Displaying 20 results from an estimated 40 matches for "fpbx".
2012 Oct 23
2
Call drop weirdness
I'm running Asterisk 10.7.0 with three sip trunks to my call
termination provider. For the most part everything works great.
However, at apparently random times and usually about 20 mins or so
into the call, the outbound audio stream dies. The call stays
connected and the inbound audio works fine. The thing is, it happens
on such an irregular basis (once or twice per day) that I can't get
2012 Jan 05
1
Blind transfers being cancelled by asterisk & hanging up on remote caller
...e native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw)
This error repeats 10-11 times quickly (within 1 second) prior to the SIP CANCEL. (see sample of logs below)
3. Other interesting lines from the asterisk full log:
VERBOSE[25592] app_dial.c: -- Connected line update to SIP/fpbx-1-b0c4ceff-00002104 prevented.
[Jan 5 20:51:30] VERBOSE[25592] app_dial.c: -- SIP/fpbx-1-b0c4ceff-00002104 requested special control 20, passing it to SIP/21-00002105
[Jan 5 20:51:30] VERBOSE[25592] app_macro.c: == Spawn extension (macro-dial-one, s, 42) exited non-zero on 'SIP/fpbx-1-b...
2016 Oct 15
2
Registered successfully, but after a minute or so no SIP messages anymore
...cord-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Content-Length: 0
<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP
172.20.40.6;branch=z9hG4bK56ac....
2010 Mar 15
1
dnd
I did a clean install to freepbx 2.6.1 and now when i do *76 i get a 1 second
flash on the screen then the phone hangs up. the FOP says it is on DND
but some ext are still getting calls. once i do a *76 FOP still says I
am on dnd. I am running asterisk 1.6.0.21.
before i was getting a message like dnd activated and dnd deactivated.
i posted this on the freepbx site and here is what i got
2016 Oct 15
3
Registered successfully, but after a minute or so no SIP messages anymore
ping times are fine as well:
[root at freepbx asterisk]# ping sipgate.de
PING sipgate.de (217.10.79.9) 56(84) bytes of data.
64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57
2011 Aug 16
1
Asterisk -> Office 365 Unified Messaging... anyone done it?
....3.4:5061;branch=z9hG4bK16884162
Max-Forwards: 70
From: "Unknown" <sip:Unknown at 1.2.3.4>;tag=as438c582c
To: <sip:um.outlook.com>
Contact: <sip:Unknown at 1.2.3.4:5061;transport=TLS>
Call-ID: 67f260947dae7c27121ca30e5ee9d3ef at 1.2.3.4:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.5.0)
Date: Fri, 12 Aug 2011 06:00:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0 ---
[2011-08-11 23:00:27] VERBOSE[17375] chan_sip.c:
<--- SIP read from TLS:65.55.174.100:5061 --->
SIP/2.0 200 OK...
2013 Jul 16
0
Help with decyphering DND status
...92.168.6.9:5060;branch=z9hG4bK56d306e4;rport
Max-Forwards: 70
From: <sip:41712 at 192.168.6.9;>;tag=as149ada79
To: <sip:41720 at 192.168.6.9>;tag=tyybvtkyiy
Contact: <sip:41712 at 192.168.6.9:5060>
Call-ID: 455FE451B390010B19F8CAE0BE485B25-ju0bdbwv885i
CSeq: 191 NOTIFY
User-Agent: FPBX-2.11.0(11.4.0)
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 206
<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="89"
state="full" entity="sip:41712 at 1...
2018 Dec 26
2
Voice mail: MWI problem / pjsip (13.24.0)
...FY
Accept: application/simple-message-summary
Content-Length: 0
SIP/2.0 200 OK
Expires: 3600
Contact: <sip:192.168.16.70:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER,
REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Server: FPBX-13.0.195.19(13.24.0)
Content-Length: 0
----------------------------------------------------------------------
After REGISTER:
----------------------------------------------------------------------
NOTIFY
Contact: <sip:1234 at 192.168.16.70:5060>
Call-ID: 7a5ee160-47a1-4efa-b190-66843d64...
