I've done some additional testing, without doing any changes on the
Cisco side, I reloaded SipX and reconfigured it and I had the exact same
issue where I could not call from the Cisco CME to the SipX extensions.
Luckily, I took snapshots of every related page before I reloaded SipX
and was able to follow it and get it to work again.
Here is what the steps are:
1. The obvious, make sure that Sip Trunking is check under Server
roles and that the service is running
2. I like to add a Domain Name Alias with the IP address of the
server although I know this should not matter
3. Add an unmanaged gateway pointing to the Sip Trunk (on my case it
was a Cisco CME router)
1. All I included was a name, IP address, set port to 5060 and
leave everything else on default.
4. Go to the SipBridge and remove the incoming calls destination and
leave it blank (make sure to send profile after doing this)
5. Add a dial plan pointing to the gateway: In here I added two
custom dial plan entries, one with the extension number I want to
dial and another with the number on the remote extension the
system see. IE, on my case we have 4 digits extensions on the CME
but when calling from the CME out to the SipX the CME uses the 10
digits E164 number as the source extension. When the SipX tries to
connect your incoming call to the SipX phone, the SipX tells it to
connect to that number, therefore if your phone does not know how
to connect to the full number it won't complete the call.
That last two are very important thing to remember and also, at least
in my case, I found out that the Sip Trunking service takes much longer
than all other services to restart so be patient and check the services tab.
Please let me know if this helps,
Jhony
Mark Wood wrote:
Unfortunately this did not work for me. The Asterisk system allows me
to set the allow/disallow codecs, so I disallow all and then allow
ulaw. No change, tried to allow all in the asterisk, no change, I can
still dial the autoattendant but not any extensions. From the Sipxecs
system I can still dial any of my Asterisk stations.
*From:* Jhony Perez [mailto:jpe...@zbzoom.net]
*Sent:* Monday, May 18, 2009 10:05 AM
*To:* Dale Worley
*Cc:* sipx-users@list.sipfoundry.org
*Subject:* Re: [sipx-users] SipXecs 4 with Cisco gateway issues - Resolved
Dale Worley wrote:
On Mon, 2009-05-18 at 10:18 -0400, Jhony Perez wrote:
After trying many things I found what I think it was the last piece of
the puzzle, on the Cisco gateway dial-peer (call leg) that points to the
SipXecs, I specified what codec to use, G711 and now I'm able to call
just fine. After doing some research I found that the Cisco gateway
defaults to G729, don't know if 4.0 has built-in support to G729 but
that seems to have fix my issue.
The Cisco should be set up to offer (and accept) any codec that it
supports. You shouldn't be configuring it to use only one of the codecs
that it can use.
Dale
I didn't specify that on the incoming dial-peer but only on the
outgoing to SipXecs, incoming it is up to the calling side to propose
what codec to use based on what's configure as the prefer codec. After
troubleshooting my issue, I found that with SipXecs 4.0 unless I
specify G711 as my prefer codec, everytime I call from a handset
connected to the Cisco CME to a handset connected to the SipXecs 4.0
I'd get fast busy but as soon as I set up the prefer codec on the
Cisco to G711 it works.
I'll try removing this and adding it a few times including setting the
prefer to G729 and see what happens, if anything would be good to know.
I'll keep you all posted.
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