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 Asterisk config overload... Technical support, how-to guides, troubleshooting, and general assistance, from beginner to seasoned pro, this is where to discuss Asterisk, the most powerful open source PBX. |
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December 27th, 2006, 09:16 PM
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 Re: Asterisk config overload... open Asterisk CLI when trying to dial from analog phone and post output. also check if your analog line is registered with the server by typing: sip show peers in CLI.     |
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December 27th, 2006, 09:40 PM
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| Re: Asterisk config overload... Calling my cell from the analog phone(rings once on analog, then switches to busy (no ring on cell)): *CLI> -- Executing Dial("SIP/3000-08972000", "SIP/MYCELL#@pstn-spa3k|60|") in new stack -- Called MYCELL#@pstn-spa3k -- SIP/pstn-spa3k-0897f000 is ringing -- SIP/pstn-spa3k-0897f000 answered SIP/3000-08972000 -- Attempting native bridge of SIP/3000-08972000 and SIP/pstn-spa3k-0897f000 == Spawn extension (home, MYCELL#, 1) exited non-zero on 'SIP/3000-08972000' pstn-spa3k/asterisk 10.1.100.102 5061 Unmonitored 4000 (Unspecified) D 0 Unmonitored 3000/3000 10.1.100.102 D 5060 Unmonitored phone01/anyname (Unspecified) D 0 Unmonitored fwd-outgoing/MYFWD# 69.90.155.70 5060 Unmonitored 5 sip peers [5 online , 0 offline] *shrug* |
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