Inbound calls not connecting

Asterisk Asterisk SIP
1

Hi all,

With the log below, can anyone tel why the call is not connecting. Outgoing calls works fine but I make a inbound call, I get the log info below and you can see that the call is answered and plays the necessary file. But the problem is that from the callers mobile phone is only indicate calling (with no sound) until the Hangup stage in the dialplan and then the call ends.

My cloud sever is not behind but for troublshooting purpose I tried using nat=force_rport,comedia but still no luck.
I event tried to open all ports (in Iptable) in this log for the specific IP but still no luck.
Sip Provider also saying that everything is okay from their end.

Also at end of the call, it show re-transmission timed out, which is probably because ack is not returned back after 200 message.

<--- SIP read from UDP:provider_SIP_SignalIP:5068 --->
INVITE sip:xxxxxxxxxx@xxxx SIP/2.0
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400
Route: <sip:provider_GW:5060;transport=udp;lr>
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>
CSeq: 1 INVITE
P-Access-Network-Info: GEN-ACCESS;"area-number=+2201"
Max-Forwards: 70
Contact: <sip:provider_SIP_SignalIP:5060>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
P-Asserted-Identity: <tel:xxxxxxx>
P-Early-Media: supported
Supported: 100rel,timer,histinfo,precondition
Min-SE: 90
Session-Expires: 1800;refresher=uac
Content-Length: 561
Content-Type: application/sdp

v=0
o=HuaweiSoftx3000 1076374170 1076374171 IN IP4 provider_SIP_SignalIP
s=SipCall
c=IN IP4 providerIP
t=0 0
m=audio 31384 RTP/AVP 108 102 8 0 18 4 116 3
a=rtpmap:108 AMR/8000
a=fmtp:108 mode-change-neighbor=1;mode-change-period=2
a=rtpmap:102 AMR/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:116 telephone-event/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=maxptime:20
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv
a=3gOoBTC
<------------->
--- (18 headers 22 lines) ---
Sending to provider_SIP_SignalIP:5068 (no NAT)
Sending to provider_SIP_SignalIP:5068 (no NAT)
Using INVITE request as basis request - j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
Found peer 'xxxxl' for 'xxxxxxx' from provider_SIP_SignalIP:5068
[Feb 25 11:56:26] ERROR[6400][C-00000002]: sip/reqresp_parser.c:929 get_name_and_number: can not parse name and number from sip header.
  == Using SIP RTP CoS mark 5
Got SDP version 1076374171 and unique parts [HuaweiSoftx3000 1076374170 IN IP4 provider_SIP_SignalIP]
Found RTP audio format 108
Found RTP audio format 102
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 116
Found RTP audio format 3
Found unknown media description format AMR for ID 108
Found unknown media description format AMR for ID 102
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 116
Found audio description format GSM for ID 3
Capabilities: us - (alaw), peer - audio=(ulaw|gsm|g723|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7f19e40193d0 -- Strict RTP learning after remote address set to: providerIP:31384
Peer audio RTP is at port providerIP:31384
Looking for xxxxxxxxxx in xxxxlio (domain xxxx)
sip_route_dump: route/path hop: <sip:provider_SIP_SignalIP:5060>

<--- Transmitting (NAT) to provider_SIP_SignalIP:5068 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Length: 0


<------------>
    -- Executing [xxxxxxxxxx@xxxxlio:1] Answer("SIP/xxxxl-00000001", "") in new stack
Audio is at 14638
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to provider_SIP_SignalIP:5068 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>;tag=as1ddfceb4
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 256

