Calls wont go trough to available agent.

[jessiekidfernando]

Hello Everyone,

We are having issues where answered calls wont go through to any agent. It will just hang up after. Regardless whatever routing extension to use. We are only using one server as dialer/database/webserver.

Here's the agi response I am getting. Anyone?

-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_local_optimize.agi
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_request: agi-VDAD_local_optimize.agi
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_channel: Local/916317918378@default-0000001a;1
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_language: en
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_type: Local
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_uniqueid: 1557858934.157
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_version: 13.21.0-vici
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_callerid: unknown
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_calleridname: V5141435330000064967
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_callingpres: 0
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_callingani2: 0
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_callington: 0
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_callingtns: 0
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_dnid: unknown
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_rdnis: unknown
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_context: default
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_extension: 138367
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_priority: 1
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_enhanced: 0.0
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_accountcode:
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_threadid: 140627828975360
<Local/916317918378@default-0000001a;1>AGI Tx >> agi_arg_1: V5141435330000064967
<Local/916317918378@default-0000001a;1>AGI Tx >>
-- Channel SIP/twilionew-00000027 joined 'simple_bridge' basic-bridge <739fa0ad-ff8f-4282-a53d-669a2c62fb41>
-- Channel Local/916317918378@default-0000001a;2 joined 'simple_bridge' basic-bridge <739fa0ad-ff8f-4282-a53d-669a2c62fb41>
-- <Local/916317918378@default-0000001a;1>AGI Script agi-VDAD_local_optimize.agi completed, returning 0
-- Executing [138367@default:2] Wait("Local/916317918378@default-0000001a;1", "2") in new stack
> 0x7fe6b8084820 -- Strict RTP switching to RTP target address 34.203.250.61:16200 as source
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing [138367@default:3] Hangup("Local/916317918378@default-0000001a;1", "") in new stack
== Spawn extension (default, 138367, 3) exited non-zero on 'Local/916317918378@default-0000001a;1'
[May 14 14:35:37] WARNING[11529][C-00000042]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
-- Executing [h@default:1] AGI("Local/916317918378@default-0000001a;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in


Thanks in advance.

[williamconley]

asterisk manager interface (AMI) version?

public or private network?

[jessiekidfernando]

Hello William,

Asterisk 13.21.0-vici built by root @ ip-172-23-12-178.ec2.internal on a x86_64 running Linux on 2018-08-28 18:14:05 UTC.

That's private network.

[williamconley]

asterisk manager interface (AMI) version?

AMI not Asterisk.

Code: Select all

telnet localhost 5038

Version should be the last line.

Code: Select all

Action: Logoff



Note that you have to hit enter twice to complete the command for logoff.

Is the call in question from off the private network or inside the private network?

If outside, what is NAT set to for this connection/carrier? is sip.conf's externip value set properly? is debug mode set for agi-VDAD_local_optimize.agi? Are you getting this output from the asterisk screen (screen -r asterisk) so you get perl error output?

[jessiekidfernando]

root@ip-172-23-12-178 agc]# telnet localhost 5038
Trying ::1...
telnet: connect to address ::1: Connection refused
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/2.10.4

[jessiekidfernando]

Is the call in question from off the private network or inside the private network? private network

If outside, what is NAT set to for this connection/carrier?
is sip.conf's externip value set properly? yes
is debug mode set for agi-VDAD_local_optimize.agi? how
Are you getting this output from the asterisk screen (screen -r asterisk) so you get perl error output? I am getting this yes from asterisk -r right after enabled the agi (agi set debug on).

[williamconley]

Code: Select all

ps aux | grep SCREEN

which scripts are running in screens?

I am getting this yes from asterisk -r right after enabled the agi (agi set debug on).

Code: Select all

screen -r asterisk

ctrl-a then ctrl-d to exit

this screen provides perl errors, the other does not. (Beware: This is the real asterisk console. If you begin typing and stop ... Vicidial will attempt to type after what you typed and that could be problematic, not to mention what happens if you hit "ctrl-c"!)

Is the call in question from off the private network or inside the private network?

private network

Is NAT set to NO?

[jessiekidfernando]

@Williamconley,

This is was I got. Yes the NAT is set to NO.

