TIPS inbound dropped calls, help plz !!

kdaffef02 · Jul 3, 2013

I am running an instance with Piaf Purple 2.0644 and it is working fine, but from time to time () , few inbound calls no longer go through (dropped after 2 sec - that's equal to such 20 calls / day of 200) ??. Calls are shown in the growl popup desktop notification, but no trace of such calls in CDR reports ??.

Here is my config :
- ──────────────────SYSTEM INFORMATION *VERIFIED*─────────────────────┐
│ Asterisk = ONLINE | Dahdi = ONLINE | MySQL = ONLINE │
│ SSH = ONLINE | Apache = ONLINE | Iptables = ONLINE │
│ Fail2ban = ONLINE | Internet = ONLINE | Ip6Tables = ONLINE │
│ Disk Free = ADEQUATE| Mem Free = ADEQUATE| NTPD = ONLINE │
│ SendMail = ONLINE | Samba = OFFLINE | Webmin = ONLINE │
│ Ethernet0 = ONLINE | Ethernet1 = N/A | Wlan0 = N/A │
│ │
│ PIAF Installed Version = 2.0.6.4 under *HARDWARE* │
│ FreePBX Version = 2.10.1.9 │
│ Running Asterisk Version = 1.8.22.0 │
│ Asterisk Source Version = 1.8.22.0 │
│ Dahdi Source Version = 2.7.0 │
│ Libpri Source Version = 1.4.12 │
│ IP Address = 192.168.2.25 on eth0 │
│ Operating System = CentOS release 6.4 (Final) │
│ Kernel Version = 2.6.32-358.11.1.el6.i686 - 32 Bit

- Piaf is up to date (programs, source and fixes) yesterday morning with no error,
- Adsl box : everything seems ok (draytek 2820n updated to last fw) - Sip alg : disabled
- ISP hotline says that when calls are presented to Piaf server, trunks seem to be "not connected", so that calls are routed to secondary exterior phone number ... (no thing more),
- All my trunks are connected with no "visible" failure
- All Sip setting are well set with an external ip and a working local network.
- With sip debug on + trunk configuration

Is there any chance to fix this ??. Any help is high appreciated !

Thank you
----------------------------------------------------------------------------------------------------------------------------

Code:

<------------->
[2013-07-03 12:57:40] VERBOSE[1778] chan_sip.c: --- (10 headers 0 lines) ---
[2013-07-03 12:57:40] VERBOSE[1778] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[2013-07-03 12:57:40] VERBOSE[1778] chan_sip.c:
<--- SIP read from UDP:91.121.129.20:5060 --->
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Call-ID: [email protected]
Contact: <sip:10.7.1.49:5060>
Content-Type: application/sdp
CSeq: 15154821 INVITE
From: "0664366197" <sip:[email protected];user=phone>;tag=10989-WZ-00e6ef6a-13b334676
Max-Forwards: 29
Record-Route: <sip:91.121.129.20:5060;lr>
To: <sip:[email protected];user=phone>
Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-RYMC-03db62f7-2865975a
Allow: UPDATE,REFER,INFO
User-Agent: Cirpack/v4.42a (gw_sip)
Content-Length: 445
 
v=0
o=cp10 137284906053 137284906053 IN IP4 10.7.1.132
s=SIP Call
c=IN IP4 91.121.128.139
t=0 0
m=audio 33176 RTP/AVP 0 8 18 4 125 111 101
b=AS:82
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000/1
a=fmtp:4 annexa=no
a=rtpmap:125 CLEARMODE/8000/1
a=rtpmap:111 iLBC/8000/1
a=fmtp:111 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
 
[2013-07-03 12:57:40] VERBOSE[1778] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2013-07-03 12:57:40] VERBOSE[1778] chan_sip.c: Peer audio RTP is at port 91.121.128.139:33176
[2013-07-03 12:57:40] VERBOSE[1778] chan_sip.c: Looking for 0033183646966 in from-trunk (domain xxx.xxx.xx.xxx)
[2013-07-03 12:57:40] VERBOSE[1778] chan_sip.c: list_route: hop: <sip:91.121.129.20:5060;lr>
[2013-07-03 12:57:40] VERBOSE[1778] chan_sip.c:
<--- Transmitting (NAT) to 91.121.129.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-RYMC-03db62f7-2865975a;received=91.121.129.20;rport=5060
Record-Route: <sip:91.121.129.20:5060;lr>
From: "0664366197" <sip:[email protected];user=phone>;tag=10989-WZ-00e6ef6a-13b334676
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 15154821 INVITE
Server: FPBX-2.10.1(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
 
