@pbx wrote:

I have a server with Asterisk 12.0.76.4 and 6.12.65-32. When I try to download the PBX Updater module I get the error. I have tried to manually run updates as below but does not make any difference. I assume I need to somehow update the database to reflect the correct version so I can continue with the upgrade? I have installed a fresh distro and restored backup but no difference.

amportal a ma download framework
amportal a ma install framework
amportal a ma download core
amportal a ma install core

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@digiteltlc wrote:

Perhaps strange question.....

My extension is ringing
Can I place an incoming call throuhg a SIP trunk to pickup my extension ringing ??

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@andersonhaulage wrote:

I went to update my system and noticed 3 of my modules had the orange "Renew" button. Great, renewed all 3 (EPM, Ext. Routing and Phone Apps). The activation has changed to reflrct the 1 year of updates. However, it also shows REST API having updates only until 2016-10-08. I can't seem to find how to renew this.

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@amztransit wrote:

I'm toying with the idea of moving our setup to a VPS.

I've tried the backup/restore module using our current setup and a VM to test migration. I can't get it to transfer over. Even dragging over all of the templates in backup, and then restoring, I can't even get so much as our extensions to load in.

Is using all 6 templates appropriate for a migration? Anything else I might be overlooking? Version is the same, but modules have not been loaded into the new VM setup. Hoping we can use backup/restore to help with that.

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@zirophyz wrote:

Hi all,

So, I've built my fair share of applications, call flows, menu's and queues in Cisco UCCX scripts and one thing I use often is an IF statement to take action based on a certain condition or data. I don't really know where to start in order to get the same sort of thing in FreePBX, nor do I even know what to search for!

The idea is that a menu will give callers several unique options, for example, an English School's Sales department has options for students ringing from China, South America, Europe etc. Each 'region' selection puts you through to a single agent, the expert in that market. In reality, the options destination is a ring-group with one (maybe two) extensions/agents in it. If they miss the call, the caller gets a new menu to either leave a voicemail to speak to someone else (that option sends callers to another "General Sales" ring group).

The problem is - if a caller wants to leave a voicemail message; how would I be able to identify which IVR selection they made so as to send them to the correct mailbox?

In a UCCX script, I would set some data on that call - if they selected China, then I could set a variable Country="China", and later in the script use an IF statement such as "If (Country==China) is True then Redirect Call to China_Sales_Voicemail".

Is there some way that I could do something similar in FreePBX? I have some ideas in my head such as;
- Duplicate the IVR for each set of agents so that I can set the correct mailbox option (seems inefficient to have like 5 of the same menu)
- Create a group mailbox, and allow Users to access this Voicemail in their UCP (this could annoy users, as the South America agents would get false positives for China VM's etc).
- Voicemail Blasting?

Here's a crude diagram of what I am trying to achieve.

Thanks in advance,

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@ericbramini wrote:

Hey All,

I am noticing that UCP new user emails and password reset emails contain https in the url and then specify the non-secure port that I specify in port management. If I specify port 81, 80, etc, my browser chokes on accessing a secure protocol using an insecure port. I tried disabling the non-secure port in port management but then I get this:

To login to the following services:

    User Control Panel: [https://pbx.mydomain.com:disabled]https://pbx.mydomain.com:disabled

Password Reset Link (Valid Until: 10:21:49 AM):
[https://pbx.mydomain.com:disabled/?forgot=9ea427cfb78bee26aa2a7aed1d1d27dd]https://pbx.mydomain.com:disabled/?forgot=9ea427cfb78bee26aa2a7aed1d1d27dd

I am at the latest modules that could possibly have something to do with this functionality:
Framework (13.0.188.9)
UCP (13.0.41.2)
User Mangement (13.0.73.3)
Sysadmin (13.0.67)

I wish it would choose to publish the secure ports if specified and then the https protocol in the URL would be happy. If others have similar problems, I will submit it as a bug but I didn't want to fire it off without asking around first in-case I was causing this somehow.

-Eric

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@sentinelace wrote:

I have a hosted Freepbx. I get random calls from extension "abc123" that do not exist. I have the firewall setup to only allow connections to our IP. Any idea why this is happening?

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@neuronetv wrote:

I don't have £500 to spend on a digium pci card and I was wondering if there are cheaper alternatives worth buying. I only want one or two fxo ports on a legacy pci card (not pci-e). I've seen some cards on ebay around the £30 - £50 mark (tdm110p, tdm400p) but I wondered has anyone tried these? are they any good and do they work ok in freepbx 13? I could afford up to £150 if there are good alternatives but I know little about what's available in that price range. thanks for any advice.

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@pkozul wrote:

Hi there,

If my video intercom (ext 8001) calls my laptop (ext 8005), the call includes video and all works well. But if I do things the other way around (laltop calls video intercom), the call does not include video. I have included the debug info below showing the call without video.

I can see an error in there:

No joint capabilities for 'video' media stream between our configuration((h264|h263p|h263|h261)) and incoming SDP((ulaw|ilbc))

BTW, since I am using PJSIP, I needed to manually enable video for each extension by setting each extension's Allowed Codecs setting to 'h264&h263p&h263&h261&ulaw&alaw&gsm&g726'. In other words, I needed to explicitly list h264 in there to get any video working at all.

Any ideas?

Thanks,
Pete

<--- Received SIP request (641 bytes) from UDP:192.168.1.1:5060 --->
OPTIONS sip:mydomain.net SIP/2.0
Call-ID: 1dcfce881510532bf6eaa37f217dc761@0:0:0:0:0:0:0:0
CSeq: 1484 OPTIONS
From: "8005" <sip:8005@mydomain.net>;tag=7b49e2c5
To: "8005" <sip:8005@mydomain.net>
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-353534-3998fdb1f040812d7bd6512ada2eeb4f
Max-Forwards: 70
Contact: "8005" <sip:8005@192.168.1.1:5060;transport=udp;registering_acc=mydomain_net>
User-Agent: Jitsi2.8.5426Windows 8
Allow: INFO,UPDATE,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE
Allow-Events: refer,conference,remote-control,presence,presence.winfo,message-summary
Content-Length: 0


