Topics created by ranahashem

@ranahashem

Scheduling destruction of SIP dialog ‘401fcf1687ec4601a3a3278d6227db07’ in 6400 ms (Method: SUBSCRIBE)

<— SIP read from UDP:192.168.1.4:55702 —>
SUBSCRIBE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:55702;rport;branch=z9hG4bKPj127792c779874087b9e3965 ce1e390c2
Max-Forwards: 70
From: sip:[email protected];tag=8c3eda753e0845bf8ebfa94c8884d64d
To: sip:[email protected]
Contact: sip:[email protected]:55702;ob
Call-ID: 401fcf1687ec4601a3a3278d6227db07
CSeq: 12288 SUBSCRIBE
Event: presence
Expires: 600
Supported: replaces, 100rel, timer, norefersub
Accept: application/pidf+xml, application/xpidf+xml
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.20.6
Authorization: Digest username=“108”, realm=“asterisk”, nonce=“3f0d59a1”, uri=“s ip:[email protected]:5060”, response=“003e71376f0034f4bcbbc47bd83a0e8a”, algorithm =MD5
Content-Length: 0

<------------->
— (16 headers 0 lines) —
Creating new subscription
Sending to 192.168.1.4:55702 (NAT)
Found peer ‘108’ for ‘108’ from 192.168.1.4:55702
Looking for 101 in from-internal (domain 192.168.1.6)
Scheduling destruction of SIP dialog ‘401fcf1687ec4601a3a3278d6227db07’ in 61000 0 ms (Method: SUBSCRIBE)

<— Transmitting (NAT) to 192.168.1.4:55702 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:55702;branch=z9hG4bKPj127792c779874087b9e3965ce1e39 0c2;received=192.168.1.4;rport=55702
From: sip:[email protected];tag=8c3eda753e0845bf8ebfa94c8884d64d
To: sip:[email protected];tag=as47f06dc0
Call-ID: 401fcf1687ec4601a3a3278d6227db07
CSeq: 12288 SUBSCRIBE
Server: FPBX-15.0.17.34(17.9.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Expires: 600
Contact: sip:[email protected]:5060;expires=600
Content-Length: 0

<------------>
Reliably Transmitting (NAT) to 192.168.1.4:55702:
NOTIFY sip:[email protected]:55702;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK12c62c7c;rport
Max-Forwards: 70
From: sip:[email protected];tag=as47f06dc0
To: sip:[email protected];tag=8c3eda753e0845bf8ebfa94c8884d64d
Contact: sip:[email protected]:5060
Call-ID: 401fcf1687ec4601a3a3278d6227db07
CSeq: 102 NOTIFY
User-Agent: FPBX-15.0.17.34(17.9.3)
Subscription-State: active
Event: presence
Content-Type: application/pidf+xml
Content-Length: 524

<?xml version="1.0" encoding="ISO-8859-1"?>

pp:person
ep:activitiesep:away/</ep:activities>
</pp:person>
Unavailable

sip:[email protected]
closed

@ranahashem

sip_general_custom.conf
accept_outofcall_message=yes
outofcall_message_context=dialplan_name
auth_message_requests=yes
…

sip_general_additional.conf
accept_outofcall_message=yes
auth_message_requests=no
outofcall_message_context=dpma_message_context
faxdetect=no
vmexten=*97
useragent=FPBX-15.0.17.34(17.9.3)
language=en
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g726
allow=g722
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
limitonpeers=yes
accept_outofcall_message=yes
outofcall_message_context=astsms
auth_message_requests=yes
context=from-sip-external
callerid=Unknown
tcpenable=no
callevents=yes
jbenable=no
checkmwi=10
maxexpiry=3600
minexpiry=60
srvlookup=no
tlsenable=no
allowguest=yes
notifyhold=yes
rtptimeout=30
canreinvite=no
tlsbindaddr=[::]:5161
rtpkeepalive=0
videosupport=no
defaultexpiry=120
notifyringing=yes
maxcallbitrate=384
rtpholdtimeout=300
g726nonstandard=no
registertimeout=20
tlsclientmethod=tlsv1
registerattempts=0
nat=force_rport,comedia
ALLOW_SIP_ANON=no
udpbindaddr=0.0.0.0:5060
tlscafile=/etc/pki/tls/certs/ca-bundle.crt
externip=104.145.12.182
localnet=192.168.1.6/24
…

sip_additional.conf
[100]
deny=0.0.0.0/0.0.0.0
secret=12345
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
defaultuser=
trustrpid=yes
user_eq_phone=no
sendrpid=pai
type=friend
session-timers=accept
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/100
accountcode=
permit=0.0.0.0/0.0.0.0
callerid=Omer RIT <100>
recordonfeature=apprecord
recordofffeature=apprecord
callcounter=yes
faxdetect=no

[101]
deny=0.0.0.0/0.0.0.0
secret=123
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
defaultuser=
trustrpid=yes
user_eq_phone=no
sendrpid=pai
type=friend
session-timers=accept
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/101
accountcode=
permit=0.0.0.0/0.0.0.0
callerid=Ahmed RIT <101>
recordonfeature=apprecord
recordofffeature=apprecord
callcounter=yes
faxdetect=no

@ranahashem said in linephone softphone file sharing:

@scottalanmiller
I didn't understand why you can't be pampered, is it because you will be restricted to using the linephone as the only soft phone, or is there another reason for that?

Why do you want to use a phone for file sharing? It doesn't make sense to us because there are much better ways to share files.

@JaredBusch said in linphone: remove/hide “default identity”:

@scottalanmiller said in linphone: remove/hide “default identity”:

@JaredBusch said in linphone: remove/hide “default identity”:

@Dashrender said in linphone: remove/hide “default identity”:

@scottalanmiller said in linphone: remove/hide “default identity”:

@Dashrender said in linphone: remove/hide “default identity”:

@scottalanmiller said in linphone: remove/hide “default identity”:

Most enterprise PBXs, and FreePBX is no exception, give you the ENTIRE system. It's an "appliance." You don't download it as software and install on top of an OS as if it were an office suite or note taking application (although that's possible.)

At least not any more, and not for several years... but there was a time, not THAT long ago that you did.

Pretty long ago. Like mid-2000s I'd say. That's more than a generation in IT terms.

That doesn't seem right - I recall building my first FreePBX and that was only like 5-7 years max and you had to install from scripts - they didn't have a DL ISO for install.

If I had to guess, you did PBX in a Flash from Nerdvittles.

That was a scripted install on top of CentOS. But it was still nothing more manual than a single script.

I remember that. "In a Flash", haha.

Compared to the manual processes that existed before then, it was good.

In the end I had issues with the crap that the system pre-setup. It was all at the novice or hobbyist. Not business.

Definitely, it was always very hokey. Way, way too many gizmos and whatevers and way too little "feels enterprise stable."