Hi all

I have a system with FreePBX 2.11.0.11, with several Cisco 7970 phones.

Now, i’m trying to add a Cisco cp-8851 with very similar xml configuration, only changing fiwmare (sip88xx.12-1-1SR1-4), but i always get the same message in the phone: “phone is registering” … and then unprovisioned

Here is a portion from the log on the phone:

9679 DEB Nov 19 19:23:20.722080 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_gen_deviceEvent: g_deviceInfo.ins_state=2
9680 NOT Nov 19 19:23:20.722101 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_gen_deviceEvent: event type : SERVER_TRANSPORT
9681 DEB Nov 19 19:23:20.722120 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_gen_deviceEvent: ref_count=1 
9682 DEB Nov 19 19:23:20.722141 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_gen_deviceEvent: name=SEP5006AB0876B6 : privacy=0 : hlog=1 : dnd_state=0 : mwi_lamp=0 
9683 DEB Nov 19 19:23:20.722162 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_gen_deviceEvent: dnd_type=01 : ins_state=02 : cucm_mode=FFFFFFFF : ins_cause=01 
9684 DEB Nov 19 19:23:20.722233 (677:901) JAVA-SNAPSHOT-RELEASE: CCAPI_Device_releaseDeviceInfo:  reference pointer=4580cfd0
9685 NOT Nov 19 19:23:20.722406 (677:902) JAVA-SIP : sip_transport_init_ti_addrs : ip_mode is:0.
9686 NOT Nov 19 19:23:20.722438 (677:902) JAVA-SIPCC-SIP_TRANS: sip_transport_init_ti_addr:  Entered
9687 DEB Nov 19 19:23:20.722469 (677:902) JAVA-PLAT : MED_API : platGetLocalIPAddr :Hi,It contains IPV6_INTEGRATION
9688 DEB Nov 19 19:23:20.722498 (677:902) JAVA-SIPCC-SIP_TRANS: sip_get_local_ip_addr: dst_addr: 8.8.8.8
9689 DEB Nov 19 19:23:20.722524 (677:902) JAVA-SIPCC-SIP_TRANS: sip_get_local_ip_addr: src_addr: 192.168.1.227
9690 DEB Nov 19 19:23:20.722598 (677:902) JAVA-[getDeployMode] deploy-mode:1 
9691 DEB Nov 19 19:23:20.722636 (677:902) JAVA-SIPCC-SIP_TRANS: sip_transport_getaddrinfo: 192.168.1.234 is already an IPv4 address
9692 DEB Nov 19 19:23:20.722660 (677:902) JAVA-PLAT : MED_API : platGetLocalIPAddr :Hi,It contains IPV6_INTEGRATION
9693 NOT Nov 19 19:23:20.722681 (677:902) JAVA-SIPCC-SIP_TRANS: sip_transport_init_ti_addr:  Entered
9694 NOT Nov 19 19:23:20.722702 (677:902) JAVA-SIPCC-SIP_TRANS: sip_transport_init_ti_addr: Unexpected value specified for ip_type : 0
9695 ERR Nov 19 19:23:20.722725 (677:902) JAVA-SIPCC-SIP_TRANS: sip_transport_get_ti_addr: No active CUCM found using primary CUCM
9696 ERR Nov 19 19:23:20.722746 (677:902) JAVA-SIPCC-SIP_TRANS: sip_transport_get_ti_addr: No active CUCM found using primary CUCM
9697 NOT Nov 19 19:23:20.722769 (677:902) JAVA-SIPCC-UI_API: ui_set_ccm_conn_status: ***********CUCM 192.168.1.234 Not connected***********
9698 DEB Nov 19 19:23:20.722873 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_update_ccm_status: entry ccm 192.168.1.234 status=0
9699 DEB Nov 19 19:23:20.722908 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_update_ccm_status: Find cucm with addr= 192.168.1.234,one of the cucms' name=192.168.1.234 (ipv6:INVALID_IPV6)
9700 DEB Nov 19 19:23:20.722934 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_update_ccm_status: server(ipv4) 192.168.1.234 is now status=0, index=0
9701 DEB Nov 19 19:23:20.722968 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_update_ccm_status: Find cucm with addr= 192.168.1.234,one of the cucms' name=INVALID (ipv6:INVALID_IPV6)
9702 DEB Nov 19 19:23:20.722992 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_update_ccm_status: Find cucm with addr= 192.168.1.234,one of the cucms' name=INVALID (ipv6:INVALID_IPV6)
9703 DEB Nov 19 19:23:20.723015 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_update_ccm_status: Find cucm with addr= 192.168.1.234,one of the cucms' name=INVALID (ipv6:INVALID_IPV6)
9704 DEB Nov 19 19:23:20.723035 (677:901) JAVA-SIPCC-SIP_CC_PROV: ccsnap_update_ccm_status: Find cucm with addr= 192.168.1.234,one of the cucms' name=INVALID (ipv6:INVALID_IPV6)
9705 DEB Nov 19 19:23:20.723067 (677:901) JAVA-SNAPSHOT-CREATE: CCAPI_Device_getDeviceInfo:  g_deviceInfo.ins_state=2
9706 DEB Nov 19 19:23:20.723097 (677:901) JAVA-SNAPSHOT-CREATE: CCAPI_Device_getDeviceInfo:  deviceInfo->sis_name=
9707 DEB Nov 19 19:23:20.723118 (677:901) JAVA-SNAPSHOT-CREATE: CCAPI_Device_getDeviceInfo:  reference pointer=4580cfd0
9708 DEB Nov 19 19:23:20.723144 (677:901) JAVA-SNAPSHOT-CREATE: CCAPI_Device_getDeviceInfo:  deviceInfo->ins_state=2