<--- Transmitting SIP response (575 bytes) to UDP:192.168.1.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;received=192.168.1.1;branch=z9hG4bK-353534-3998fdb1f040812d7bd6512ada2eeb4f
Call-ID: 1dcfce881510532bf6eaa37f217dc761@0:0:0:0:0:0:0:0
From: "8005" <sip:8005@mydomain.net>;tag=7b49e2c5
To: "8005" <sip:8005@mydomain.net>;tag=z9hG4bK-353534-3998fdb1f040812d7bd6512ada2eeb4f
CSeq: 1484 OPTIONS
WWW-Authenticate: Digest  realm="asterisk",nonce="1476602018/a24308ade57c599a10f6ac5fc78c3371",opaque="71df285a0c4bcc2d",algorithm=md5,qop="auth"
Server: FPBX-13.0.188.8(13.11.2)
Content-Length:  0


<--- Received SIP request (1424 bytes) from UDP:192.168.1.1:5060 --->
INVITE sip:8001@mydomain.net SIP/2.0
Call-ID: a12a40a29bdf6119d9a33f8a95524db6@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
From: "8005" <sip:8005@mydomain.net>;tag=632a848f
To: <sip:8001@mydomain.net>
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-353534-553fa82d926557ea2b0c35cba85a5af3
Max-Forwards: 70
Contact: "8005" <sip:8005@192.168.1.1:5060;transport=udp;registering_acc=mydomain_net>
User-Agent: Jitsi2.8.5426Windows 8
Content-Type: application/sdp
Content-Length: 933

v=0
o=8005-jitsi.org 0 0 IN IP4 192.168.1.1
s=-
c=IN IP4 192.168.1.1
t=0 0
m=audio 5114 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 4 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 usedtx=1
a=rtpmap:97 SILK/24000
a=rtpmap:98 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:100 speex/32000
a=rtpmap:102 speex/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:104 speex/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=rtpmap:101 telephone-event/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics
m=video 5116 RTP/AVP 105 99
a=recvonly
a=rtpmap:105 H264/90000
a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1
a=imageattr:105 send * recv [x=[0-1366],y=[0-768]]
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01f
a=imageattr:99 send * recv [x=[0-1366],y=[0-768]]

<--- Transmitting SIP response (564 bytes) to UDP:192.168.1.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;received=192.168.1.1;branch=z9hG4bK-353534-553fa82d926557ea2b0c35cba85a5af3
Call-ID: a12a40a29bdf6119d9a33f8a95524db6@0:0:0:0:0:0:0:0
From: "8005" <sip:8005@mydomain.net>;tag=632a848f
To: <sip:8001@mydomain.net>;tag=z9hG4bK-353534-553fa82d926557ea2b0c35cba85a5af3
CSeq: 1 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1476602019/7aa91e420a34d2adb4b2be27abd9fcb2",opaque="3c3fbc6344a11717",algorithm=md5,qop="auth"
Server: FPBX-13.0.188.8(13.11.2)
Content-Length:  0


<--- Received SIP request (377 bytes) from UDP:192.168.1.1:5060 --->
ACK sip:8001@mydomain.net SIP/2.0
Call-ID: a12a40a29bdf6119d9a33f8a95524db6@0:0:0:0:0:0:0:0
Max-Forwards: 70
From: "8005" <sip:8005@mydomain.net>;tag=632a848f
To: <sip:8001@mydomain.net>;tag=z9hG4bK-353534-553fa82d926557ea2b0c35cba85a5af3
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-353534-553fa82d926557ea2b0c35cba85a5af3
CSeq: 1 ACK
Content-Length: 0


<--- Received SIP request (1681 bytes) from UDP:192.168.1.1:5060 --->
INVITE sip:8001@mydomain.net SIP/2.0
Call-ID: a12a40a29bdf6119d9a33f8a95524db6@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
From: "8005" <sip:8005@mydomain.net>;tag=632a848f
To: <sip:8001@mydomain.net>
Max-Forwards: 70
Contact: "8005" <sip:8005@192.168.1.1:5060;transport=udp;registering_acc=mydomain_net>
User-Agent: Jitsi2.8.5426Windows 8
Content-Type: application/sdp
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-353534-155af64740ec711b81d78256433f2fb8
Authorization: Digest username="8005",realm="asterisk",nonce="1476602019/7aa91e420a34d2adb4b2be27abd9fcb2",uri="sip:8001@mydomain.net",response="dea5ae3b76aac4c45ab32655b1f63531",algorithm=md5,opaque="3c3fbc6344a11717",qop=auth,cnonce="xyz",nc=00000001
Content-Length: 933

v=0
o=8005-jitsi.org 0 0 IN IP4 192.168.1.1
s=-
c=IN IP4 192.168.1.1
t=0 0
m=audio 5114 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 4 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 usedtx=1
a=rtpmap:97 SILK/24000
a=rtpmap:98 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:100 speex/32000
a=rtpmap:102 speex/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:104 speex/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=rtpmap:101 telephone-event/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics
m=video 5116 RTP/AVP 105 99
a=recvonly
a=rtpmap:105 H264/90000
a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1
a=imageattr:105 send * recv [x=[0-1366],y=[0-768]]
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01f
a=imageattr:99 send * recv [x=[0-1366],y=[0-768]]

<--- Transmitting SIP response (359 bytes) to UDP:192.168.1.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;received=192.168.1.1;branch=z9hG4bK-353534-155af64740ec711b81d78256433f2fb8
Call-ID: a12a40a29bdf6119d9a33f8a95524db6@0:0:0:0:0:0:0:0
From: "8005" <sip:8005@mydomain.net>;tag=632a848f
To: <sip:8001@mydomain.net>
CSeq: 2 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Content-Length:  0


