Re: [asterisk-users] CALLERID on pjsip doesn't work?

[Andrew Ivins]

On 1 July 2016 at 17:41, Joshua Colp  wrote:
>
>
>> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>)
>> same => n,Dial(PJSIP/phone123, 30)
>>
>
> Your exten line has no priority, is that how it is in your dialplan?
>

Actually no, I stole that line from an earlier email to this list. Mine has
a priority.


> If not you can isolate things a bit further by trying the following:
>
> Set(CALLERID(all)=Jon Doe <+123456789>)
>
> Or individually:
>
> Set(CALLERID(name)=Jon Doe)
> Set(CALLERID(num)=+123456789)
>

Tried many permutations of this, and the only thing I can get to happen is
to make the call present as Anonymous by changing the pres-name/pres-num
setting.

It's not a production system, dialplan is pretty simple:

same => _X.1,Set(CALLERID(name-pres)=allowed)
same => n,Set(CALLERID(num-pres)=allowed)
same => n,Set(CALLERID(name)=Fred)
same => n,Set(CALLERID(num)=6123)
same => n,Dial(PJSIP/DEADDEADBEEF, 30)
same => n,Hangup()

DEADDEADBEEF is the name of the endpoint and the endpoint works. I use MAC
addresses and plan to dynamically map extensions to them later on (kind of
like user mode in freepbx).

In the console, if I log the value of CALLERID, it is what I expect to it
to be.

In the pjsip debug, the callerid I am trying to set doesn't appear anywhere.

I'm using your Sorcery stuff backing into astb for pjsip, but I've done a
little script to dump it back into text so I can override it in the config
file. Therefore it's a bit verbose. Thanks for looking.

[DEADDEADBEEF]
type=aor
support_path=true
default_expiration=3600
qualify_timeout=3.00
mailboxes=
minimum_expiration=60
outbound_proxy=
voicemail_extension=
maximum_expiration=7200
qualify_frequency=0
authenticate_qualify=false
contact=
max_contacts=1
remove_existing=true

[DEADDEADBEEF]
type=auth
md5_cred=
realm=
auth_type=userpass
password=4D7D9A7F1822
nonce_lifetime=32
username=507B495E565B

[DEADDEADBEEF]
type=endpoint
timers_sess_expires=1800
device_state_busy_at=0
dtls_cipher=
from_domain=
dtls_rekey=0
dtls_fingerprint=SHA-256
direct_media_method=invite
send_rpid=false
pickup_group=
sdp_session=Asterisk
dtls_verify=No
message_context=
mailboxes=
named_pickup_group=
record_on_feature=automixmon
dtls_private_key=
named_call_group=
t38_udptl_maxdatagram=0
media_encryption_optimistic=false
aors=DEADDEADBEEF
rpid_immediate=false
outbound_proxy=
identify_by=username
inband_progress=false
rtp_symmetric=false
transport=transport-udp
rtp_keepalive=0
t38_udptl_ec=none
fax_detect=false
t38_udptl_nat=false
allow_transfer=true
tos_video=0
srtp_tag_32=false
timers_min_se=90
call_group=
sub_min_expiry=0
100rel=yes
direct_media=true
rtp_timeout_hold=0
g726_non_standard=false
dtmf_mode=rfc4733
voicemail_extension=
rtp_timeout=0
dtls_cert_file=
media_encryption=no
media_use_received_transport=false
direct_media_glare_mitigation=none
trust_id_inbound=false
force_avp=false
record_off_feature=automixmon
send_diversion=true
language=
mwi_from_user=
rtp_ipv6=false
ice_support=false
callerid=unknown
aggregate_mwi=true
one_touch_recording=false
cos_video=0
accountcode=
allow=(g722|ulaw|alaw)
rewrite_contact=false
t38_udptl_ipv6=false
tone_zone=
user_eq_phone=false
allow_subscribe=true
rtp_engine=asterisk
auth=DEADDEADBEEF
from_user=DEADDEADBEEF
bind_rtp_to_media_address=false
disable_direct_media_on_nat=false
set_var=
use_ptime=false
outbound_auth=
media_address=
tos_audio=0
dtls_ca_path=
dtls_setup=active
force_rport=false
connected_line_method=invite
callerid_tag=
timers=yes
sdp_owner=-
trust_id_outbound=false
use_avpf=false
context=default
moh_suggest=default
send_pai=false
t38_udptl=false
dtls_ca_file=
callerid_privacy=allowed_not_screened
mwi_subscribe_replaces_unsolicited=false
cos_audio=0
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[asterisk-users] Please help me understand lines and extensions little better

[Ivan Demkovitch]

Hello,

Probably obvious to most of you but I’m still learning VOIP concept(and 
telephony inside organization in general)

I have bunch of unrelated questions on my mind :)

- I picked SPA504G phones as “phone to go” for our small company. They are 
cheap and available. Any pros/cons you can come up with against this device?

I setup direct extensions in Asterisk, they work great. But under conference 
settings - I have to set which SIP devices participate. It just sends calls to 
those devices and they come in as main extension.
It’s not super-convenient. Main problem is - I want to know if user came from 
SALES/SUPPORT or DIRECT line.

-Now I’m getting more into phone setup little more and I think I’m not doing it 
right.
Current on a phone I setup only 1 extensions with SIP ID/Password and then I 
assign this extension 1 to all 4 “Line keys”

I understand this is how I can switch between calls. Let’s say one call comes 
in - I can be on call and next call comes in - I can press “Line 2” button and 
pickup. Correct? 

So, now I got idea about Line key 3 & 4.
If my main SIP ID 
XXX

I can set 2 more id’s:
XXX_SLS
XXX_SPT

Then I can register them under ext 3/4 on a phone and set line keys to ext 3/4
Of course, I will add XXX_SLS and XXX_SPT under appropriate queues.conf

This way I will setup more extensions but users will have much better 
visibility of who’s calling.

Is this correct/common approach? Just trying to understand what is the best way 
to setup phones/extensions..
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