v=0
o=root 1476618138 1476618138 IN IP4 provider_GW
s=Asterisk PBX 16.16.0
c=IN IP4 provider_GW
t=0 0
m=audio 14638 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:provider_SIP_SignalIP:5068 --->
INVITE sip:xxxxxxxxxx@xxxx SIP/2.0
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400
Route: <sip:provider_GW:5060;transport=udp;lr>
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>
CSeq: 1 INVITE
P-Access-Network-Info: GEN-ACCESS;"area-number=+2201"
Max-Forwards: 70
Contact: <sip:provider_SIP_SignalIP:5060>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
P-Asserted-Identity: <tel:xxxxxxx>
P-Early-Media: supported
Supported: 100rel,timer,histinfo,precondition
Min-SE: 90
Session-Expires: 1800;refresher=uac
Content-Length: 561
Content-Type: application/sdp

v=0
o=HuaweiSoftx3000 1076374170 1076374171 IN IP4 provider_SIP_SignalIP
s=SipCall
c=IN IP4 providerIP
t=0 0
m=audio 31384 RTP/AVP 108 102 8 0 18 4 116 3
a=rtpmap:108 AMR/8000
a=fmtp:108 mode-change-neighbor=1;mode-change-period=2
a=rtpmap:102 AMR/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:116 telephone-event/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=maxptime:20
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv
a=3gOoBTC
<------------->
--- (18 headers 22 lines) ---
Ignoring this INVITE request

<--- Transmitting (NAT) to provider_SIP_SignalIP:5068 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Length: 0


<------------>
Audio is at 14638
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to provider_SIP_SignalIP:5068 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>;tag=as1ddfceb4
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 256

v=0
o=root 1476618138 1476618139 IN IP4 provider_GW
s=Asterisk PBX 16.16.0
c=IN IP4 provider_GW
t=0 0
m=audio 14638 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
Retransmitting #1 (NAT) to provider_SIP_SignalIP:5068:
SIP/2.0 200 OK
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>;tag=as1ddfceb4
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 256

v=0
o=root 1476618138 1476618138 IN IP4 provider_GW
s=Asterisk PBX 16.16.0
c=IN IP4 provider_GW
t=0 0
m=audio 14638 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Executing [xxxxxxxxxx@xxxxlio:2] Wait("SIP/xxxxl-00000001", "2") in new stack

<--- SIP read from UDP:provider_SIP_SignalIP:5068 --->
INVITE sip:xxxxxxxxxx@xxxx SIP/2.0
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400
Route: <sip:provider_GW:5060;transport=udp;lr>
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>
CSeq: 1 INVITE
P-Access-Network-Info: GEN-ACCESS;"area-number=+2201"
Max-Forwards: 70
Contact: <sip:provider_SIP_SignalIP:5060>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
P-Asserted-Identity: <tel:xxxxxxx>
P-Early-Media: supported
Supported: 100rel,timer,histinfo,precondition
Min-SE: 90
Session-Expires: 1800;refresher=uac
Content-Length: 561
Content-Type: application/sdp

v=0
o=HuaweiSoftx3000 1076374170 1076374171 IN IP4 provider_SIP_SignalIP
s=SipCall
c=IN IP4 providerIP
t=0 0
m=audio 31384 RTP/AVP 108 102 8 0 18 4 116 3
a=rtpmap:108 AMR/8000
a=fmtp:108 mode-change-neighbor=1;mode-change-period=2
a=rtpmap:102 AMR/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:116 telephone-event/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=maxptime:20
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv
a=3gOoBTC
<------------->
--- (18 headers 22 lines) ---
Ignoring this INVITE request

<--- Transmitting (NAT) to provider_SIP_SignalIP:5068 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Length: 0


<------------>
Audio is at 14638
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to provider_SIP_SignalIP:5068 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>;tag=as1ddfceb4
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 256

v=0
o=root 1476618138 1476618140 IN IP4 provider_GW
s=Asterisk PBX 16.16.0
c=IN IP4 provider_GW
t=0 0
m=audio 14638 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
Retransmitting #2 (NAT) to provider_SIP_SignalIP:5068:
SIP/2.0 200 OK
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>;tag=as1ddfceb4
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 256

v=0
o=root 1476618138 1476618138 IN IP4 provider_GW
s=Asterisk PBX 16.16.0
c=IN IP4 provider_GW
t=0 0
m=audio 14638 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Executing [xxxxxxxxxx@xxxxlio:3] Playback("SIP/xxxxl-00000001", "tt-monkeys") in new stack
    -- <SIP/xxxxl-00000001> Playing 'tt-monkeys.alaw' (language 'en')

