Incoming calls not working Elastix 1.6.2-2

eiger3970

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#1
Hi Elastix forum,
after an upgrade to Elastix 1.6.2-2, my incoming calls aren't working.
All trunks and routes are the same as on my old Elastix version.

I have repeated a yum update but still not working. My asterisk -r command displays the following from a mobile incoming call to the PBX:
<------------>
-- Executing [xxxxxxxxxx@from-sip-external:1] NoOp("SIP/xxxxxxxxxx-08a97668", "Received incoming SIP connection from unknown peer to xxxxxxxxxx") in new stack
-- Executing [xxxxxxxxxx@from-sip-external:2] Set("SIP/xxxxxxxxxx-08a97668", "DID=xxxxxxxxxx") in new stack
-- Executing [xxxxxxxxxx@from-sip-external:3] Goto("SIP/xxxxxxxxxx-08a97668", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/xxxxxxxxxx-08a97668", "0?from-trunk|0755938793|1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/xxxxxxxxxx-08a97668", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2010-05-29 03:04:21 UTC.
-- Executing [s@from-sip-external:3] Answer("SIP/xxxxxxxxxx-08a97668", "") in new stack
Audio is at 192.168.1.103 port 18718
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 202.169.178.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.169.178.10:5060;branch=z9hG4bK1caa9b20a4c008425-d099-0;received=202.169.178.10
From: "xxxxxxxxxx"<sip:xxxxxxxxxx@202.169.178.10:5060>;tag=700f73f8-co534-INS001
To: <sip:xxxxxxxxxx@192.168.1.103:5060>;tag=as79267859
Call-ID: 1f05-409-724197052737-TCP-IMG03-3-210.80.190.202
CSeq: 53401 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:xxxxxxxxxx@192.168.1.103>
Content-Type: application/sdp
Content-Length: 309

v=0
o=root 2744 2744 IN IP4 192.168.1.103
s=session
c=IN IP4 192.168.1.103
t=0 0
m=audio 18718 RTP/AVP 8 0 96 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:eek:ff - - - -
a=ptime:20
a=sendrecv

<------------>
-- Executing [s@from-sip-external:4] Wait("SIP/xxxxxxxxxx-08a97668", "2") in new stack
xxxxxxxxxx*CLI>
<--- SIP read from 202.169.178.10:5060 --->
ACK sip:xxxxxxxxxx@121.208.10.172:5060 SIP/2.0
Via: SIP/2.0/UDP 202.169.178.10:5060;branch=z9hG4bK1caa9b20a4c008425-d099-1
Max-Forwards: 70
To: <sip:xxxxxxxxxx@192.168.1.103:5060>;tag=as79267859
From: "xxxxxxxxxx"<sip:xxxxxxxxxx@202.169.178.10:5060>;tag=700f73f8-co534-INS001
Call-ID: 1f05-409-724197052737-TCP-IMG03-3-210.80.190.202
CSeq: 53401 ACK
User-Agent: ENSR2.5.4
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
-- Executing [s@from-sip-external:5] Playback("SIP/xxxxxxxxxx-08a97668", "ss-noservice") in new stack
-- <SIP/xxxxxxxxxx-08a97668> Playing 'ss-noservice' (language 'en')
-- Executing [s@from-sip-external:6] PlayTones("SIP/xxxxxxxxxx-08a97668", "congestion") in new stack
== Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/xxxxxxxxxx-08a97668'
-- Executing [h@from-sip-external:1] NoOp("SIP/xxxxxxxxxx-08a97668", "Hangup") in new stack
-- Executing [h@from-sip-external:2] Set("SIP/xxxxxxxxxx-08a97668", "DID=s") in new stack
-- Executing [h@from-sip-external:3] Goto("SIP/xxxxxxxxxx-08a97668", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/xxxxxxxxxx-08a97668", "0?from-trunk|s|1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/xxxxxxxxxx-08a97668", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2010-05-29 03:04:28 UTC.
-- Executing [s@from-sip-external:3] Answer("SIP/xxxxxxxxxx-08a97668", "") in new stack
== Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/xxxxxxxxxx-08a97668'
Scheduling destruction of SIP dialog '1f05-409-724197052737-TCP-IMG03-3-210.80.190.202' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:xxxxxxxxxx@202.169.178.10:5060> for address/port to send to
set_destination: set destination to 202.169.178.10, port 5060
Reliably Transmitting (NAT) to 202.169.178.10:5060:
BYE sip:xxxxxxxxxx@202.169.178.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK1bc42afd;rport
From: <sip:xxxxxxxxxx@192.168.1.103:5060>;tag=as79267859
To: "xxxxxxxxxx"<sip:0403420075@202.169.178.10:5060>;tag=700f73f8-co534-INS001
Call-ID: 1f05-409-724197052737-TCP-IMG03-3-210.80.190.202
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0
 

danardf

Joined
Dec 3, 2007
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#2
Hi.

If you do a simple submit on every incoming routes?
Or maybe to remake them?
 

eiger3970

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#3
Thanks DanardF, after the reinstallation, I found I had missed the Elastix\PBX\General Settings\Allow Anonymous Inbound SIP Calls? settings which was on no. Changed to yes and all fixed!

The only final issue is the poor quality of the VoIP calls as the reinstallation removed my G.729 codec. I really don't want to pay another $10 for the G.729 codec.

The ilbc codec is free but it's only source code and I need object code. Does Elastix have a download for ilbc as object code?
 

Patrick_elx

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#4
you probably want to create a proper inbound route for all your trunk and avoid treat the anonymous call as an exception if you really need it or completely disable it.

If you had a license for your g729 codecs previously, you can reinstall them with the same license key you had before.


ilbc can not be provided compiled anymore to not infringe the license.
But if you do a seach on this forum, you will find a compiled version of it that you can just copy to the proper folder.
 

eiger3970

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#5
Well, finally fixed the the g729 by finding another post showing how to download the binaries. Thanks to the help which ruled out a few steps to take me to the fix.
 

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