> Putting these cryptic notes out here doesn't allow anyone to give you any
> assistance, and doesn't get you closer to fixing this issue.
I'm kinda thinking out loud when I do that, posting the findings I have at the
time, just
in case someone might notice something and have a lead.
> The bu
Solution;
Took user info out of First and Last Name section of Users account.
Having a name in there breaks outgoing calls on the mediant.
Is there any way of having sipx strip that info over not being able to enter it?
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So, the problems seem to be down to this;
Abnormal Disconnect cause:50#GWAPP_REQUESTED_FAC_NOT_SUBSCRIBED
Nov 28 00:03:03 192.168.10.26 ( lgr_psbrdif)(2323 ) pstn send -->
PlaceCall:
Trunk:0 BChannel:8 ConnID:0 SrcPN=651x11 SrcSN= DstPN=1651x81 DstSN=
SrcNT=0
SrcNP=0 SrcPres=0
The busy out problem seems to be related to;
Nov 27 23:43:32 192.168.10.26 ( lgr_TrnkGrp)(1162 ) #1:TrunkGroup set
BusyOut to
0, reason: 48. BusyOut feature disabled, request ignored. [Time: 06:22:13]
If I'm reading this right, BusyOut will not happen because there is something
else wr
> We can make incoming calls but cannot make outgoing calls, always busy. The
> above error
That error isn't the problem. I guess I don't quite understand what sipx notes
are telling
me. When creating the gateway, the side note says to simply create the device
and pstn
lines, that everything e
TrunkGroup set BusyOut to 0, reason: 48. BusyOut
We can make incoming calls but cannot make outgoing calls, always busy. The
above error
seems to indicate the problem but can't find anything for reason 48 on the net.
Changing this setting doesn't seem to make any difference.
I fired up the old server and everyone registered immediately.
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sip
I see some other errors in here, such as BASE_DOMAIN for example.
"2009-11-28T03:25:26.716232Z":32:SIP:INFO:uc70.mydomain.com:SipRegistrar:B754CB90:SipRegis
trar:"[150-ISN] SipRedirectorISN::readConfig dialing prefix is empty"
"2009-11-28T03:25:26.716494Z":33:SIP:CRIT:uc70.mydomain.com:SipRegistra
Updated this and it made no difference. Wish it was forums, I could rename the
subjects on
some of these threads.
On Fri, 27 Nov 2009 21:08:19 -0600, m...@grounded.net wrote:
> Could it be something with the SSL cert? I just noticed this as well.
>
> "2009-11-
> 28T03:04:23.410421Z":49:SIP:NOT
On Fri, 27 Nov 2009 19:08:01 -0800, Todd Hodgen wrote:
> Check Internet Calling and make sure you have your Intranet domains and
> subnets defined correctly also. You do have domain in there and not FQDN,
> right?
Yes, I have the same settings as when things were working, two domain names, no
ho
Could it be something with the SSL cert? I just noticed this as well.
"2009-11-28T03:04:23.410421Z":49:SIP:NOTICE:uc70.mydomain.com:SipRegistrar:B74E7B90:SipReg
istrar:"BranchId::setSecret reset identifier key; previously generated branch
ids will not
be recognized as local."
_
Yes, nothing changed that I can think of and I simply restored. The only
difference would
be that the new server I installed DNS/DHCP on it while the old one was using
external
DNS.
This is baffling us and confusing as well.
Mike
___
sipx-users m
I turned some of the logging up to debug and now see;
"2009-11-28T02:36:14.837942Z":1737:OUTGOING:INFO:uc70.mydomain.com:SipRegistrarServer:B6AF
7B90:SipRegistrar:"SipUserAgent::sendTcp TCP SIP User Agent sent
message:\nRemote
Host:192.168.10.70 Port: 5060\nSIP/2.0 401 Unauthorized\r
I am seeing some errors in the sipregistrar.log
"2009-11-28T02:03:59.352184Z":3:SIP:WARNING:uc70.mydomain.com:SipRegistrar:B7519B90:SipReg
istrar:"LocationDB::load failed to load \"/var/sipxdata/sipdb/location.xml\""
"2009-11-28T02:03:59.352510Z":4:SIP:CRIT:uc70.mydomain.com:SipRegistrar:B7519B90:
I'm also watching the sipxbridge log. Not much or nothing obvious to me in
there.
