Problem When Using Polycom with 2 Lines

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Asterisk Users 2 Comments

Hi,

Has anyone had a problem setting up two registrations (on the same Asterisk server) on one Polycom phone?
When the user tries to make a call on the 2nd line, it works fine.

But when they try the first line, the CLI says:-

Using INVITE request as basis request – href=”mailto:9f5fe9a5-215d0f3a-b2fbe6b7@192.168.1.138″>9f5fe9a5-215d0f3a-b2fbe6b7@192.168.1.138
Found peer client _202′ <--- Which is incorrect, it should be client_201.
And then:
[2010-11-15 10:46:29] WARNING[26082]: chan_sip.c:9063 check_auth: username mismatch, have < client _202>, digest has < client _201>
[2010-11-15 10:46:29] NOTICE[26082]: chan_sip.c:15079 handle_request_invite: Failed to authenticate user ” client line 1″ ;tag=E3231B61-69F3BDD6

The IP and port for client_201 and client _202 are the same.

Any ideas?
Thanks
Dan

2 thoughts on - Problem When Using Polycom with 2 Lines

  • On Mon, Nov 15, 2010 at 6:05 AM, Dan Journo
    wrote:
    href=”mailto:9f5fe9a5-215d0f3a-b2fbe6b7@192.168.1.138″>9f5fe9a5-215d0f3a-b2fbe6b7@192.168.1.138

    I have a Polycom 550 with 2 lines registered to Asterisk 1.8. I’m
    using extensions 2000 and 2001. It worked with Asterisk 1.6.2 as well.
    I do use FreePBX to configure Asterisk.

    Ryan

  • Hi!

    href=”mailto:9f5fe9a5-215d0f3a-b2fbe6b7@192.168.1.138″>9f5fe9a5-215d0f3a-b2fbe6b7@192.168.1.138 Found peer client _202′ < --- In short: Asterisk matches by IP address and assigns the INIVTE to the last entry in sip.conf
    with that IP.

    In more detail: When Asterisk receives an incoming SIP call, the SIP Channel Module

    * first tries to find a [user] section matching the caller name (From: username),
    * then tries to find a [peer] section matching the caller’s IP address.
    * If no matching user or peer is found, the call is sent to the context defined in the [general]
    section of sip.conf.

    Source: http://www.voip-info.org/wiki/view/Asterisk+SIP+channels

    “As of Asterisk 1.2, there is no reason to actually use ‘user’ entries
    any more at all; you can use ‘type=peer’ for everything and the behavior
    will be much more consistent.

    All configuration options supported under ‘type=user’ are also
    supported under ‘type=peer’.

    The difference between friend and peer is the same as defining _both_ a
    user and peer, since that is what ‘type=friend’ does internally.

    The only benefit of type=user is when you _want_ to match on username
    regardless of IP the calls originate from. If the peer is registering to
    you, you don’t need it. If they are on a fixed IP, you don’t need it.
    ‘type=peer’ is _never_ matched on username for incoming calls, only
    matched on IP address/port number (unless you use insecure=port or higher).”

    Source: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer

    Philipp