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FPBX between T1 and existing Phone system

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I am in the process of replacing a Mitel PBX system with a FreePBX system. The main reason for this is to allow multi-site dialing. I have the PBX system, with a Sangoma A104D interface. The current Mitel has 2 T1 interfaces, connected to the provider. To ease the migration, we are going to be running dual systems for a while, so I need to configure the FreePBX box as a man in the middle.

I know I need to configure the Interface for PRI_CPE connected to the provider, and PRI_NET for connecting to the Mitel.

I have set the PRI_CPE (Group1) as from-pstn, and PRI_NET (Group2) as from-internal.

My question is where to go from here for the basic configuration of just "pass everything thru" and "do nothing else".

All calls coming from the Group 1 should connect to Group2, and vice-versa. This seems like a very simple, basic problem, but for the life of me I cannot figure out what I should change. I would appreciate any ideas.


Multiple Locations and phone lines

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We are setup with 10 POTS lines in one location and 3 POTS in a second location. We want to allow either location while connected to be passed a line as needed. Has anyone done this with this software. I am reviewing the Xtreme 300 for our installation.

GigaSet C610a IP and MWI

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Have freepbx running with a GigaSet C610a IP connected to it. As I have multiple accounts I have setup all 6 lines on the GigaSet so I can see what outside number was called (named extensions with the outside number).
Also for most of the lines I have voicemail enabled. I have found a post somewhere online how to link all extensions to the main extension (601) voicemailbox. So in case somebody leaves a voicemail there is no need to listen to all separate boxes.

Somehow without configuring the display of the Siemens is able to show there is a message left by showing a voicemail symbol in the display. So far so good. This morning I received a call by a scammer (800-Notes) that left a 1 second message. The call was made to the main account that leads to extension 601.
Problem that I now ran into is that the Siemens was showing that there were 6 voicemail messages. Needles to say that when I removed the message (1 new) in mailbox 601 the MWI disappeared.

Any clues?

John

Best phone to use with FreePBX

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I would like to know what phone is best for FreePBX. What phone is easily provisioned by FreePBX and just works 100% with no hassle.

Cisco 8961 Configuration Troubles

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So here's a little backstory - I have 3 Cisco 6900 (6921 and 6941) working awesome with our system (PBX In a flash) - I decided to purchase a Cisco 8961 to test out, assuming since it was the same "family" of the new 9.x firmware that it would be relatively simple (in Cisco terms) to get working. Not so much.

I unpacked it, set up to point to my update server - updated the firmware to the same relative revision (in my case 9.3.1.x) as the 6900 phones. That went fine. I then copied a 6941 xml config file, editing the bits inside to point to the right extension, etc...and loaded that up. The phone appears to register, the lines show up on the buttons as expected. I can place calls to other extensions or external phones with no problem, and can hear audio both ways. HOWEVER...........

When I place a call INBOUND to the 8961 from any phone, it will not ring but instead the caller hears the congestion tones. I have a sip trace, and can post a copy of the xml config I am using, but I am not sure the bet way to insert them here. Below is my first attempt at the sip trace. In sip show peers, the phone doesnt really show up, as you will see:

SIP SHOW PEERS INFO:
8961 is extension 206 and 906:
---------------------------------------
pbx*CLI> sip show peers
Name/username              Host                                    Dyn Forcerport ACL Port     Status
100                        (Unspecified)                            D              A  0        UNKNOWN
1001                       (Unspecified)                            D              A  0        UNKNOWN
201/201                    10.10.50.2                               D              A  5060     Unmonitored
202                        (Unspecified)                            D              A  0        Unmonitored
206/206                    (Unspecified)                            D   N          A  0        Unmonitored
207/207                    10.10.50.4                               D              A  5060     Unmonitored
208/208                    10.10.50.6                               D              A  5060     Unmonitored
211/211                    10.10.50.10                              D   N          A  5060     OK (92 ms)
213/213                    10.10.50.13                              D   N          A  35420    Unmonitored
214/214                    10.10.50.11                              D   N          A  35851    Unmonitored
215/215                    10.10.50.12                              D   N          A  1041     Unmonitored
3001                       (Unspecified)                            D              A  0        UNKNOWN
301/301                    10.10.50.4                               D              A  5060     Unmonitored
3401/3401                  10.10.81.12                              D              A  5060     Unmonitored
3402/3402                  10.10.81.14                              D              A  5060     OK (35 ms)
901/901                    10.10.50.2                               D              A  5060     Unmonitored
902                        (Unspecified)                            D              A  0        Unmonitored
906/906                    (Unspecified)                            D   N          A  0        Unmonitored
907/907                    10.10.50.4                               D              A  5060     Unmonitored
908/908                    10.10.50.6                               D              A  5060     Unmonitored
913/913                    10.10.50.13                              D   N          A  35420    Unmonitored
914/914                    10.10.50.11                              D   N          A  35851    Unmonitored
915/915                    10.10.50.12                              D   N          A  1041     Unmonitored
9401/9401                  10.10.81.12                              D              A  5060     Unmonitored
9402/9402                  10.10.81.14                              D              A  5060     OK (31 ms)
vitel-inbound/bravonoj     66.241.96.164                                N             5060     Unmonitored
vitel-outbound/bravonoj    64.2.142.215                                 N             5060     Unmonitored
27 sip peers [Monitored: 3 online, 3 offline Unmonitored: 17 online, 4 offline]
       > doing dnsmgr_lookup for 'inbound24.vitelity.net'
       > doing dnsmgr_lookup for 'inbound24.vitelity.net'
pbx*CLI>
------------------------------------------------------------------------------------------------------------
------------------------------------------------------------------------------------------------------------
------------------------------------------------------------------------------------------------------------
------------------------------------------------------------------------------------------------------------
pbx*CLI> sip show tcp
Address                                         Transport   Type
10.10.50.11:35851                               TCP       Server
10.10.50.14:53123                               TCP       Server
10.10.50.13:35420                               TCP       Server
10.10.50.12:1041                                TCP       Server
------------------------------------------------------------------------------------------------------------
------------------------------------------------------------------------------------------------------------
------------------------------------------------------------------------------------------------------------
------------------------------------------------------------------------------------------------------------
--------------BELOW IS THE SIP TRACE/SIP DEBUG FROM THE TIME OF THE CALL.  ---------------------------------
--------------211 is the phone that I used to call the 8961 (at ext. 206)  ---------------------------------
------------------------------------------------------------------------------------------------------------
------------------------------------------------------------------------------------------------------------
------------------------------------------------------------------------------------------------------------
------------------------------------------------------------------------------------------------------------

<--- SIP read from UDP:10.10.50.10:5060 --->
INVITE sip:206@10.10.60.6:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.50.10:5060;branch=z9hG4bK9bde9cd1
Max-Forwards: 70
To: <sip:206@10.10.60.6>
From: <sip:211@10.10.60.6>;tag=3799445153
Call-ID:

CSeq: 1 INVITE
Contact: <sip:211@10.10.50.10:5060>
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY
Content-Type: application/sdp
User-Agent: Panasonic_KX-TGP500B04/12.02 (0080f0b6e9e4)
Content-Length: 313

v=0
o=- 1399056903 1399056903 IN IP4 10.10.50.10
s=-
c=IN IP4 10.10.50.10
t=0 0
m=audio 16048 RTP/AVP 9 8 2 18 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
<------------->
--- (12 headers 15 lines) ---
Sending to 10.10.50.10:5060 (NAT)
Using INVITE request as basis request -

Found peer '211' for '211' from 10.10.50.10:5060

<--- Reliably Transmitting (NAT) to 10.10.50.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.10.50.10:5060;branch=z9hG4bK9bde9cd1;received=10.10.50.10;rport=5060
From: <sip:211@10.10.60.6>;tag=3799445153
To: <sip:206@10.10.60.6>;tag=as0f115b43
Call-ID:

CSeq: 1 INVITE
Server: FPBX-2.9.0(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0fc14ec5"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '953f43a0-8760f11a2ca1a22ff99c0080f0b6e9e4@10.10.50.10' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:10.10.50.10:5060 --->
ACK sip:206@10.10.60.6:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.50.10:5060;branch=z9hG4bK9bde9cd1
Max-Forwards: 70
To: <sip:206@10.10.60.6>;tag=as0f115b43
From: <sip:211@10.10.60.6>;tag=3799445153
Call-ID:

CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:10.10.50.10:5060 --->
INVITE sip:206@10.10.60.6:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.50.10:5060;branch=z9hG4bK8afa4fbf
Max-Forwards: 70
To: <sip:206@10.10.60.6>
From: <sip:211@10.10.60.6>;tag=3799445153
Call-ID:

CSeq: 2 INVITE
Contact: <sip:211@10.10.50.10:5060>
Authorization: Digest realm="asterisk", nonce="0fc14ec5", algorithm=MD5, uri="sip:206@10.10.60.6:5060", username="211", response="bb36ef5424afe4c018f9724105a7bfdb"
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY
Content-Type: application/sdp
User-Agent: Panasonic_KX-TGP500B04/12.02 (0080f0b6e9e4)
Content-Length: 313

v=0
o=- 1399056903 1399056903 IN IP4 10.10.50.10
s=-
c=IN IP4 10.10.50.10
t=0 0
m=audio 16048 RTP/AVP 9 8 2 18 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
<------------->
--- (13 headers 15 lines) ---
Sending to 10.10.50.10:5060 (NAT)
Using INVITE request as basis request -

Found peer '211' for '211' from 10.10.50.10:5060
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format G726-32 for ID 2
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x28000e (gsm|ulaw|alaw|h263|h264), peer - audio=0x190c (ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.10.50.10:16048
Peer doesn't provide video
Looking for 206 in from-internal (domain 10.10.60.6)
list_route: hop: <sip:211@10.10.50.10:5060>

<--- Transmitting (NAT) to 10.10.50.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.50.10:5060;branch=z9hG4bK8afa4fbf;received=10.10.50.10;rport=5060
From: <sip:211@10.10.60.6>;tag=3799445153
To: <sip:206@10.10.60.6>
Call-ID:

CSeq: 2 INVITE
Server: FPBX-2.9.0(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:206@10.10.60.6:5060>
Content-Length: 0