    -- Executing [8001@from-internal:1] GotoIf("PJSIP/8005-00000004", "1?ext-local,8001,1:followme-check,8001,1") in new stack
    -- Goto (ext-local,8001,1)
    -- Executing [8001@ext-local:1] Set("PJSIP/8005-00000004", "__RINGTIMER=15") in new stack
    -- Executing [8001@ext-local:2] Macro("PJSIP/8005-00000004", "exten-vm,novm,8001,0,0,0") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("PJSIP/8005-00000004", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("PJSIP/8005-00000004", "TOUCH_MONITOR=1476602019.4") in new stack
    -- Executing [s@macro-user-callerid:2] Set("PJSIP/8005-00000004", "AMPUSER=8005") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("PJSIP/8005-00000004", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("PJSIP/8005-00000004", "1?Set(REALCALLERIDNUM=8005)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("PJSIP/8005-00000004", "AMPUSER=8005") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("PJSIP/8005-00000004", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("PJSIP/8005-00000004", "AMPUSERCIDNAME=Laptop") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("PJSIP/8005-00000004", "0?report") in new stack
    -- Executing [s@macro-user-callerid:9] Set("PJSIP/8005-00000004", "AMPUSERCID=8005") in new stack
    -- Executing [s@macro-user-callerid:10] Set("PJSIP/8005-00000004", "__DIAL_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-user-callerid:11] Set("PJSIP/8005-00000004", "CALLERID(all)="Laptop" <8005>") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("PJSIP/8005-00000004", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:13] ExecIf("PJSIP/8005-00000004", "0?Set(GROUP(concurrency_limit)=8005)") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("PJSIP/8005-00000004", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:15] Set("PJSIP/8005-00000004", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:16] GotoIf("PJSIP/8005-00000004", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,27)
    -- Executing [s@macro-user-callerid:27] Set("PJSIP/8005-00000004", "CALLERID(number)=8005") in new stack
    -- Executing [s@macro-user-callerid:28] Set("PJSIP/8005-00000004", "CALLERID(name)=Laptop") in new stack
    -- Executing [s@macro-user-callerid:29] GotoIf("PJSIP/8005-00000004", "0?cnum") in new stack
    -- Executing [s@macro-user-callerid:30] Set("PJSIP/8005-00000004", "CDR(cnam)=Laptop") in new stack
    -- Executing [s@macro-user-callerid:31] Set("PJSIP/8005-00000004", "CDR(cnum)=8005") in new stack
    -- Executing [s@macro-user-callerid:32] Set("PJSIP/8005-00000004", "CHANNEL(language)=en") in new stack
    -- Executing [s@macro-exten-vm:2] Set("PJSIP/8005-00000004", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("PJSIP/8005-00000004", "__EXTTOCALL=8001") in new stack
    -- Executing [s@macro-exten-vm:4] Set("PJSIP/8005-00000004", "__PICKUPMARK=8001") in new stack
    -- Executing [s@macro-exten-vm:5] Set("PJSIP/8005-00000004", "RT=") in new stack
    -- Executing [s@macro-exten-vm:6] Gosub("PJSIP/8005-00000004", "sub-record-check,s,1(exten,8001,dontcare)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("PJSIP/8005-00000004", "0?initialized") in new stack
    -- Executing [s@sub-record-check:2] Set("PJSIP/8005-00000004", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:3] Set("PJSIP/8005-00000004", "NOW=1476602019") in new stack
    -- Executing [s@sub-record-check:4] Set("PJSIP/8005-00000004", "__DAY=16") in new stack
    -- Executing [s@sub-record-check:5] Set("PJSIP/8005-00000004", "__MONTH=10") in new stack
    -- Executing [s@sub-record-check:6] Set("PJSIP/8005-00000004", "__YEAR=2016") in new stack
    -- Executing [s@sub-record-check:7] Set("PJSIP/8005-00000004", "__TIMESTR=20161016-181339") in new stack
    -- Executing [s@sub-record-check:8] Set("PJSIP/8005-00000004", "__FROMEXTEN=8005") in new stack
    -- Executing [s@sub-record-check:9] Set("PJSIP/8005-00000004", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:10] NoOp("PJSIP/8005-00000004", "Recordings initialized") in new stack
    -- Executing [s@sub-record-check:11] ExecIf("PJSIP/8005-00000004", "0?Set(ARG3=dontcare)") in new stack
    -- Executing [s@sub-record-check:12] Set("PJSIP/8005-00000004", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:13] ExecIf("PJSIP/8005-00000004", "0?Set(REC_STATUS=NO)") in new stack
    -- Executing [s@sub-record-check:14] GotoIf("PJSIP/8005-00000004", "5?checkaction") in new stack
    -- Goto (sub-record-check,s,17)
    -- Executing [s@sub-record-check:17] GotoIf("PJSIP/8005-00000004", "1?sub-record-check,exten,1") in new stack
    -- Goto (sub-record-check,exten,1)
    -- Executing [exten@sub-record-check:1] NoOp("PJSIP/8005-00000004", "Exten Recording Check between 8005 and 8001") in new stack
    -- Executing [exten@sub-record-check:2] Set("PJSIP/8005-00000004", "CALLTYPE=internal") in new stack
    -- Executing [exten@sub-record-check:3] ExecIf("PJSIP/8005-00000004", "0?Set(CALLTYPE=)") in new stack
    -- Executing [exten@sub-record-check:4] Set("PJSIP/8005-00000004", "CALLEE=dontcare") in new stack
    -- Executing [exten@sub-record-check:5] ExecIf("PJSIP/8005-00000004", "0?Set(CALLEE=dontcare)") in new stack
    -- Executing [exten@sub-record-check:6] GotoIf("PJSIP/8005-00000004", "0?callee") in new stack
    -- Executing [exten@sub-record-check:7] GotoIf("PJSIP/8005-00000004", "1?caller") in new stack
    -- Goto (sub-record-check,exten,13)
    -- Executing [exten@sub-record-check:13] Set("PJSIP/8005-00000004", "RECMODE=dontcare") in new stack
    -- Executing [exten@sub-record-check:14] ExecIf("PJSIP/8005-00000004", "0?Set(RECMODE=dontcare)") in new stack
    -- Executing [exten@sub-record-check:15] ExecIf("PJSIP/8005-00000004", "1?Set(RECMODE=dontcare)") in new stack
    -- Executing [exten@sub-record-check:16] Gosub("PJSIP/8005-00000004", "recordcheck,1(dontcare,internal,8001)") in new stack
    -- Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/8005-00000004", "Starting recording check against dontcare") in new stack
    -- Executing [recordcheck@sub-record-check:2] Goto("PJSIP/8005-00000004", "dontcare") in new stack
    -- Goto (sub-record-check,recordcheck,3)
    -- Executing [recordcheck@sub-record-check:3] Return("PJSIP/8005-00000004", "") in new stack
    -- Executing [exten@sub-record-check:17] Return("PJSIP/8005-00000004", "") in new stack
    -- Executing [s@macro-exten-vm:7] GotoIf("PJSIP/8005-00000004", "1?macrodial") in new stack
    -- Goto (macro-exten-vm,s,13)
    -- Executing [s@macro-exten-vm:13] GosubIf("PJSIP/8005-00000004", "0?clrheader,1()") in new stack
    -- Executing [s@macro-exten-vm:14] Macro("PJSIP/8005-00000004", "dial-one,,Ttr,8001") in new stack
    -- Executing [s@macro-dial-one:1] Set("PJSIP/8005-00000004", "DEXTEN=8001") in new stack
    -- Executing [s@macro-dial-one:2] Set("PJSIP/8005-00000004", "DIALSTATUS_CW=") in new stack
    -- Executing [s@macro-dial-one:3] GosubIf("PJSIP/8005-00000004", "0?screen,1()") in new stack
    -- Executing [s@macro-dial-one:4] GosubIf("PJSIP/8005-00000004", "0?cf,1()") in new stack
    -- Executing [s@macro-dial-one:5] GotoIf("PJSIP/8005-00000004", "1?skip1") in new stack
    -- Goto (macro-dial-one,s,8)
    -- Executing [s@macro-dial-one:8] GotoIf("PJSIP/8005-00000004", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:9] GotoIf("PJSIP/8005-00000004", "0?continue") in new stack
    -- Executing [s@macro-dial-one:10] Set("PJSIP/8005-00000004", "EXTHASCW=ENABLED") in new stack
    -- Executing [s@macro-dial-one:11] GotoIf("PJSIP/8005-00000004", "0?next1:cwinusebusy") in new stack
    -- Goto (macro-dial-one,s,23)