May i need to upgrade to a newer freepbx version? Or what’s the problem? I can give you further details or logs

Could you help me? Thank you

Since i am going to try and use the LE cert to avoid the whole self sign issue – if i go into etc/asterisk/keys i see the “certificatename”.crt

Assume that is the file i upload into the phone

Any other settings need to be adjusted on the Trusted Cert page?

My phone is a T54s Firmware Version 70.84.0.10

I didn’t have to make any modifications on the Trusted Cert page aside of loading the ca.crt file, where I was putting a self-signed certificate into Asterisk.

I can’t tell you exactly what’s going to happen here, because I don’t have a publicly signed cert running against my test server to which I’m connecting the Yealink, and I’m not running your Yealink firmwre version or exact model. I can only speak to my general experience.

One prerequisite, as it pertains to telephones, is time. Deskphones don’t have batteries in them, for good reasons. So, they have to be set to an accurate clock, not January 1, 1970, in order to validate anything using OpenSSL. Make sure your device was able to source time from an NTP server, or via some other method.

Where a server has a publicly signed certificate on it, that includes the full chain, and where a client contains a root CA that can be used to validate that server certificate, everything should be copacetic. The full chain is important, because most client devices that aren’t web browsers, i.e. telephones, will call openssl to validate the server’s cert. OpenSSL will be pointed at some built-in bundle of root CAs, stored on the phone, that it’ll search against. If it finds one that’ll work, everything’s golden.

If the server doesn’t have a publicly signed certificate, your options are to either tell the phone to ignore server certificate validation, or to load the server’s root CA onto the phone - that’s what I did by dropping the ca.crt file from ast_tls_cert into the phone’s web page, Trusted Certificates.

Was posted today.

Very simple: the more beta users/testers, the sooner it’ll be moved to stable.

Can you post a call log of a failed call? Separate internal and external log.

https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs

is it beta or Alpha. Those are two very different things. I have a production system, but will run beta if its more on the stable end.

So I have setup the following:

ILG settings:
Create = 41
01 Group Type: SIP
02 Line Type: Tie
03 Service Type: DIT
04 Private Service Type : Q-SIG
11 DID Digits: 3 (I have 3 digit dialing on the Toshiba system)

NOTE There are a ton of other default settings which were left alone.

OLG settings:
Create = 41
01 Group Type: SIP
02 Trunk Type: Tie/E&M
03 Service Type: Standard

NOTE There are a bunch of other settings there which I left as default.

SIP Trunk:
ID: 3
01 Equipment: 0206
02 LAN interface number: 1
03 SIP Trunk Channels: 4 (Licensed for 6)

Service Def:
ID: 1
01 Registration Mode: Client
02 ILG: 41 (same as above)
03 OLG: 41 (same as above)
04 Effective Channel Number: 4 (licensed for 6 testing 4…not sure if that is my problem)
05 Domain Name: /
06 SIP Server: /
35 SIP Trunk Message Option: SIP Server IP Address
36 SIP Trunk Message to Header OPtion: SIP Server IP Address
37 SIP Trunk Register Message From Header Option: IPU IP Address
38 SIP Trunk Register Message To Header Option: SIP Server IP Address (other option is FQDN) - I am not sure if this is configured properly

Service Assignment:
00 Channel Group: 3
02 Service Index: 1

URI:

00 SIP URI Trunk Service Index: 1
01 SIP URI Index: 1
02 SIP URI: MandPOffice
03 SIP URI Username: MandPOffice
04 SIP URI password:
05 SIP URI Channel Group: 3
06 SIP URI Attribution: main
Then clicked add.

