No audio - DIDWW to Asterisk SIP Trunk

networker · October 24, 2016, 7:06am

hi,

I’ve a SIP trunk between DIDWW to Asterisk SIP Trunk. The call comes in to the Asterisk but there is no IVR sound.

Asterisk Trunk Configuration

context=from-trunk
dtmfmode=rfc2833
dtmf=rfc2833
host=X.X.X.X
insecure=very
type=peer
nat=never (I tried “yes” and “route” - nothing helps)
allow=g729 (I tried ulaw, alaw & all - nothing helps)

Please find the logs below from Asterisk.

-- Executing [300@from-pstn:1] NoOp("SIP/DIDWW-0000002a", "Catch-All DID Mat                                                                                        ch - Found 300 - You probably want a DID for this.") in new stack
-- Executing [300@from-pstn:2] Goto("SIP/DIDWW-0000002a", "ext-did,s,1") in                                                                                         new stack
-- Goto (ext-did,s,1)
-- Executing [s@ext-did:1] Set("SIP/DIDWW-0000002a", "__FROM_DID=s") in new                                                                                         stack
-- Executing [s@ext-did:2] Gosub("SIP/DIDWW-0000002a", "app-blacklist-check,                                                                                        s,1") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/DIDWW-0000002a", "0?black                                                                                        listed") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/DIDWW-0000002a", "CALLED_BLA                                                                                        CKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/DIDWW-0000002a", "") in n                                                                                        ew stack
-- Executing [s@ext-did:3] ExecIf("SIP/DIDWW-0000002a", "0 ?Set(CALLERID(nam                                                                                        e)=+966114405457)") in new stack
-- Executing [s@ext-did:4] Set("SIP/DIDWW-0000002a", "__CALLINGPRES_SV=allow                                                                                        ed_not_screened") in new stack
-- Executing [s@ext-did:5] Set("SIP/DIDWW-0000002a", "CALLERPRES()=allowed_n                                                                                        ot_screened") in new stack
-- Executing [s@ext-did:6] Goto("SIP/DIDWW-0000002a", "ivr-3,s,1") in new st                                                                                        ack
-- Goto (ivr-3,s,1)
-- Executing [s@ivr-3:1] Set("SIP/DIDWW-0000002a", "MSG=custom/Welcome") in                                                                                         new stack
-- Executing [s@ivr-3:2] Set("SIP/DIDWW-0000002a", "LOOPCOUNT=0") in new sta                                                                                        ck
-- Executing [s@ivr-3:3] Set("SIP/DIDWW-0000002a", "__DIR-CONTEXT=") in new                                                                                         stack
-- Executing [s@ivr-3:4] Set("SIP/DIDWW-0000002a", "_IVR_CONTEXT_ivr-3=") in                                                                                         new stack
-- Executing [s@ivr-3:5] Set("SIP/DIDWW-0000002a", "_IVR_CONTEXT=ivr-3") in                                                                                         new stack
-- Executing [s@ivr-3:6] GotoIf("SIP/DIDWW-0000002a", "0?begin") in new stac                                                                                        k
-- Executing [s@ivr-3:7] Answer("SIP/DIDWW-0000002a", "") in new stack
-- Executing [s@ivr-3:8] Wait("SIP/DIDWW-0000002a", "1") in new stack
-- Executing [s@ivr-3:9] Set("SIP/DIDWW-0000002a", "TIMEOUT(digit)=3") in ne                                                                                        w stack
-- Digit timeout set to 3.000
-- Executing [s@ivr-3:10] Set("SIP/DIDWW-0000002a", "TIMEOUT(response)=10")                                                                                         in new stack
-- Response timeout set to 10.000
-- Executing [s@ivr-3:11] Set("SIP/DIDWW-0000002a", "__IVR_RETVM=") in new s                                                                                        tack
-- Executing [s@ivr-3:12] ExecIf("SIP/DIDWW-0000002a", "1?Background(custom/                                                                                        Welcome)") in new stack
-- <SIP/DIDWW-0000002a> Playing 'custom/Welcome.slin' (language 'en')
-- Executing [s@ivr-3:13] WaitExten("SIP/DIDWW-0000002a", ",") in new stack
-- Timeout on SIP/DIDWW-0000002a, going to 't'
-- Executing [t@ivr-3:1] Set("SIP/DIDWW-0000002a", "MSG=custom/Welcome") in new stack
-- Executing [t@ivr-3:2] Goto("SIP/DIDWW-0000002a", "loop,1") in new stack
-- Goto (ivr-3,loop,1)
-- Executing [loop@ivr-3:1] Set("SIP/DIDWW-0000002a", "LOOPCOUNT=1") in new stack
-- Executing [loop@ivr-3:2] GotoIf("SIP/DIDWW-0000002a", "0?hang,1") in new stack
-- Executing [loop@ivr-3:3] Goto("SIP/DIDWW-0000002a", "ivr-3,s,begin") in new stack
-- Goto (ivr-3,s,9)
-- Executing [s@ivr-3:9] Set("SIP/DIDWW-0000002a", "TIMEOUT(digit)=3") in new stack
-- Digit timeout set to 3.000
-- Executing [s@ivr-3:10] Set("SIP/DIDWW-0000002a", "TIMEOUT(response)=10") in new stack
-- Response timeout set to 10.000
-- Executing [s@ivr-3:11] Set("SIP/DIDWW-0000002a", "__IVR_RETVM=") in new stack
-- Executing [s@ivr-3:12] ExecIf("SIP/DIDWW-0000002a", "1?Background(custom/Welcome)") in new stack
-- <SIP/DIDWW-0000002a> Playing 'custom/Welcome.slin' (language 'en')