    -- Executing [s@macro-dial-one:23] GotoIf("PJSIP/8005-00000004", "0?next3:continue") in new stack
    -- Goto (macro-dial-one,s,25)
    -- Executing [s@macro-dial-one:25] GotoIf("PJSIP/8005-00000004", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:26] GosubIf("PJSIP/8005-00000004", "1?dstring,1():dlocal,1()") in new stack
    -- Executing [dstring@macro-dial-one:1] Set("PJSIP/8005-00000004", "DSTRING=") in new stack
    -- Executing [dstring@macro-dial-one:2] Set("PJSIP/8005-00000004", "DEVICES=8001") in new stack
    -- Executing [dstring@macro-dial-one:3] ExecIf("PJSIP/8005-00000004", "0?Return()") in new stack
    -- Executing [dstring@macro-dial-one:4] ExecIf("PJSIP/8005-00000004", "0?Set(DEVICES=001)") in new stack
    -- Executing [dstring@macro-dial-one:5] Set("PJSIP/8005-00000004", "LOOPCNT=1") in new stack
    -- Executing [dstring@macro-dial-one:6] Set("PJSIP/8005-00000004", "ITER=1") in new stack
    -- Executing [dstring@macro-dial-one:7] Set("PJSIP/8005-00000004", "THISDIAL=PJSIP/8001") in new stack
    -- Executing [dstring@macro-dial-one:8] GosubIf("PJSIP/8005-00000004", "1?zap2dahdi,1()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("PJSIP/8005-00000004", "0?Return()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:2] Set("PJSIP/8005-00000004", "NEWDIAL=") in new stack
    -- Executing [zap2dahdi@macro-dial-one:3] Set("PJSIP/8005-00000004", "LOOPCNT2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:4] Set("PJSIP/8005-00000004", "ITER2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:5] Set("PJSIP/8005-00000004", "THISPART2=PJSIP/8001") in new stack
    -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("PJSIP/8005-00000004", "0?Set(THISPART2=DAHDIIP/8001)") in new stack
    -- Executing [zap2dahdi@macro-dial-one:7] Set("PJSIP/8005-00000004", "NEWDIAL=PJSIP/8001&") in new stack
    -- Executing [zap2dahdi@macro-dial-one:8] Set("PJSIP/8005-00000004", "ITER2=2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("PJSIP/8005-00000004", "0?begin2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:10] Set("PJSIP/8005-00000004", "THISDIAL=PJSIP/8001") in new stack
    -- Executing [zap2dahdi@macro-dial-one:11] Return("PJSIP/8005-00000004", "") in new stack
    -- Executing [dstring@macro-dial-one:9] GotoIf("PJSIP/8005-00000004", "0?docheck") in new stack
    -- Executing [dstring@macro-dial-one:10] NoOp("PJSIP/8005-00000004", "Debug: Found PJSIP Destination PJSIP/8001, updating with PJSIP_DIAL_CONTACTS") in new stack
    -- Executing [dstring@macro-dial-one:11] Set("PJSIP/8005-00000004", "THISDIAL=PJSIP/8001/sip:8001@192.168.1.110:5060") in new stack
    -- Executing [dstring@macro-dial-one:12] GotoIf("PJSIP/8005-00000004", "0?skipset") in new stack
    -- Executing [dstring@macro-dial-one:13] Set("PJSIP/8005-00000004", "DSTRING=PJSIP/8001/sip:8001@192.168.1.110:5060&") in new stack
    -- Executing [dstring@macro-dial-one:14] Set("PJSIP/8005-00000004", "ITER=2") in new stack
    -- Executing [dstring@macro-dial-one:15] GotoIf("PJSIP/8005-00000004", "0?begin") in new stack
    -- Executing [dstring@macro-dial-one:16] ExecIf("PJSIP/8005-00000004", "0?Return()") in new stack
    -- Executing [dstring@macro-dial-one:17] Set("PJSIP/8005-00000004", "DSTRING=PJSIP/8001/sip:8001@192.168.1.110:5060") in new stack
    -- Executing [dstring@macro-dial-one:18] Return("PJSIP/8005-00000004", "") in new stack
    -- Executing [s@macro-dial-one:27] GotoIf("PJSIP/8005-00000004", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:28] GotoIf("PJSIP/8005-00000004", "0?skiptrace") in new stack
    -- Executing [s@macro-dial-one:29] GosubIf("PJSIP/8005-00000004", "1?ctset,1():ctclear,1()") in new stack
    -- Executing [ctset@macro-dial-one:1] Set("PJSIP/8005-00000004", "DB(CALLTRACE/8001)=8005") in new stack
    -- Executing [ctset@macro-dial-one:2] Return("PJSIP/8005-00000004", "") in new stack
    -- Executing [s@macro-dial-one:30] Set("PJSIP/8005-00000004", "D_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-dial-one:31] NoOp("PJSIP/8005-00000004", "Blind Transfer: , Attended Transfer: , User: 8005, Alert Info: ") in new stack
    -- Executing [s@macro-dial-one:32] ExecIf("PJSIP/8005-00000004", "1?Set(ALERT_INFO=)") in new stack
    -- Executing [s@macro-dial-one:33] ExecIf("PJSIP/8005-00000004", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [s@macro-dial-one:34] ExecIf("PJSIP/8005-00000004", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [s@macro-dial-one:35] GosubIf("PJSIP/8005-00000004", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
    -- Executing [s@macro-dial-one:36] ExecIf("PJSIP/8005-00000004", "0?Set(CHANNEL(musicclass)=)") in new stack
    -- Executing [s@macro-dial-one:37] GosubIf("PJSIP/8005-00000004", "0?qwait,1()") in new stack
    -- Executing [s@macro-dial-one:38] Set("PJSIP/8005-00000004", "__CWIGNORE=") in new stack
    -- Executing [s@macro-dial-one:39] Set("PJSIP/8005-00000004", "__KEEPCID=TRUE") in new stack
    -- Executing [s@macro-dial-one:40] GotoIf("PJSIP/8005-00000004", "0?usegoto,1") in new stack
    -- Executing [s@macro-dial-one:41] GotoIf("PJSIP/8005-00000004", "0?godial") in new stack
    -- Executing [s@macro-dial-one:42] Gosub("PJSIP/8005-00000004", "sub-presencestate-display,s,1(8001)") in new stack
    -- Executing [s@sub-presencestate-display:1] Goto("PJSIP/8005-00000004", "state-not_set,1") in new stack
    -- Goto (sub-presencestate-display,state-not_set,1)
    -- Executing [state-not_set@sub-presencestate-display:1] Set("PJSIP/8005-00000004", "PRESENCESTATE_DISPLAY=") in new stack
    -- Executing [state-not_set@sub-presencestate-display:2] Return("PJSIP/8005-00000004", "") in new stack
    -- Executing [s@macro-dial-one:43] Set("PJSIP/8005-00000004", "CONNECTEDLINE(name,i)=Doorbell") in new stack
    -- Executing [s@macro-dial-one:44] Set("PJSIP/8005-00000004", "CONNECTEDLINE(num)=8001") in new stack
    -- Executing [s@macro-dial-one:45] Set("PJSIP/8005-00000004", "D_OPTIONS=TtrI") in new stack
    -- Executing [s@macro-dial-one:46] Macro("PJSIP/8005-00000004", "dialout-one-predial-hook,") in new stack
    -- Executing [s@macro-dialout-one-predial-hook:1] MacroExit("PJSIP/8005-00000004", "") in new stack
    -- Executing [s@macro-dial-one:47] ExecIf("PJSIP/8005-00000004", "0?Set(D_OPTIONS=trII)") in new stack
    -- Executing [s@macro-dial-one:48] Dial("PJSIP/8005-00000004", "PJSIP/8001/sip:8001@192.168.1.110:5060,,TtrIb(func-apply-sipheaders^s^1)") in new stack
    -- PJSIP/8001-00000005 Internal Gosub(func-apply-sipheaders,s,1) start
    -- Executing [s@func-apply-sipheaders:1] NoOp("PJSIP/8001-00000005", "Applying SIP Headers to channel") in new stack
    -- Executing [s@func-apply-sipheaders:2] Set("PJSIP/8001-00000005", "SIPHEADERKEYS=") in new stack
    -- Executing [s@func-apply-sipheaders:3] While("PJSIP/8001-00000005", "0") in new stack
    -- Jumping to priority 6
    -- Executing [s@func-apply-sipheaders:7] Return("PJSIP/8001-00000005", "") in new stack
  == Spawn extension (from-internal, 8001, 1) exited non-zero on 'PJSIP/8001-00000005'
    -- PJSIP/8001-00000005 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
    -- Called PJSIP/8001/sip:8001@192.168.1.110:5060
<--- Transmitting SIP response (547 bytes) to UDP:192.168.1.1:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;received=192.168.1.1;branch=z9hG4bK-353534-155af64740ec711b81d78256433f2fb8
Call-ID: a12a40a29bdf6119d9a33f8a95524db6@0:0:0:0:0:0:0:0
From: "8005" <sip:8005@mydomain.net>;tag=632a848f
To: <sip:8001@mydomain.net>;tag=8253e5a2-79ff-4e8c-828e-e0f57b174e8e
CSeq: 2 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:192.168.1.220:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Content-Length:  0