<--- SIP read from UDP:provider_SIP_SignalIP:5068 --->
INVITE sip:xxxxxxxxxx@xxxx SIP/2.0
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400
Route: <sip:provider_GW:5060;transport=udp;lr>
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>
CSeq: 1 INVITE
P-Access-Network-Info: GEN-ACCESS;"area-number=+2201"
Max-Forwards: 70
Contact: <sip:provider_SIP_SignalIP:5060>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
P-Asserted-Identity: <tel:xxxxxxx>
P-Early-Media: supported
Supported: 100rel,timer,histinfo,precondition
Min-SE: 90
Session-Expires: 1800;refresher=uac
Content-Length: 561
Content-Type: application/sdp

v=0
o=HuaweiSoftx3000 1076374170 1076374171 IN IP4 provider_SIP_SignalIP
s=SipCall
c=IN IP4 providerIP
t=0 0
m=audio 31384 RTP/AVP 108 102 8 0 18 4 116 3
a=rtpmap:108 AMR/8000
a=fmtp:108 mode-change-neighbor=1;mode-change-period=2
a=rtpmap:102 AMR/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:116 telephone-event/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=maxptime:20
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv
a=3gOoBTC
<------------->
--- (18 headers 22 lines) ---
Ignoring this INVITE request

<--- Transmitting (NAT) to provider_SIP_SignalIP:5068 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Length: 0


<------------>
Audio is at 14638
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to provider_SIP_SignalIP:5068 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>;tag=as1ddfceb4
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 256

v=0
o=root 1476618138 1476618141 IN IP4 provider_GW
s=Asterisk PBX 16.16.0
c=IN IP4 provider_GW
t=0 0
m=audio 14638 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
Retransmitting #3 (NAT) to provider_SIP_SignalIP:5068:
SIP/2.0 200 OK
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>;tag=as1ddfceb4
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 256

v=0
o=root 1476618138 1476618138 IN IP4 provider_GW
s=Asterisk PBX 16.16.0
c=IN IP4 provider_GW
t=0 0
m=audio 14638 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:provider_SIP_SignalIP:5066 --->
OPTIONS sip:provider_GW:5060 SIP/2.0
Via: SIP/2.0/UDP provider_SIP_SignalIP:5066;branch=z9hG4bKxzm2xtfxztny00t23tmft0y3s;X-DispMsg=1402
Call-ID: 57115xstxt3ntnhz5mhm53h5fz0z7f20@10.18.5.64
From: <sip:provider_SIP_SignalIP:5060>;tag=32s01n31-CC-1008-OFC-45
To: <sip:provider_GW:5060>
CSeq: 1 OPTIONS
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to provider_SIP_SignalIP:5066 (no NAT)
Looking for s in phoneglue (domain provider_GW)

<--- Transmitting (no NAT) to provider_SIP_SignalIP:5066 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP provider_SIP_SignalIP:5066;branch=z9hG4bKxzm2xtfxztny00t23tmft0y3s;X-DispMsg=1402;received=provider_SIP_SignalIP
From: <sip:provider_SIP_SignalIP:5060>;tag=32s01n31-CC-1008-OFC-45
To: <sip:provider_GW:5060>;tag=as39f17b8a
Call-ID: 57115xstxt3ntnhz5mhm53h5fz0z7f20@10.18.5.64
CSeq: 1 OPTIONS
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '57115xstxt3ntnhz5mhm53h5fz0z7f20@10.18.5.64' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:provider_SIP_SignalIP:5068 --->
INVITE sip:xxxxxxxxxx@xxxx SIP/2.0
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400
Route: <sip:provider_GW:5060;transport=udp;lr>
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>
CSeq: 1 INVITE
P-Access-Network-Info: GEN-ACCESS;"area-number=+2201"
Max-Forwards: 70
Contact: <sip:provider_SIP_SignalIP:5060>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
P-Asserted-Identity: <tel:xxxxxxx>
P-Early-Media: supported
Supported: 100rel,timer,histinfo,precondition
Min-SE: 90
Session-Expires: 1800;refresher=uac
Content-Length: 561
Content-Type: application/sdp