2015 Aug 18
5
Asterisk 13 chan_sip trunk appending @string to dialled number
...456 at CUBE
chan_sip.c: Got SIP response 500 "Internal Server Error" back from 172.22.4.12:5060
In the SIP SDP;
INVITE sip:0429920437%40CUBE at 172.22.4.12 SIP/2.0.
To: <sip:0429920437%40CUBE at 172.22.4.12>.
As you can see, the @CUBE carries over into the SIP URI as %40CUBE. The FPBX trunk name and outbound route were called CUBE (afaik, purely descriptive) but I changed them to something different and the @CUBE persisted. I'm really not sure where this is coming from, and why.
Here is my trunk configuration;
PEER
type=friend
qualify=yes
nat=no
insecure=port,invite
host=...
2016 Jan 20
2
Incoming webrtc call succeeds in Firefox but fails in Google Chrome
...0.7.2
I set up my SIP configuration to have two SIP accounts. Account 1000 is the Linphone and 1001 is the webrtc:
[general]
faxdetect=no
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-2.11.0(11.20.0)
disallow=all
allow=g723
allow=ulaw
allow=gsm
allow=alaw
allow=g729
allow=speex
allow=g722
allow=h264
allow=h263p
allow=h263
allow=h261
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscipher=ALL
tlsclientmethod=tlsv1
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.cr...
2017 Feb 13
2
First SIP-registering succeeds, second doesn't
...14c0d37-ef26-4dc5-b112-caf0c12a51f1
To: sip:custumoerIDe0 at sipgate.de
Call-ID: 6cf97519-e593-4a7e-bfe4-29141604665d
CSeq: 2367 REGISTER
Contact:
Expires: 300
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Max-Forwards: 70
User-Agent: FPBX-13.0.190.12(13.13.1)
Content-Length: 0
2017/02/10 20:40:59.300873 217.10.79.9:5060 -> 192.193.194.99:55060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
99.88.77.66:55060;rport;branch=z9hG4bKPj7d031cfc-7602-4ac1-a264-bcf8d267500b
From: sip:custumoerIDe0 at sipgate.de;tag=b14c0d37-ef26-4dc5-b112-ca...
2015 Mar 06
0
cant get incoming calls in asterisk
...br <sip%3A80081 at ser.sipcode.com.br>>*
*Contact: <sip:111111 at 200.152.125.213 <sip%3A111111 at 200.152.125.213>>*
*Call-ID: 5c385d117f894c0f2dd79a3f2129b8f5 at ser.sipcode.com.br
<5c385d117f894c0f2dd79a3f2129b8f5 at ser.sipcode.com.br>*
*CSeq: 105 INVITE*
*User-Agent: FPBX-2.9.0(1.4.41)*
*Max-Forwards: 69*
*Date: Fri, 06 Mar 2015 18:17:21 GMT*
*Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO*
*Supported: replaces*
*Content-Type: application/sdp*
*Content-Length: 338*
*v=0*
*o=root 3211 3214 IN IP4 200.152.125.213*
*s=session*
*c=IN IP4 200.1...
2012 Feb 01
2
Getting one way audio even NAT is configured
...h=z9hG4bK1fbbab95;rport
Max-Forwards: 70
From: "77057" <sip:77057 at test.localhost.com>;tag=as1fa9b502
To: <sip:173242 at 12.194.12.12>
Contact: <sip:77057 at 12.131.12.13:5060>
Call-ID: 04ce1d566f1f17a221caba261e2af4bb at test.localhost.com
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.5.0)
Date: Wed, 01 Feb 2012 16:11:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 122642112 122642112 IN IP4 12.131.12.13
s=Asterisk PBX 1.8.5.0
c=IN IP4 12.13...
2015 Jan 12
3
Polycom instant messages
...4bK484dcd1fDD872ECE;received=<CENSORED POLYCOM
IP>;rport=5060
From: "Michael" <sip:3109@<CENSORED>>;tag=D2DAE96E-D8618427
To: <sip:0100@<CENSORED>;user=phone>;tag=as3d0d8c04
Call-ID: d2c5011e-1d718f7-9203b936@<CENSORED POLYCOM IP>
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Thank you,
- --
Michael J. Englehorn
H: 952-884-6776 (Use first)
E-Mail: Michael at englehorn.com
- ----
Those parts of the system that you can h...