    -- Connected line update to PJSIP/8005-00000004 prevented.
<--- Transmitting SIP request (1175 bytes) to UDP:192.168.1.110:5060 --->
INVITE sip:8001@192.168.1.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;rport;branch=z9hG4bKPj6293df21-6ba4-4c01-ac26-889157260ca0
From: "Laptop" <sip:8005@192.168.1.220>;tag=2c9bbe4c-a2e4-4c7e-89b8-82c34e927e45
To: <sip:8001@192.168.1.110>
Contact: <sip:asterisk@192.168.1.220:5060>
Call-ID: e2b71c3a-c2b7-45c1-9061-0139ddb94e6e
CSeq: 31258 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-13.0.188.8(13.11.2)
Content-Type: application/sdp
Content-Length:   500

v=0
o=- 2130796319 2130796319 IN IP4 192.168.1.220
s=Asterisk
c=IN IP4 192.168.1.220
t=0 0
m=audio 18162 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 12826 RTP/AVP 99 98 34 31
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01F
a=rtpmap:98 h263-1998/90000
a=rtpmap:34 H263/90000
a=rtpmap:31 H261/90000
a=sendrecv

<--- Received SIP response (377 bytes) from UDP:192.168.1.110:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.220:5060;rport=5060;branch=z9hG4bKPj6293df21-6ba4-4c01-ac26-889157260ca0
From: "Laptop" <sip:8005@192.168.1.220>;tag=2c9bbe4c-a2e4-4c7e-89b8-82c34e927e45
To: <sip:8001@192.168.1.110>
Call-ID: e2b71c3a-c2b7-45c1-9061-0139ddb94e6e
CSeq: 31258 INVITE
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0


<--- Received SIP response (500 bytes) from UDP:192.168.1.110:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.220:5060;rport=5060;branch=z9hG4bKPj6293df21-6ba4-4c01-ac26-889157260ca0
From: "Laptop" <sip:8005@192.168.1.220>;tag=2c9bbe4c-a2e4-4c7e-89b8-82c34e927e45
To: <sip:8001@192.168.1.110>;tag=156114505
Call-ID: e2b71c3a-c2b7-45c1-9061-0139ddb94e6e
CSeq: 31258 INVITE
Contact: <sip:8001@192.168.1.110:5060>
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
MaxRingingTime: 30
MaxConnectingTime: 120
DependentInfo: 0.0.0.0
Content-Length: 0


