Problem with trunk SIP configuration

Get help with installing, upgrading and running Asterisk.

Moderators: muppetmaster, Moderator, Support

Problem with trunk SIP configuration

Postby morpheus51 » Mon May 23, 2011 12:46 am

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
morpheus51
Newsterisk
 
Posts: 2
Joined: Fri May 20, 2011 7:17 am

Re: Problem with trunk SIP configuration

Postby david55 » Mon May 23, 2011 1:00 am

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.
david55
Moves Like Spencer
 
Posts: 10699
Joined: Fri Sep 26, 2008 5:03 am

Re: Problem with trunk SIP configuration

Postby morpheus51 » Mon May 23, 2011 1:14 am

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
morpheus51
Newsterisk
 
Posts: 2
Joined: Fri May 20, 2011 7:17 am

Re: Problem with trunk SIP configuration

Postby thor » Tue May 31, 2011 1:29 am

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.
thor
Oldsterisk
 
Posts: 238
Joined: Thu Mar 18, 2010 12:19 pm

Re: Problem with trunk SIP configuration

Postby david55 » Tue May 31, 2011 4:06 am

thor wrote:not sure why this thread is in asterisk general section


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)!
david55
Moves Like Spencer
 
Posts: 10699
Joined: Fri Sep 26, 2008 5:03 am


Return to Asterisk Support

Who is online

Users browsing this forum: charmz, Google [Bot] and 28 guests