QUESTION Aastra 57i working remotely

[LesD]

I am trying to get one of my local phones to register with a a remote new system (Virtualbox VM) I am configuring and it is not registering.

I have never tried setting up a remote phone before.

I have temporarily turned off iptables (service iptables stop) in the VM.

On the remote router I am forwarding ports 10000-20000, 5060 and 69 (t*f*t*p) to the VM.

I have also used Bulk Extensions export from my local system and imported it to the VM so the extensions there should be identical to those on my local system - including the same passwords.

Also, I have cloned the contents of my tftpboot folder to the remote VM so the cfg files are all there if needed.

I then took one of my working 57i phones and made the following changes to the config via the browser:

Line 1:
Proxy server: FQDN of remote site
Registrar server: FQDN of remote site

Configuration Server:
t*f*t*p server: FQDN of remote site

Anything I have overlooked?

 

[LesD]

Done that.

Forgot to look before in the log before but it is now showing the following errors:

[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/sip_additional.conf': Found
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/sip_custom_post.conf': Found
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/501.inc': Found
[2014-06-08 00:44:41] WARNING[28701] config.c: Category addition requested, but category '501' does not exist, line 2 of /etc/asterisk/501.inc
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/701.inc': Found
[2014-06-08 00:44:41] WARNING[28701] config.c: Category addition requested, but category '701' does not exist, line 2 of /etc/asterisk/701.inc
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/702.inc': Found
[2014-06-08 00:44:41] WARNING[28701] config.c: Category addition requested, but category '702' does not exist, line 2 of /etc/asterisk/702.inc
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/703.inc': Found
[2014-06-08 00:44:41] WARNING[28701] config.c: Category addition requested, but category '703' does not exist, line 2 of /etc/asterisk/703.inc
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/704.inc': Found
[2014-06-08 00:44:41] WARNING[28701] config.c: Category addition requested, but category '704' does not exist, line 2 of /etc/asterisk/704.inc
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/705.inc': Found
[2014-06-08 00:44:41] WARNING[28701] config.c: Category addition requested, but category '705' does not exist, line 2 of /etc/asterisk/705.inc
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/706.inc': Found
[2014-06-08 00:44:41] WARNING[28701] config.c: Category addition requested, but category '706' does not exist, line 2 of /etc/asterisk/706.inc
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/707.inc': Found
[2014-06-08 00:44:41] WARNING[28701] config.c: Category addition requested, but category '707' does not exist, line 2 of /etc/asterisk/707.inc
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/708.inc': Found
[2014-06-08 00:44:41] WARNING[28701] config.c: Category addition requested, but category '708' does not exist, line 2 of /etc/asterisk/708.inc
[2014-06-08 00:44:41] WARNING[28701] config.c: Maximum Include level (10) exceeded
[2014-06-08 00:44:41] ERROR[28701] config.c: The file '709.inc' was listed as a #include but it does not exist.
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/users.conf': Found
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/phoneprov.conf': Found
[2014-06-08 00:44:41] VERBOSE[28701] loader.c: -- Reloading module 'res_rtp_asterisk.so' (Asterisk RTP Stack)
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/rtp.conf': Found
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/rtp_additional.conf': Found
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/rtp_custom.conf': Found
[2014-06-08 00:44:41] VERBOSE[28701] res_rtp_asterisk.c: == RTP Allocating from port range 10000 -> 20000
[2014-06-08 00:44:41] VERBOSE[28701] loader.c: -- Reloading module 'res_stun_monitor.so' (STUN Network Monitor)
[2014-06-08 00:44:41] VERBOSE[28701] loader.c: -- Reloading module 'res_xmpp.so' (Asterisk XMPP Interface)
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/xmpp.conf': Found
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/xmpp_general_custom.conf': Found
[2014-06-08 00:44:41] VERBOSE[28701] config.c: == Parsing '/etc/asterisk/xmpp_custom.conf': Found
[2014-06-08 00:45:47] NOTICE[1569] acl.c: SIP Peer ACL: Rejecting 'xx.xx.xx.xx' due to a failure to pass ACL '(BASELINE)'
[2014-06-08 00:45:47] NOTICE[1569] chan_sip.c: Registration from '<sip:[email protected]:5060>' failed for 'xx.xx.xx.xx:44292' - Device does not match ACL

The last error is repeated an other 3 times.