v=0
o=HuaweiSoftx3000 1076374170 1076374171 IN IP4 provider_SIP_SignalIP
s=SipCall
c=IN IP4 providerIP
t=0 0
m=audio 31384 RTP/AVP 108 102 8 0 18 4 116 3
a=rtpmap:108 AMR/8000
a=fmtp:108 mode-change-neighbor=1;mode-change-period=2
a=rtpmap:102 AMR/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:116 telephone-event/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=maxptime:20
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv
a=3gOoBTC
<------------->
--- (18 headers 22 lines) ---
Ignoring this INVITE request

<--- Transmitting (NAT) to provider_SIP_SignalIP:5068 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Length: 0


<------------>
Audio is at 14638
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to provider_SIP_SignalIP:5068 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>;tag=as1ddfceb4
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 256

v=0
o=root 1476618138 1476618142 IN IP4 provider_GW
s=Asterisk PBX 16.16.0
c=IN IP4 provider_GW
t=0 0
m=audio 14638 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
Retransmitting #4 (NAT) to provider_SIP_SignalIP:5068:
SIP/2.0 200 OK
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>;tag=as1ddfceb4
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 256

v=0
o=root 1476618138 1476618138 IN IP4 provider_GW
s=Asterisk PBX 16.16.0
c=IN IP4 provider_GW
t=0 0
m=audio 14638 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
Retransmitting #5 (NAT) to provider_SIP_SignalIP:5068:
SIP/2.0 200 OK
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>;tag=as1ddfceb4
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 256

v=0
o=root 1476618138 1476618138 IN IP4 provider_GW
s=Asterisk PBX 16.16.0
c=IN IP4 provider_GW
t=0 0
m=audio 14638 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:provider_SIP_SignalIP:5068 --->
INVITE sip:xxxxxxxxxx@xxxx SIP/2.0
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400
Route: <sip:provider_GW:5060;transport=udp;lr>
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>
CSeq: 1 INVITE
P-Access-Network-Info: GEN-ACCESS;"area-number=+2201"
Max-Forwards: 70
Contact: <sip:provider_SIP_SignalIP:5060>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
P-Asserted-Identity: <tel:xxxxxxx>
P-Early-Media: supported
Supported: 100rel,timer,histinfo,precondition
Min-SE: 90
Session-Expires: 1800;refresher=uac
Content-Length: 561
Content-Type: application/sdp

v=0
o=HuaweiSoftx3000 1076374170 1076374171 IN IP4 provider_SIP_SignalIP
s=SipCall
c=IN IP4 providerIP
t=0 0
m=audio 31384 RTP/AVP 108 102 8 0 18 4 116 3
a=rtpmap:108 AMR/8000
a=fmtp:108 mode-change-neighbor=1;mode-change-period=2
a=rtpmap:102 AMR/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:116 telephone-event/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=maxptime:20
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv
a=3gOoBTC
<------------->
--- (18 headers 22 lines) ---
Ignoring this INVITE request

<--- Transmitting (NAT) to provider_SIP_SignalIP:5068 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Length: 0