Code: Select all

[root@ip-172-23-12-178 centos]# ps aux | grep SCREEN
root      3728  0.0  0.0 127768  1380 ?        Ss   10:58   0:00 /usr/bin/SCREEN -S astshell20190514105645
root      3752  0.0  0.0 128564  2180 ?        Ss   10:58   0:00 SCREEN -L -S asterisk
root      3960  0.0  0.0 127768  1276 ?        Ss   10:58   0:09 /usr/bin/SCREEN -d -m -S ASTupdate /usr/share/astguiclient/AST_update_AMI2.pl
root      3963  0.0  0.0 127768  1276 ?        Ss   10:58   0:04 /usr/bin/SCREEN -d -m -S ASTsend /usr/share/astguiclient/AST_manager_send.pl
root      3966  0.0  0.0 127768  1284 ?        Ss   10:58   0:00 /usr/bin/SCREEN -d -m -S ASTlisten /usr/share/astguiclient/AST_manager_listen_AMI2.pl
root      3969  0.0  0.0 127768  1284 ?        Ss   10:58   0:01 /usr/bin/SCREEN -d -m -S ASTVDauto /usr/share/astguiclient/AST_VDauto_dial.pl
root      3973  0.0  0.0 127768  1280 ?        Ss   10:58   0:00 /usr/bin/SCREEN -d -m -S ASTVDremote /usr/share/astguiclient/AST_VDremote_agents.pl --debug
root      3977  0.0  0.0 127768  1280 ?        Ss   10:58   0:03 /usr/bin/SCREEN -d -m -S ASTVDadapt /usr/share/astguiclient/AST_VDadapt.pl --debug
root      3980  0.0  0.0 127768  1280 ?        Ss   10:58   0:00 /usr/bin/SCREEN -d -m -S ASTfastlog /usr/share/astguiclient/FastAGI_log.pl --debug
root      3985  0.0  0.0 127768  1284 ?        Ss   10:58   0:00 /usr/bin/SCREEN -d -m -S ASTVDadFILL /usr/share/astguiclient/AST_VDauto_dial_FILL.pl --debug
root      3995  0.0  0.0 127768  1280 ?        Ss   10:58   0:00 /usr/bin/SCREEN -d -m -S ASTconf3way /usr/share/astguiclient/AST_conf_update_3way.pl --debug
root      4054  0.0  0.0 112708   980 pts/11   S+   17:05   0:00 grep --color=auto SCREEN
dialer.+ 16898  0.0  0.0 127768  1476 ?        Ss   12:45   0:00 SCREEN -S april16
dialer.+ 16985  0.0  0.0 127768  1476 ?        Ss   12:46   0:00 SCREEN -S april17
dialer.+ 17074  0.0  0.0 127768  1476 ?        Ss   12:46   0:00 SCREEN -S april18
dialer.+ 17184  0.0  0.0 127768  1476 ?        Ss   12:46   0:00 SCREEN -S april19

---

Code: Select all

[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_callerid: unknown
[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_calleridname: V5141707360000064962
[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_callingpres: 0
[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_callingani2: 0
[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_callington: 0
[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_callingtns: 0
[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_dnid: unknown
[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_rdnis: unknown
[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_context: default
[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_extension: 138367
[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_priority: 1
[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_enhanced: 0.0
[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_accountcode:
[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_threadid: 140627828463360
[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_arg_1: V5141707360000064962
[May 14 17:07:37] <Local/916317918378@default-0000001f;1>AGI Tx >>
[May 14 17:07:37]     -- Channel SIP/twilionew-0000002c joined 'simple_bridge' basic-bridge <5bb66cf4-52b8-4222-822b-8543670f6ff9>
[May 14 17:07:37]     -- Channel Local/916317918378@default-0000001f;2 joined 'simple_bridge' basic-bridge <5bb66cf4-52b8-4222-822b-8543670f6ff9>
1557868057.458769
1|cid_channels_recent_172023012178|SHOW TABLES LIKE "cid_channels_recent_172023012178";
SELECT dest_channel FROM cid_channels_recent_172023012178 where (caller_id_name = 'V5141707360000064962' or connected_line_name = 'V5141707360000064962' or caller_id_name = 'V5141707360000064962' or connected_line_name = 'V5141707360000064962') and ( linkedid = '1557868056.183' or dest_uniqueid = '1557868056.183' or uniqueid = '1557868056.183') and call_date > '2019-05-14 17:05:37'
NO DestChannel for callid V5141707360000064962|Local/916317918378@default-0000001f;1!!!
[May 14 17:07:37]     -- <Local/916317918378@default-0000001f;1>AGI Script agi-VDAD_local_optimize.agi completed, returning 0
[May 14 17:07:37]     -- Executing [138367@default:2] Wait("Local/916317918378@default-0000001f;1", "2") in new stack
[May 14 17:07:37]        > 0x7fe6a8019940 -- Strict RTP switching to RTP target address 34.203.250.81:17256 as source
[May 14 17:07:38]   == Manager 'sendcron' logged off from 127.0.0.1
[May 14 17:07:39]     -- Executing [138367@default:3] Hangup("Local/916317918378@default-0000001f;1", "") in new stack
[May 14 17:07:39]   == Spawn extension (default, 138367, 3) exited non-zero on 'Local/916317918378@default-0000001f;1'
[May 14 17:07:39] WARNING[4345][C-0000004c]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[May 14 17:07:39]     -- Executing [h@default:1] AGI("Local/916317918378@default-0000001f;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[May 14 17:07:39] AGI Tx >> agi_network: yes
[May 14 17:07:39] AGI Tx >> agi_network_script: call_log--HVcauses--PRI-----NODEBUG-----16--------------------)
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_request: agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_channel: Local/916317918378@default-0000001f;1
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_language: en
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_type: Local
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_uniqueid: 1557868056.183
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_version: 13.21.0-vici
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_callerid: unknown
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_calleridname: V5141707360000064962
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_callingpres: 0
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_callingani2: 0
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_callington: 0
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_callingtns: 0
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_dnid: unknown
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_rdnis: unknown
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_context: default
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_extension: h
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_priority: 1
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_enhanced: 0.0
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_accountcode:
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >> agi_threadid: 140627828463360
[May 14 17:07:39] <Local/916317918378@default-0000001f;1>AGI Tx >>

