Sip Trunk not working properly (RTP maybe)

Asterisk Asterisk SIP
1

Hello guys!

I am new to asterisk and need some help…

I have an asterisk with 2 network cards, 1 of them my local network 10.1.X.X - 255.255.0.0 and the other network of the telephone company with link sip 30.X.X.X - 255.255.255.248

I forwarded the incoming calls to my extension 5101, it rings normally. but when he answers the call he falls.

Internal connections work normally.

logs:

> [root@PBXXXXXXXX]# asterisk -rvvvvvvvvv
> Asterisk 13.22.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
> Created by Mark Spencer <markster@digium.com>
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
> This is free software, with components licensed under the GNU General Public
> License version 2 and other licenses; you are welcome to redistribute it under
> certain conditions. Type 'core show license' for details.
> =========================================================================
> Connected to Asterisk 13.22.0 currently running on PBXXXXXX (pid = 2311)
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
>        > 0x7f4ac00130d0 -- Strict RTP learning after remote address set to: 30.X.X.X:54832
>     -- Executing [8500@recebe_sip:1] Answer("SIP/4131XXXXXX-00000008", "") in new stack
>     -- Executing [8500@recebe_sip:2] NoCDR("SIP/4131XXXXXX-00000008", "") in new stack
>     -- Executing [8500@recebe_sip:3] NoOp("SIP/4131XXXXXX-00000008", "###### ENTRADA SIP SERCOMTEL  #####") in new stack
>     -- Executing [8500@recebe_sip:4] Dial("SIP/4131XXXXXX-00000008", "SIP/5101") in new stack
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
>     -- Called SIP/5101
>     -- SIP/5101-00000009 is ringing
> [2019-04-08 18:45:16] WARNING[2482]: chan_sip.c:4068 retrans_pkt: Retransmission timeout reached on transmission l5xaw50b5j6wwu9a170aja0xu787baw6@SoftX3000 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> Packet timed out after 6400ms with no response
> [2019-04-08 18:45:16] WARNING[2482]: chan_sip.c:4092 retrans_pkt: Hanging up call l5xaw50b5j6wwu9a170aja0xu787baw6@SoftX3000 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
>   == Spawn extension (recebe_sip, 8500, 4) exited non-zero on 'SIP/4131XXXXXX-00000008'

any1 can help plz?

2

Please provide:

Routing configuration.
Confirmationation that there is no NAT involved on the the 30… interface.
sip.conf, at least the general section and that for source of the call.
The SIP INVITE transaction for the failing leg, from the sip set debug on.

Redact as necessary, but not in any way that does not retain a distinction between addresses, and a distinction between public and private networks.

3

Okay, thanks for the tips. sorry really iam new with this.

here is the sip.conf and I will post the SIP INVITE transaction soon

> 
> sip.conf
> 
> [general]
> 
> #include sip_general_custom.conf
> 
> #include sip_additional.conf
> 
> #include sip_custom.conf
> 
> sip_general_additional.conf
> 
> faxdetect=no
> vmexten=*97
> context=from-sip-external
> callerid=Unknown
> notifyringing=yes
> notifyhold=yes
> tos_sip=cs3
> tos_audio=ef
> tos_video=af41
> alwaysauthreject=yes
> useragent=IPBX-2.11.0(13.22.0)
> disallow=all
> allow=alaw
> allow=gsm
> allow=ulaw
> 
> 
> sip_additional.conf
> 
> [5101]
> deny=0.0.0.0/0.0.0.0
> secret=38aa024e0000XXXXXXXXXXXXXXXXXXXXX
> dtmfmode=rfc2833
> canreinvite=no
> context=from-internal
> host=dynamic
> trustrpid=yes
> sendrpid=no
> type=friend
> nat=no
> port=5060
> qualify=yes
> qualifyfreq=60
> transport=udp
> avpf=no
> force_avp=no
> icesupport=no
> dtlsenable=no
> dtlsverify=no
> dtlssetup=actpass
> rtcp_mux=no
> encryption=no
> callgroup=
> pickupgroup=
> dial=SIP/5101
> mailbox=5101@device
> permit=0.0.0.0/0.0.0.0
> callerid=Suporte 1 <5101>
> callcounter=yes
> faxdetect=no
> 
> sip_custom.conf
> 
> [SIP_TRUNK]
> 
> type=friend
> 
> username=XXXXXXXXXX
> secret=XXXXXXXXXXXXXXXX
> domain=30.x.x.x
> host=30.x.x.x.x
> 
> qualify=yes
> 
> port=5060
> 
> nat=no
> 
> disallow=all
> 
> allow=ulaw
> 
> allow=alaw
> 
> dtmfmode=rfc2833
> 
> context=recebe_sip
> 
> reinvite=no
> 
> canreinvite=no
> 
> insecure=port,invite
4 
<--- SIP read from UDP:30.X.X.X:5060 --->
INVITE sip:85XX@10.37.X.X:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>
CSeq: 1 INVITE
P-Asserted-Identity: <sip:41999XXXXXX@30.X.X.X;user=phone>
Contact: <sip:41999XXXXXX@30.X.X.X:5060;user=phone;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R011
Supported: 100rel
Max-Forwards: 69
Content-Length: 269
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 31053767 31053767 IN IP4 30.37.X.Y
s=Sip Call
c=IN IP4 30.37.X.Y
t=0 0
m=audio 55106 RTP/AVP 8 0 18 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=no
<------------->
--- (14 headers 12 lines) ---
Sending to 30.X.X.X:5060 (no NAT)
Sending to 30.X.X.X:5060 (no NAT)
Using INVITE request as basis request - nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
Found peer '4131XXXXXX' for '41999XXXXXX' from 30.X.X.X:5060
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 97
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 telephone-event for ID 97
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 30.37.X.Y:55106
Looking for 85XX in from-trunk (domain 10.37.X.X)
sip_route_dump: route/path hop: <sip:41999XXXXXX@30.X.X.X:5060;user=phone;transport=udp>

