Thanks steve for your response
the details is below When i call from iax extension (1018) to sip extension there is no issue == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4-r110474M-AheevaCCS-3.1.0_Build-11629 currently running on srvradio (pid = 24818) Verbosity is at least 3 -- Accepting UNAUTHENTICATED call from 192.168.5.131: > requested format = ulaw, > requested prefs = (), > actual format = ulaw, > host prefs = (alaw|ulaw), > priority = mine -- Executing [MCALL106^1298455141.287500@agents:1] Set("IAX2/1018-6", "AH_TEMP=106^1298455141.287500") in new stack -- Executing [MCALL106^1298455141.287500@agents:2] NoOp("IAX2/1018-6", "[106^1298455141.287500]") in new stack -- Executing [MCALL106^1298455141.287500@agents:3] Set("IAX2/1018-6", "AH_EXTEN=106") in new stack -- Executing [MCALL106^1298455141.287500@agents:4] Set("IAX2/1018-6", "AHEEVA_TRACKNUM=1298455141.287500") in new stack -- Executing [MCALL106^1298455141.287500@agents:5] Goto("IAX2/1018-6", "agents|106|1") in new stack -- Goto (agents,106,1) -- Executing [106@agents:1] Dial("IAX2/1018-6", "SIP/106") in new stack -- Called 106 -- SIP/106-095133e8 is ringing -- SIP/106-095133e8 answered IAX2/1018-6 == Agent '1018' logged out == Spawn extension (agents, AH1018, 1) exited non-zero on 'IAX2/1018-4' == Spawn extension (agents, 106, 1) exited non-zero on 'IAX2/1018-6' -- Executing [h@agents:1] GotoIf("IAX2/1018-4", "0?3:2") in new stack -- Executing [h@agents:1] GotoIf("IAX2/1018-6", "1?3:2") in new stack -- Goto (agents,h,2) -- Executing [h@agents:2] AHEventsProxy("IAX2/1018-4", "MSG_TYPE_TERMINATE_CALL::::1298455155") in new stack AHEventsProxy: Channel [IAX2/1018-4]. Data [MSG_TYPE_TERMINATE_CALL::::1298455155] -- chan is IAX2/1018-4 AHEventsProxy: Send To CtiServer: socket:[67]. message:[41,1298455155^^^^Ipbx01^~] -- Executing [h@agents:3] Hangup("IAX2/1018-4", "") in new stack == Spawn extension (agents, h, 3) exited non-zero on 'IAX2/1018-4' -- Hungup 'IAX2/1018-4' -- Goto (agents,h,3) -- Executing [h@agents:3] Hangup("IAX2/1018-6", "") in new stack == Spawn extension (agents, h, 3) exited non-zero on 'IAX2/1018-6' -- Hungup 'IAX2/1018-6' -- Accepting UNAUTHENTICATED call from 192.168.5.131: > requested format = ulaw, > requested prefs = (), > actual format = ulaw, > host prefs = (alaw|ulaw), > priority = mine -- Executing [AH1018@agents:1] AgentLogin("IAX2/1018-9", "1018|s") in new stack -- Started music on hold, class 'none', on channel 'IAX2/1018-9' == Agent '1018' logged in (format ulaw/slin) -- Stopped music on hold on IAX2/1018-9 [Feb 23 09:59:22] NOTICE[25420]: chan_sip.c:15012 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 106 srvradio*CLI> but when i call from sip extension 106 to iax extension (1018) i got the message below == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4-r110474M-AheevaCCS-3.1.0_Build-11629 currently running on srvradio (pid = 24818) Verbosity is at least 3 [Feb 23 09:55:49] NOTICE[25420]: chan_sip.c:13952 handle_request_invite: Call from '106' to extension '1018' rejected because extension not found. srvradio*CLI> thank you for your help 2011/2/22 Danny Nicholas <da...@debsinc.com> > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve > Edwards > Sent: Tuesday, February 22, 2011 12:33 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] calls between iax and sip > > On Tue, 22 Feb 2011, salaheddine elharit wrote: > > > i have asterisk installed and i have configured a client iax and sip > > without any issue, when i call a internal extension sip from iax there > > is no problem > > > > but when i call a iax extension from sip extension the result is > > KO(wrong number) > > > > any help please > > No details, no help. > > Crank up verbosity on the CLI and see if the messages yield a clue. If > not, please post the console messages. > > Isn't Dionne Warrick a poster on this list? :) > > > -- > _____________________________________________________________________ > -- 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 >
-- _____________________________________________________________________ -- 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