[williamconley]

1) Please put code in code blocks. easier to read. I fixed this one. lol

2) There's more than that. You only got most of the agi dump, seems like you started in the middle.

3) I'd like to see the SIP debug as well (or IAX2 debug, whichever)

4) Have you tested dialing without login? Direct to/from phones? Just to be sure your asterisk is properly set up before we dive in to the rest.

5) Does the agent hear "you are the only person in this conference" when logging in?

[jessiekidfernando]

Hello Williamconley,

Apologies if I forgot to place that in a code blocks.

4) Have you tested dialing without login? Direct to/from phones? Just to be sure your asterisk is properly set up before we dive in to the rest. - Yes I was able to provision a softphone using an object type peer. Able to dialout using the defined carrier and no audio issue.

5) Does the agent hear "you are the only person in this conference" when logging in? - Yes. Agent was able to hear that message.

[jessiekidfernando]

2) There's more than that. You only got most of the agi dump, seems like you started in the middle. - Do you need the entire messages? How can I capture that if I am connected via screen -r asterisk?

3) I'd like to see the SIP debug as well (or IAX2 debug, whichever). - Do you want me to produce tcpdump for the sip traces?

[williamconley]

You make it hard to read and differentiate between "your" and "my" text by putting yours at the end of the line. I'm beginning to believe you're one of those people who add their text to the end of the ever-growing email chain in a new color each time until there's no way to make sense of it. lol (yes, I'm picking on you and hoping you being to use the Quote blocks in addition to the Code blocks ).

Screen -r asterisk has a logging facility to hard file, but the console errors for perl may not show up in them. Other options are OS and PuTTY version dependent and a bit more tricky. Ultimately, however, the goal is to capture any/all perl errors that will only show up in that one console (which is part of why that console exists).

TCPDump and/or SIP DEBUG are both viable. The goal is merely to see if there is a stated reason in the SIP handshake for terminating the call vs the more likely reason that the call is still local. If it's still local there are some fairly simple reasons. Wrong asterisk version in /etc/astguiclient.conf or admin->server during perl install.pl can cause loading of the wrong asterisk .conf files, for instance.

[jessiekidfernando]

@Williamconley,

haha you are definitely awesome. Yes I am an old school who likes to add the text on the sender's email (most of the time, I guess haha).

By the way, here's the sip debug.

Code: Select all

<--- SIP read from UDP:54.172.60.0:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 18.215.120.176:5060;received=18.215.120.176;branch=z9hG4bK642d9d26;rport=5060
From: "V5141835330000064875" <sip:+1@18.215.120.176>;tag=as1893c92b
To: <sip:+16317918378@fps-shared-uat-1.pstn.us1.twilio.com>
Call-ID: 201ebcd654cbc53950981d992fac9b1b@18.215.120.176:5060
CSeq: 102 INVITE
Server: Twilio Gateway
Content-Length: 0

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

<--- SIP read from UDP:54.172.60.0:5060 --->
SIP/2.0 200 OK
CSeq: 102 INVITE
Call-ID: 201ebcd654cbc53950981d992fac9b1b@18.215.120.176:5060
From: "V5141835330000064875" <sip:+1@18.215.120.176>;tag=as1893c92b
To: <sip:+16317918378@fps-shared-uat-1.pstn.us1.twilio.com>;tag=93531037_6772d868_a302cc04-e5bb-4150-94a1-68ebea362718
Via: SIP/2.0/UDP 18.215.120.176:5060;received=18.215.120.176;branch=z9hG4bK642d9d26;rport=5060
Record-Route: <sip:54.172.60.0:5060;lr;ftag=as1893c92b>
Server: Twilio
Contact: <sip:172.18.43.244:5060>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY
Content-Type: application/sdp
X-Twilio-CallSid: CA549efcfa2f65f908c909f92f68f2afa5
Content-Length: 240