<--- Transmitting (no NAT) to 30.X.X.X:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1;received=30.X.X.X
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:85XX@10.1.2.141:5060>
Content-Length: 0


<------------>
    -- Executing [85XX@from-trunk:1] NoOp("SIP/4131XXXXXX-0000001c", "Catch-All DID Match - Found 85XX - You probably want a DID for this.") in new stack
    -- Executing [85XX@from-trunk:2] Set("SIP/4131XXXXXX-0000001c", "__FROM_DID=85XX") in new stack
    -- Executing [85XX@from-trunk:3] Goto("SIP/4131XXXXXX-0000001c", "ext-did,s,1") in new stack
    -- Goto (ext-did,s,1)
    -- Executing [s@macro-dial-one:27] GotoIf("SIP/4131XXXXXX-0000001c", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:28] GotoIf("SIP/4131XXXXXX-0000001c", "0?skiptrace") in new stack
    -- Executing [s@macro-dial-one:29] GosubIf("SIP/4131XXXXXX-0000001c", "1?ctset,1():ctclear,1()") in new stack
    -- Executing [ctset@macro-dial-one:1] Set("SIP/4131XXXXXX-0000001c", "DB(CALLTRACE/5101)=41999XXXXXX") in new stack
    -- Executing [ctset@macro-dial-one:2] Return("SIP/4131XXXXXX-0000001c", "") in new stack
    -- Executing [s@macro-dial-one:30] Set("SIP/4131XXXXXX-0000001c", "D_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-dial-one:31] ExecIf("SIP/4131XXXXXX-0000001c", "0?SIPAddHeader(Alert-Info: )") in new stack
    -- Executing [s@macro-dial-one:32] ExecIf("SIP/4131XXXXXX-0000001c", "0?SIPAddHeader()") in new stack
    -- Executing [s@macro-dial-one:33] ExecIf("SIP/4131XXXXXX-0000001c", "1?Set(CHANNEL(musicclass)=default)") in new stack
    -- Executing [s@macro-dial-one:34] GosubIf("SIP/4131XXXXXX-0000001c", "0?qwait,1()") in new stack
    -- Executing [s@macro-dial-one:35] Set("SIP/4131XXXXXX-0000001c", "__CWIGNORE=") in new stack
    -- Executing [s@macro-dial-one:36] Set("SIP/4131XXXXXX-0000001c", "__KEEPCID=TRUE") in new stack
    -- Executing [s@macro-dial-one:37] GotoIf("SIP/4131XXXXXX-0000001c", "0?usegoto,1") in new stack
    -- Executing [s@macro-dial-one:38] GotoIf("SIP/4131XXXXXX-0000001c", "1?godial") in new stack
    -- Goto (macro-dial-one,s,43)
    -- Executing [s@macro-dial-one:43] Dial("SIP/4131XXXXXX-0000001c", "SIP/5101,,Ttr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 14086
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.1.2.X:53758:
INVITE sip:5101@10.1.2.X:53758;rinstance=f4179442ee6aa768;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.2.141:5060;branch=z9hG4bK69e6ad05
Max-Forwards: 70
From: "41999XXXXXX" <sip:41999XXXXXX@10.1.2.141>;tag=as4e56f4f0
To: <sip:5101@10.1.2.X:53758;rinstance=f4179442ee6aa768;transport=UDP>
Contact: <sip:41999XXXXXX@10.1.2.141:5060>
Call-ID: 6c83674d7fd9332924a6bd1b5de98e96@10.1.2.141:5060
CSeq: 102 INVITE
User-Agent: IPBX-2.11.0(13.22.0)
Date: Wed, 10 Apr 2019 04:34:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 270