<------------>
    -- Executing [206@from-internal:1] ExecIf("SIP/211-0000013d", "0?Set(__RINGTIMER=0)") in new stack
    -- Executing [206@from-internal:2] Macro("SIP/211-0000013d", "exten-vm,novm,206,0,0,0") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/211-0000013d", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/211-0000013d", "AMPUSER=211") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/211-0000013d", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/211-0000013d", "1?Set(REALCALLERIDNUM=211)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/211-0000013d", "AMPUSER=211") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/211-0000013d", "AMPUSERCIDNAME=Cordless 1") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/211-0000013d", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/211-0000013d", "AMPUSERCID=211") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/211-0000013d", "CALLERID(all)="Cordless 1" <211>") in new stack
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/211-0000013d", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:10] ExecIf("SIP/211-0000013d", "0?Set(GROUP(concurrency_limit)=211)") in new stack
    -- Executing [s@macro-user-callerid:11] ExecIf("SIP/211-0000013d", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/211-0000013d", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:13] Set("SIP/211-0000013d", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("SIP/211-0000013d", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,25)
    -- Executing [s@macro-user-callerid:25] Set("SIP/211-0000013d", "CALLERID(number)=211") in new stack
    -- Executing [s@macro-user-callerid:26] Set("SIP/211-0000013d", "CALLERID(name)=Cordless 1") in new stack
    -- Executing [s@macro-user-callerid:27] Set("SIP/211-0000013d", "CHANNEL(language)=en") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/211-0000013d", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/211-0000013d", "__EXTTOCALL=206") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/211-0000013d", "__PICKUPMARK=206") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/211-0000013d", "RT=") in new stack
    -- Executing [s@macro-exten-vm:6] Macro("SIP/211-0000013d", "record-enable,206,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/211-0000013d", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/211-0000013d", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/211-0000013d", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,14)
    -- Executing [s@macro-record-enable:14] GotoIf("SIP/211-0000013d", "1?IN") in new stack
    -- Goto (macro-record-enable,s,18)
    -- Executing [s@macro-record-enable:18] ExecIf("SIP/211-0000013d", "1?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:7] Macro("SIP/211-0000013d", "dial-one,,tr,206") in new stack
    -- Executing [s@macro-dial-one:1] Set("SIP/211-0000013d", "DEXTEN=206") in new stack
    -- Executing [s@macro-dial-one:2] Set("SIP/211-0000013d", "DIALSTATUS_CW=") in new stack
    -- Executing [s@macro-dial-one:3] GosubIf("SIP/211-0000013d", "0?screen,1") in new stack
    -- Executing [s@macro-dial-one:4] GosubIf("SIP/211-0000013d", "0?cf,1") in new stack
    -- Executing [s@macro-dial-one:5] GotoIf("SIP/211-0000013d", "1?skip1") in new stack
    -- Goto (macro-dial-one,s,8)
    -- Executing [s@macro-dial-one:8] GotoIf("SIP/211-0000013d", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:9] GotoIf("SIP/211-0000013d", "0?continue") in new stack
    -- Executing [s@macro-dial-one:10] Set("SIP/211-0000013d", "EXTHASCW=ENABLED") in new stack
    -- Executing [s@macro-dial-one:11] GotoIf("SIP/211-0000013d", "0?next1:cwinusebusy") in new stack
    -- Goto (macro-dial-one,s,23)
    -- Executing [s@macro-dial-one:23] GotoIf("SIP/211-0000013d", "1?next3:continue") in new stack
    -- Goto (macro-dial-one,s,24)
    -- Executing [s@macro-dial-one:24] ExecIf("SIP/211-0000013d", "0?Set(DIALSTATUS_CW=BUSY)") in new stack
    -- Executing [s@macro-dial-one:25] GotoIf("SIP/211-0000013d", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:26] GosubIf("SIP/211-0000013d", "1?dstring,1:dlocal,1") in new stack
    -- Executing [dstring@macro-dial-one:1] Set("SIP/211-0000013d", "DSTRING=") in new stack
    -- Executing [dstring@macro-dial-one:2] Set("SIP/211-0000013d", "DEVICES=206") in new stack
    -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/211-0000013d", "0?Return()") in new stack
    -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/211-0000013d", "0?Set(DEVICES=06)") in new stack
    -- Executing [dstring@macro-dial-one:5] Set("SIP/211-0000013d", "LOOPCNT=1") in new stack
    -- Executing [dstring@macro-dial-one:6] Set("SIP/211-0000013d", "ITER=1") in new stack
