Resolve a call routing issue with an Unsupported community Asterisks (trixbox ce) system
$25-50 USD / hour
完了済み
投稿日: 8年以上前
$25-50 USD / hour
Resolve issue where "Anyone calling the extension in question receives a 'not in service' recording. The phone will not ring, but will register a missed call. Outgoing calls work fine." All other phone and lines seem normal. We can provided a screen sharing session with remote web access to Asterisks system as well as network information. Network is flat with nothing but switches between phone and Asterisk system. System is supporting 70 users and probably ~8 years old.
This is a community Asterisk system so does not have any vendor support.
System Status Version: 2.6.1.3
Active Channels: SIP: 5
Efforts by our existing engineer to date but have not resulted in a solution:
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- Previous efforts on 8/21 (reset phone and rebuild tftp boot file) recorded in 234543
- PBX Status console shows x1168 registered and online at [login to view URL], which matches the configured device for TFTP boot file, no status differences from other extensions and devices
- Verify Asterisk/Trixbox configuration for Extension, Inbound Route, TFTP boot settings specific to phone/extension, no issues noted, effectively identical to numerous other working extensions/phones except for extension number and display names
- Examine all custom extension, follow-me, ZAP channel DID, Day/Night control, IVR, Queue, Ring Group, Time Condition/Group, Conference and general settings, no references to x1168 found
- Examine SIP and Extensions configuration files and compare to other extensions, x1168 is essentially identical to all other extensions
- Examine phone web interface and compare to other working extensions, no differences found except display name/extension #
- Contact XO Communications to determine if this particular extension is bound to POTS line or otherwise different from other extensions in the SIP block, awaiting callback for XO ticket
Good day! I work with the asterisk and PBX systems over 8 years.
Also have expirience in Freeswitch
I created a lot of call centers that are working successfully for several years.
I have my own call center and my own server on which i install the hybrid Vicidial\Goautodial and FreePBX\a2billing. It is configured and ready to go.I can show you it to the test.
I integrate Vtiger\Sugar CRM etc.
I have experience in supporting operators and customer of call center.
I managed call center for more than 3 years.
My last job was: Creating Survey Press 1 compaign for a large call center with simultaneous incoming and outgoing calls, and complex IVR.
I have access to the largest US wholesale VOIP provider.
I think I can provide you with good and quality services.
Hi,
I'ev been working with cisco VoIP since 2001 and with asterisk/Trixbox since 2006,
these days Elastix have taken TrixBox's place as free and GUI based asterisk, so major deployments these days are based on Elastix, but have very good experience with TrixBox CE.
I'm very new to Freelancer so trying to keep the prices as low as possible and would like to earn better reviews more then more cash, means you may get the best service for your money.
Final solutions can only be given only after having proper look at your system, but even if you decide not to grant the project to me try checking the routing as well as open/block ports to/from the extension.
Hope that would help.