"I configured multiple interfaces on the sipxecs server"
I think this is where your problems started. I don't believe this is
supported in any way.
On 2/4/2011 8:58 PM, Wayne A. Green wrote:
What is the relationship between the sipxecs server default gateway
and the auto-attendant. This questi
What is the relationship between the sipxecs server default gateway and the
auto-attendant. This question is related to a very strange problem I discovered
when changing the default gateway. The issue involved the following:
1. I configured multiple interfaces on the sipxecs server. This was rela
CRASH HERE ---> } else if (fromUser.equalsIgnoreCase("anonymous")
&&
fromDomain.equalsIgnoreCase("invalid")) {
privacyHeader = ((HeaderFactoryExt)
ProtocolObjects.headerFactory)
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This tracing was very useful
I'm missing an ACK from the SPA942, right?.
Wonder why?
Any idea?
//Niklas
__
Hello Tony,
Hello Nikolay,
many many thanks for your advices.
After reinstalling sipXecs and adding 192.0.2.0/24 to the "intranet
subnets" menu and everything is fine. I would never ever looked at
"IP-Anrufe" (that's the german spelling) without that hint. Maybe the
german translation should b
To give an update on this flowroute issue for anyone who is interested,
I have an open ticket with them and here is there latest response:
~
We have not received an update regarding this from our ULC; however, we
have added a new Tier-1 ULC to expand our coverage. These calls should
now be
In glancing at the sipxbridge log, can you confirm you can send calls out
via the cisco gateway you are using with the clear channel codec normally?
It appears that sipxbridge is complaining (maybe) because of an issue with a
failcode from the cisco gateway when sending a call outbound without the
Douglas,
I think this is probably what you need? It is just the error that
sipxbridge.log shows for the event.
If more is needed, please let me know.
Kind regards,
Peter
"2011-02-03T10:41:49.159000Z":136246:JAVA:INFO:sipx.creativevalley.nl:Thread-7370::sipXbridge:"[SIPTransaction.java
For the second time in a looong time I have the problem that only 50% of
the calls from a GW to a SIP-phone come through, the other 50% go to
voicemail directly.
I have an HA setup and the GW is distributing the calls round-robin to the
2 SipX servers that form an HA-cluster.
Also calls from pho
you should only add what is local to your network that does not have to
negotiate NAT. I don't think by default 192.0.2.0/24 is included in the
standard settings. You should "erase" what is there and add JUST what your
local subnets are (that do not have to traverse nat) at this time.
According to
I will check.
Peter
Smart Future cv,
Buys Ballotstraat 14,
3572 ZR Utrecht,
The Netherlands
phone: +31 308 793 512
fax: +31 847 156 296
mobile: +31 620 749 471
petervanders...@smart-future.nl
On Feb 4, 2011, at 14:13 , Douglas Hubler wrote:
> On Fri, Feb 4, 2011 at 4:49 AM, Peter van der Sal
--Am Freitag, 4. Februar 2011 08:15 -0500 Tony Graziano
schrieb:
> Yes. His local subnet is not in a private "acceptable" range. He needs to
> define it.
If there's "magic" about private subnets you may consider adding
192.0.2.0/24, 198.51.100.0/24 and 203.0.113.0/24. These are also somewhat
Yes. His local subnet is not in a private "acceptable" range. He needs to
define it.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax:
As far as I understand smartnode 4638 is bri voip gateway.
If you plan to connect to PSTN only through this bri gw, and do not plan to
use sip-trunks to ITSP's, you may want to disable siptrunking role for the
sipx.
Anyway read about sipxbridge and siptrunking on the wiki. And about far/near
end na
On Fri, Feb 4, 2011 at 4:49 AM, Peter van der Salm
wrote:
> Then SipXproxy sends and INVITE to sipXbridge to setup the call to
> 0642769062
> sipXbridge returns a 500 Server Internal Error.
I would think /var/log/sipxpbx/sipxbridge.log should have an error message.
__
> Ok, now I see all signalling messages in pcap files.
> (unfortunately your xml trace file does not contain the whole call...).
Hmm, there is nothing in the xml file I left out. I will try to create a
new one.
> I do not change my opinion:
Don't take me wrong, I have no idea about SIP / VoIP s
Some more meditation:
Lets consider call 200 -> 201.
Phone 200 (192.0.2.254) gets 200Ok message (packet 21 in your trace).
And according to sip standard it must send Ack message to address, specified in
Record-Route header.
But Record-Route header points to this phone itself.
So... Ack message n
Ok, now I see all signalling messages in pcap files.
(unfortunately your xml trace file does not contain the whole call...).
I do not change my opinion:
You do not have voice because one of your polycom phones (200) does not send
Ack message.
Or to be precise: phone 201 sends 200 ok, but does no
Hello Tony,
> HD audio or not, if the phones are on the same LAN segment (or sipx is
> told to treat the media as local via intranet subnets) all media between
> handsets is peer to peer. This is not unique to polycom, nor G722 (HD)
> codecs).
since I'm new to VoIP I think my trace is ok or is s
On Fri, Feb 4, 2011 at 5:51 AM, Claas Hilbrecht <
claas.hilbrecht+maillinglists.sipx...@linum.com
> wrote:
> Hello Nikolay,
>
> thanks for the response.
>
>
> I would guess that you do not have voice when calling 200->201 because
>> there is no Ack sent by "200-phone". And in the trace there is o
Really the info you provide is not enough.
Can you provide xml traces suitable for sipviewer of good and bad calls? It
would be a good starting point to trobleshoot a problem.
http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Si
pviewer
Rgds,
Nikolay.
> -Original Messa
I would guess that you do not have voice when calling 200->201 because there is
no Ack sent by "200-phone".
And in the trace there is one way RTP. I would say that you should have "one
way audio", not "no audio".
I don't use polycom phones and I can't advice if you use appropriate firmware
versi
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Hi again.
Have done more investigations.
If I do the call from xlite client it's work fine.
If I do exactly t
Thanks Dave. I think I will try it just before loading the 4.4 ISO.
Wouldn't hurt to learn from this exercise, and have a good exit strategy for
when I make a mistake.
Thanks for the help on this.
From: Dave Deutschman [mailto:ddeutsch...@innovational.net]
Sent: Thursday, February 03, 201
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