    -- PJSIP/8001-00000005 is ringing
<--- Transmitting SIP response (547 bytes) to UDP:192.168.1.1:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;received=192.168.1.1;branch=z9hG4bK-353534-155af64740ec711b81d78256433f2fb8
Call-ID: a12a40a29bdf6119d9a33f8a95524db6@0:0:0:0:0:0:0:0
From: "8005" <sip:8005@mydomain.net>;tag=632a848f
To: <sip:8001@mydomain.net>;tag=8253e5a2-79ff-4e8c-828e-e0f57b174e8e
CSeq: 2 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Contact: <sip:192.168.1.220:5060>
Content-Length:  0


<--- Received SIP response (714 bytes) from UDP:192.168.1.110:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.220:5060;rport=5060;branch=z9hG4bKPj6293df21-6ba4-4c01-ac26-889157260ca0
From: "Laptop" <sip:8005@192.168.1.220>;tag=2c9bbe4c-a2e4-4c7e-89b8-82c34e927e45
To: <sip:8001@192.168.1.110>;tag=156114505
Call-ID: e2b71c3a-c2b7-45c1-9061-0139ddb94e6e
CSeq: 31258 INVITE
Contact: <sip:8001@192.168.1.110:5060>
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Type: application/sdp
Content-Length:   252

v=0
o=0 0 0 IN IP4 192.168.1.110
s=Dahua VT 1.5
c=IN IP4 192.168.1.110
t=0 0
m=video 20001 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
a=sendrecv
m=audio 20000 RTP/AVP 97 0
a=rtpmap:97 PCM/16000
a=rtpmap:0 PCMU/8000
a=sendrecv

[2016-10-16 18:13:39] NOTICE[10668]: res_pjsip_sdp_rtp.c:335 set_caps: No joint capabilities for 'video' media stream between our configuration((h264|h263p|h263|h261)) and incoming SDP((ulaw|ilbc))
<--- Transmitting SIP request (411 bytes) to UDP:192.168.1.110:5060 --->
ACK sip:8001@192.168.1.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;rport;branch=z9hG4bKPj48694f45-0615-4ee5-987d-26f538003651
From: "Laptop" <sip:8005@192.168.1.220>;tag=2c9bbe4c-a2e4-4c7e-89b8-82c34e927e45
To: <sip:8001@192.168.1.110>;tag=156114505
Call-ID: e2b71c3a-c2b7-45c1-9061-0139ddb94e6e
CSeq: 31258 ACK
Max-Forwards: 70
User-Agent: FPBX-13.0.188.8(13.11.2)
Content-Length:  0


    -- PJSIP/8001-00000005 answered PJSIP/8005-00000004
<--- Transmitting SIP response (992 bytes) to UDP:192.168.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;received=192.168.1.1;branch=z9hG4bK-353534-155af64740ec711b81d78256433f2fb8
Call-ID: a12a40a29bdf6119d9a33f8a95524db6@0:0:0:0:0:0:0:0
From: "8005" <sip:8005@mydomain.net>;tag=632a848f
To: <sip:8001@mydomain.net>;tag=8253e5a2-79ff-4e8c-828e-e0f57b174e8e
CSeq: 2 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Contact: <sip:192.168.1.220:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   368

v=0
o=- 0 2 IN IP4 192.168.1.220
s=Asterisk
c=IN IP4 192.168.1.220
t=0 0
m=audio 16374 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 10656 RTP/AVP 105
a=rtpmap:105 H264/90000
a=fmtp:105 profile-level-id=4DE01F
a=sendonly

    -- Channel PJSIP/8001-00000005 joined 'simple_bridge' basic-bridge <675ee3cf-59d8-4bb1-a8ae-2fd2c383bcd8>
    -- Channel PJSIP/8005-00000004 joined 'simple_bridge' basic-bridge <675ee3cf-59d8-4bb1-a8ae-2fd2c383bcd8>
<--- Received SIP request (748 bytes) from UDP:192.168.1.1:5060 --->
ACK sip:192.168.1.220:5060 SIP/2.0
Call-ID: a12a40a29bdf6119d9a33f8a95524db6@0:0:0:0:0:0:0:0
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-353534-9402b889ac05ec17fcd0f77bb28a447a
From: "8005" <sip:8005@mydomain.net>;tag=632a848f
To: <sip:8001@mydomain.net>;tag=8253e5a2-79ff-4e8c-828e-e0f57b174e8e
Max-Forwards: 70
Authorization: Digest username="8005",realm="asterisk",nonce="1476602019/7aa91e420a34d2adb4b2be27abd9fcb2",uri="sip:8001@mydomain.net",response="dea5ae3b76aac4c45ab32655b1f63531",algorithm=md5,opaque="3c3fbc6344a11717",qop=auth,cnonce="xyz",nc=00000001
Contact: "8005" <sip:8005@192.168.1.1:5060;transport=udp;registering_acc=mydomain_net>
User-Agent: Jitsi2.8.5426Windows 8
Content-Length: 0

Posts: 1

Participants: 1

Read full topic

@mvogel4949 wrote:

So I want to utilize failover internet. But when internet carrier #2 takes over my wan ip no longer matches what is found in the Asterisk SIP Settings and calls are disconnected with no audio. Do I switch to DHCP in the Chan SIP tab and input the DDNS found in system admin? Thanks!

Posts: 4

Participants: 2

Read full topic

@sorvani wrote:

Steps to replicate
Create IAX2 trunks between servers and use DNS names instead of IP addresses

For purpose of testing this, I have a simplified IAX2 config.

type=friend
qualify=yes
host=pbx.site1.com
deny=all
context=from-internal
allow=ulaw

Right after you apply config you will see this. Repeatedly.

[2016-10-17 10:10:52] ERROR[23988]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
[2016-10-17 10:10:52] ERROR[23988]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
[2016-10-17 10:10:52] ERROR[23988]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
[2016-10-17 10:10:52] ERROR[23988]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
[2016-10-17 10:10:52] ERROR[23988]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
[2016-10-17 10:10:52] ERROR[23988]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
[2016-10-17 10:10:57] ERROR[22771]: netsock2.c:524 ast_sockaddr_hash: Unknown address family '0'.
[2016-10-17 10:10:57] ERROR[22764]: netsock2.c:524 ast_sockaddr_hash: Unknown address family '0'.
[2016-10-17 10:10:57] ERROR[22773]: netsock2.c:524 ast_sockaddr_hash: Unknown address family '0'.

