Problem with trunk SIP configuration

Asterisk Asterisk Support
1

Hello,
I have an old astersik server (1.4) and i migrate us in 1.8.I have quantum SIP gateway
Installation of asterisk 1.8+freepbx is ok but my old trunk configuration is not ok.
My configuration is :

Trunk Name: 4pvs_lines_out
Outbound Caller ID: my tel number
CID Options:
Maximum Channels: 8

Outgoing Settings
Trunk Name: 4pvs_line_out
PEER Details:
type=peer
host=10.0.1.184
fromuser=9999
secret=9999
context=pvs-ipphone
disallow=all
insecure=port,invite

Incoming Settings
USER Context: 9999
USER Details:
type=friend
host=10.0.1.184&dynamic
context=from-pays-external
disallow=all canreinvite=yes
call-limit=50 allow=alaw
insecure=port,invite

Outgoing call is ok but incomming call is not ok
My log is :
WARNING[1131]: chan_sip.c:13450 check_auth: username mismatch, have <interne_externe>, digest has <9999>
[May 18 08:37:18] NOTICE[1131]: chan_sip.c:21256 handle_request_invite: Failed to authenticate device sip:xxxxxxxxx@10.0.1.152;tag=a0001b8-874

what’s the problem?help me please
Thanks

2

  1. You probably have a user or friend, when you should have a peer.

  2. the section name doesn’t match the name by which it is known to the service provider.

3

in my old asterisk(1.4) i have this configuration and is ok but if i use this config in asterisk 1.8 is not ok.
Effectively my peer is in type friend/friend but in 1.8 is not ok.
What changes should I do?
Sorry for my english

General Settings
Trunk Description: 4pvs_lines_out
Outbound Caller ID: my tel number
Maximum Channels: 8

Trunk Name: 4pvs_line_out
PEER Details:
username=9999
type=friend
secret=9999
host=10.0.1.184
disallow=all context=pvs-ipphone
canreinvite=yes c
all-limit=50 allow=alaw

Incoming Settings
USER Context: 9999
USER Details:
username=9998
type=friend
secret=9999
host=10.0.1.184&dynamic
disallow=all
context=from-pays-external
canreinvite=yes
call-limit=50
allow=alaw

help me please

4

This is what you get for using FreePBX trunk configurator (not sure why this thread is in asterisk general section)
FreePBX lulls you into thinking you have incoming and outgoing “trunks” ,
while in reality only one of them is needed.

You also should not be using type=friend, but this was already mentioned.

5

The only FreePBX section seems to pre-suppose AsteriskNow and even then:

  • a lot of people fail to find the AsteriskNow forum;
  • AsteriskNow users tend to operate in ask-only mode (very few technical questions seem to get answered)!