Outgoing calls Elastix => OXO Bad Gateway

[rhfn]

Hello!

I tried to setup a SIP-Trunk between Elastix and Alcatel OmniPCX Office (I am new to all this). Calling from Elastix to Alcatel internal subscribers
and from Alcatel to Elastix works, also external incoming calls get routed to the Elastix appliance (via DDI). Talking to each
other also works good. But now I am stuck... got the problem, that when I dial an external number from my SIP-Phone connected to Elastix I can't get out. The error in SIP trace log is like "Got SIP response 502 "Bad Gateway" back from 'Alcatel IP Address'" and "Dial failed for some reason with DIALSTATUS = CONGESTION"HANGUPCAUSE = 27"). My Alcatel OXO Configuration is like the one posted by tot3nkopf, SIP-Extension is 111.

Asterisk Outgoing Trunk Configuration:

Code: Select all

host='Alcatel IP Address'
type=peer
qualify=yes
transfer=yes
dtmf=rfc3325
immediate=no
insecure=very

Full SIP-Trace of an outgoing external Call:

Code: Select all

[May 15 14:00:50] VERBOSE[3845] netsock.c:   == Using SIP RTP TOS bits 184
[May 15 14:00:50] VERBOSE[3845] netsock.c:   == Using SIP RTP CoS mark 5
[May 15 14:00:50] VERBOSE[3845] app_dial.c:     -- Called OXO/00152xxxxx876
[May 15 14:00:50] VERBOSE[3171] chan_sip.c:     -- Got SIP response 502 "Bad Gateway" back from 'Alcatel IP Address'
[May 15 14:00:50] VERBOSE[3845] app_dial.c:     -- SIP/OXO-00000023 is circuit-busy
[May 15 14:00:50] VERBOSE[3845] app_dial.c:   == Everyone is busy/congested at this time (1:0/1/0)
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/111-00000022", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 27") in new stack
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Executing [s@macro-dialout-trunk:21] Goto("SIP/111-00000022", "s-CONGESTION,1") in new stack
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Goto (macro-dialout-trunk,s-CONGESTION,1)
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/111-00000022", "RC=27") in new stack
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/111-00000022", "27,1") in new stack
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Goto (macro-dialout-trunk,27,1)
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Executing [27@macro-dialout-trunk:1] Goto("SIP/111-00000022", "continue,1") in new stack
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Goto (macro-dialout-trunk,continue,1)
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/111-00000022", "0?noreport") in new stack
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Executing [continue@macro-dialout-trunk:2] AGI("SIP/111-00000022", "trunklog") in new stack
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/111-00000022", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 27 - failing through to other trunks") in new stack
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Executing [continue@macro-dialout-trunk:4] Set("SIP/111-00000022", "CALLERID(number)=111") in new stack
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Executing [0015206475846@from-internal:5] Macro("SIP/111-00000022", "outisbusy,") in new stack
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Executing [s@macro-outisbusy:1] Progress("SIP/111-00000022", "") in new stack
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Executing [s@macro-outisbusy:2] GotoIf("SIP/111-00000022", "0?emergency,1") in new stack
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Executing [s@macro-outisbusy:3] GotoIf("SIP/111-00000022", "0?intracompany,1") in new stack
[May 15 14:00:50] VERBOSE[3845] pbx.c:     -- Executing [s@macro-outisbusy:4] Playback("SIP/111-00000022", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
[May 15 14:00:50] VERBOSE[3845] file.c:     -- <SIP/111-00000022> Playing 'all-circuits-busy-now.gsm' (language 'en')
[May 15 14:00:52] VERBOSE[3845] file.c:     -- <SIP/111-00000022> Playing 'pls-try-call-later.gsm' (language 'en')
[May 15 14:00:53] VERBOSE[3171] chan_sip.c:     -- Registered SIP '111' at 10.44.172.109 port 53303
[May 15 14:00:54] VERBOSE[3845] pbx.c:     -- Executing [s@macro-outisbusy:5] Congestion("SIP/111-00000022", "20") in new stack
[May 15 14:00:54] WARNING[3845] channel.c: Prodding channel 'SIP/111-00000022' failed
[May 15 14:00:54] VERBOSE[3845] app_macro.c:   == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/111-00000022' in macro 'outisbusy'
[May 15 14:00:54] VERBOSE[3845] pbx.c:   == Spawn extension (from-internal, 0015206475846, 5) exited non-zero on 'SIP/111-00000022'
[May 15 14:00:54] VERBOSE[3845] pbx.c:     -- Executing [h@from-internal:1] Macro("SIP/111-00000022", "hangupcall") in new stack
[May 15 14:00:54] VERBOSE[3845] pbx.c:     -- Executing [s@macro-hangupcall:1] GotoIf("SIP/111-00000022", "1?skiprg") in new stack
[May 15 14:00:54] VERBOSE[3845] pbx.c:     -- Goto (macro-hangupcall,s,4)
[May 15 14:00:54] VERBOSE[3845] pbx.c:     -- Executing [s@macro-hangupcall:4] GotoIf("SIP/111-00000022", "1?skipblkvm") in new stack
[May 15 14:00:54] VERBOSE[3845] pbx.c:     -- Goto (macro-hangupcall,s,7)
[May 15 14:00:54] VERBOSE[3845] pbx.c:     -- Executing [s@macro-hangupcall:7] GotoIf("SIP/111-00000022", "1?theend") in new stack
[May 15 14:00:54] VERBOSE[3845] pbx.c:     -- Goto (macro-hangupcall,s,9)
[May 15 14:00:54] VERBOSE[3845] pbx.c:     -- Executing [s@macro-hangupcall:9] Hangup("SIP/111-00000022", "") in new stack
[May 15 14:00:54] VERBOSE[3845] app_macro.c:   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/111-00000022' in macro 'hangupcall'
[May 15 14:00:54] VERBOSE[3845] pbx.c:   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/111-00000022'