Other URI Entry (just for testing):
01 SIP URI Index: 2
02 SIP URI: 266
03 SIP URI Username: blank
04 SIP URI password: blank
05 SIP URI Channel Group: 3
06 SIP URI Attribution: sub

NOTE The 266 above is an extension from the Toshiba system and does NOT exist in the Switchvox/FreePBX.

Toshiba Flex access code:
00 Access Code: 87
01 Feature Name: Line group access code- one access code for each OLG
02 OLG Number: 41 (as entered above)

Now this is where I differ from…well everyone here. Enter Switchvox land…:
(I see no where to enter PJSIP) - so I enter in a VOIP provider:

UID: same as SIP above
PWD: same as SIP above
IP: of the MIPU card lets just call it (192.168.0.43)
Then I set it to peer rather than provider in the settings.

Now in the server side it is showing SIP connected and “working”. I then also added and inbound and outbound call rule on my Switchvox.

Outbound:
(paraphrase)
If ext 266 is dialed then send call over .

Inbound:
(paraphrase)
If ANY extensions on Switchvox come in on then route to that extension AFTER trimming the first tow digits

NOTE I was not sure if the leading twe digits get sent or not (ie. the felx access code digits of 87).

My results:

1. SIP is connecting between Switchvox & Toshiba
2. Switchvox is NOT able to dial to the Toshiba (error 85 - refers to any failed call over SIP)

• ISDN Error: Received status of CHANUNAVAIL with a cause code of 21

3. Toshiba calling Switchvox recieves a busy error

I noted:

Off the top of my head you may need to enable trunk to trunk transfers in that Toshiba’s cos for that sip trunk (I would make it a cos that you don’t use for anything else as that is not a good idea to turn that on for normal trunks). You may have to setup DISA on the Toshiba in order to make that work.

All indicators are that this is likely an issue with the Toshiba and NOT the Switchvox. Any thoughts?

a) there is an alias root but it is disabled
b) I did remove the trunk-check script.

So now 99.5 of all messages are cleared. Funtionally my problem is solved, but I never addressed the original issue: why are root emails being sent out for delivery and not being sent to the root mailbox locally? This doesn’t appear to be impacting anything besides not being able to see those messages, and unless my Relay exceeds its 24hr limit I’ll never know there’s a problem.

It’s the first time you mention here about a Voicemail issue. You originally asked for helped with upgrading modules.

But if you are asking me, I’d say you should make a full backup of your PBX, and then upgrade to FreePBX 14 (through command line, instructions on the Wiki)

Once upgraded, make sure all modules are up to date, and use it with EPM.

I’m getting a lot better at FreePBX, but this one is weird.

One of our Users has Follow-Me set on their extension. One of a few & it works fine. Normally.

Recently, one of the outside crew called that ext. by DID & instead of ringing through to the cell (via Follow-Me) or going to our internal Voicemail, it went to the VM at the Provider (VoIP.ms). This Provider-VM issue has been a BIG problem in the past, but I thought I’d beaten it into a bit-bucket. Now, other than this oddity, VoIP.ms only takes VMs when we’re actually offline.

From the logs, it would appear the call flow went (Cell#) -> DID -> FollowMe -> VoIP.ms VM. (yes, I know there’s no log for that last step) The two co-workers had exchanged calls a couple of times before & after this moment within a half-hour. I cannot confirm the odd call rang the FM target cell #.

This may not be a Follow-Me problem, but I can’t see any other reason. Obviously we can’t reproduce this on demand, and it only happened the one time.

I’m asking any experts in ancient FPBX who may be reading this to put on your Hypothetical Cap & postulate on what conditions might cause such behavior. I’ll do the heavy lifting, I’d just appreciate a clue as to where I should look.

If it helps, one other oddity I noticed is one time, I dialed an internal 3-digit extension & got the same VoIP.ms VM prompt.

So I’d appreciate a general dissertation on what (other than offline) causes calls to divert to the Provider’s VM instead of getting into FreePBX.

The biggest problem I have is, none of the problems I find are reproducible. Often they only happen one time ever. I’m responsible for answering ‘why’, so thank you in advance for pointing my tired old head into the right quadrant of the Universe.

:^)

Jim

Any firmware after X.80.X.X is supposed to have the LE root cert.

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Well i just dont know then

Can you guys recommend someone who i can pay for an hour or two of their time to vpn into my FPBX system and help me finish this up?

