Ring Group- no audio both ways(dead air)

Firewall settings are correct.
CLI shows as normal.

Setup:

1800xxxxxxx is called, this is call is connected another company. Company forwards the call to our system. We then take the call to a ring group to an external #. Using this method results in a dead air connection.

What will work is if i dont go through that 1800 number. Using a testing DID that it have I can call the #. Hits our server and connects to the external #. Audio works fine.

I am completely baffled by this as we had this working before we cut over to a new server. (Old server suffered a HDD fail)

sounds like a reinvite problem, or a codec problem. maybe a a bit of debug will help

not 100% if this is what youre asking for

Audio is at 32720
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.15.132:5060:
INVITE sip:107@192.168.15.132:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.45:5060;branch=z9hG4bK2b5d4195
Max-Forwards: 70
From: “8057294176” sip:18057294176@192.168.15.45;tag=as330b13d8
To: sip:107@192.168.15.132:5060;transport=udp
Contact: sip:18057294176@192.168.15.45:5060
Call-ID: 5a920cd556d5959575524b566d1753c2@192.168.15.45:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.0rc1(1.8.11)
Date: Sat, 05 Jan 2013 02:44:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 786846933 786846933 IN IP4 192.168.15.45
s=Asterisk PBX 1.8.11-cert1
c=IN IP4 192.168.15.45
t=0 0
m=audio 32720 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


-- Called SIP/107

<— SIP read from UDP:192.168.15.132:5060 —>
SIP/2.0 100 Trying
Call-ID: 5a920cd556d5959575524b566d1753c2@192.168.15.45:5060
CSeq: 102 INVITE
From: “8057294176” sip:18057294176@192.168.15.45;tag=as330b13d8
To: sip:107@192.168.15.132:5060;tag=549bbddb863d963
Via: SIP/2.0/UDP 192.168.15.45:5060;branch=z9hG4bK2b5d4195
Content-Length: 0
User-Agent: Aastra 480i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:192.168.15.132:5060 —>
SIP/2.0 180 Ringing
Call-ID: 5a920cd556d5959575524b566d1753c2@192.168.15.45:5060
CSeq: 102 INVITE
From: “8057294176” sip:18057294176@192.168.15.45;tag=as330b13d8
To: sip:107@192.168.15.132:5060;tag=549bbddb863d963
Via: SIP/2.0/UDP 192.168.15.45:5060;branch=z9hG4bK2b5d4195
Content-Length: 0
Call-Info: sip:192.168.15.45;appearance-index=1
Allow-Events: talk, hold, conference
Contact: Rodney Miller sip:107@192.168.15.132:5060
User-Agent: Aastra 480i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

<------------->
— (11 headers 0 lines) —
list_route: hop: sip:107@192.168.15.132:5060
– SIP/107-00000046 is ringing

<— SIP read from UDP:192.168.15.132:5060 —>
SIP/2.0 200 OK
Call-ID: 5a920cd556d5959575524b566d1753c2@192.168.15.45:5060
CSeq: 102 INVITE
From: “8057294176” sip:18057294176@192.168.15.45;tag=as330b13d8
To: sip:107@192.168.15.132:5060;tag=549bbddb863d963
Via: SIP/2.0/UDP 192.168.15.45:5060;branch=z9hG4bK2b5d4195
Content-Length: 231
Call-Info: sip:192.168.15.45;appearance-index=1
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
Content-Type: application/sdp
Supported: replaces
Contact: Rodney Miller sip:107@192.168.15.132:5060
User-Agent: Aastra 480i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

v=0
o=MxSIP 0 1295515283 IN IP4 192.168.15.132
s=SIP Call
c=IN IP4 192.168.15.132
t=0 0
m=audio 3000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
<------------->
— (14 headers 11 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.15.132:3000
list_route: hop: sip:107@192.168.15.132:5060
set_destination: Parsing sip:107@192.168.15.132:5060 for address/port to send to
set_destination: set destination to 192.168.15.132:5060
Transmitting (no NAT) to 192.168.15.132:5060:
ACK sip:107@192.168.15.132:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.45:5060;branch=z9hG4bK01ac976a
Max-Forwards: 70
From: “8057294176” sip:18057294176@192.168.15.45;tag=as330b13d8
To: sip:107@192.168.15.132:5060;transport=udp;tag=549bbddb863d963
Contact: sip:18057294176@192.168.15.45:5060
Call-ID: 5a920cd556d5959575524b566d1753c2@192.168.15.45:5060
CSeq: 102 ACK
User-Agent: FPBX-2.10.0rc1(1.8.11)
Content-Length: 0

Can you please tell more about the actual topology? Please include the IP addresses used in the VoIP system.

I would also agree with the notion of a potential codec issue. It could be quite a few things with the little bit of information posted. The issue involves 5 potential transfers/phone systems/carriers and we have a partial output from one.

I find a lot of SIP carriers are not always 100%. I have seen some oddities over the years of doing this. A good example would be 30 seconds of dead air when calling area code 646 numbers, only over a specific carrier however (calls were completing). There are known issues in certain parts of the US like East Texas, Arkansas etc where local termination is an issue and controlled by, or not service by any other significant backbone carriers (XO, Cogent, Level 3, Global Crossing etc.) and it creates some significant local termination/carrier issues.

I would suggest, from experience, either a conference call involving everyone and organization for the appropriate SIP traces is in order, or porting the 888 number to another carrier and banking on the the DID not being under the same circumstances creating the current situation.