Hi Mike,
the wiki is outdated. Sipxecs supports SNOM firmware 7.x since version
4.0 and since that time xml is the file format supported. I had write
access to the old wiki and I remember I changed it simply adding the xml
extension.
Don't know why the new wiki did not pick-up the change.
For
Hello,
I have sipXecs(4.4.0- 2011-09-02EDT15:26:52 domU-12-31-39-15-62-46)
Installed and everything was working correctly until about 2 weeks ago.
Basically the issue we are experiencing is when a outside caller calls in
and is blind transferred to a external number outside of sipexec the audio
i
This is at a decent sized healthcare company and we have hundreds of
calls a day with many that go over several hours. It definitely does not
happen the majority of the time. The session timer is the default 1800.
It is tough to troubleshoot the intermittent problem...
On 9/26/2011 1:42 PM, Ton
I recall this a while back. I think it was flowroute or vitelity. We
had a long discussion about it. The issue stemmed from how they
handled the session timer, as it was running out and how/whether they
acknowledged it. You might find more information on it in the user
archives. The trace is one si
Yes - I sent Patton packet captures and Verizon was disconnecting the call.
For kicks, I'll turn off the AOC stuff on the interface for Verizon and report
back.
Here is their story:
== Please reply above this line ==
I'm not sure if I'm onto something or not, but I do see a difference
I am running 4.2.1 - sipXconfig (4.2.1-018971 2010-08-17T02:20:18 build20)
Polycom 450s and 550s with 3.2.4 firmware
No firewall
No IP NAT
Sipxbridge with Verizon VoIP services
180 users at this site, no remote workers
Sipx server: 10.87.20.5
Caller ID on all outbound calls is 6157778201
I know I
there's already a note about this in the snom page in the wiki (
http://wiki.sipfoundry.org)
Thanks,
Mike
On Mon, Sep 26, 2011 at 12:07 PM, Domenico Chierico
wrote:
>
>
>
>
> > Be careful that snom 7.x phone will look for a xml file. The path will
> be:
> > http://{HOSTNAME}:8090/phone/profile
> Be careful that snom 7.x phone will look for a xml file. The path will be:
> http://{HOSTNAME}:8090/phone/profile/docroot/{mac}.xml
yes and the better solution i've found for this is an entry into dhcpd config
file like this:
class "snom" {
match if substring (hardware, 1, 3) = 00:
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also...
It appears you are trying to authenticate with the host part set for an ip
address. Why is it that you are not using the sipdomain instead of the IP?
Session Initiation Protocol
Status-Line: SIP/2.0 401 Unauthorized
Status-Code: 401
[Resent Packet: False]
[Requ
are you serious in that you are running sipXecs/3.10.2? can you consider
building a test 4.4.0 system to test against?
On Mon, Sep 26, 2011 at 8:40 AM, Audrey Simpson wrote:
> **
> I've attached the packets.
>
> invite = to a sip trunk, after the 407
> register = attempt to re-reg after above ca
Thanks Kumaran, I modify dial plan setting and the alias function OK.
2011/9/26 Kumaran T
> **
> HI Cristian,
> If you create the alias starts with 8 and contain 4 digit along with
> 8(ie 8797).It will go to IVR only because in dial plan the default VM
> setting will be "Internal station
Hi Cristian,
Please check the issue http://track.sipfoundry.org/browse/XX-9334.
If the you dial 8797 then IVR will played like "user 797 is not
available please leave the message" if 797 user is valid...Otherwise it
will play "That extension is not valid"..
Regards,
Kumaran T
O
HI Cristian,
If you create the alias starts with 8 and contain 4 digit along
with 8(ie 8797).It will go to IVR only because in dial plan the default
VM setting will be "Internal station extension length =3" and prefix is
8.So if 8 followed by 3 digit will go to voicemail of 797...Please ch
Also running sipxproxy 4.4.0-287.gb0a66. Added alaias to user 4498 of 14498
and it rang to the device for 20 seconds and then went to IVR as it should.
On Mon, Sep 26, 2011 at 8:05 AM, Cristian Luna wrote:
> Hello
>
> I am using sipxproxy-4.4.0-287, the alias not function for me.
>
> I have s
Hello
I am using sipxproxy-4.4.0-287, the alias not function for me.
I have setup an alias 8797 on a user extension 18797, the call ring in
device but the call is answered for the IVR. In the trace I see the dialog
SIP and the INVITE is to the IVR first.
Cristian Luna
___
I don't think anyone would want to tinker around trying to fix two versions
ago...
CDR is a postgres function. The "speedials" are stored on the disk
separately but depending on whether they are speeddials and/or BLF entries,
they get loaded from different sources. Speeddials are loaded from the p
Hi folks,
In the last week or so I'm seeing a strange problem with our SipX (4.0.1)
installation.
The CDR (Call Detail Records) module refuses to restart. It seems to go through
the testing phase fine when I restart it from the web interface, but gets stuck
on 'Starting'. Nothing has changed i
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