[tot3nkopf]

Maybe you missed joining, rights for SIP TG or configuration for transit...
If you use ARS for instance in OXO you should have an entry in public or private numbering plan (depending on how the SIP TG with Elastix is declared (public or private):
start 0 end 0 base ARS drop

You should add a 0 in outbound dialing from Elastix:
number dialed by user in elastix --> 2222222
number sent by elastix to oxo ---> 02222222

[rhfn]

Thank you tot3nkopf. Problem has been solved. There was a fault in the Incoming/Outgoing Numbering Format in OXO. Now it works like a charm. Still some FAX-Problems over SIP, but working on it.

[tot3nkopf]

In ARS gateway parameters set the Fax parameter from T38 to G711. Take care also to have same framing on both sides.
It will work just fine with Hylafax.

[luis.molinas]

Hi guys! "I have problem to call to external numbers through OXO".

//If you use ARS for instance in OXO you should have an entry in public or private numbering plan (depending on how the SIP TG with Elastix is declared (public or private):
start 0 end 0 base ARS drop

You should add a 0 in outbound dialing from Elastix:
number dialed by user in elastix --> 2222222
number sent by elastix to oxo ---> 02222222//

I have declared my SIP TG with Elastix as Private. So, i need to create in private numbering plan with "Secundary trunk group" funtion??

[tot3nkopf]

Yes. Why u ask and do not try?

[luis.molinas]

I tried, but it doesn't works.

Example:
The number "0" is my prefix for call to external number.

ext number: 0981730125
I dialed: 00981730125

In the Sip trace, the OXO respond "404 not found".

I attached the SIP capture. Can I attach the OXO data base too?

screenshot example.JPG

[cavagnaro]

Joining between trunks ok? Prefix created on correct zone?

[luis.molinas]

Hi Cavagnaro, sorry but i'm not familiar with OXO. how/where i can to check the "joining between trunks?

Regards.

[luis.molinas]

Cavagnaro,

joining.JPG

I attached the sceenshot of the "Joining", is rigth this configuration?