    -- Executing [dstring@macro-dial-one:7] Set("SIP/211-0000013d", "THISDIAL=SIP/206") in new stack
    -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/211-0000013d", "1?zap2dahdi,1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/211-0000013d", "0?Return()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/211-0000013d", "NEWDIAL=") in new stack
    -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/211-0000013d", "LOOPCNT2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/211-0000013d", "ITER2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/211-0000013d", "THISPART2=SIP/206") in new stack
    -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/211-0000013d", "0?Set(THISPART2=DAHDI/206)") in new stack
    -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/211-0000013d", "NEWDIAL=SIP/206&") in new stack
    -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/211-0000013d", "ITER2=2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/211-0000013d", "0?begin2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/211-0000013d", "THISDIAL=SIP/206") in new stack
    -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/211-0000013d", "") in new stack
    -- Executing [dstring@macro-dial-one:9] Set("SIP/211-0000013d", "DSTRING=SIP/206&") in new stack
    -- Executing [dstring@macro-dial-one:10] Set("SIP/211-0000013d", "ITER=2") in new stack
    -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/211-0000013d", "0?begin") in new stack
    -- Executing [dstring@macro-dial-one:12] Set("SIP/211-0000013d", "DSTRING=SIP/206") in new stack
    -- Executing [dstring@macro-dial-one:13] Return("SIP/211-0000013d", "") in new stack
    -- Executing [s@macro-dial-one:27] GotoIf("SIP/211-0000013d", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:28] GotoIf("SIP/211-0000013d", "0?skiptrace") in new stack
    -- Executing [s@macro-dial-one:29] GosubIf("SIP/211-0000013d", "1?ctset,1:ctclear,1") in new stack
    -- Executing [ctset@macro-dial-one:1] Set("SIP/211-0000013d", "DB(CALLTRACE/206)=211") in new stack
    -- Executing [ctset@macro-dial-one:2] Return("SIP/211-0000013d", "") in new stack
    -- Executing [s@macro-dial-one:30] Set("SIP/211-0000013d", "D_OPTIONS=tr") in new stack
    -- Executing [s@macro-dial-one:31] ExecIf("SIP/211-0000013d", "0?SIPAddHeader(Alert-Info: )") in new stack
    -- Executing [s@macro-dial-one:32] ExecIf("SIP/211-0000013d", "0?SIPAddHeader()") in new stack
    -- Executing [s@macro-dial-one:33] ExecIf("SIP/211-0000013d", "0?Set(CHANNEL(musicclass)=)") in new stack
    -- Executing [s@macro-dial-one:34] GosubIf("SIP/211-0000013d", "0?qwait,1") in new stack
    -- Executing [s@macro-dial-one:35] Set("SIP/211-0000013d", "__CWIGNORE=") in new stack
    -- Executing [s@macro-dial-one:36] Set("SIP/211-0000013d", "__KEEPCID=TRUE") in new stack
    -- Executing [s@macro-dial-one:37] GotoIf("SIP/211-0000013d", "0?usegoto,1") in new stack
    -- Executing [s@macro-dial-one:38] GotoIf("SIP/211-0000013d", "0?godial") in new stack
    -- Executing [s@macro-dial-one:39] Set("SIP/211-0000013d", "CONNECTEDLINE(name,i)=Living Room") in new stack
    -- Executing [s@macro-dial-one:40] Set("SIP/211-0000013d", "CONNECTEDLINE(num)=206") in new stack
    -- Executing [s@macro-dial-one:41] Set("SIP/211-0000013d", "D_OPTIONS=trI") in new stack
    -- Executing [s@macro-dial-one:42] Dial("SIP/211-0000013d", "SIP/206,,trI") in new stack
Really destroying SIP dialog '7ced58c179de45910b660cd94d77f529@[::1]:5060' Method: INVITE
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dial-one:43] ExecIf("SIP/211-0000013d", "0?Set(DIALSTATUS=)") in new stack
    -- Executing [s@macro-dial-one:44] GosubIf("SIP/211-0000013d", "0?s-CHANUNAVAIL,1") in new stack
    -- Executing [s@macro-dial-one:45] MacroExit("SIP/211-0000013d", "") in new stack
    -- Executing [s@macro-exten-vm:8] Set("SIP/211-0000013d", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-exten-vm:9] GosubIf("SIP/211-0000013d", "0?docfu,1") in new stack
    -- Executing [s@macro-exten-vm:10] GosubIf("SIP/211-0000013d", "0?docfb,1") in new stack
    -- Executing [s@macro-exten-vm:11] Set("SIP/211-0000013d", "DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-exten-vm:12] ExecIf("SIP/211-0000013d", "0?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:13] GotoIf("SIP/211-0000013d", "1?s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:1] GotoIf("SIP/211-0000013d", "0?exit,1") in new stack
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:2] PlayTones("SIP/211-0000013d", "congestion") in new stack
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:3] Congestion("SIP/211-0000013d", "10") in new stack