So there are two sets of errors. Not sure if the 1st is causing the 2nd or not.

My googling has not yet come up with anything I understand.

 

[LesD]

Managed to get it to register. On the extension definition, I had to add 'aa.bb.cc.dd/255.255.255.255' in the 'permit' field where aa.bb.cc.dd is the IP from where the phone is registering.

TM3 turned back on and so far it still connects.

Is the first error I listed above (ERROR[28701] config.c: The file '709.inc' was listed as a #include but it does not exist.) of any significance?

 

[LesD]

The next problem is that I can't make any outgoing calls.

The log (below) shows something very strange.

Instead of sending the call to an outbound route, it seems to be fed through as an Asteridex lookup which then fails. I hear nothing on the phone and the CDR shows the call as answered with "Channel:" blank.

I must have done something silly somewhere.

[2014-06-08 16:05:44] VERBOSE[1569][C-00000009] netsock2.c:  == Using SIP RTP TOS bits 184
[2014-06-08 16:05:44] VERBOSE[1569][C-00000009] netsock2.c:  == Using SIP RTP CoS mark 5
[2014-06-08 16:05:44] VERBOSE[4849][C-00000009] pbx.c:    -- Executing [004420xxxxxxxx@from-internal:1] SayDigits("SIP/214-00000009", "4420xxxxxxxx,") in new stack
[2014-06-08 16:05:44] VERBOSE[4849][C-00000009] file.c:    -- <SIP/214-00000009> Playing 'digits/4.gsm' (language 'en')
[2014-06-08 16:05:44] VERBOSE[4849][C-00000009] file.c:    -- <SIP/214-00000009> Playing 'digits/4.gsm' (language 'en')
[2014-06-08 16:05:45] VERBOSE[4849][C-00000009] file.c:    -- <SIP/214-00000009> Playing 'digits/2.gsm' (language 'en')
[2014-06-08 16:05:46] VERBOSE[4849][C-00000009] file.c:    -- <SIP/214-00000009> Playing 'digits/0.gsm' (language 'en')
[2014-06-08 16:05:47] VERBOSE[4849][C-00000009] file.c:    -- <SIP/214-00000009> Playing 'digits/x.gsm' (language 'en')
[2014-06-08 16:05:47] VERBOSE[4849][C-00000009] file.c:    -- <SIP/214-00000009> Playing 'digits/x.gsm' (language 'en')
[2014-06-08 16:05:48] VERBOSE[4849][C-00000009] file.c:    -- <SIP/214-00000009> Playing 'digits/x.gsm' (language 'en')
[2014-06-08 16:05:48] VERBOSE[4849][C-00000009] file.c:    -- <SIP/214-00000009> Playing 'digits/x.gsm' (language 'en')
[2014-06-08 16:05:49] VERBOSE[4849][C-00000009] file.c:    -- <SIP/214-00000009> Playing 'digits/x.gsm' (language 'en')
[2014-06-08 16:05:50] VERBOSE[4849][C-00000009] file.c:    -- <SIP/214-00000009> Playing 'digits/x.gsm' (language 'en')
[2014-06-08 16:05:51] VERBOSE[4849][C-00000009] file.c:    -- <SIP/214-00000009> Playing 'digits/x.gsm' (language 'en')
[2014-06-08 16:05:51] VERBOSE[4849][C-00000009] file.c:    -- <SIP/214-00000009> Playing 'digits/x.gsm' (language 'en')
[2014-06-08 16:05:52] VERBOSE[4849][C-00000009] pbx.c:    -- Executing [004420xxxxxxxx@from-internal:2] EAGI("SIP/214-00000009", "asteridex.agi,004420xxxxxxxx") in new stack
[2014-06-08 16:05:52] VERBOSE[4849][C-00000009] res_agi.c:    -- Launched AGI Script /var/lib/asterisk/agi-bin/asteridex.agi