"2009-11-28T00:28:37.043000Z":4:JAVA:INFO:uc70.mydomain.com:main::Gateway:"---
REGISTERING"
"2009-11-28T00:29:41.904000Z":1:JAVA:INFO:uc70.mydomain.com:main::SipXbridgeXmlRpc
Server
> 1. The configuration to turn on remote workers (where you put in your public
> ip and say server behind nat, etc.) Was not turned on
Internet Calling is Enabled.
On sipx, NAT Traversal options are both on.
> 2. The sip passwords are different and need to be re-entered on one end so
> both ends
Then they are not registering because:
1. The configuration to turn on remote workers (where you put in your public
ip and say server behind nat, etc.) Was not turned on
Or
2. The sip passwords are different and need to be re-entered on one end so
both ends match.
To
On Fri, 27 Nov 2009 20:26:06 -0500, Tony Graziano wrote:
> Its all basic. Assuming the ip address didn't change internally, you should
> also make sure the sip passwords stayed the same.
No IP changes and internal phones registered as soon as I restored but remotes
aren't
registering.
Its all basic. Assuming the ip address didn't change internally, you should
also make sure the sip passwords stayed the same.
If it changed (ip), your firewalls need to point to the new private ip.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email
Put together another server since I had to move hardware anyhow, included
dhcp/dns servers
this time then restored my config. All tests pass now incuding dhcp based 100%
and only
NAPTR fails since I don't have this in the internal DNS server, was told I
would not need
this, only on the public
On Fri, 27 Nov 2009 18:39:33 -0600, Josh Patten wrote:
> Totally normal. This simply means that the SRV record isn't in the local
> device, so it queries the external DNS server (IE the one you defined in
> the configuration). Nothing to worry about, all my Audiocodes devices do
> this.
That's wha
Totally normal. This simply means that the SRV record isn't in the local
device, so it queries the external DNS server (IE the one you defined in
the configuration). Nothing to worry about, all my Audiocodes devices do
this.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
M
I think your tftp server name should be an ip address.
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: 434.984.8427
Helpdesk Contrac
Some of the options in my dhcpd.conf file seem to be a little different from
the ones
you've suggested above.
Here are mine;
option sip-servers-name code 120 = text;
option routers 192.168.1.1;
option subnet-mask 255.255.0.0;
option domain-name "mydomain.com";
I keep seeing the following in the mediant logs yet each side can see each
other. What am
I missing?
Nov 27 17:09:51 192.168.10.26 ( sip_stack)(9575 ) Starting resolution
of server
number 0. Resolving mydomain.com [Time: 17:48:11]
Nov 27 17:09:51 192.168.10.26 (lgr_dns_resolver)(957
Nitin hotmail.com> writes:
>
> Hello
>
> I have setup Sipx, able to register poycoms - everything works well.
> I have encountered 2 problems -
>
> Problem 1
> Vegastream 50 Europa(ISDN), which will not dial. I have gone through the
dial
> rules a million times, still it wont.
>
> Here is w
I just set up sipXecs for the first time, and it seems very cool. However,
I can't seem to get outbound calling to work through and ITSP. I'm using
vbuzzer.com, which isn't in the supported ITSP's list, but I have had it
working with freeswitch for a while, and I didn't have to do anything
specia
If you read the issues being tracked at the tracker:
track.sipfoundry.org
You'll be able to understand more. I think the core focus is to replace the
VM system and add the features already mentioned. Further upgrades will be
built upon that fs engine instead.