<--- Reliably Transmitting (NAT) to 10.10.50.10:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.10.50.10:5060;branch=z9hG4bK8afa4fbf;received=10.10.50.10;rport=5060
From: <sip:211@10.10.60.6>;tag=3799445153
To: <sip:206@10.10.60.6>;tag=as2f8b2192
Call-ID:

CSeq: 2 INVITE
Server: FPBX-2.9.0(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 20
Content-Length: 0


<------------>
  == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/211-0000013d' in macro 'exten-vm'
  == Spawn extension (from-internal, 206, 2) exited non-zero on 'SIP/211-0000013d'
    -- Executing [h@from-internal:1] Hangup("SIP/211-0000013d", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/211-0000013d'

<--- SIP read from UDP:10.10.50.10:5060 --->
ACK sip:206@10.10.60.6:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.50.10:5060;branch=z9hG4bK8afa4fbf
Max-Forwards: 70
To: <sip:206@10.10.60.6>;tag=as2f8b2192
From: <sip:211@10.10.60.6>;tag=3799445153
Call-ID:

CSeq: 2 ACK
Authorization: Digest realm="asterisk", nonce="0fc14ec5", algorithm=MD5, uri="sip:206@10.10.60.6:5060", username="211", response="afdc81e1df0b3ba97366574edbea51db"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '953f43a0-8760f11a2ca1a22ff99c0080f0b6e9e4@10.10.50.10' Method: ACK
Reliably Transmitting (NAT) to 10.10.50.10:5060:
OPTIONS sip:211@10.10.50.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.60.6:5060;branch=z9hG4bK32d44700;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.10.60.6>;tag=as2af44884
To: <sip:211@10.10.50.10:5060>
Contact: <sip:Unknown@10.10.60.6:5060>
Call-ID:
:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.13.0)
Date: Fri, 02 May 2014 18:55:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.10.50.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.60.6:5060;branch=z9hG4bK32d44700;rport=5060
To: <sip:211@10.10.50.10>;tag=808883192
From: "Unknown" <sip:Unknown@10.10.60.6>;tag=as2af44884
Call-ID:
:5060
CSeq: 102 OPTIONS
Contact: <sip:10.10.50.10:5060>
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '767765e606a28430600a44351cc06b96@10.10.60.6:5060' Method: OPTIONS
pbx*CLI> sip set debug off
SIP Debugging Disabled
       > doing dnsmgr_lookup for 'inbound24.vitelity.net'
       > doing dnsmgr_lookup for 'inbound24.vitelity.net'
       > doing dnsmgr_lookup for 'inbound24.vitelity.net'
       > doing dnsmgr_lookup for 'inbound24.vitelity.net'
pbx*CLI> exit
Executing last minute cleanups

Forward DID Indication to Extensions

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Hi,
i'm trying to do a special deployment on FreePBX.
I'm running a VPS with FreePBX, installed via Script. All works fine and i can get trunk adn extension registered.
Since i've got some trunks, i need to manage them via DIDs and via Inboud Routes all works.
I run this to have a real mobility (my house dsl isn't good so if i need to connect in mobility i don't have to register to my home's asterisk server but i will register to the VPS, that's better).

After some days i find useful to install in my house an old Cisco SPA9000 (that i had in garage for some years) with six old SPA942 phones.
I use this SPA9000 with the PSTN (via SPA3102) registered as LINE2 and the FreePBX "trunk" via SPA9000's LINE1.
All works fine, inbound and outbound, but the SPA9000 can't manage the Ring Groups. This because it doesn't receive the DID from FreePBX.

IS there any way or any workaround to forward the incomind DID to the extension? I don't want to apply this rule to all the FreePBX's exntensions (extensions that i use on mobiles doesn't need it).

I'm running FreePBX 2.11.0.37 and Asterisk's version is 11.9.0
All runs on Centos 6.5

Thank you.
Alberto - Italy

ip trunk registration timeout - get notified by mail

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hi,
sometimes my pbx fails to register on ip trunk: is there a way to receive a mail if the registration trunk goes on timeout?

Callback when operator is ready

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The idea is that if the call is not answered because of that all operators are busy, it forwarded to voice message that he would call back as soon relieve the operator, and when the system sees the free operator, it first dials the operator, voice prompt informs the operator that is dialup client ADD, and then call an outside party.
How to implement it through FreePBX, please, throw the idea.


How To: Signalise "Busy" if one Extension in Follow-me is busy

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Hi

Details to the current System:
Asterisk: 1.8.21.0
FreePBX: 2.9.0
Debian: 6
------------------------------
I have currently the following challange:
- "How to configure FreePBX so that an incoming call is declared as "busy" to the caller while two extensions are set in a "follow-me" or "ring group" and just one of them is actualy busy?"

Or client has some separate inboundrouts for his employees like
0xxxx6030
0xxxx6031
a.s.o...

Some of them have a SNOM370 VoIP-Phone and a Gigaset610IP DECT-Phone.
In this example the employee with inboundroute "...6031":
Extension "31" is the Snom, Extension "41" is the DECT-Phone.
Calls on Inboundrout "6031" are given to the Follow-me-Rule "31". In this Follow-Me both Extensions are listet.

In this actual szenario both extensions are ringing. The problem now is that if this employee is busy either on the snom or the DECT, he won't take of the other call. The call will be terminated after 20 secounds. For the Caller this seems like this person is not in the office or what so ever.

