Alan, Thank you very much for your response. I just tried your suggestion and it doesn't seem to have helped. Now not only the extensions, but 0, 100, & 101 go straight to a busy signal as well. I have tried using various dial plans with varying degrees of success, but none perfect. I could be looking at things wrong, but it looks to me like the dial plan in sipX for a SPA-2102 (listed as Linksys Ata2102 in sipX) is the exact same as the default or factory dial plan. Also (I don't know if this helps or not) I can dial 8xxx, meaning 8 plus the extension. Now that doesn't ring for any device, it just goes straight to the prompt "The owner of extension X is not available." I am assuming that is normal behavior, or is it? What are you supposed to hear when you dial 8 plus an extension? Again, I really appreciate your help, this has been very frustrating. Cheers, Clint
_____ From: [email protected] [mailto:[email protected]] On Behalf Of [email protected] Sent: Monday, May 04, 2009 7:25 PM To: [email protected] Subject: Re: [sipx-users] Can not dial extensions via LinkSys SPA-2102or LinkSys SPA-3102 Have you tried to drop the dial plans from within the SPA's and have them use the sipX dial plan, as the SPA linksys ATA/routers will use their default dial plan if not told otherwise. Alan Quoting Clint Anderson <[email protected]>: > Problem Summary: Can not dial extensions phones connected via LinkSys > SPA-2102 or LinkSys SPA-3102 > > First: I apologize in advance if I missed the answer to this somewhere, but > I haven't been able to find anything anywhere that has worked. Also, I > tried to post properly and give enough information without going overboard, > I am sorry if it is still too much (or too little). > > > Configuration: > . Fresh install of Fedora 8 and sipXecs 4.0 > . sipXconfig (4.0.0-015321 2009-04-28T13:02:58 ecs-fc8) > . Latest firmware on both LinkSys SPA adapters. > > . SPA-3102 > Software Version: 5.1.10(GW) > Hardware Version: 1.4.5(a) > Connections: > Network: Internal private dedicated VoIP network > PSTN Line: Local POTS Line (Ext. 214) > Line 1: Southwestern Bell corded phone (Ext. 215) > > . SPA-2102 > Software Version: 5.2.5 > Hardware Version: 1.3.5(a) > Connections: > Network: Internal private dedicated VoIP network > Line 1: Phillips CD 440 DECT 6.0 Cordless (Ext. 220) > Line 2: GE Slimline corded phone (Ext. 221) > > . Polycom SoundPoint 320 > Connections: > Network: Internal private dedicated VoIP network > Ext. 250 > > . X-Lite CounterPath SoftPhone > Version: 3.0 Build 47546 > Connections: > Network: Internal private Non-VoIP network > Ext. 260 > > . sipXecs > Users setup for each extension, User ID 215, User ID 220, etc. > Phones / adapters setup and assigned to users > All phones, adapters, devices, etc. are registered and showing under > "Diagnostics > Registrations", no issues or errors. > > . Me: Experienced in computers (more Windows based then Linux) but no > previous experience with VoIP, sip, sipXecs, etc. > > > Problem Details: > ------------------------------------------------------------ >> From Polycom (Ext. 250) I can direct dial any extension (e.g. press 215, > 220, 260 etc.) and it rings straight through to the extension, including any > phone connected to either LinkSys SPA, no problems. > >> From X-Lite SoftPhone (Ext. 260) I can direct dial any extension (e.g. press > 215, 220, 250, etc.) and it rings straight through to the extension, > including any phone connected to either LinkSys SPA, no problems. > >> From anything connected to an SPA, corded phones on Ext. 215 & 221 or the > cordless on Ext. 220, I can dial 0, 100, & 101 without any problems. I can > dial 100 and dial an extension (e.g. press 100, wait for the attendant to > answer, and then press 250) and get transferred and it rings through to the > extension, no problems. But, if I try to direct dial any extension (e.g. > press 220), all I get is a busy signal. > > Bottom line, from the Polycom and the softphone, I can direct dial any > extension. From any phone connected through an SPA, I can not direct dial > any extension. > > > Things I have noticed: > ------------------------------------------------------------ > . I can get out to my POTS line via the PSTN Line on the SPA-3102 from the > Polycom and the soft phone. > . Calls that fail from the SPAs show up under "Diagnostics > Call Detail > Records > Historic" with a Duration of "0 seconds" and a Status of "Failed" > > > Things I have tried: > ------------------------------------------------------------ > . User resets on the SPAs > . Factory resets on the SPAs > . Increasing the DTMF Playback level on the SPAs > . Increasing the DTMF Playback length on the SPAs > . Reduced the RTP Packet Size to 0.020, and also tried 0.010 on the SPAs > . Under the "SIP Timer Values" on the SPAs, change the following: > SIP T1: .7 > Reg Max Expires: 1200 > Reg Retry Intvl: 30 > Reg Retry Long Intvl: 110 > > . Updating the firmware on the SPAs > . Reinstalling Fedora & sipXecs > . Digitmaps I have tried on the SPAs: > Original: (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) >> From the Polycom: > ([2-9]11|0T|100|101|011xxx.T|9011xxx.T|1[2-9]xxxxxxxxx|91[2-9]xxxxxxxxx|9[2- > 9]xxxxxx|*xx|[8]xxx|[2-7]xx) > Bare bones: (0|100|101|260|[2-7]xx|[8]xxx) > Other variations of desperation (and some others not listed): > ([1-8]xxS0|9[2-9]xxxxxx|91[2-9]xx[2-9]xxxxxx|9[69]11S0) > ([2-9]11S0|[8]xxx|[2-7]xx|0S0|100|101|011xxx.T|9011xxx.T|1[2-9]xxxxxxxxx|91[ > 2-9]xxxxxxxxx|9[2-9]xxxxxx|*xx) > (*xx|[3469]11S0|[8]xxxS0|[2-7]xxS0|0S0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxx > xxxxxxxx.) > > > > Any suggestions would be greatly appreciated and thank you all very much for > your help and your time. > > Cheers, > Clint > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users <http://webmail.slingshot.co.nz/services/go.php?url=http%3A%2F%2Flist.sipfou ndry.org%2Farchive%2Fsipx-users> > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users <http://webmail.slingshot.co.nz/services/go.php?url=http%3A%2F%2Flist.sipfou ndry.org%2Fmailman%2Flistinfo%2Fsipx-users>
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