[asterisk-users] Asterisk and Mitel 3300 ICP

[Joesph O]

Good morning, it now works, failure was due to a misconfigured/misunderstood
Class of Restriction Group Assignment for the SIP Trunk Routes on the
3300ICP.

Now Asterisk can call the world through the Mitel and incoming calls (DID,
operator transfers etc) to Asterisk via the 3300ICP, all work.

Interesting side note - both phone systems have same range of
extensions e.g100 - 299 (just an example)and we created routes on both
to point to the
other for the range, of course, an extension should only exist on one at any
time. therefore, if an extension does not exist locally, it is routed to the
other and vice versa, this way, we keep same range of extensions and this
has helped with migrating users who do not want to trade their loved
Extension & DID number for anything. will continue to test and share
results.

Joesph O.


On 7/9/07, Joesph O <asterisk at me.net.ng> wrote:
>
> Good day everyone,
>
> I have Asterisk and Mitel 3300 ICP communicating via SIP. Calls to and
> from extensions on both sides are completing successfully (details on config
> coming soon). Problem is that calls from Asterisk to PSTN via E1 on Mitel
> 3300 ICP are rejected. What do I need to do to get Asterisk to route PSTN
> calls through it successfully?
>
> Here is an extract of the log on Asterisk whenever I try to call PSTN
> through 3300ICP, in this case, Extension 2540 on Asterisk called 2345678, 9
> is a leading digit -
>
> Jul 7 16:48:07 DEBUG[6860] chan_sip.c: Outgoing Call for 92345678
> Jul 7 16:48:07 VERBOSE[6860] logger.c: -- Called Mitel3300ICP/92345678
> Jul 7 16:48:07 DEBUG[6860] channel.c: Prodding channel 'SIP/2540-b7904a98'
> Jul 7 16:48:07 VERBOSE[6860] logger.c: -- Started music on hold, class
> '24', on SIP/2540-b7904a98
> Jul 7 16:48:07 DEBUG[6860] channel.c: Scheduling timer at 160 sample
> intervals
> Jul 7 16:48:07 DEBUG[6270] chan_sip.c: (Provisional) Stopping
> retransmission (but retaining packet) on '
> 038846853db765e636e57d3e7f1d5dac at 192.168.1.101' Request 102: Found
> Jul 7 16:48:07 DEBUG[6270] chan_sip.c: Acked pending invite 102
> Jul 7 16:48:07 DEBUG[6270] chan_sip.c: Stopping retransmission on '
> 038846853db765e636e57d3e7f1d5dac at 192.168.1.101' of Request 102: Match
> Found
> Jul 7 16:48:07 DEBUG[6860] channel.c: Generator got voice, switching to
> phase locked mode
> Jul 7 16:48:07 DEBUG[6860] channel.c: Scheduling timer at 0 sample
> intervals
> Jul 7 16:48:07 DEBUG[6270] chan_sip.c: (Provisional) Stopping
> retransmission (but retaining packet) on '
> 038846853db765e636e57d3e7f1d5dac at 192.168.1.101' Request 103: Found
> Jul 7 16:48:08 DEBUG[6270] chan_sip.c: Acked pending invite 103
> Jul 7 16:48:08 DEBUG[6270] chan_sip.c: Stopping retransmission on '
> 038846853db765e636e57d3e7f1d5dac at 192.168.1.101' of Request 103: Match
> Found
> Jul 7 16:48:08 WARNING[6270] chan_sip.c: Forbidden - wrong password on
> authentication for INVITE to '"Tester" < sip:2540 at 192.168.1.101>;tag=as07fef065'
> Jul 7 16:48:08 VERBOSE[6860] logger.c: -- SIP/Mitel3300ICP-0832de50 is
> circuit-busy
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070718/965eb108/attachment.htm