Topic: Linksys SPA-3102 Outbound

[Teviot]

Hi S & All

I have a outbound issue that I'm nor sure how to fix.  I have check everything that I can think of and still can't get an outbound number to dial and connect. Can anyone remind me what I have forgotten?

Code: [Select]

pbx*CLI>
    -- Executing [1300xxxxxx@internal:1] AGI("SIP/201-09e44258", "selintra|OutCluster|1300xxxxxx") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing [1300xxxxxx@qrxvtmny:1] AGI("SIP/201-09e44258", "selintra|OutRoute|13 Numbers") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Dial) Options: (SIP/1300xxxxxx@telstra||T)
    -- Called 1300xxxxxx@telstra
[May  6 10:21:28] WARNING[5134]: chan_sip.c:12334 handle_response_invite: Received response: "Forbidden" from '"ext201" <sip:201@192.168.1.1>;tag=as61e817de'
    -- SIP/telstra-09e424f0 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- AGI Script Executing Application: (Background) Options: (were-sorry)
[May  6 10:21:28] WARNING[1948]: file.c:602 ast_openstream_full: File were-sorry does not exist in any format
[May  6 10:21:28] WARNING[1948]: file.c:912 ast_streamfile: Unable to open were-sorry (format 0x100 (g729)): No such file or directory
[May  6 10:21:28] WARNING[1948]: pbx.c:5722 pbx_builtin_background: ast_streamfile failed on SIP/201-09e44258 for were-sorry
    -- AGI Script Executing Application: (Background) Options: (call-cannot-complete)
[May  6 10:21:28] WARNING[1948]: file.c:602 ast_openstream_full: File call-cannot-complete does not exist in any format
[May  6 10:21:28] WARNING[1948]: file.c:912 ast_streamfile: Unable to open call-cannot-complete (format 0x100 (g729)): No such file or directory
[May  6 10:21:28] WARNING[1948]: pbx.c:5722 pbx_builtin_background: ast_streamfile failed on SIP/201-09e44258 for call-cannot-complete
    -- AGI Script Executing Application: (Background) Options: (please-hang-up-and-try-again)
[May  6 10:21:28] WARNING[1948]: file.c:602 ast_openstream_full: File please-hang-up-and-try-again does not exist in any format
[May  6 10:21:28] WARNING[1948]: file.c:912 ast_streamfile: Unable to open please-hang-up-and-try-again (format 0x100 (g729)): No such file or directory
[May  6 10:21:28] WARNING[1948]: pbx.c:5722 pbx_builtin_background: ast_streamfile failed on SIP/201-09e44258 for please-hang-up-and-try-again
    -- AGI Script selintra completed, returning 0
  == Auto fallthrough, channel 'SIP/201-09e44258' status is 'CONGESTION'
    -- Executing [h@qrxvtmny:1] Hangup("SIP/201-09e44258", "") in new stack
  == Spawn extension (qrxvtmny, h, 1) exited non-zero on 'SIP/201-09e44258'

I know that there is something I have missed but can't think of it.

[SARK devs]

OK...

Before we get t9 the call fail, you are getting errors because you don't have Asterisk additional sounds installed.  When a call fails, SAIL trys to play an autoattendent message but the files aren't installed on your system.  You can eithetr install them or set the "Play tone on..." switches in Globals panel to YES.  This will force SAIL to play a congestion tone instead of a message.

Your call is being sent to Telstra but Telstra us bouncing it with "Forbidden" (usually a SIP 403 response).   This means that Telstra doesn't like your request for some reason.  You might want to check the SIP settings in the trunk and make sure that they are the same as Telstra recommends.  I also see that you have used a route name with a space in it.   This is usually not allowed.  You seem to have gotten away with it but it isn't good practice and will be forbidden in V3.0

Kind Regards

S 

[Teviot]

I have 2 PBX's operating.  Both have the same infomation in the trunk "telstra".  The only difference that I can see is 2 different locations and the IP addresses in the config of the trunk. And of course location "A" works and location "B" (the one I'm working on) does not work.

This is the config for Location "B"

[telstra]

Code: [Select]

type=peer
host=dynamic
qualify=3000
canreinvite=no
username=telstra
secret=asterisk
disallow=all
allow=g729
allow=alaw
allow=ulaw
[telstra]

Code: [Select]

type=user
context=mainmenu
[provisioning]

Code: [Select]

["spa$MAC.cfg"
<Proxy_2_> 192.168.1.1
</Proxy_2_>
<Outbound_Proxy_2_> 192.168.1.1
</Outbound_Proxy_2_>
<User_ID_2_> $didsip
</User_ID_2_>
<Password_2_> password
</Password_2_>
<Display_Name_2_>
</Display_Name_2_>
<Dial_Plan_2_2_> (S0&lt;:$didsip&gt;)
</Dial_Plan_2_2_>
<FAX_Passthru_Method_2_> None
</FAX_Passthru_Method_2_>
<Line_1_VoIP_Caller_DP_2_> none
</Line_1_VoIP_Caller_DP_2_>
<VoIP_Caller_1_DP_2_> 2
</VoIP_Caller_1_DP_2_>
<PSTN_Ring_Thru_Line_1_2_> no
</PSTN_Ring_Thru_Line_1_2_>
<PSTN_CID_For_VoIP_CID_2_> yes
</PSTN_CID_For_VoIP_CID_2_>
<PSTN_Caller_Default_DP_2_> 2
</PSTN_Caller_Default_DP_2_>
<PSTN_Caller_1_DP_2_> 2
</PSTN_Caller_1_DP_2_>
<PSTN_Answer_Delay_2_> 2
</PSTN_Answer_Delay_2_>
<Min_CPC_Duration_2_> 0.09
</Min_CPC_Duration_2_>
<FXO_Port_Impedance_2_> Global
</FXO_Port_Impedance_2_>
<SPA_To_PSTN_Gain_2_> 3
</SPA_To_PSTN_Gain_2_>
<PSTN_To_SPA_Gain_2_> 5
</PSTN_To_SPA_Gain_2_>
<On-Hook_Speed_2_> 3 ms (ETSI)
</On-Hook_Speed_2_>
<Ring1_Cadence> 60(.4/.2,.4/2)
</Ring1_Cadence>
<Ring2_Cadence> 60(1/2)
</Ring2_Cadence>
<Ring3_Cadence> 60(.25/.25,.25/.25,.25/1.75)
</Ring3_Cadence>
<Ring4_Cadence> 60(.4/.8)
</Ring4_Cadence>
<Ring5_Cadence> 60(2/4)
</Ring5_Cadence>
<FXS_Port_Input_Gain> 0
</FXS_Port_Input_Gain>
<Caller_ID_Method> Bellcore(N.Amer,China)
</Caller_ID_Method>
<FXS_Port_Impedance> 220+820||120nF
</FXS_Port_Impedance>
<FXS_Port_Output_Gain> 7
</FXS_Port_Output_Gain>
<Time_Zone> GMT+10
</Time_Zone>

[PWDasterisk]

I don't know if this is relevant to your situation but I had two SIP trunks stop working earlier this week after doing a software update. They were previously working and all the trunk data was correct in the SAIL interface. The debug messages showed data in the SDP fields that didn't match what was in the trunk stanzas. I had to delete the trunks, do a commit, then did a signal-event post-upgrade; signal-event reboot to rewrite the update again, then recreated the trunks using the same info, did a commit and they worked fine...

[Teviot]

Thanks PWDasterisk

Will give that a try next week as I have a full skedual for the weekend