Sebastian Milioto
2010-Mar-19 14:50 UTC
[asterisk-users] SPA3102 + asterisk drop call and loop (was SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?) )
Ok, I downgraded spa3102 to 3.3.6. Now when I make a call from pstn and call is established asterisk seems to drop the call. However I still hearing ringback on pstn side, call is established again, and asterisk drops the call again, like a loop. -- Executing [preat_admin at nodo:1] Playback("SIP/PSTN-08214948", "horario-atencion/our-business-hours-are") in new stack -- <SIP/PSTN-08214948> Playing 'horario-atencion/our-business-hours-are' (language 'es') == Spawn extension (nodo, preat_admin, 1) exited non-zero on 'SIP/PSTN-08214948' -- Executing [preat_admin at nodo:1] Playback("SIP/PSTN-08214948", "horario-atencion/our-business-hours-are") in new stack -- <SIP/PSTN-08214948> Playing 'horario-atencion/our-business-hours-are' (language 'es') == Spawn extension (nodo, preat_admin, 1) exited non-zero on 'SIP/PSTN-08214948' -- Executing [preat_admin at nodo:1] Playback("SIP/PSTN-08214948", "horario-atencion/our-business-hours-are") in new stack -- <SIP/PSTN-08214948> Playing 'horario-atencion/our-business-hours-are' (language 'es') == Spawn extension (nodo, preat_admin, 1) exited non-zero on 'SIP/PSTN-08214948' I've read this had happen to other people, however I can't find how they solved it. It seems to be a codec problem.. however I've already tried configuring g729a,g711u, and g711a in spa3102 with no success.. Can anybody help me with that, please? Sebastian On Fri, Mar 19, 2010 at 10:16 AM, Sebastian Milioto <smilioto at gmail.com>wrote:> Thanks! > > > On Thu, Mar 18, 2010 at 5:04 PM, Joseph <syscon780 at gmail.com> wrote: > >> On 03/18/10 16:22, Sebastian Milioto wrote: >> >Somebody has 5.1.7 firmware for SPA3102? >> >I'm having issues with inbound/outbound calls using asterisk through >> SPA3102 >> >with firmware 5.1.10. I've read it has a codec bug, since it doesn't care >> >about what you set up in Preferred Codec. >> > >> >Any help will be appreciated. >> > >> >Sebastian >> >> You will find it here: >> http://prov.802.cz/fw/ >> >> Ever since the Linksys took over from Sipura and now by Cisco, thoese >> devices are of very poor quality. >> Two of SPA3102 died on me within two years, in addition lots of echo >> impossible to eliminate. >> >> I've switched/replaced my Linksys/Sipura units with Audiocodes MP-114 but >> they are not perfect either. >> Though, I can say they don't have/generate any echo problems and fixes go >> through without any problem (which I can not say the same about >> Linksys/Sipura >> units.) >> >> -- >> Joseph >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100319/20a1efe1/attachment-0001.htm