<------------>
Audio is at 14638
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to provider_SIP_SignalIP:5068 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>;tag=as1ddfceb4
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 256

v=0
o=root 1476618138 1476618143 IN IP4 provider_GW
s=Asterisk PBX 16.16.0
c=IN IP4 provider_GW
t=0 0
m=audio 14638 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
Retransmitting #6 (NAT) to provider_SIP_SignalIP:5068:
SIP/2.0 200 OK
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>;tag=as1ddfceb4
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 256

v=0
o=root 1476618138 1476618138 IN IP4 provider_GW
s=Asterisk PBX 16.16.0
c=IN IP4 provider_GW
t=0 0
m=audio 14638 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Executing [xxxxxxxxxx@xxxxlio:4] Hangup("SIP/xxxxl-00000001", "") in new stack
  == Spawn extension (xxxxlio, xxxxxxxxxx, 4) exited non-zero on 'SIP/xxxxl-00000001'
Scheduling destruction of SIP dialog 'j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64' in 32000 ms (Method: INVITE)
Retransmitting #7 (NAT) to provider_SIP_SignalIP:5068:
SIP/2.0 200 OK
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>;tag=as1ddfceb4
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 256

v=0
o=root 1476618138 1476618138 IN IP4 provider_GW
s=Asterisk PBX 16.16.0
c=IN IP4 provider_GW
t=0 0
m=audio 14638 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:provider_SIP_SignalIP:5066 --->
OPTIONS sip:provider_GW:5060 SIP/2.0
Via: SIP/2.0/UDP provider_SIP_SignalIP:5066;branch=z9hG4bKnzxx5ffhyf5f0n27yy0ezmhey;X-DispMsg=1402
Call-ID: 0mmfzy513ne3ms7ym77xz2zzxs2shn0e@10.18.5.64
From: <sip:provider_SIP_SignalIP:5060>;tag=mxx07523-CC-1008-OFC-45
To: <sip:provider_GW:5060>
CSeq: 1 OPTIONS
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to provider_SIP_SignalIP:5066 (no NAT)
Looking for s in phoneglue (domain provider_GW)

<--- Transmitting (no NAT) to provider_SIP_SignalIP:5066 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP provider_SIP_SignalIP:5066;branch=z9hG4bKnzxx5ffhyf5f0n27yy0ezmhey;X-DispMsg=1402;received=provider_SIP_SignalIP
From: <sip:provider_SIP_SignalIP:5060>;tag=mxx07523-CC-1008-OFC-45
To: <sip:provider_GW:5060>;tag=as2a6e5b94
Call-ID: 0mmfzy513ne3ms7ym77xz2zzxs2shn0e@10.18.5.64
CSeq: 1 OPTIONS
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '0mmfzy513ne3ms7ym77xz2zzxs2shn0e@10.18.5.64' in 32000 ms (Method: OPTIONS)
Retransmitting #8 (NAT) to provider_SIP_SignalIP:5068:
SIP/2.0 200 OK
Via: SIP/2.0/UDP provider_SIP_SignalIP:5068;branch=z9hG4bKyvbawvavqrw2xjkaaky1arbk3;X-DispMsg=1400;received=provider_SIP_SignalIP;rport=5068
From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=q01ay24v-CC-1013-OFC-45
To: "xxxxxxx"<sip:xxxxxxx@xxxx>;tag=as1ddfceb4
Call-ID: j14w2awr3yba0q2bv3bkbxykaj2avr2w@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 16.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:xxxxxxxxxx@provider_GW:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 256

v=0
o=root 1476618138 1476618138 IN IP4 provider_GW
s=Asterisk PBX 16.16.0
c=IN IP4 provider_GW
t=0 0
m=audio 14638 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
2

Because the 200 OK isn’t reaching the provider, or isn’t being recognized by them.

3

Oh thanks for the reply, the SIP provider said that my Asterisk keep retransmitting 200 to them. So they are seeing the 200 but for some reason they not responding with ack and so my Asterisk keep retransmitting the 200 to them over and over.

Since they seeing the 200 as they said, is there any adjustment i can make on Asterisk for them to recognize this 200 from my end?