The Target now is that if the employee is busy on either one of the phones, an additional incoming call shouldn't even ring but the system should declare the line as busy and not ringing on either one of the phones.

How could this be realised?

I haven't found any good solution so far.
The challange increases as the mainnumber "...6030" leads to a ringgroup which includes the extension 31. Which meens we must make shure that the solution doesn't interfear with other inboundrouts.

Example: The solution bases on a check on the extension 31 itself, which is included in the main ringgroup; this would lead to a drop of any other incomming calls if only one of the extensions is busy...

I hope you see my problem here Acute

Ring multiple extensions from external call

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Hello,

I added the following to my extensions_custom.conf :

exten => 2204,1,Dial(SIP/2204&SIP/2272)
When I call the extension internally both extensions ring however when a call comes in externally (from the IVR) only the extension 2204 rings. I would like both the to ring.
Would anyone know what I need to modify to make this work. Any help would be greatly appreciated.
Thank You

setup barix instreamer as MOH source

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I'm wanting to setup an audio stream from a barix instreamer as a MOH source. I'm finding plenty of info on how to pick up an mp3 stream, I've done that before, with mixed results, mostly due to mpg123 dying for no good reason..

The barix unit is capable of encoding g.711u 8khz mono natively, to me, it seems it would be better to feed asterisk a native stream rather than have to run through mpg123 for transcoding. Has anyone else done this? I'm not coming up with much in my searches.

digium phones and multicasting

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does anyone know if the Digium phones now support multicasting - want to use it for paging with the d40/d50 phones.

Pick up waiting call from Queue and avoid ringing periodically

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Hi all !
I'm new in this community and also a bit new with FreePbx Smile

I'm looking for making a trick with freepbx.
This is my situation :
I've only one phone behind the PBX, so the PBX is mainly used as queueing calls waiting for the phone to answer.

Queueing is working well, but this is the ring behavior which is causing problem.
The person with the phone can hangup the call if he's busy. So the caller is staying in the Queue with a wonderful song ;).
My problem is that the phone will ring again when the ring time elapsed, and then I must hangup the phone again.

Would It be possible after having hanged up, to avoid ringing again the same phone (i.e : extension), to let me pickup back the caller from the Queue when I'm ready ?

Thanks a lot for any answer,
Best regards,
Renaud.

DID switching in inbound route.

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Here is a question. I don't know if it is possible.
Say I have multiple DID with various Sip providers.
I create an Inbound route with a DID from a provider who has 4 Channels. So at any given time the 4 simultaneous calls can come in on that DID. I want to see if I can automatically forward the DID to another DID (SIP provider) at the time when the 4th Channel is ringing (3 channels already occupied).
I know this is probably not going to have a solution. However, if someone can suggest on how to achieve this, that would be simply great.

Greeting message when transfer to Voice mail using *XXX

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I am using Asterisk 1.8.20 FreePbx 2.8.1 with Yealink T46 sip phone. When I have a caller that I want to transfer directly to an extension's voice mail I dial *3XX ( I am using extension numbers 301-3XX) the caller is connected to extension 3XX voice mail, but the greeting is NOT played. Is there a way to have the Greeting play ?

Thanks


Auto Answer "Bot"

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Hello,
I want to make somehow an Extension that automatically Pick up incoming calls and play a sound file that says:"Hit *2100 to call the CEO, hit 2101* to call logistic etc....

I cant find any solution, but i am sure it is possible to make this live.

I have FreePBX 2.11.

384 Analog extensions on one Atom server - High Density FXS - Channel Bank D4

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For some reason dense analog solutions seem to give everyone a headache.

Well thanks to the xLEC's getting out of the fractional T1 business (kind of hard to sell a sub 1M Internet connection these days) there are literally thousands of channel banks for sale at outrageous prices.

A channel bank is a very simple device. It takes a channelized T1 (from a gateway or a DAHDI card) and breaks out the constituent 24 analog channels. For our use we use FXS ports but you could use a mixture of FXO/FXS/GS/E&M etc. if you want to bring it TIE trunks from an old PBX or POTS lines.

