SPA3102 as Voip Gateway

Dear friends, I have Freepbx with some voip Trunks but i also have one PSTN line in my office that i also need to connect to my Freepbx (I’m using FreePBX 15.0.16.44 completly updadet). To do this I purchased a Cisco SPA3102 voip gateway. Following this link https://wiki.freepbx.org/pages/viewpage.action?pageId=55476525, i was able to setup both freepbx and SPA3102. Instead of creating a SIP trunk I used the note at the botton of this arcticle using PJSIP Trunk. It’s is working well for receiving calls. When I call my PSTN number, in a few secounds Freepbx receiveis it normaly. I can pick up the call perfectly. But When i try dialing to some number using this trunk I receive the message that “call cannot be completed”). In outboud routes, I have setted up a route (pressing 0) to access my PSTN line. This looks to be no working, once i dial for example 0981533000 (0 as access code and 981533000 - my mobile phone), i get this message and call is not completed.

Here is my CLI log when i’m trying to call (I’m calling from my SPA3102 FXS Port connected to extension 207 at freepbx localy):

– Executing [0981533000@from-internal:1] ResetCDR(“PJSIP/207-00000006”, “”) in new stack
_ – Executing [0981533000@from-internal:2] NoCDR(“PJSIP/207-00000006”, “”) in new stack_
_ – Executing [0981533000@from-internal:3] Progress(“PJSIP/207-00000006”, “”) in new stack_
_ – Executing [0981533000@from-internal:4] Wait(“PJSIP/207-00000006”, “1”) in new stack_
_ > 0x7fe8e0330220 – Strict RTP learning after remote address set to: 192.168.15.235:16482_
_ > 0x7fe8e0330220 – Strict RTP switching to RTP target address 192.168.15.235:16482 as source_
_ – Executing [0981533000@from-internal:5] Playback(“PJSIP/207-00000006”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack_
_ – <PJSIP/207-00000006> Playing ‘silence/1.ulaw’ (language ‘en’)_
_ – <PJSIP/207-00000006> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)_
_ – <PJSIP/207-00000006> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)_
_ > 0x7fe8e0330220 – Strict RTP learning complete - Locking on source address 192.168.15.235:16482_
_ – Executing [0981533000@from-internal:6] Wait(“PJSIP/207-00000006”, “1”) in new stack_
_ – Executing [0981533000@from-internal:7] Congestion(“PJSIP/207-00000006”, “20”) in new stack_
_ == Spawn extension (from-internal, 0981533000, 7) exited non-zero on ‘PJSIP/207-00000006’_
_ – Executing [h@from-internal:1] Macro(“PJSIP/207-00000006”, “hangupcall”) in new stack_
_ – Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/207-00000006”, “1?theend”) in new stack_
_ – Goto (macro-hangupcall,s,3)_
_ – Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/207-00000006”, “0?Set(CDR(recordingfile)=)”) in new stack_
_ – Executing [s@macro-hangupcall:4] NoOp(“PJSIP/207-00000006”, " montior file= ") in new stack_
_ – Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/207-00000006”, “1?skipagi”) in new stack_
_ – Goto (macro-hangupcall,s,7)_
_ – Executing [s@macro-hangupcall:7] Hangup(“PJSIP/207-00000006”, “”) in new stack_
_ == Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/207-00000006’ in macro ‘hangupcall’_
_ == Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/207-00000006’_

Hope someone can help me on how solving it, many thanks!!

You don’t have an outbound route with a pattern match for the number you’re dialing.

You are right! Solved! Many thanks

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