[2014-06-08 16:05:52] VERBOSE[4849][C-00000009] res_agi.c: asteridex.agi,004420xxxxxxxx: EXTEN IS: 4420xxxxxxxx
[2014-06-08 16:05:52] VERBOSE[4849][C-00000009] res_agi.c: asteridex.agi,004420xxxxxxxx: Ready for AsteriDex lookup...
[2014-06-08 16:05:52] VERBOSE[4849][C-00000009] res_agi.c: asteridex.agi,004420xxxxxxxx: Unable to find an AsteriDex match.
[2014-06-08 16:05:52] VERBOSE[4849][C-00000009] res_agi.c:    -- <SIP/214-00000009>AGI Script asteridex.agi completed, returning 0
[2014-06-08 16:05:52] VERBOSE[4849][C-00000009] pbx.c:    -- Executing [004420xxxxxxxx@from-internal:3] GotoIf("SIP/214-00000009", "1?97") in new stack
[2014-06-08 16:05:52] VERBOSE[4849][C-00000009] pbx.c:    -- Goto (from-internal,004420xxxxxxxx,97)
[2014-06-08 16:05:52] VERBOSE[4849][C-00000009] pbx.c:    -- Executing [004420xxxxxxxx@from-internal:97] Playback("SIP/214-00000009", "num-not-in-db") in new stack
[2014-06-08 16:05:52] VERBOSE[4849][C-00000009] file.c:    -- <SIP/214-00000009> Playing 'num-not-in-db.gsm' (language 'en')
[2014-06-08 16:05:54] VERBOSE[4849][C-00000009] pbx.c:    -- Executing [004420xxxxxxxx@from-internal:98] Playback("SIP/214-00000009", "goodbye") in new stack
[2014-06-08 16:05:54] VERBOSE[4849][C-00000009] file.c:    -- <SIP/214-00000009> Playing 'goodbye.gsm' (language 'en')
[2014-06-08 16:05:55] VERBOSE[4849][C-00000009] pbx.c:    -- Executing [004420xxxxxxxx@from-internal:99] Hangup("SIP/214-00000009", "") in new stack
[2014-06-08 16:05:55] VERBOSE[4849][C-00000009] pbx.c:  == Spawn extension (from-internal, 004420xxxxxxxx, 99) exited non-zero on 'SIP/214-00000009'
[2014-06-08 16:05:55] VERBOSE[4849][C-00000009] pbx.c:    -- Executing [h@from-internal:1] Hangup("SIP/214-00000009", "") in new stack
[2014-06-08 16:05:55] VERBOSE[4849][C-00000009] pbx.c:  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/214-00000009'
[2014-06-08 16:06:08] WARNING[1569] chan_sip.c: Retransmission timeout reached on transmission 1d5ac12e2a82cbe8 for seqno 30407 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 15809ms with no response

PS: I have an outgoing trunk defined but not yet an incoming one. I have both an incoming and an outgoing route defined.

 

[LesD]

Thanks. Yes, trying a number without the 00 works.

extensions_custom.conf does contain a block of entries

exten => _00.,1,SayDigits(${EXTEN:2},) ; extensions dialed with 00 prefix get looked up in AsteriDex
exten => _00.,2,EAGI(asteridex.agi,${EXTEN})
exten => _00.,3,GotoIf($["${DIAL:0:2}" = "00"]?97)
exten => _00.,4,NoOp(Number to Dial: ${DIAL})
exten => _00.,5,NoOp(Person to Dial: ${DUDE})
exten => _00.,6,Flite("Connecting to: ${DUDE}. One moment please.")
exten => _00.,7,Goto(outbound-allroutes,${DIAL},1)
exten => _00.,8,Hangup()
exten => _00.,97,Playback(num-not-in-db)
exten => _00.,98,Playback(goodbye)
exten => _00.,99,Hangup()

I do now remember hitting this problem before but that was about a century ago - or if not that then at least in 2010/11.

