Linking Asterisk Servers - partial success

Discussion in 'UK VOIP' started by Matt, Apr 13, 2007.

  1. Matt

    Matt Guest

    Hi,

    We've linked the servers and can call both sets of extensions etc.

    If I try to call out over one of the trunks on the remote server the
    call fails and I see

    NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
    attempt from xx.xx.xx.xx, request '1571@default' does not exist.

    I was trying to call out on the trunk to the trunks voicemail services
    (1571 - voipfone).

    Any ideas what I've got wrong, or missed out?

    Cheers


    Matthew
    Matt, Apr 13, 2007
    #1
    1. Advertising

  2. Matt

    Desk Rabbit Guest

    Matt wrote:
    > Hi,
    >
    > We've linked the servers and can call both sets of extensions etc.
    >
    > If I try to call out over one of the trunks on the remote server the
    > call fails and I see
    >
    > NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
    > attempt from xx.xx.xx.xx, request '1571@default' does not exist.
    >
    > I was trying to call out on the trunk to the trunks voicemail services
    > (1571 - voipfone).
    >
    > Any ideas what I've got wrong, or missed out?
    >
    > Cheers
    >
    >
    > Matthew
    >

    What version of FreePBX?
    What version of Asterisk?
    Desk Rabbit, Apr 13, 2007
    #2
    1. Advertising

  3. Matt

    Matt Guest

    On Apr 13, 11:51 am, Desk Rabbit <> wrote:
    > Matt wrote:
    > > Hi,

    >
    > > We've linked the servers and can call both sets of extensions etc.

    >
    > > If I try to call out over one of the trunks on the remote server the
    > > call fails and I see

    >
    > > NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
    > > attempt from xx.xx.xx.xx, request '1571@default' does not exist.

    >
    > > I was trying to call out on the trunk to the trunks voicemail services
    > > (1571 - voipfone).

    >
    > > Any ideas what I've got wrong, or missed out?

    >
    > > Cheers

    >
    > > Matthew

    >
    > What version of FreePBX?
    > What version of Asterisk?


    Cheers for the reply Desk Rabbit.

    The local end is freepbx 2.2.0 with asterisk 1.2.7.1

    And the remote end is the Digium Asterisk GUI (beta) on Asterisk
    1.4.2.

    The problem occurs when I try to dial out over the remote voipfone
    trunk. The trunk all works ok for the users at their end.

    Thanks again

    Matthew
    Matt, Apr 13, 2007
    #3
  4. Matt

    Matt Guest

    On Apr 13, 12:14 pm, "Matt" <> wrote:
    > On Apr 13, 11:51 am, Desk Rabbit <> wrote:
    >
    >
    >
    > > Matt wrote:
    > > > Hi,

    >
    > > > We've linked the servers and can call both sets of extensions etc.

    >
    > > > If I try to call out over one of the trunks on the remote server the
    > > > call fails and I see

    >
    > > > NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
    > > > attempt from xx.xx.xx.xx, request '1571@default' does not exist.

    >
    > > > I was trying to call out on the trunk to the trunks voicemail services
    > > > (1571 - voipfone).

    >
    > > > Any ideas what I've got wrong, or missed out?

    >
    > > > Cheers

    >
    > > > Matthew

    >
    > > What version of FreePBX?
    > > What version of Asterisk?

    >
    > Cheers for the reply Desk Rabbit.
    >
    > The local end is freepbx 2.2.0 with asterisk 1.2.7.1
    >
    > And the remote end is the Digium Asterisk GUI (beta) on Asterisk
    > 1.4.2.
    >
    > The problem occurs when I try to dial out over the remote voipfone
    > trunk. The trunk all works ok for the users at their end.
    >
    > Thanks again
    >
    > Matthew


    I should also add that the remote server users can successfully make
    outgoing calls over the local trunks here
    Matt, Apr 13, 2007
    #4
  5. Matt

    Desk Rabbit Guest

    Matt wrote:
    > On Apr 13, 11:51 am, Desk Rabbit <> wrote:
    >> Matt wrote:
    >>> Hi,
    >>> We've linked the servers and can call both sets of extensions etc.
    >>> If I try to call out over one of the trunks on the remote server the
    >>> call fails and I see
    >>> NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
    >>> attempt from xx.xx.xx.xx, request '1571@default' does not exist.
    >>> I was trying to call out on the trunk to the trunks voicemail services
    >>> (1571 - voipfone).
    >>> Any ideas what I've got wrong, or missed out?
    >>> Cheers
    >>> Matthew

    >> What version of FreePBX?
    >> What version of Asterisk?