Showing the peers results in no IP listed.

fpbx*CLI> iax2 show peers
Name/Username    Host                                           Mask                                      Port           Status      Description
bundy_to_site1                                              (S)  255.255.255.255                                          UNREACHABLE
bundy_to_site2                                           (S)  255.255.255.255                                          UNREACHABLE
bundy_to_site3                                           (S)  255.255.255.255                                          UNREACHABLE
3 iax2 peers [0 online, 3 offline, 0 unmonitored]

Eventually, DNS will refresh.

[2016-10-17 10:14:45] ERROR[22726]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
[2016-10-17 10:14:45] NOTICE[22726]: dnsmgr.c:226 dnsmgr_refresh: dnssrv: host 'pbx.site2.com' changed from  to 12.XXX.XXX.XXX:0
[2016-10-17 10:14:45] ERROR[22726]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
[2016-10-17 10:14:45] NOTICE[22726]: dnsmgr.c:226 dnsmgr_refresh: dnssrv: host 'hpbx.site3.com' changed from  to 67.XXX.XXX.XXX:0
[2016-10-17 10:14:45] ERROR[22726]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
[2016-10-17 10:14:45] NOTICE[22726]: dnsmgr.c:226 dnsmgr_refresh: dnssrv: host 'pbx.site1.com' changed from  to 68.XXX.XXX.XXX:0

But the trunks do not come online.

fpbx*CLI> iax2 show peers
Name/Username    Host                                           Mask                                      Port           Status      Description
bundy_to_site1     68.XXX.XXX.XXX                            (S)  255.255.255.255                           0              UNREACHABLE
bundy_to_site2  12.XXX.XXX.XXX                            (S)  255.255.255.255                           0              UNREACHABLE
bundy_to_site3  67.XXX.XXX.XXX                           (S)  255.255.255.255                           0              UNREACHABLE
3 iax2 peers [0 online, 3 offline, 0 unmonitored]

looking at a single peer

fpbx*CLI> iax2 show peer bundy_to_site1


  * Name       : bundy_to_site1
  Description  :
  Secret       : <Not set>
  Context      : from-internal
  Parking lot  :
  Mailbox      :
  Dynamic      : No
  Callnum limit: 0
  Calltoken req: No
  Trunk        : No
  Encryption   : No
  Callerid     : "" <>
  Expire       : -1
  ACL          : Yes
  Addr->IP     : 68.XXX.XXX.XXX Port 0
  Defaddr->IP  : (null) Port (null)
  Username     :
  Codecs       : (gsm|ulaw|alaw)
  Codec Order  : (ulaw|alaw|gsm)
  Status       : UNREACHABLE
  Qualify      : every 60000ms when OK, every 10000ms when UNREACHABLE (sample smoothing Off)

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@DarkQuark wrote:

Greetings, I have a system that I rebuilt from scratch and did a restore on. Post restore everything seems to be fine with the exception of outbound calling. It fails and I think it is a codec mismatch although EVERYTHING is manually set to ulaw at this point. The old system previously ran on 729 and I have a sneaking suspicion that even though it was changed to ulaw it was still running 729. And now that I have restored to a new system that has no 729 installed its failing on the outbound.

The system in question (and the previous one) were from the latest v13 Distro and are current in patches/updates.

Everything is set to ULAW , system, phones and trunks.

Here are some logging that is of importance I believe.

Thank you in advance

==========
[2016-10-17 15:39:11] WARNING[6693][C-00000004] channel.c: Unable to find a codec translation path: (gsm|alaw|ulaw) -> (g729)
[2016-10-17 15:39:11] WARNING[6693][C-00000004] file.c: Unable to open agent-loggedoff (format (g729)): No such file or directory
[2016-10-17 15:39:11] WARNING[6693][C-00000004] app_playback.c: Playback failed on SIP/501-0000000b for agent-loggedoff
===========
[2016-10-17 15:59:25] WARNING[28580] res_odbc.c: res_odbc: Error SQLConnect=-1 errno=1045 [unixODBC][MySQL][ODBC 5.1 Driver]Access denied for user 'freepbxuser'@'localhost' (using password:
[2016-10-17 15:59:25] NOTICE[28580] res_odbc.c: Registered ODBC class 'asteriskcdrdb' dsn->[MySQL-asteriskcdrdb]
===========
[2016-10-17 16:45:15] WARNING[12927][C-0000000c]: channel.c:6405 ast_channel_make_compatible_helper: No path to translate from SIP/outbound-0000001f to SIP/508-0000001e

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@bksales wrote:

i need to remove "+1" from the inbound caller id. we added "trustrpid=yes" to our inbound trunks and the rpid is showing up with the +1 in it.
i thought i could simply create a new scheme but it does not look like i can remove the "+". any suggestions?

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@woodpecker505 wrote:

When trying to upgrade any FreePBX module, I can check for online updates, but when trying to download and install, the I always get the following:

Please wait while module actions are performed
Downloading fw_ari Error(s) downloading fw_ari:

** Unable to connect to servers from URLs provided: http://mirror1.freepbx.org/modules/packages/fw_ari/fw_ari-2.11.1.5.tgz?installid=1043cfbab76a26c299a05b9891af95f1&brandid=freepbxdistro&depolymentid=75969960&astver=11.14.2&phpver=5.3.28&distro=freepbxdistro&distrover=5.211.65-21&fpbxver=2.11.0&ucount=21,http://mirror2.freepbx.org/modules/packages/fw_ari/fw_ari-2.11.1.5.tgz?installid=1043cfbab76a26c299a05b9891af95f1&brandid=freepbxdistro&depolymentid=75969960&astver=11.14.2&phpver=5.3.28&distro=freepbxdistro&distrover=5.211.65-21&fpbxver=2.11.0&ucount=21**

It does not help to set in Advanced System Settings the "Use wget For Module Admin" . My system is not behind Proxy, but uses NAT and in fact when performing via CLI the following:

http://mirror1.freepbx.org/modules/packages/fw_ari/fw_ari-2.11.1.5.tgz

then the file can be downloaded on the Linux server which is running FreePBX.

REMARK : in the response of the failing upgrade there is a line: depolymentid= should this not be deploymentid ? is that the reason of the failure maybe ?

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@conexatech wrote:

Freepbx: 13.0.188.9
Asterisk: 13.11.2
OS: CentOS 6
Host: Google Cloud

Having an issue with calls,

When a call comes in the extension rings and then once answered immediately hangs up and when trying to place an outbound it wont go through.

I can see in the logs that it's sending a "BYE" and I see a warning:

[2016-10-17 16:41:01] WARNING[7875][C-00000001] chan_sip.c: Insufficient information in SDP (c=)...