v=0
o=root 535343699 535343699 IN IP4 10.1.2.141
s=Asterisk PBX 13.22.0
c=IN IP4 10.1.2.141
t=0 0
m=audio 14086 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=maxptime:150
a=sendrecv

---
    -- Called SIP/5101

<--- Transmitting (no NAT) to 30.X.X.X:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1;received=30.X.X.X
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>;tag=as4f3257f7
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:85XX@10.1.2.141:5060>
Content-Length: 0


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

<--- SIP read from UDP:10.1.2.X:53758 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.2.141:5060;branch=z9hG4bK69e6ad05
To: <sip:5101@10.1.2.X:53758;rinstance=f4179442ee6aa768;transport=UDP>
From: "41999XXXXXX" <sip:41999XXXXXX@10.1.2.141>;tag=as4e56f4f0
Call-ID: 6c83674d7fd9332924a6bd1b5de98e96@10.1.2.141:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:30.X.X.X:5060 --->
INVITE sip:85XX@10.37.X.X:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>
CSeq: 1 INVITE
P-Asserted-Identity: <sip:41999XXXXXX@30.X.X.X;user=phone>
Contact: <sip:41999XXXXXX@30.X.X.X:5060;user=phone;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R011
Supported: 100rel
Max-Forwards: 69
Content-Length: 269
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 31053767 31053767 IN IP4 30.37.X.Y
s=Sip Call
c=IN IP4 30.37.X.Y
t=0 0
m=audio 55106 RTP/AVP 8 0 18 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=no
<------------->
--- (14 headers 12 lines) ---
Ignoring this INVITE request

<--- Transmitting (no NAT) to 30.X.X.X:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1;received=30.X.X.X
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:85XX@10.1.2.141:5060>
Content-Length: 0


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

<--- SIP read from UDP:10.1.2.X:53758 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.2.141:5060;branch=z9hG4bK69e6ad05
Contact: <sip:5101@10.1.2.X:53758>
To: <sip:5101@10.1.2.X:53758;rinstance=f4179442ee6aa768;transport=UDP>;tag=e6508312
From: "41999XXXXXX" <sip:41999XXXXXX@10.1.2.141>;tag=as4e56f4f0
Call-ID: 6c83674d7fd9332924a6bd1b5de98e96@10.1.2.141:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.2.28 rv2.8.115
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:5101@10.1.2.X:53758>
    -- SIP/5101-0000001d is ringing

<--- Transmitting (no NAT) to 30.X.X.X:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1;received=30.X.X.X
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>;tag=as4f3257f7
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:85XX@10.1.2.141:5060>
Content-Length: 0


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

<--- SIP read from UDP:30.X.X.X:5060 --->
INVITE sip:85XX@10.37.X.X:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>
CSeq: 1 INVITE
P-Asserted-Identity: <sip:41999XXXXXX@30.X.X.X;user=phone>
Contact: <sip:41999XXXXXX@30.X.X.X:5060;user=phone;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R011
Supported: 100rel
Max-Forwards: 69
Content-Length: 269
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 31053767 31053767 IN IP4 30.37.X.Y
s=Sip Call
c=IN IP4 30.37.X.Y
t=0 0
m=audio 55106 RTP/AVP 8 0 18 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=no
<------------->
--- (14 headers 12 lines) ---
Ignoring this INVITE request

<--- Transmitting (no NAT) to 30.X.X.X:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1;received=30.X.X.X
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:85XX@10.1.2.141:5060>
Content-Length: 0


