (trixbox/asterisknow/...) And in due time the same document might be interesting for the integration into freeswitch.

New Member

Could you provide any idea how to integrate SPA8800 and CUCM?

New Member

hi pat, are you able to show me how to connect the spa8800 to an analog ptsn?> primarly so the outgoing failover works via the analog lines?

New Member

Did anything ever come of these extra documents people were requesting?  I have the Asterisk document but would love to see the UC500 or CUCM integration document.

Cisco Employee

Hi Patrick,

There is no special configuration. The SPA8800's relays are normally closed so when power is applied, the relays open. When power fails or is removed, the relays close thus directly connection the Phone1 phone with the LINE1 PSTN line.

Regards,

Patrick

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Cisco Employee

Hi mreed,

No. I checked about two weeks ago, late Sept 2010 and found that the documents were never written. If you have something, or know someone who has something documented, please feel free to share with the community.

Regards,

Patrick

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New Member

I'm working on getting it working with CUCM right now.  I'll post something if I get it.

New Member

I'm about to attempt getting one to work with CUCM.  Did anyone have any luck?

New Member

I did get the FXO ports to work with CUCM.  I only use it for outbound dialing on the FXO ports though. Here are some notes I started for configuring the device.

The real trick is setting up a new SIP Security Profile in CUCM.  Create a new profile and set the Outgoing Transport Type to UDP.  Then select that when creating the SIP trunk for the 8800.  You have to create one Trunk per FXO port.  You set the UDP port to match the one specified for each FXO port on the 8800, ie 5061, 5062,5063,5064  

Connect to PC AUX port

Browse to http://192.168.0.1/admin/advanced

Click Wan Status

Set Static IP

Connect device to correct subnet using the Ethernet port

Click Voice Tab - Click System Set Password

Voice Line Tab

Set Proxy address to voice server

Use Outbound Proxy - No

New Member

Hi

I have SPA8800 routing 2 PSTN lines to our Asterisk Server. The PSTN lines are configured as Incomming Trunks in Asterisk and all functions work as expected. My problem is consistently poor voice quality on the outgoing voice only. i.e. When we take an inbound call from the PSTN,  we hear good quality voice, but the external caller complains of poor quality describing the sound as low volume and clipped. Can you advise any settings that I should look at to address this?

Thanks

David

New Member

Hello,

We have SPA8800 configured according document: I have 2 questions:

1) (problem): Calls from anyone using a SIP Soft phone on a PC can route calls to the PSTN on the FXO ports: Is there a posibility to restrict only to Ip form Asterisk server? I tried registering, but this results dificult on my Asterisk, as I need then to have host=dynamic in the peer definition in the sip.conf and in my particular situation results imposible...

In the SPA8800 SIP settings on the line tab, I see a YES/NO value with name "Restrict Source IP" what is this and where do I put the IP to restict to?

2) (curiosity): in the DP for the FXO, we use (SO:nnnnnnnn@192.168.x.x:5060>) .... what stands the SO for, and what other options are there?

thanks,

Sven

Silver

Hi Sven,

Since you have question instead of a comment I suggest you post this question in the discussion area located here for help: https://supportforums.cisco.com/community/netpro/small-business/voiceandconferencing/ata

Regards,
Cindy Toy
Cisco Small Business Community Manager
for Cisco Small Business Products
www.cisco.com/go/smallbizsupport

Silver

Hi David,

Since you have question instead of a comment I suggest you post this  question in the discussion area located here for help:  https://supportforums.cisco.com/community/netpro/small-business/voiceandconferencing/ata

Regards,

Cindy Toy

Cisco Small Business Community Manager

for Cisco Small Business Products

www.cisco.com/go/smallbizsupport

New Member

Hello Patrick.

I have a SPA8800 connected to two PSTN lines and two analog phones and all of them register as trunks and extensions on a asterisk server, where many other IP phones also connect. The SPA works as a gateway between the internal VoIP phones and the PSTN.

Last year, there was a power failure and we still could make and receive calls to and from the PSTN. The idea of the two analog phones was to provide communications when out of VoIP.

However, when the sip proxy is unreachable (on the same LAN, but out of service), I noticed that I can make calls from the analog phones to the pstn (Auto PSTN fallback set to YES). However, on the same circunstance, incoming calls do not ring on the analog phones.

How can I configure the box in order to make incoming PSTN calls answered by the associated analog phone when the sip proxy is unavailable?

Thanks. Regards,

Fernando

Silver

Hi Patrick,

Since you have question instead of a comment I suggest you post this question in the discussion area located here for help:  https://supportforums.cisco.com/community/netpro/small-business/voiceandconferencing/ata

The discussion area has more moderators online that may be able to help answer your question.

Regards,

Cindy Toy

Cisco Small Business Community Manager

for Cisco Small Business Products

www.cisco.com/go/smallbizsupport

New Member

Hi Patrick.

Im new here.

Right now i just set up my IP PBX using AsteriskNOW.

Can you help me how to configure my PBX to make call to PSTN from IP Phone.

 

VIP Gold

Thos is forum related to Cisco products, not the Asterisk. So it would be better you will ask elswhere - if I remember correctly, there are comunity forums on AsteriskNOW vendor site. Even in the case you have a question related to a Cisco product, you should start new discussion, not to comment existing document with question unrelated to it.

Thank you.

 

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