When I try that number I get a message on the device: "Connection time-out"
I get the same message for other local numbers also. Message: 13 Date: Tue, 4 Aug 2009 16:22:11 -0500 From: "Danny Nicholas" <da...@debsinc.com> Subject: Re: [asterisk-users] Calling issue for non-extension numbers To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <68dc215933a34b169ff8c21f01596...@db0002> Content-Type: text/plain; charset="us-ascii" It is probably a dialplan or timeout issue. What happens if you do 80055511212# ? _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kayton Sapale Sent: Tuesday, August 04, 2009 4:13 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Calling issue for non-extension numbers Hi all, Thanks to the previous replies that helped me with this before, but I got side-tracked in the middle of trying to figure this out, so apologies for posting the same issue. I use a Nokia e71, with an asterisk server and am having an issue dialing certain numbers. When I attempt to dial a local number, like xxx-xxx-xxxx, I cannot connect. What shows in the asterisk debug is the following: Found peer '104' However, if I try to dial an extension that is configured on the asterisk server, the call goes through fine. When I use another device to connect the server (another nokia actually) and dial a local number like xxx-xxx-xxxx, I see this in the debug dialog: Found peer '103' Found RTP audio format 96 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 98 Found RTP audio format 13 Peer audio RTP is at port 192.168.111.183:49152 Found unknown media description format AMR for ID 96 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format iLBC for ID 97 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 98 Found audio description format CN for ID 13 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.111.183:49152 Looking for 6789940793 in DLPN_Free_Outbound (domain sip.speartek.com) list_route: hop: <mailto:sip:1...@192.168.111.183> <sip:1...@192.168.111.183> It appears that my device cannot connect to the server when dialing certain numbers. Does anyone have any idea about this? Thanks, Kayton _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users