Confused as though might be an issue with trunk connection but the calls are coming in, here is the pastebin link for the asterisk log with SIP Debugging on:

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@tlarrea wrote:

It seems as though the ringallv2 strategy is behaving the same as the ringall strategy. From what I've read, ringallv2 should initial a call to the primary extension and then after the initial ring time, without terminating the call to primary extension, inititate call to other extensions in the follow-me list.

So my configuration is as follows:
FreePBX: 13.0.188.8
Asterisk: 13.11.2

Capture1.PNG745x442 22.6 KB

Looking at the log file, it seems to execute a hangup after 7s (initial call duration) and the initiates the new call to the follow me list. Wondering if there's something I've got configured wrong.

[2016-10-18 09:22:25] VERBOSE[18190][C-00001527] res_agi.c: dialparties.agi: Starting New Dialparties.agi
[2016-10-18 09:22:25] VERBOSE[18190][C-00001527] res_agi.c: dialparties.agi: Caller ID name is 'Primary Reception' number is '260'
[2016-10-18 09:22:25] VERBOSE[18190][C-00001527] res_agi.c: dialparties.agi: Methodology of ring is 'ringallv2'
[2016-10-18 09:22:25] VERBOSE[18190][C-00001527] res_agi.c: dialparties.agi: Added extension 303 to extension map
[2016-10-18 09:22:25] VERBOSE[18190][C-00001527] res_agi.c: dialparties.agi: Extension 302 cf is disabled
[2016-10-18 09:22:25] VERBOSE[18190][C-00001527] res_agi.c: dialparties.agi: Extension 303 cf is disabled
[2016-10-18 09:22:25] VERBOSE[18190][C-00001527] res_agi.c: dialparties.agi: Extension 302 do not disturb is disabled
[2016-10-18 09:22:25] VERBOSE[18190][C-00001527] res_agi.c: dialparties.agi: Extension 303 do not disturb is disabled
[2016-10-18 09:22:25] VERBOSE[18190][C-00001527] res_agi.c: dialparties.agi: Discovered PJSIP Endpoint PJSIP/302

[2016-10-18 09:22:25] VERBOSE[18192][C-00001527] app_dial.c: Called PJSIP/302/sip:302@192.168.6.1:5060
[2016-10-18 09:22:25] VERBOSE[18190][C-00001527] app_dial.c: Local/FMPR-302@from-internal-000000f7;1 is ringing
[2016-10-18 09:22:25] VERBOSE[18190][C-00001527] app_dial.c: Connected line update to PJSIP/260-0000270f prevented.
[2016-10-18 09:22:25] VERBOSE[18192][C-00001527] app_dial.c: PJSIP/302-00002710 is ringing
[2016-10-18 09:22:25] VERBOSE[18190][C-00001527] app_dial.c: Local/FMPR-302@from-internal-000000f7;1 is ringing

[2016-10-18 09:22:32] VERBOSE[18193][C-00001527] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
[2016-10-18 09:22:32] VERBOSE[18192][C-00001527] app_dial.c: Nobody picked up in 7000 ms

[2016-10-18 09:22:32] VERBOSE[18192][C-00001527] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Local/FMPR-302@from-internal-000000f7;2' in macro 'hangupcall'
[2016-10-18 09:22:32] VERBOSE[18192][C-00001527] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'Local/FMPR-302@from-internal-000000f7;2'
[2016-10-18 09:22:32] VERBOSE[18193][C-00001527] res_agi.c: dialparties.agi: Starting New Dialparties.agi
[2016-10-18 09:22:32] VERBOSE[18193][C-00001527] res_agi.c: dialparties.agi: Caller ID name is 'Primary Reception' number is '260'
[2016-10-18 09:22:32] VERBOSE[18193][C-00001527] res_agi.c: dialparties.agi: Methodology of ring is 'ringall'
[2016-10-18 09:22:32] VERBOSE[18193][C-00001527] res_agi.c: dialparties.agi: Added extension 303 to extension map
[2016-10-18 09:22:32] VERBOSE[18193][C-00001527] res_agi.c: dialparties.agi: Extension 303 cf is disabled
[2016-10-18 09:22:32] VERBOSE[18193][C-00001527] res_agi.c: dialparties.agi: Extension 303 do not disturb is disabled
[2016-10-18 09:22:32] VERBOSE[18193][C-00001527] res_agi.c: dialparties.agi: Discovered PJSIP Endpoint PJSIP/303
[2016-10-18 09:22:32] VERBOSE[18193][C-00001527] res_agi.c: dialparties.agi: Ended up with real PJSIP Dial string PJSIP/303/sip:303@192.168.6.2:5060
[2016-10-18 09:22:32] VERBOSE[18193][C-00001527] res_agi.c: dialparties.agi: dbset CALLTRACE/303 to 260
[2016-10-18 09:22:32] VERBOSE[18193][C-00001527] res_agi.c: dialparties.agi: Filtered ARG3: 303
[2016-10-18 09:22:32] VERBOSE[18193][C-00001527] res_agi.c: AGI Script dialparties.agi completed, returning 0

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@dcitelecom wrote:

I have a shared database and want to connect 2 servers. Both are running the integrated responsive firewall. I need to open port 3306 on the shared database server so that the other machine can access it. Adding the IP as 111.222.333.444/32 as a trusted zone works but seems a bit overkill to allow all traffic when I just want to allow one port. Is there a better way?

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@xramm wrote:

Hi ,
I am looking for Fax solution in FreePBX.
Actually my question is what the exact requirements for Fax system in my FreePBX is ?
Is Faxpro compatible with analog Fax machines ? or do I have to use Digital Fax machine ?

Thanks in advance,
H. Aydin Ucar

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@chacaman wrote:

Issue: Queue Droping Calls
Freepbx: 10.13.66-13
Asterisk: 11.22.0
stage:
Stations: 300
Pstn trunks: 3 (72 channels)
Maximo called I / O: 30
i migrate from Elastix 2.5 to Freepbx In the Elastix pbx everything worked well now when I receive a call distribution queue calls dropp, even cut me out of the cli of asterisk or the asterisk services restart.

note: this error drop queue calls and normals calls.

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@vespino wrote:

I just recently bought a W52p base with 2 (W52H) handsets. Settings this up was easy. What I was wondering is whether it's possible to use the lookup function I'm using for my desktop phones. When I get a call the desktop phones show the name (if found on the website I'm using of course) but the wireless phones only show the number.

The value altered by "Set CallerID" does find its way to the wireless phones. Also my remote phonebook entries are shown in the screen when called.

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