That said, my above issue proves that it does not work. The T46G on the desk in front of me right now has firmware 28.83.0.50.

But if I put a remote address book on a button using https it will fail to load.

<134>Nov 19 13:51:57 GUI [1262:1294]: DIR <6+info  > 117.929.497:CContactRemoteAction::Exec()
<134>Nov 19 13:51:57 GUI [1262:1294]: WIND<6+info  > 117.930.414:[DCMN]create handle success.
<134>Nov 19 13:51:57 GUI [1262:1294]: TASK<6+info  > 117.933.215:set download option ssl trust[1],maxdownload size[1572864] connect timeout[10],download timeout[0]
<134>Nov 19 13:51:57 GUI [1262:1294]: TASK<6+info  > 117.934.181:http download url[https://pbx.mydomain.com/ylmenu.xml] user[****] passwd[****] http_autp[0] method[0]
<134>Nov 19 13:51:57 GUI [1262:1294]: WIND<6+info  > 117.934.515:[DCMN]download to file...
<134>Nov 19 13:51:57 GUI [1262:1294]: WIND<6+info  > 117.934.676:[DCMN]Use new short connect.
<134>Nov 19 13:51:57 GUI [1262:1294]: WIND<6+info  > 117.935.491:[DCMN]HTTP request use auth = 0.
<134>Nov 19 13:51:57 GUI [1262:1294]: WIND<6+info  > 117.935.901:[DCMN]ssl cipher:AES:!ADH:!LOW:!EXPORT:!NULL
<134>Nov 19 13:51:57 GUI [1262:1294]: WIND<6+info  > 117.936.248:[DCMN]I will write to file: /tmp/RemoteDirectory.0000000165.xml
<134>Nov 19 13:51:57 GUI [1262:1262]: POPB<6+info  > 117.947.799:CMsgBoxManager::AddMessageBox
<134>Nov 19 13:51:57 GUI [1262:1262]: POPB<6+info  > 117.948.336:Add CCommonMessageBox::PrintMsgBoxInfo: ?`23Loading, please wait...
<134>Nov 19 13:51:57 GUI [1262:1262]: POPB<6+info  > 117.948.754:CMsgBoxManager Update MsgBox Info
<134>Nov 19 13:51:57 GUI [1262:1262]: BKLT<6+info  > 117.949.455:BackLightManager::AddEvent [1]
<134>Nov 19 13:51:57 GUI [1262:1262]: UIMG<6+info  > 117.967.509:[UIManager] Dialog(CDlgPickupMenu) is not cached or not used!
<134>Nov 19 13:51:57 GUI [1262:1262]: IDUI<6+info  > 117.968.260:QEvent::Hide
<134>Nov 19 13:51:57 GUI [1262:1262]: DIRU<6+info  > 117.970.784:OnBrowserUIShow(FrameList:0x24bdd0)
<134>Nov 19 13:51:57 GUI [1262:1262]: DIRU<6+info  > 117.971.640:CDirectoryListDelegate::OnLoadData(0x24bdd0 0 5)
<134>Nov 19 13:51:57 GUI [1262:1262]: DIRU<6+info  > 117.974.414:CCommonDirCtrl::LoadListData(Index:0 Count:5)
<134>Nov 19 13:51:57 GUI [1262:1262]: DIRU<6+info  > 117.974.889:Dir_GetContactListByGroupId(Type:5 GroupId:2 Count:5 Index:0)
<134>Nov 19 13:51:57 GUI [1262:1262]: DIR <6+info  > 117.975.430:GetContactList pData->m_nTotalCount[0]
<134>Nov 19 13:51:57 GUI [1262:1262]: DIRU<6+info  > 117.975.826:CCommonUIHelper::FillData(TotalCount:0 GetCount:0 Index:0)
<134>Nov 19 13:51:58 GUI [1262:1294]: WIND<6+info  > 118.008.537:[DCMN]CURL Info: TLSv1.2 (OUT), TLS handshake, Client hello (1):
<134>Nov 19 13:51:58 ipp [1145]: IIPP<6+info  > 118.013.264:EptEvent event 2 streamid 4 slot_num 65535 index 0
<134>Nov 19 13:51:58 GUI [1262:1294]: WIND<6+info  > 118.074.764:[DCMN]CURL Info: TLSv1.2 (IN), TLS handshake, Server hello (2):
<134>Nov 19 13:51:58 GUI [1262:1294]: WIND<6+info  > 118.075.933:[DCMN]CURL Info: TLSv1.2 (IN), TLS handshake, Certificate (11):
<134>Nov 19 13:51:58 GUI [1262:1294]: WIND<6+info  > 118.175.279:[DCMN]CURL Info: TLSv1.2 (IN), TLS handshake, Server key exchange (12):
<134>Nov 19 13:51:58 GUI [1262:1262]: BKLT<6+info  > 118.208.114:Operation[1] status[0] IsFailBackMode[0] to power saving
<134>Nov 19 13:51:58 GUI [1262:1262]: GBIH<6+info  > 118.209.767:Global Key Up:(1000031)
<134>Nov 19 13:51:58 GUI [1262:1262]: BKLT<6+info  > 118.210.509:BackLightManager::AddEvent [1]
<134>Nov 19 13:51:58 GUI [1262:1262]: BKLT<6+info  > 118.225.374:Operation[0] status[0] IsFailBackMode[0] to power saving
<134>Nov 19 13:51:58 GUI [1262:1262]: BKLT<6+info  > 118.226.493:Operation[1] status[0] IsFailBackMode[0] to power saving
<134>Nov 19 13:51:58 GUI [1262:1294]: WIND<6+info  > 118.270.908:[DCMN]CURL Info: TLSv1.2 (IN), TLS handshake, Server finished (14):
<134>Nov 19 13:52:30 GUI [1262:1294]: WIND<6+info  > 150.920.802:[DCMN]CURL Info: TLSv1.2 (OUT), TLS handshake, Client key exchange (16):
<134>Nov 19 13:52:30 GUI [1262:1294]: WIND<6+info  > 150.923.071:[DCMN]CURL Info: TLSv1.2 (OUT), TLS change cipher, Client hello (1):
<134>Nov 19 13:52:30 GUI [1262:1294]: WIND<6+info  > 150.927.044:[DCMN]CURL Info: TLSv1.2 (OUT), TLS handshake, Finished (20):
<134>Nov 19 13:52:30 GUI [1262:1294]: WIND<6+info  > 150.928.590:[DCMN]CURL Info: Unknown SSL protocol error in connection to pbx.mydomain.com:443 
<134>Nov 19 13:52:30 GUI [1262:1294]: WIND<6+info  > 150.934.848:[DCMN]Connect is short Cleanup curl.
<131>Nov 19 13:52:30 GUI [1262:1294]: WIND<3+error > 150.936.293:[DCMN]download common error, errcode:35.
<134>Nov 19 13:52:30 GUI [1262:1294]: WIND<6+info  > 150.936.834:[DCMN]download common error, remove file.