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

<--- SIP read from UDP:30.X.X.X:5060 --->
INVITE sip:85XX@10.37.X.X:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>
CSeq: 1 INVITE
P-Asserted-Identity: <sip:41999XXXXXX@30.X.X.X;user=phone>
Contact: <sip:41999XXXXXX@30.X.X.X:5060;user=phone;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R011
Supported: 100rel
Max-Forwards: 69
Content-Length: 269
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 31053767 31053767 IN IP4 30.37.X.Y
s=Sip Call
c=IN IP4 30.37.X.Y
t=0 0
m=audio 55106 RTP/AVP 8 0 18 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=no
<------------->
--- (14 headers 12 lines) ---
Ignoring this INVITE request

<--- Transmitting (no NAT) to 30.X.X.X:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1;received=30.X.X.X
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:85XX@10.1.2.141:5060>
Content-Length: 0


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

<--- SIP read from UDP:30.X.X.X:5060 --->
INVITE sip:85XX@10.37.X.X:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>
CSeq: 1 INVITE
P-Asserted-Identity: <sip:41999XXXXXX@30.X.X.X;user=phone>
Contact: <sip:41999XXXXXX@30.X.X.X:5060;user=phone;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R011
Supported: 100rel
Max-Forwards: 69
Content-Length: 269
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 31053767 31053767 IN IP4 30.37.X.Y
s=Sip Call
c=IN IP4 30.37.X.Y
t=0 0
m=audio 55106 RTP/AVP 8 0 18 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=no
<------------->
--- (14 headers 12 lines) ---
Ignoring this INVITE request

<--- Transmitting (no NAT) to 30.X.X.X:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1;received=30.X.X.X
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:85XX@10.1.2.141:5060>
Content-Length: 0


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

<--- SIP read from UDP:10.1.2.X:53758 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.141:5060;branch=z9hG4bK69e6ad05
Contact: <sip:5101@10.1.2.X:53758>
To: <sip:5101@10.1.2.X:53758;rinstance=f4179442ee6aa768;transport=UDP>;tag=e6508312
From: "41999XXXXXX" <sip:41999XXXXXX@10.1.2.141>;tag=as4e56f4f0
Call-ID: 6c83674d7fd9332924a6bd1b5de98e96@10.1.2.141:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.2.28 rv2.8.115
Allow-Events: presence, kpml, talk
Content-Length: 592

v=0
o=Z 0 1 IN IP4 10.1.2.X
s=Z
c=IN IP4 10.1.2.X
t=0 0
m=audio 8000 RTP/AVP 0 106 9 3 111 8 97 110 112 102 101 98 100 99
a=rtpmap:106 opus/48000/2
a=fmtp:106 minptime=20; cbr=1; maxaveragebitrate=40000; useinbandfec=1
a=rtpmap:111 speex/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:112 speex/32000
a=rtpmap:102 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:100 telephone-event/16000
a=fmtp:100 0-16
a=rtpmap:99 telephone-event/32000
a=fmtp:99 0-16
a=sendrecv
<------------->
--- (12 headers 23 lines) ---
Found RTP audio format 0
Found RTP audio format 106
Found RTP audio format 9
Found RTP audio format 3
Found RTP audio format 111
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 112
Found RTP audio format 102
Found RTP audio format 101
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 99
Found audio description format opus for ID 106
Found audio description format speex for ID 111
Found audio description format iLBC for ID 97
Found audio description format speex for ID 110
Found audio description format speex for ID 112
Found audio description format G726-32 for ID 102
Found audio description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 98
Found unknown media description format telephone-event for ID 100
Found unknown media description format telephone-event for ID 99
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|gsm|alaw|g722|ilbc|g726|opus|speex|speex16|speex32)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.1.2.X:8000
sip_route_dump: route/path hop: <sip:5101@10.1.2.X:53758>
set_destination: Parsing <sip:5101@10.1.2.X:53758> for address/port to send to
set_destination: set destination to 10.1.2.X:53758
Transmitting (no NAT) to 10.1.2.X:53758:
ACK sip:5101@10.1.2.X:53758 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.141:5060;branch=z9hG4bK41acbd56
Max-Forwards: 70
From: "41999XXXXXX" <sip:41999XXXXXX@10.1.2.141>;tag=as4e56f4f0
To: <sip:5101@10.1.2.X:53758;rinstance=f4179442ee6aa768;transport=UDP>;tag=e6508312
Contact: <sip:41999XXXXXX@10.1.2.141:5060>
Call-ID: 6c83674d7fd9332924a6bd1b5de98e96@10.1.2.141:5060
CSeq: 102 ACK
User-Agent: IPBX-2.11.0(13.22.0)
Content-Length: 0