 
<------------>
[2013-07-03 12:57:40] VERBOSE[28889] pbx.c: -- Executing [0033183646966@from-trunk:1] Set("SIP/1-Office-02-000000c4", "__FROM_DID=0033183646966") in new stack
[2013-07-03 12:57:40] VERBOSE[28889] pbx.c: -- Executing [0033183646966@from-trunk:2] Gosub("SIP/1-Office-02-000000c4", "app-blacklist-check,s,1()") in new stack
[2013-07-03 12:57:40] VERBOSE[28889] pbx.c: -- Executing [s@app-blacklist-check:1] GotoIf("SIP/1-Office-02-000000c4", "0?blacklisted") in new stack
[2013-07-03 12:57:40] VERBOSE[28889] pbx.c: -- Executing [s@app-blacklist-check:2] Set("SIP/1-Office-02-000000c4", "CALLED_BLACKLIST=1") in new stack
[2013-07-03 12:57:40] VERBOSE[28889] pbx.c: -- Executing [s@app-blacklist-check:3] Return("SIP/1-Office-02-000000c4", "") in new stack
[2013-07-03 12:57:40] VERBOSE[28889] pbx.c: -- Executing [0033183646966@from-trunk:3] Gosub("SIP/1-Office-02-000000c4", "cidlookup,cidlookup_1,1()") in new stack
[2013-07-03 12:57:40] VERBOSE[28889] pbx.c: -- Executing [cidlookup_1@cidlookup:1] Set("SIP/1-Office-02-000000c4", "CURLOPT(httptimeout)=7") in new stack
[2013-07-03 12:57:45] VERBOSE[1778] chan_sip.c:
<--- SIP read from UDP:91.121.129.20:5060 --->
CANCEL sip:[email protected]:5060;transport=udp SIP/2.0
Call-ID: [email protected]
CSeq: 15154821 CANCEL
From: "0664366197" <sip:[email protected];user=phone>;tag=10989-WZ-00e6ef6a-13b334676
Max-Forwards: 29
To: <sip:[email protected];user=phone>
Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-RYMC-03db62f7-2865975a
Reason: q.850;cause=27
User-Agent: Cirpack/v4.42a (gw_sip)
Content-Length: 0
 
<------------->
[2013-07-03 12:57:45] VERBOSE[1778] chan_sip.c: --- (10 headers 0 lines) ---
[2013-07-03 12:57:45] VERBOSE[1778] chan_sip.c: Sending to 91.121.129.20:5060 (NAT)
[2013-07-03 12:57:45] VERBOSE[1778] chan_sip.c:
<--- Reliably Transmitting (NAT) to 91.121.129.20:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-RYMC-03db62f7-2865975a;received=91.121.129.20;rport=5060
From: "0664366197" <sip:[email protected];user=phone>;tag=10989-WZ-00e6ef6a-13b334676
To: <sip:[email protected];user=phone>;tag=as3f6461d4
Call-ID: [email protected]
CSeq: 15154821 INVITE
Server: FPBX-2.10.1(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
 
 
<------------>
[2013-07-03 12:57:45] VERBOSE[1778] chan_sip.c:
<--- Transmitting (NAT) to 91.121.129.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-RYMC-03db62f7-2865975a;received=91.121.129.20;rport=5060
From: "0664366197" <sip:[email protected];user=phone>;tag=10989-WZ-00e6ef6a-13b334676
To: <sip:[email protected];user=phone>;tag=as3f6461d4
Call-ID: [email protected]
CSeq: 15154821 CANCEL
Server: FPBX-2.10.1(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
 
 
<------------>
[2013-07-03 12:57:45] VERBOSE[1778] chan_sip.c:
<--- SIP read from UDP:91.121.129.20:5060 --->
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Call-ID: [email protected]
CSeq: 15154821 ACK
From: "0664366197" <sip:[email protected];user=phone>;tag=10989-WZ-00e6ef6a-13b334676
Max-Forwards: 29
To: <sip:[email protected];user=phone>;tag=as3f6461d4
Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-RYMC-03db62f7-2865975a
Content-Length: 0

kdaffef02 · Jul 9, 2013

no member by this side ??
Thank you for your help !!

Stewart · Jul 15, 2013

So, your trunks show as registered, but they see you offline? There's a few places that could be a problem. I'd start by asking them why they see you offline. If you are sending registration packets and respond with your ACK, then you are on and registered. They shouldn't be polling you, you should be polling them. I'm assuming the issue isn't the registration but rather that they attempt to initiate the call to you and don't get a response. That would be a firewall issue. Check the logs and see if anything from their IP is being blocked. You may need to open port 5060 limited to just their IP if your firewall isn't functioning the way you need it to.

kdaffef02 · Jul 15, 2013

Thank a lot stewart for your answer.
As i didn't resolved the problem, i tried o une my second server (Piaf 20641 + fpbx 2.90). The resulat, i got just few
dropped let's say 4 - 5 dropped a calls/day). that's better, but i will explore you idea "firewall issue" ?.
- My adsl rouder mode is : Draytek vigor 2820n dualband
I have no specif firewall configuration : just the basic manufacturer one

Question : how we can check its firewall traffic between modem and Piaf ??.
In the other hand, i noticed this in router online table :
ADSL Information ( ADSL Firmware Version: 211011_A)
RX CRC errs 1498

Or may be, you mean by fiewall, Piaf Iptables ??
Thank you for help,

Kamel

kdaffef02 · Jul 17, 2013

Well, i will try to use a traffic capture : Piaf Server <> Adsl Modem with wireshark.

But after reading this other topic, i get "tcpdump: syntax error"

Is that :
"tcpdump -s 0 -w /tmp/mycapture.cap src or dst [ISPIP] and port 5060 or port portrange 10000-20000"

Any help will be very appreciated

kdaffef02 · Jul 23, 2013

Leemason has answered by giving the right syntax for this command (without {}):

{tcpdump -s 0 -w /tmp/mycapture.cap 'udp port 5060 or udp portrange 10000-20000'}

thanks to him,

the 0 between -s and w is zero !

TIPS inbound dropped calls, help plz !!

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