It should really get removed from the current builds. Even in the US they have numbers starting 00 so it must be biting others as well.

I have removed that block and 00 now works.

Thanks again.

 

[LesD]

Hopefully this will be the last problem.

I can now make a call on the phone I have remotely connected to the remote system and I call my own number and indeed the phone rings here.

Someone picks it up and there is silence at both ends.

But I do not think this is the normal one-way sound problem.

The CDR shows the call coming into the ring-group and being answered but there is no following CDR entry showing the extension that picked up the call.

It is as if the call just dies at the ring group - but not till it is actually picked up (I did try leaving it to ring for a while and it did continue to ring till the phone was picked up).

Extract from the Remote end log: (214 is the extension number of the remotely registered phone making the outgoing call)

[2014-06-08 19:57:31] VERBOSE[8451][C-00000001] netsock2.c:  == Using SIP RTP TOS bits 184
[2014-06-08 19:57:31] VERBOSE[8451][C-00000001] netsock2.c:  == Using SIP RTP CoS mark 5
[2014-06-08 19:57:31] VERBOSE[8451][C-00000001] app_dial.c:    -- Called SIP/DesserVoipOut-T/004420xxxxxxxx
[2014-06-08 19:57:31] VERBOSE[8451][C-00000001] app_dial.c:    -- SIP/DesserVoipOut-T-00000003 is making progress passing it to SIP/214-00000002
[2014-06-08 19:57:46] VERBOSE[8451][C-00000001] app_dial.c:    -- SIP/DesserVoipOut-T-00000003 answered SIP/214-00000002
[2014-06-08 19:57:55] VERBOSE[8451][C-00000001] pbx.c:    -- Executing [h@macro-dialout-trunk:1] Macro("SIP/214-00000002", "hangupcall,") in new stack
[2014-06-08 19:57:55] VERBOSE[8451][C-00000001] pbx.c:    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/214-00000002", "1?theend") in new stack
[2014-06-08 19:57:55] VERBOSE[8451][C-00000001] pbx.c:    -- Goto (macro-hangupcall,s,3)
[2014-06-08 19:57:55] VERBOSE[8451][C-00000001] pbx.c:    -- Executing [s@macro-hangupcall:3] ExecIf("SIP/214-00000002", "0?Set(CDR(recordingfile)=)") in new stack
[2014-06-08 19:57:55] VERBOSE[8451][C-00000001] pbx.c:    -- Executing [s@macro-hangupcall:4] Hangup("SIP/214-00000002", "") in new stack
[2014-06-08 19:57:55] VERBOSE[8451][C-00000001] app_macro.c:  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/214-00000002' in macro 'hangupcall'
[2014-06-08 19:57:55] VERBOSE[8451][C-00000001] pbx.c:  == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/214-00000002'
[2014-06-08 19:57:55] VERBOSE[8451][C-00000001] app_macro.c:  == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/214-00000002' in macro 'dialout-trunk'
[2014-06-08 19:57:55] VERBOSE[8451][C-00000001] pbx.c:  == Spawn extension (from-internal, xxxxxxxx, 6) exited non-zero on 'SIP/214-00000002'
[2014-06-08 19:57:58] WARNING[8363] chan_sip.c: Retransmission timeout reached on transmission 8b1b9eb2aefedf59 for seqno 8739 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 12481ms with no response

I have not been able to pick out anything strange from the log at my end. There seems to be a simple hangup.

(I do have an an other issue which may or may not be related. The CID showing up on the phone when the call comes in is a random numerical string. The log at the far end shows the CID correctly. The log at my end shows the wrong CID. I suspect this is a service provider issue).

I have looked at https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions but do not really understand it.

 

[LesD]

I have simulated an outgoing call using the Wakeup call module and it seems to work fine - at least I can hear the message and key press tones work properly.

So this issue must still be to do with how this phone is working remotely.