Attached is a picture of the rack. This is 16 T1's. Each of the 2U Cisco routers have 8 T1 Multiflex (voice and date) WIC cards in them. They are setup SIP and are simply SIP extensions from Asterisk's perspective

The 1U router at the bottom has two more T1 interfaces and is the gateway for the PRI's. It converts the PRI's to SIP.

The little silver 1I box at the bottom is a 48V DC inverter. The channel banks run on either DC or AC. To me why rectify the power twice so I run it all DC (the thing on the side of the rack is a fuse panel) but it's really not necessary and unless you are familiar with -48 wiring and practices I would recommend the normal route and a UPS.

So anyway, this stuff is fun, the next time you need to do an installation with a large quantity of analog extensions look into this solution. You will differentiate yourself from the competition and deliver real value in the process.

Here is a PIC

Channel_Banks.jpg

Trunk settings for Freephoneline.ca/Fongo.ca

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Hi everyone! Well, since this is my first-ever post here I thought I would make it a helpful one.

I have been browsing the forums here for years and a common thread that keeps popping up is "Trunk settings not working for inbound or outbound calls"

I thought I would post the settings I have finally found that work correctly with Freephoneline.ca/Fongo and will probably work with almost all other SIP providers. (Please note the ALMOST.)

I ran a fresh install of PIAF Green with Asterisk 11.10.0 and FreePBX 2.11.0.37 today and aside from creating 2 DAHDI FXS extensions with my digium TDM400P and 2 SIP trunks along with inbound and outbound routes, everything else is stock vanilla. On to the good stuff:

Under the Trunk settings, fill out as per usual but when you get to the "Outgoing" section, try this:

username=1NXXNXXXXXX
secret=[insertSIPpasswordhere]
host=voip.freephoneline.ca
context=from-trunk
type=friend
insecure=port,invite
disallow=all
allow=ulaw&alaw&gsm&g729
fromdomain=freephoneline.ca

"Incoming" should be this:

secret=[insertSIPpasswordhere]
type=user
context=from-trunk
insecure=port,invite
fromdomain=freephoneline.ca

Registration string should be this:

1NXXNXXXXXX:[insertSIPpasswordhere]@voip.freephoneline.ca/1NXXNXXXXXX

Now a couple of notes:
Freephoneline, like most SIP/DID providers have a touchy registration timer. If it's too short they will hit you with a "Too many requests" error and prevent your trunk from registering. You will have to lengthen it under "Asterisk SIP Settings" to RegisterTimeout=45 and RegisterExpiry=240

Another note is that you must add under "Asterisk SIP Settings" at the bottom where it says "Other SIP settings" put in useragent and then after the equals (=) put Mozilla or WRTP54Gv3.0.6.1

I'm new here but I Hope this helps. Any criticism will be taken constructively unless you talk about my wife or my momma, thems' is fighting words bub! Wink

Cheers,

Repherb

Cisco 7911G upgrade

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This is potentially a @Cisco@ issue rather than 7911 specific.
And I'm hoping someone has an obvious solution for me.
I have what appears to be a valid config file.
I have TFTP setup
The phone sits on registering forever.

In the TFTP logs I see over and over again:
07:47:06 server in.tftpd[24529]: RRQ from filename CTLSEP.tlv
07:47:06 server in.tftpd[24530]: RRQ from filename SEP.cnf.xml
07:47:12 server in.tftpd[24531]: RRQ from filename SIP11.9-3-1SR3-1S.loads
07:47:26 server in.tftpd[24534]: RRQ from filename CTLSEP.tlv
07:47:26 server in.tftpd[24535]: RRQ from filename SEP.cnf.xml

The phone web interface shows:
App Load ID jar11sccp.8-3-2SR1.sbn
Boot Load ID tnp11.3-0-1-32.bin
Version SCCP11.8-3-2SR1S
Hardware Revision 7.1

So it looks to me like it never does the firmware update
(also, the date is stuck in 2007)
Debug shows:
04:20:44 31: Name=SEP Load= SCCP11.8-3-2SR1S File Auth Fail: SIP11.9-3-1SR3-1S
04:20:44 25: Name=SEP Load= SCCP11.8-3-2SR1S Last=Initialized

Can anyone point me at anything that might address this?

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