    >
    > Cheers for the reply Desk Rabbit.
    >
    > The local end is freepbx 2.2.0 with asterisk 1.2.7.1

    Thats fine and thats what I have here at both ends.


    > And the remote end is the Digium Asterisk GUI (beta) on Asterisk
    > 1.4.2.

    Ah! At this point you are on your own. Sorry.


    > The problem occurs when I try to dial out over the remote voipfone
    > trunk. The trunk all works ok for the users at their end.


    The only thing I can think of is the conext. FreePBX uses
    "from-internal". It looks like (And I am guessing here) your remote site
    is using "default".
    Desk Rabbit, Apr 13, 2007
    #5
  6. Matt

    Matt Guest

    On Apr 13, 12:41 pm, Desk Rabbit <> wrote:
    > Matt wrote:
    > > On Apr 13, 11:51 am, Desk Rabbit <> wrote:
    > >> Matt wrote:
    > >>> Hi,
    > >>> We've linked the servers and can call both sets of extensions etc.
    > >>> If I try to call out over one of the trunks on the remote server the
    > >>> call fails and I see
    > >>> NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
    > >>> attempt from xx.xx.xx.xx, request '1571@default' does not exist.
    > >>> I was trying to call out on the trunk to the trunks voicemail services
    > >>> (1571 - voipfone).
    > >>> Any ideas what I've got wrong, or missed out?
    > >>> Cheers
    > >>> Matthew
    > >> What version of FreePBX?
    > >> What version of Asterisk?

    >
    > > Cheers for the reply Desk Rabbit.

    >
    > > The local end is freepbx 2.2.0 with asterisk 1.2.7.1

    >
    > Thats fine and thats what I have here at both ends.
    >
    > > And the remote end is the Digium Asterisk GUI (beta) on Asterisk
    > > 1.4.2.

    >
    > Ah! At this point you are on your own. Sorry.
    >
    > > The problem occurs when I try to dial out over the remote voipfone
    > > trunk. The trunk all works ok for the users at their end.

    >
    > The only thing I can think of is the conext. FreePBX uses
    > "from-internal". It looks like (And I am guessing here) your remote site
    > is using "default".


    I'm trying to persuade them to move away from the asterisk gui!

    I'll check the contexts - looks like a good point to start.

    Thanks!
    Matt, Apr 13, 2007
    #6
  7. Matt

    alexd Guest

    Matt wrote:

    >> >>> NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
    >> >>> attempt from xx.xx.xx.xx, request '1571@default' does not exist.
    >> >>> I was trying to call out on the trunk to the trunks voicemail
    >> >>> services (1571 - voipfone).


    > I'm trying to persuade them to move away from the asterisk gui!


    I've not used it myself - is it any good?

    I'm bound to think that any GUI that tries to capture the functionality of
    Asterisk's text config files - which is practically a programming language
    in itself - is going to be a disappointment on some level.

    > I'll check the contexts - looks like a good point to start.


    Also, are the local users who can dial 1571 dialling 9 first, perchance? How
    are you routing the call? @default doesn't look right. For example [vanilla
    Asterisk], in my sip.conf, my Sipgate context is called 'sipgate' and my
    voip.co.uk context is called 'voipcouk'. And to use the Sipgate trunk, I
    dial 89 then the number. So if I 8910000, the leading 89 gets stripped off,
    and Asterisk says:

    Executing Dial("SIP/6012-00637c30", "SIP/10000@sipgate|60|tr") in new stack

    Dial 9 for voip.co.uk trunk, 9 gets stripped off:

    Executing Dial("SIP/6012-00637c30", "SIP/08000850643@voipcouk|60|o") in new
    stack

    So your dial requests should look like 1571@voipfone, or whatever your
    Voipfone trunk context is. You should be able to see what it's doing by
    opening an Asterisk and making sure the verbosity is at least 3 ['set
    verbose 3'].

