Re: [asterisk-users] All trunk are busy please try your call again later

[dave cantera]

lolu, while you are making the call., capture and post your CLI> output ...  this is easy to do since you are using putty. login to your pbx and start asterisk, use the below command: # asterisk -vvvr then make the call.  hilite the text on the putty terminal and paste it into the body of the email to the list...  sorry if I'm making these instruction too basic... pbv01*CLI>     -- Executing [EMAIL PROTECTED]:1] Wait("SIP/202-b753da18", "1") in new stack     -- Executing [EMAIL PROTECTED]:2] Answer("SIP/202-b753da18", "") in new stack     -- Executing [EMAIL PROTECTED]:3] NoOp("SIP/202-b753da18", "DEBUG: CALLERID=") in new stack     -- Executing [EMAIL PROTECTED]:4] Notify("SIP/202-b753da18", "8000000202|x202|300/192.168.15.100") in new stack     -- Notify: sending '8000000202|x202|300' to 192.168.15.100:40000     -- Executing [EMAIL PROTECTED]:5] AGI("SIP/202-b753da18", "agi-callpop4.sh||red") in new stack     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-callpop4.sh     -- AGI Script agi-callpop4.sh completed, returning 0     -- Executing [EMAIL PROTECTED]:6] NoOp("SIP/202-b753da18", "AGISTATUS is >FAILURE<") in new stack     -- Executing [EMAIL PROTECTED]:7] NoOp("SIP/202-b753da18", "DEBUG: EXTEN=300") in new stack     -- Executing [EMAIL PROTECTED]:8] Dial("SIP/202-b753da18", "SIP/300|15|rt") in new stack     -- Called 300     -- SIP/300-09e062e8 is ringing   == Spawn extension (local-sip, 300, 8) exited non-zero on 'SIP/202-b753da18' daveC Lolu Gbenga wrote: Thanks Please am using putty to again access to my Linux asterisk box. How can i use tcpdump to get your request on the exact Ethernet port and port number. I will appreciate your  reply. Best Regards On Dec 18, 2007 4:02 PM, dave cantera <[EMAIL PROTECTED]> wrote: lolu, sounds more like a telco/itsp problem then *. I would    tcpdump -i eth0 port 5060 to make sure it is actually going out... change 5060 if you have changed your port to your itsp, of course. to see what is going on as well as the other debugging notes mentioned in this thread. daveC Lolu Gbenga wrote: > Good Day all > > Please I am having some issues on my voip asterisk server > > I make internal calls on extensions configured ie extension 192 can > call extension 195 etc > > But each time i try to make calls outside the extension ie calling a > GSM or an external line ,i always hear this response "all trunk calls > are busy please try your call again later" > > Please how can i resolve this problem . > > I will appreciate your response. > > Best Regards > > Success > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >    http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.2 - Release Date: 12/14/2007 12:00 AM -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894

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