Because of the " chan_sip.c: Retransmission timeout reached on transmission…" that comes at the end of the call. I was also thinking that perhaps that they seeing the 200 and trying to send ack to my asterisk but something is blocking it. So I had to open all the ports in the log above but no luck.
my rtp port range is now 10000 to 30000, perhaps i use iptables and open about 3 ports within this range?
The SIP provider has 2 IPs for media, i open 3 ports for the 2 media IPs?

4

Nothing should need to be done, so there is unlikely to be any setting to change things. You need to find out why they are not recognizing the 200 OK. Maybe a router is mangling it?

5

An Asterisk server of version 13 is connected to this same SIP provider and inbound and outbound are all working fine.

I did adapt the same setting in sip.conf and iptable on this Asterisk 13 to my Asterisk 16 but still.

From SIP debug log below in the Asterisk 13, it shows “silenceSupp:off “ but in my Asterisk 16 this is not in the debug log.
So the SIP provider suggest I add “silenceSupp:off “ in my Asterisk 16 config but I don’t know how.
Any help or advise please?
The link is also running through a VPN tunnel, not sure if it could have anything to do with this?

> 
> > > Audio is at 14780
> > > Adding codec alaw to SDP
> > > 
> > > <--- Reliably Transmitting (NAT) to xxxxxxx:5067 --->
> > > SIP/2.0 200 OK
> > > Via: SIP/2.0/UDP xxxxxxx:5067;branch=z9hG4bKqa2h0aqanqta2tpq0jnjna0k0;X-DispMsg=1401;received=xxxxxxx;rport=5067
> > > From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=jb0akrhj-CC-1000-OFC-8
> > > To: "xxxxxxx"<sip:xxxxxxx@xxxxxxxx>;tag=as1db4ddd3
> > > Call-ID: 0ktrrtqjknsq00c2snhu2ajprhsrhnsc@10.18.5.64
> > > CSeq: 1 INVITE
> > > Server: Asterisk PBX 13.19.2
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> > > Supported: replaces, timer
> > > Session-Expires: 1800;refresher=uac
> > > Contact: <sip:account@AsteriskserverIP:5060>
> > > Content-Type: application/sdp
> > > Require: timer
> > > Content-Length: 225
> > > 
> > > v=0
> > > o=root 1370995353 1370995353 IN IP4 AsteriskserverIP
> > > s=Asterisk PBX 13.19.2
> > > c=IN IP4 AsteriskserverIP
> > > t=0 0
> > > m=audio 14780 RTP/AVP 8
> > > a=rtpmap:8 PCMA/8000
> > > a=silenceSupp:off - - - -
> > > a=ptime:20
> > > a=maxptime:150
> > > a=sendrecv
> > > 
> > > <------------>
> > >        > 0x7fc53400cc60 -- Strict RTP switching to RTP target address xxxxxxx:36392 as source
> > >     -- Executing [account@providerNm:2] Set("SIP/providerNm-00000020", "FOO="xxxxxxx"<sip:xxxxxxx") in new stack
> > >     -- Executing [account@providerNm:3] Set("SIP/providerNm-00000020", "FOO=xxxxxxx") in new stack
> > >     -- Executing [account@providerNm:4] Goto("SIP/providerNm-00000020", "xxxxxxx,1") in new stack
> > >     -- Goto (providerNm,xxxxxxx,1)
> > >     -- Executing [xxxxxxx@providerNm:1] Answer("SIP/providerNm-00000020", "") in new stack
> > >     -- Executing [xxxxxxx@providerNm:2] Voximal("SIP/providerNm-00000020", "http://nemaivr.sbc4d.com/start.vxml.php?DNID=account&MSISDN=xxxxxxx&service=mis") in new stack
> > > Retransmitting #1 (NAT) to xxxxxxx:5067:
> > > SIP/2.0 200 OK
> > > Via: SIP/2.0/UDP xxxxxxx:5067;branch=z9hG4bKqa2h0aqanqta2tpq0jnjna0k0;X-DispMsg=1401;received=xxxxxxx;rport=5067
> > > From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=jb0akrhj-CC-1000-OFC-8
> > > To: "xxxxxxx"<sip:xxxxxxx@xxxxxxxx>;tag=as1db4ddd3
> > > Call-ID: 0ktrrtqjknsq00c2snhu2ajprhsrhnsc@10.18.5.64
> > > CSeq: 1 INVITE
> > > Server: Asterisk PBX 13.19.2
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> > > Supported: replaces, timer
> > > Session-Expires: 1800;refresher=uac
> > > Contact: <sip:account@AsteriskserverIP:5060>
> > > Content-Type: application/sdp
> > > Require: timer
> > > Content-Length: 225
> > > 
> > > v=0
> > > o=root 1370995353 1370995353 IN IP4 AsteriskserverIP
> > > s=Asterisk PBX 13.19.2
> > > c=IN IP4 AsteriskserverIP
> > > t=0 0
> > > m=audio 14780 RTP/AVP 8
> > > a=rtpmap:8 PCMA/8000
> > > a=silenceSupp:off - - - -
> > > a=ptime:20
> > > a=maxptime:150
> > > a=sendrecv
> > > 
> > > ---
> > > 
> > > <--- SIP read from UDP:xxxxxxx:5067 --->
> > > ACK sip:account@AsteriskserverIP:5060 SIP/2.0
> > > Via: SIP/2.0/UDP xxxxxxx:5067;branch=z9hG4bKj2p2rjnkpaescbccjhbqtp0bc;X-DispMsg=1401
> > > Call-ID: 0ktrrtqjknsq00c2snhu2ajprhsrhnsc@10.18.5.64
> > > From: "xxxxxxx"<sip:xxxxxxx@Asterisk-PBX.com>;tag=jb0akrhj-CC-1000-OFC-8
> > > To: "xxxxxxx"<sip:xxxxxxx@xxxxxxxx>;tag=as1db4ddd3
> > > CSeq: 1 ACK
> > > Max-Forwards: 70
> > > Content-Length: 0
6