v=0
o=root 2004511276 2004511276 IN IP4 34.203.251.151
s=Twilio Media Gateway
c=IN IP4 34.203.251.151
t=0 0
m=audio 11862 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 34.203.251.151:11862
sip_route_dump: route/path hop: <sip:54.172.60.0:5060;lr;ftag=as1893c92b>
Transmitting (NAT) to 54.172.60.0:5060:
ACK sip:172.18.43.244:5060 SIP/2.0
Via: SIP/2.0/UDP 18.215.120.176:5060;branch=z9hG4bK082b899a;rport
Route: <sip:54.172.60.0:5060;lr;ftag=as1893c92b>
Max-Forwards: 70
From: "V5141835330000064875" <sip:+1@18.215.120.176>;tag=as1893c92b
To: <sip:+16317918378@fps-shared-uat-1.pstn.us1.twilio.com>;tag=93531037_6772d868_a302cc04-e5bb-4150-94a1-68ebea362718
Contact: <sip:+1@18.215.120.176:5060>
Call-ID: 201ebcd654cbc53950981d992fac9b1b@18.215.120.176:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.21.0-vici
Content-Length: 0


---
[May 14 18:35:36] WARNING[18349][C-00000002]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
Scheduling destruction of SIP dialog '201ebcd654cbc53950981d992fac9b1b@18.215.120.176:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 54.172.60.0:5060:
BYE sip:172.18.43.244:5060 SIP/2.0
Via: SIP/2.0/UDP 18.215.120.176:5060;branch=z9hG4bK538a0e10;rport
Route: <sip:54.172.60.0:5060;lr;ftag=as1893c92b>
Max-Forwards: 70
From: "V5141835330000064875" <sip:+1@18.215.120.176>;tag=as1893c92b
To: <sip:+16317918378@fps-shared-uat-1.pstn.us1.twilio.com>;tag=93531037_6772d868_a302cc04-e5bb-4150-94a1-68ebea362718
Call-ID: 201ebcd654cbc53950981d992fac9b1b@18.215.120.176:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.21.0-vici
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:54.172.60.0:5060 --->
SIP/2.0 200 OK
CSeq: 103 BYE
Call-ID: 201ebcd654cbc53950981d992fac9b1b@18.215.120.176:5060
From: "V5141835330000064875" <sip:+1@18.215.120.176>;tag=as1893c92b
To: <sip:+16317918378@fps-shared-uat-1.pstn.us1.twilio.com>;tag=93531037_6772d868_a302cc04-e5bb-4150-94a1-68ebea362718
Via: SIP/2.0/UDP 18.215.120.176:5060;received=18.215.120.176;branch=z9hG4bK538a0e10;rport=5060
Server: Twilio
X-Twilio-CallSid: CA549efcfa2f65f908c909f92f68f2afa5
Content-Length: 0

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


[williamconley]

Try another carrier. Just for fun. Twilio is an oddity.

But that raises another question: Why you lie to me?

Is the call in question from off the private network or inside the private network?

private network

You are not making "private IP" calls to Twilio. NAT is definitely involved or vicidial is on a public IP. Therefor we must explore firewall and NAT values.

[jessiekidfernando]

Sorry I didnt lie to you.. I have misunderstood your questions. My apologies on that part. When you say we should explore firewall and nat values, why I can make manual outbound calls?

[williamconley]

autodialed calls are invoked differently look at the "vicidial_manager" table. this is a very different invite method compared to accepting a call from one sip account and generating one on another.

but I digress: asterisk version in /etc/astguiclient.conf (and has it ever been changed)? and asterisk version from admin->servers (and has it ever been changed)? particular attention should be paid to any changes after perl install.pl. Also have you recompiled asterisk and (as part of the process) rewritten any of the .conf files? verifying all the .conf files are for the correct version of asterisk/vici is important in any install, but in a manual install it becomes a necessity.

usually a local call will be terminated because sound never initiated. but sometimes the agi script will only think that because versions of files are wrong.

[PManley]

We are seeing a very similar problem at two different sites. In both cases, our agents are outside the Firewall. Also in both cases, using Wireshark on the Agent PC, we see a successful REGISTER transaction between the (Zoiper) softphone and the Vici/Asterisk server. We also see that the Vici/Asterisk server sending a successful SIP OPTIONS transaction. However, upon agent login, the the softphone on the Agent PC is not receiving an INVITE request to initiate the session.

The greater oddity is that once in a while the INVITE request comes through normally.

Is this the same problem? Or should I open a new request?

Any ideas??

[williamconley]

registrations expire.

but routers close ports also.

if those two transactions happen in the wrong order (eg: if the router closes the port before the registration expires ...), the call will fail as the INVITE bounces off the firewall, even though the phone believes it is still registered.

add a keepalive or shorten the keepalive or reduce the registration expiration or increase the open port time in the router.