– Executing [s@ivr-3:13] WaitExten(“SIP/DIDWW-0000002a”, “,”) in new stack
== Spawn extension (ivr-3, s, 13) exited non-zero on ‘SIP/DIDWW-0000002a’
– Executing [h@ivr-3:1] Hangup(“SIP/DIDWW-0000002a”, “”) in new stack
== Spawn extension (ivr-3, h, 1) exited non-zero on ‘SIP/DIDWW-0000002a’

Regards,

david551 · October 24, 2016, 8:16am

The IVR is working. I suspect you have a NAT or firewall problem that is preventing media getting through.

If you are inside NAT, the key options are those which tell Asterisk how to find its own address, not the nat= one.

Incidentally note that very is deprecated or obsolete as a parameter of insecure. Also the default is to allow all codecs, so allowing one explicitly without disallowing others has no effect.

You can use “rtp set debug on” to see if media is arriving and “sip set debug on” to see what addresses are being exchanged.

networker · October 24, 2016, 9:32am

Appreciate your response…

What can I use instead of "Insecure=very"
I made the codec to allow=all

As per your suggestion, I ran the debugs and the output follows…

Retransmitting #8 (NAT) to 46.19.210.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 46.19.210.14;branch=z9hG4bKnvoTBaT0;received=46.19.210.14;rport=5060
From: +966114405457 sip:+966114405457@46.19.210.14;tag=4376B38B-580DCB53000AD6B3-203AE700
To: sip:300@178.20.146.250;tag=as4c18471b
Call-ID: 11-2B9BC43D-580DCB53000AD73E-203AE700
CSeq: 10 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:300@178.20.146.250:5060
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 472582435 472582435 IN IP4 178.20.146.250
s=Asterisk PBX 1.8.7.0
c=IN IP4 178.20.146.250
t=0 0
m=audio 11360 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

v=0
o=root 472582435 472582435 IN IP4 178.20.146.250
s=Asterisk PBX 1.8.7.0
c=IN IP4 178.20.146.250
t=0 0
m=audio 11360 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

ElastixCLI> sip set debug off
SIP Debugging Disabled
== Spawn extension (ivr-3, s, 13) exited non-zero on ‘SIP/DIDWW-00000005’
– Executing [h@ivr-3:1] Hangup(“SIP/DIDWW-00000005”, “”) in new stack
== Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/DIDWW-00000005’
ElastixCLI> sip set debug off
SIP Debugging Disabled
Elastix*CLI>

regards,

david551 · October 24, 2016, 9:58am

Please mark the log as unformatted text, otherwise everything in angle brackets gets hidden.

insecure does two things:

with invite, doesn’t try to authenticate incoming calls. This tends to be necessary with ITSP’s, as they generally only operate one way authentication. However, in recent versions of Asterisk, I consider that using remotesecret, rather than secret, is a less obscure way of doing this.
with port, ignores the source port number. This is particularly necessary for TCP, but might be necessary in some other cases.

You need to evaluate your specific requirements and use an appropriate combination of these options,but only the minimum needed as they remove security checks.

david551 · October 24, 2016, 10:09am

As well as not receiving any inbound media, which indicates that 178.20.146.25011360 is not routable, to you, from the peer, you are also not receiving an ACK, which suggests that either the hidden information in your Contact header is not acceptable to them, or that 46.19.210.14:5060 doesn’t actually route back to them. The lack of media could also be because 46.19.210.14:5060 is not routable from you.

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No audio - DIDWW to Asterisk SIP Trunk

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