To understand why they are ignoring the 200, we’d need to see the INVITE as well as the 200, and we would need to see them as as they are at the provider system, as it is most unlikely that they will mismatch at Asterisk.

7

Hi,
I was talking to them but they want me to sort this out:

The log below is from an Asterisk 13 Server connected to the same SIP provider and its working fine. The log from this Server include “a=silenceSupp:off - - - -” but in my Server this field is not included. So they want me to turn off SilenceSupp in my Asterisk 16 so this field “a=silenceSupp:off - - - -” can be included in the transmission. How can i do this?

v=0
o=root xxxxxxxxxxx xxxxxxxx IN IP4 xxxxxxxx
s=Asterisk PBX 13.19.2
c=IN IP4 xxxxxxxxx
t=0 0
m=audio 14780 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=sendrecv

8

If that is the reason, their SIP implementation is broken. The correct way to handle an unacceptable SDP combination in 200 OK is to send ACK, and then immediately send BYE. Not sending ACK is not a permissible thing to do.

Also, Asterisk doesn’t really support silence suppression, so offering it as off seems the only correct thing to do.

9

Oh thanks, please in which file can i turn off silence suppression?

10

It’s an optional setting that is only defined in the context of ATM based networks. It doesn’t appear in the source code of Asterisk.

I think you provider is clutching at straws. Most SIP user agents are not going to send this parameter.

As I’ve already stated, unacceptable, or invalid, SDP is not a valid reason for completely ignoring a final response. If the provider really is ignoring it for this reason, their system is badly broken. Valid reasons for ignoring final responses would have to be present in the SIP header (or at IP level), not in the SIP body.

11

Well noted and thanks

13

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