VoIP Phones - Asterisk: Difference between revisions
Content deleted Content added
m Explain why you can get 2 copies of each inoming call Tags: Mobile edit Mobile web edit |
m Don't feel comfortable with live number in example. No point in allowing ulaw because AAISP don't use it |
||
| Line 30: | Line 30: | ||
In pjsip.conf: |
In pjsip.conf: |
||
[ |
[reg_441234567890] |
||
type = registration |
type = registration |
||
retry_interval = 20 |
retry_interval = 20 |
||
| Line 39: | Line 39: | ||
contact_user = maininbound |
contact_user = maininbound |
||
expiration = 120 |
expiration = 120 |
||
outbound_auth = |
outbound_auth = auth_reg_441234567890 |
||
client_uri = sip:+ |
client_uri = sip:+441234567890@voiceless.aa.net.uk |
||
server_uri = sip:voiceless.aa.net.uk |
server_uri = sip:voiceless.aa.net.uk |
||
[ |
[auth_reg_441234567890] |
||
type = auth |
type = auth |
||
password = |
password = SecretPasswordGoesHere |
||
username = + |
username = +441234567890 |
||
[aaisptrunk] |
[aaisptrunk] |
||
type = aor |
type = aor |
||
contact = sip:+ |
contact = sip:+441234567890@voiceless.aa.net.uk |
||
qualify_frequency=20 |
qualify_frequency=20 |
||
[aaisptrunk_servera] |
[aaisptrunk_servera] |
||
type = aor |
type = aor |
||
contact = sip:+ |
contact = sip:+441234567890@a.voiceless.aa.net.uk |
||
qualify_frequency=20 |
qualify_frequency=20 |
||
[aaisptrunk_serverb] |
[aaisptrunk_serverb] |
||
type = aor |
type = aor |
||
contact = sip:+ |
contact = sip:+441234567890@b.voiceless.aa.net.uk |
||
qualify_frequency=20 |
qualify_frequency=20 |
||
| Line 74: | Line 74: | ||
disallow = all |
disallow = all |
||
allow = alaw |
allow = alaw |
||
allow = ulaw |
|||
direct_media = no |
direct_media = no |
||
rtp_symmetric = yes |
rtp_symmetric = yes |
||
aors = aaisptrunk,aaisptrunk_servera,aaisptrunk_serverb |
aors = aaisptrunk,aaisptrunk_servera,aaisptrunk_serverb |
||
outbound_auth= |
outbound_auth=auth_reg_441234567890 |
||
The "contact_user" option in the registration section sets the context for incoming calls to Asterisk, in this example calls come into the context "maininbound" in extensions.conf |
The "contact_user" option in the registration section sets the context for incoming calls to Asterisk, in this example calls come into the context "maininbound" in extensions.conf |
||
| Line 88: | Line 87: | ||
exten = maininbound,n,Voicemail(222@default,us) |
exten = maininbound,n,Voicemail(222@default,us) |
||
You can dial out via the trunk with: |
You can dial out via the trunk with (probably in a context like "from-internal"): |
||
[mainoutbound] |
|||
exten => _X.,1,Dial(PJSIP/${EXTEN}@aaisptrunk,,) |
exten => _X.,1,Dial(PJSIP/${EXTEN}@aaisptrunk,,) |
||