    --
    <http://ale.cx/> (AIM:troffasky) ()
    00:22:52 up 4:41, 2 users, load average: 0.03, 0.06, 0.07
    Yes. I'm just guessing.
    alexd, Apr 14, 2007
    #7
  8. Matt

    Matt Guest

    On Apr 14, 12:38 am, alexd <> wrote:
    > Matt wrote:
    > >> >>> NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
    > >> >>> attempt from xx.xx.xx.xx, request '1571@default' does not exist.
    > >> >>> I was trying to call out on the trunk to the trunks voicemail
    > >> >>> services (1571 - voipfone).

    > > I'm trying to persuade them to move away from the asterisk gui!

    >
    > I've not used it myself - is it any good?


    Seems a bit buggy at the moment, and without some of the freepbx
    functionality.


    >
    > I'm bound to think that any GUI that tries to capture the functionality of
    > Asterisk's text config files - which is practically a programming language
    > in itself - is going to be a disappointment on some level.
    >

    Think I'll stick with Freepbx at this end!

    > > I'll check the contexts - looks like a good point to start.

    >
    > Also, are the local users who can dial 1571 dialling 9 first, perchance? How
    > are you routing the call? @default doesn't look right. For example [vanilla
    > Asterisk], in my sip.conf, my Sipgate context is called 'sipgate' and my
    > voip.co.uk context is called 'voipcouk'. And to use the Sipgate trunk, I
    > dial 89 then the number. So if I 8910000, the leading 89 gets stripped off,
    > and Asterisk says:
    >
    > Executing Dial("SIP/6012-00637c30", "SIP/10000@sipgate|60|tr") in new stack
    >
    > Dial 9 for voip.co.uk trunk, 9 gets stripped off:
    >
    > Executing Dial("SIP/6012-00637c30", "SIP/08000850643@voipcouk|60|o") in new
    > stack
    >
    > So your dial requests should look like 1571@voipfone, or whatever your
    > Voipfone trunk context is. You should be able to see what it's doing by
    > opening an Asterisk and making sure the verbosity is at least 3 ['set
    > verbose 3'].
    >

    Thats what I thought. All the local users can dial out without any
    problems, without a 9.

    The settings for the trunks are so simplistic in the gui though.....

    When I get a chance I'll sit and have another watch of the cli...

    Cheers

    Matthew
    Matt, Apr 14, 2007
    #8
    1. Advertising

Want to reply to this thread or ask your own question?

It takes just 2 minutes to sign up (and it's free!). Just click the sign up button to choose a username and then you can ask your own questions on the forum.
Similar Threads
  1. Peter
    Replies:
    4
    Views:
    5,057
    Party Girl
    Jul 8, 2003
  2. Sparks
    Replies:
    17
    Views:
    10,788
    sameerms18
    Nov 18, 2011
  3. =?Utf-8?B?ZG91Z2hib3kzMQ==?=

    Implementing dhcp servers and dns servers

    =?Utf-8?B?ZG91Z2hib3kzMQ==?=, Jun 16, 2006, in forum: MCSE
    Replies:
    20
    Views:
    3,081
    Guest
    Jun 24, 2006
  4. Matt

    Linking 2 asterisk servers

    Matt, Apr 11, 2007, in forum: UK VOIP
    Replies:
    4
    Views:
    872
    Desk Rabbit
    Apr 12, 2007
  5. Bill T.

    Re: macro photography - partial success and mystery

    Bill T., Feb 18, 2010, in forum: Digital Photography
    Replies:
    3
    Views:
    387
    Bill T.
    Feb 19, 2010
Loading...

Share This Page