---
    -- SIP/5101-0000001d answered SIP/4131XXXXXX-0000001c
Audio is at 11978
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 30.X.X.X:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1;received=30.X.X.X
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>;tag=as4f3257f7
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:85XX@10.1.2.141:5060>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 694506252 694506252 IN IP4 10.1.2.141
s=Asterisk PBX 13.22.0
c=IN IP4 10.1.2.141
t=0 0
m=audio 11978 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
    -- Channel SIP/5101-0000001d joined 'simple_bridge' basic-bridge <5998feb8-4f08-402b-907f-c35eaee943fb>
    -- Channel SIP/4131XXXXXX-0000001c joined 'simple_bridge' basic-bridge <5998feb8-4f08-402b-907f-c35eaee943fb>
Retransmitting #1 (no NAT) to 30.X.X.X:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1;received=30.X.X.X
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>;tag=as4f3257f7
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:85XX@10.1.2.141:5060>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 694506252 694506252 IN IP4 10.1.2.141
s=Asterisk PBX 13.22.0
c=IN IP4 10.1.2.141
t=0 0
m=audio 11978 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
Retransmitting #2 (no NAT) to 30.X.X.X:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1;received=30.X.X.X
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>;tag=as4f3257f7
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:85XX@10.1.2.141:5060>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 694506252 694506252 IN IP4 10.1.2.141
s=Asterisk PBX 13.22.0
c=IN IP4 10.1.2.141
t=0 0
m=audio 11978 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


<------------>
Retransmitting #9 (no NAT) to 30.X.X.X:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1;received=30.X.X.X
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>;tag=as4f3257f7
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:85XX@10.1.2.141:5060>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 694506252 694506252 IN IP4 10.1.2.141
s=Asterisk PBX 13.22.0
c=IN IP4 10.1.2.141
t=0 0
m=audio 11978 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
Retransmitting #10 (no NAT) to 30.X.X.X:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 30.X.X.X:5060;branch=z9hG4bKaors8900e0fg8r81n720.1;received=30.X.X.X
From: <sip:41999XXXXXX@30.X.X.X;user=phone>;tag=8nx2wuxa-CC-26
To: <sip:85XX@10.37.X.X;user=phone>;tag=as4f3257f7
Call-ID: nluubj8b16naln82875a10lxbbb91jw8@SoftX3000
CSeq: 1 INVITE
Server: IPBX-2.11.0(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:85XX@10.1.2.141:5060>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 694506252 694506252 IN IP4 10.1.2.141
s=Asterisk PBX 13.22.0
c=IN IP4 10.1.2.141
t=0 0
m=audio 11978 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
[2019-04-10 01:34:47] WARNING[2483]: chan_sip.c:4068 retrans_pkt: Retransmission timeout reached on transmission nluubj8b16naln82875a10lxbbb91jw8@SoftX3000 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
[2019-04-10 01:34:47] WARNING[2483]: chan_sip.c:4092 retrans_pkt: Hanging up call nluubj8b16naln82875a10lxbbb91jw8@SoftX3000 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    -- Channel SIP/4131XXXXXX-0000001c left 'simple_bridge' basic-bridge <5998feb8-4f08-402b-907f-c35eaee943fb>
    -- Channel SIP/5101-0000001d left 'simple_bridge' basic-bridge <5998feb8-4f08-402b-907f-c35eaee943fb>
Scheduling destruction of SIP dialog '6c83674d7fd9332924a6bd1b5de98e96@10.1.2.141:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:5101@10.1.2.X:53758> for address/port to send to
set_destination: set destination to 10.1.2.X:53758
Reliably Transmitting (no NAT) to 10.1.2.X:53758:
BYE sip:5101@10.1.2.X:53758 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.141:5060;branch=z9hG4bK7ad35dca
Max-Forwards: 70
From: "41999XXXXXX" <sip:41999XXXXXX@10.1.2.141>;tag=as4e56f4f0
To: <sip:5101@10.1.2.X:53758;rinstance=f4179442ee6aa768;transport=UDP>;tag=e6508312
Call-ID: 6c83674d7fd9332924a6bd1b5de98e96@10.1.2.141:5060
CSeq: 103 BYE
User-Agent: IPBX-2.11.0(13.22.0)
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
5

Your responses to 30.X.X.X are not getting there or are being so badly mangled on the way that the peer cannot match them to the original INVITE. As a result, the peer is resending that INVITE.

6

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