2014 Dec 21
3
PJSIP ports, multiple IP addresses and wrong owner
...this is due to the fact that Asterisk / PJSIP produces a wrong owner record. A typical INVITE:
No. Time Source Destination Protocol Length Info
9225 7.503015 192.168.20.48 xx.xxx.xx.xxx SIP/SDP 886 Request: INVITE sip:004982349663847 at fpbx.de |
Frame 9225: 886 bytes on wire (7088 bits), 886 bytes captured (7088 bits)
Ethernet II, Src: MS-NLB-PhysServer-01_01:01:05:01 (02:01:01:01:05:01), Dst: D-Link_03:a4:18 (00:1b:11:03:a4:18)
Internet Protocol Version 4, Src: 192.168.20.48 (192.168.20.48), Dst: xx.xxx.xx.xxx (xx.xxx.xx.xxx)
User...
2007 Aug 15
1
CDR billsec greater than duration
Hi all,
I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1
Doing a select in the CDR table I noticed there are some calls with
billsec greater than duration, duration is always 0 in those calls.
How can this happens ? Am I missing something ?
Tnx in advance
Regards
Edoardo Serra
WeBRainstorm S.r.l.
2014 Feb 26
1
SIP 603 Declined error message
...bK8066eb6f589ce3126b652973b4b00
Record-Route: <sip:172.17.184.46;transport=tcp;lr>
From: "Haley, Scott" <sip:3145152244 at edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00
To: <sip:51104 at edj.devjones.com>
Call-ID: 8066eb6f589ce3125b652973b4b00
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1200;refresher=uac
Contact: <sip:51104 at 192.168.122.51:5060;transport=TCP>
Content-Length: 0
<------------>
-- Executing [51104 at from-t...
2014 Dec 22
0
PJSIP ports, multiple IP addresses and wrong owner
...the fact that Asterisk / PJSIP produces a wrong owner record. A typical INVITE:
>
> No. Time Source Destination Protocol Length Info
> 9225 7.503015 192.168.20.48 xx.xxx.xx.xxx SIP/SDP 886 Request: INVITE sip:004982349663847 at fpbx.de |
>
> Frame 9225: 886 bytes on wire (7088 bits), 886 bytes captured (7088 bits)
> Ethernet II, Src: MS-NLB-PhysServer-01_01:01:05:01 (02:01:01:01:05:01), Dst: D-Link_03:a4:18 (00:1b:11:03:a4:18)
> Internet Protocol Version 4, Src: 192.168.20.48 (192.168.20.48), Dst: xx.xxx.xx.xxx (xx...
2015 Aug 14
2
chan_sip.c: Retransmission timeout reached on transmission
...bK06c2c701
Max-Forwards: 70
From: "PBX-DID" <sip:outbound-trunk at PROVIDER-IP>;tag=as27ef83ae
To: <sip:dialed_number at PROVIDER-IP>
Contact: <sip:outbound-trunk at PBX-PUBLIC_IP:5060>
Call-ID: 6b9ad82d4673fdab722f9e53411a767d at PROVIDER-IP
CSeq: 103 INVITE
User-Agent: FPBX-2.8.1(1.8.11.0)
Proxy-Authorization: Digest username="outbound-trunk", realm="SoftSwitch",
algorithm=MD5, uri="sip:dialed_number at PROVIDER-IP",
nonce="d1b5806808a0888112190722408572932332",
response="40c94f3c04e87e3382c7652d1f012dc9"
Date: Thu, 13...
2015 Aug 18
2
Asterisk 13 chan_sip trunk appending @string to dialled number
...the SIP SDP;
INVITE sip:0429920437%40CUBE at 172.22.4.12<mailto:sip%3A0429920437%2540CUBE at 172.22.4.12> SIP/2.0.
To: <sip:0429920437%40CUBE at 172.22.4.12<mailto:sip%3A0429920437%2540CUBE at 172.22.4.12>>.
As you can see, the @CUBE carries over into the SIP URI as %40CUBE. The FPBX trunk name and outbound route were called CUBE (afaik, purely descriptive) but I changed them to something different and the @CUBE persisted. I?m really not sure where this is coming from, and why.
Here is my trunk configuration;
PEER
type=friend
qualify=yes
nat=no
insecure=port,invite
host=172....