Hello,
I’m testing/evaluating FreePBX Distro 3.211.63-5 (x64) with Asterisk 11 and DAHDI 2.11.14 installed on a bare metal box with HFC-S Cologne Chip based ISDN PCI Card (single port) to connect with an single BRI (not PRI) Euro-ISDN PSTN land-line (configured with PtMP signalling mode by the TELCO’s provider and in DAHDI).
I noticed that all outgoing calls to PSTN (to Mobile or Land-Line numbers) fail.
Here two examples:
The first one shows an Extension (as example: 100) that is trying to call a Mobile Phone number 3994709093. For reference the first 0 digit is used to dial out to PSTN as
set on the Outbound Dial Prefix section under Dialed Number Manipulation Rules section of the only one Trunk available: the g0/DAHDI Trunk the system had.
[font=Courier]
localhostCLI>
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [03994709093@from-internal:1] ResetCDR(“SIP/100-00000005”, “”) in new stack
– Executing [03994709093@from-internal:2] NoCDR(“SIP/100-00000005”, “”) in new stack
– Executing [03994709093@from-internal:3] Progress(“SIP/100-00000005”, “”) in new stack
– Executing [03994709093@from-internal:4] Wait(“SIP/100-00000005”, “1”) in new stack
– Executing [03994709093@from-internal:5] Progress(“SIP/100-00000005”, “”) in new stack
– Executing [03994709093@from-internal:6] Playback(“SIP/100-00000005”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
– <SIP/100-00000005> Playing ‘silence/1.ulaw’ (language ‘en’)
– <SIP/100-00000005> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
– <SIP/100-00000005> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)
– Executing [03994709093@from-internal:7] Wait(“SIP/100-00000005”, “1”) in new stack
– Executing [03994709093@from-internal:8] Congestion(“SIP/100-00000005”, “20”) in new stack
[2013-02-06 15:02:55] WARNING[3064][C-00000005]: channel.c:4810 ast_prod: Prodding channel ‘SIP/100-00000005’ failed
== Spawn extension (from-internal, 03994709093, 8) exited non-zero on ‘SIP/100-00000005’
– Executing [h@from-internal:1] Hangup(“SIP/100-00000005”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000005’
localhostCLI>
[/font]
On the second one the same Extension is calling a PSTN land-line number 0444333695 (in which the part 0444 represents the local Area Code prefix, see here for details).
As above the first 0 digit is used to dial out to PSTN as it was set on the Outbound Dial Prefix section under Dialed Number Manipulation Rules section of the only one Trunk available: the g0/DAHDI Trunk the system had.
So the caller goes out from the FreePBX by calling 00444333695:
[font=Courier]
localhostCLI>
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [00444333695@from-internal:1] ResetCDR(“SIP/100-00000007”, “”) in new stack
– Executing [00444333695@from-internal:2] NoCDR(“SIP/100-00000007”, “”) in new stack
– Executing [00444333695@from-internal:3] Progress(“SIP/100-00000007”, “”) in new stack
– Executing [00444333695@from-internal:4] Wait(“SIP/100-00000007”, “1”) in new stack
– Executing [00444333695@from-internal:5] Progress(“SIP/100-00000007”, “”) in new stack
– Executing [00444333695@from-internal:6] Playback(“SIP/100-00000007”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
– <SIP/100-00000007> Playing ‘silence/1.ulaw’ (language ‘en’)
– <SIP/100-00000007> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
– <SIP/100-00000007> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)
– Executing [00444333695@from-internal:7] Wait(“SIP/100-00000007”, “1”) in new stack
– Executing [00444333695@from-internal:8] Congestion(“SIP/100-00000007”, “20”) in new stack
[2013-02-06 15:05:51] WARNING[3070][C-00000007]: channel.c:4810 ast_prod: Prodding channel ‘SIP/100-00000007’ failed
== Spawn extension (from-internal, 00444333695, 8) exited non-zero on ‘SIP/100-00000007’
– Executing [h@from-internal:1] Hangup(“SIP/100-00000007”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000007’
localhostCLI>
[/font]
It seems that the only one Trunk available is not requested by the Outgoing Route defined (but I defined it to use g0/DAHDI).
I tried a lot of others combinations (prefixing the called number without its 0 digit or addit two 0 digits!) but the errors and logs reported were always exactly the same.
I can post configuration files (Asterisk/DAHDI) if requested.
Best regards, Davide.