Posting a sip debug will probably be helpfull aswell as you can see exactly where the traffic is being sent and what the response was.
Andy Beak wrote: > Hi, > > Thanks, I added that. I'll ask my network provider if they received > these message tomorrow morning. That will narrow things down to either > an Asterisk configuration or a network routing issue. > > There is not really a caller, I'm trying to use Asterisk as an Automated > Voice Message server to dial phone numbers and play an mp3. > > I'm using my mobile phone to test on and it doesn't ring. Asterisk > gives the following message immediately after reading the .call file > from the spool directory: > > -- Attempting call on SIP/MTN-NEW/mynumber for application > MP3Player(/myfile) (Retry 1) > == Using SIP RTP CoS mark 5 > > Channel SIP/MTN-NEW-00000001 was never answered. > [Jul 20 18:07:37] NOTICE[22259]: pbx_spool.c:339 attempt_thread: Call > failed to go through, reason (8) Congestion (circuits busy) > > Because the phone doesn't ring and the error message appears immediately > I don't think it's a timeout issue. > > Will reading the source for pbx_spool.c at line 339 give any clues as to > what's happening or will that be a waste of time? > > Cheers, > Andy > > > On 20/07/2010 05:42 PM, Gareth Blades wrote: >> If you add qualify=yes to the setting in sip.conf it will send a sip >> message to the peer every 60 seconds to check if it is alive. >> If you try to make a call while the peer is not alive it will fail >> immediatly rather than the caller hearing silence while your box waits >> for a reply timeout. >> >> Andy Beak wrote: >> >>> Hi, >>> >>> No that is the correct address. I know it is an internal IP. >>> >>> We have our machine hosted in racks at our SIP providers data center. >>> >>> They've patched a new port to our cabinet and linked that to a gateway >>> (172.28.20.105). >>> >>> As long as we use that gateway (and the IP address they assigned to us) >>> our traffic will reach their SBC. >>> >>> I've confirmed that traceroute follows the path it is supposed to: >>> >>> traceroute to 192.168.34.1 (192.168.34.1), 30 hops max, 60 byte packets >>> 1 192.168.0.1 (192.168.0.1) 0.656 ms 0.562 ms 0.501 ms >>> 2 172.28.20.105 (172.28.20.105) 1.211 ms 1.209 ms 1.196 ms >>> 3 192.168.34.5 (192.168.34.5) 23.270 ms 23.269 ms 23.328 ms >>> 4 * * * >>> 5 * * * >>> 6 * * *^C >>> >>> Is there a way to test in Asterisk if it is able to reach a particular >>> IP address? Do you think that there is a routing problem here? >>> >>> Thanks, >>> Andy >>> >>> >>> >>> >>> On 20/07/2010 04:58 PM, Zeeshan Zakaria wrote: >>> >>>> This "host=192.168.34.1" is where you'll put your provider's IP >>>> address. Currently you are using some local address which is not your >>>> provider's IP address. Where did you get it from? Call your providrr >>>> and ask them the IP address of the server where you'll be sending your >>>> calls. >>>> >>>> Zeeshan A Zakaria >>>> >>>> -- >>>> www.ilovetovoip.com<http://www.ilovetovoip.com> >>>> >>>> >>>>> On 2010-07-20 10:27 AM, "Andy Beak"<[email protected] >>>>> <mailto:[email protected]>> wrote: >>>>> >>>>> Hi, >>>>> >>>>> I set my list to subscribe to digest and I can't see how to reply to >>>>> your reply without starting a new thread. >>>>> >>>>> There is no need for SIP username and password because the provider >>>>> authenticates me on my IP address. >>>>> >>>>> I thought that "host=192.168.34.1" would be the sip provider IP >>>>> address. >>>>> >>>>> At this point I don't need to accept incoming calls or place >>>>> VOIP-to-VOIP. All I need to do is connect to their PBX to place a >>>>> call to a cellphone. >>>>> >>>>> I reread all the documentation I could find and couldn't see where >>>>> else in sip.conf I should set the provider IP. >>>>> >>>>> Thanks for your reply, >>>>> Andy >>>>> >>>>> >>>>> >>>>> >>>>>> In your sip.conf, there is no mention of your sip provider's IP >>>>>> >>>>> address, username and secret (pa... >>>>> >>>>> www.ilovetovoip.com<http://www.ilovetovoip.com> >>>>> <http://www.ilovetovoip.com> >>>>> >>>>> >>>>> >>>>> >>>>>> On 2010-07-20 5:09 AM, "Andy Beak"<andr...@xxxxxxxxxxxxxxx >>>>>> >>>>> <mailto:andr...@xxxxxxxxxxxxxxx<mailto:andr...@xxxxxxxxxxxxxxx>>> >>>>> wr... >>>>> >>>>> -- >>>>> _____________________________________________________________________ >>>>> -- Bandwidth and Colocation Provided byhttp://www.api-digital.com >>>>> <http://www.api-digital.com> -- >>>>> >>>>> >>>>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>>>> http://www.aste... >>>>> >>>>> >>>>> -- >>>>> _____________________________________________________________________ >>>>> -- 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