The PCAP captured by the phone itself shows the Server key exchange and hello completing, which matches the above log.

Then some weird 20+ seconds delay while it fails to figure out the client key and cipher.

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I’m testing a T42S right now. it seems to be working. Give me a minute.

I will say I did demo Switchvox and I had issues getting the SIP trunk to work at all, which is why I went with FreePBX. I wasn’t sure how to add it in order for it to work correctly.

In terms of dialing, are you dialing from Toshiba -> Switchvox by dialing 87+Ext?

In terms of Switchvox dialing to Toshiba, is there any options to set the context? from-internal?

The SIP is showing connected on Switchvox but it appears based on the logs that the Toshiba is blocking the call. The Toshiba is acking the SIP call from Switchvox and then rejecting it. I am wondering is this might be an issue of the extensions internally not “seeing” a proper path to SIP trunk… (on the Toshiba side)

Toshiba’s calls when dialing 87+ext are not working at all. In fact their call is not even hitting the Switchvox.

I have also noted the following error in the switchvox logs:

res_pjsip_registrar.c: AOR ‘MandPOffice’ not found for endpoint ‘sip_provider_104’

Oh and for the record I too prefer FreePBX Image may be NSFW.
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Ok, just tested everything and figured it out.

The Yealink T4XS series only work when you set the SSL method to SSLv2 or SSLv3
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Once I changed that and rebooted the phone registered.

I then changed the RTP encryption settings in the extension I was testing.
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And updated the phone to this.

account.1.sip_server.1.address = pbx.mydomain.com
account.1.sip_server.1.port = 5061
account.1.password = yourpassword
account.1.user_name = yourextension
account.1.sip_server.1.transport_type = 2
account.1.srtp_encryption = 2

Is there a way to have the phone ring when using a “System Recording”? I have a small plumbing service business I run from home & would like to know when someone is calling like it does when using voicemail during business hours. We use the system recordings when we close for holidays and such. It goes straight to voicemail now.
Is there a better way to do this? Right now all calls goes to 1 extension with a voicemail.

Here is what is set up for special hours when we are closed:
Announcements > Our Message > Destination : VoiceMail - Ext. 100 (Instructions Only)

It is Alpla.