I would encourage you to read both t
*Bump*
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
Josh Patten wrote:
Here are
the three advantages I've seen to using the new FreeSWITCH
voicemail system over the old one so far:
Better Performance
IMAP integration
HD Audio
Are there other functional a
At the same time there is no assurance that trying sipXbridge in 4.0.4
will fix anything in this respect. My hands are tied. I'm not trying to
shirk my open source responsibilities (yes, everyone that uses open
source software, in my opinion, is responsible for contributing back in
some way) bu
I think its possible this has been resolved, but there is no way to tell
until you use the stable code (4.0.4) that has resolved a few sipxbridge
issues that were not addressed in the patches.
I don't think the logs in your issue will be looked at until you post a
recent log with 4.0.4. I might be
Josh wrote:
> Are there plans to add configuration support for the
> SoundPoint IP 335 to 4.2? The only configuration differences
> between the 335 and 330/331 are that the 335 supports HD
> Voice/G.722 and uses an RJ-9 headset port.
> The nicest thing about this new model, though, is the
> ad
Unfortunately I have provided all I can provide (my boss has mandated
not to use sipXbridge until the problem has been resolved, and is in
the process of ordering a Mediant 1000). I posted a second shapshot for
both sipX servers in that ticket that I don't think anyone has looked
at with severa
Unfortunately, this description does not help me a lot in trying to
resolve any problems you might be experiencing. There are too many
interacting variables at play. If you can isolate the problem ( for
example run a single problematic call flow with sipxbridge in the
middle between sipx and Aster
On Fri, 2009-11-27 at 10:53 -0500, Andres Jaramillo wrote:
> Looking in the Mediant1000 side, i just found this:
>
> [WARNING] RegistrationController::RegisterResponse - AKA was required,
> but failed. Can not work with this proxy !!!
>
> someone have seen this kind of error, or maybe what could
I've got debug running on the mediant and sending syslog to a syslog server.
I'm watching
the logs.
Here is the very start of a call which isn't working. Here I am actually
dialing a 612
area code. The number however, comes into the mediant as a 651;
Nov 27 09:59:14 192.168.10.26 ( lgr_psbr
On Fri, 27 Nov 2009 02:12:06 -0800, Todd Hodgen wrote:
> The "Terminal Adapter" in this case will be a Channel Service Unit or CSU.
That's a good thought and is something I wanted to make sure about also. The
PRI
terminates directly onto the mediant. There was an adtran in the mix at one
point
I'm installing SipXecs on SUSE, however i notice that in the section of
4.0.4 there are only 4.0.3 rpm's.
Where are 4.0.4 rpm's for SUSE?
thank you,
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Are there plans to add configuration support for the SoundPoint IP 335
to 4.2? The only configuration differences between the 335 and 330/331
are that the 335 supports HD Voice/G.722 and uses an RJ-9 headset port.
The nicest thing about this new model, though, is the addition of a
backlight to
Looking in the Mediant1000 side, i just found this:
[WARNING] RegistrationController::RegisterResponse - AKA was required,
but failed. Can not work with this proxy !!!
someone have seen this kind of error, or maybe what could means ?
Thanks !!!
2009/11/24 Tony Graziano :
> I would think it shou
Its a workaround for a cludgy aastra phone.
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: 434.984.8427
Helpdesk Contract Customers
Yeah, I tried just about everything I could think of, from adding a
domain alias for the IP of the server and registering directly to the
IP, to setting the outbound proxy to the IP of the sipX server and I
never could get it to register. It's sitting on a shelf gathering dust
right now...
Jos
On Fri, 2009-11-27 at 09:13 -0600, Robert B wrote:
>
> Isn't the handset supposed to look for the SRV record for the
> registration? If I change registration to 2...@pbx-1.domain.tld, the
> phone registers just fine.
Yes, it is supposed to look for the SRV record first.
I don't know enough abo
Well sure, but isn't that breaking the whole purpose of DNS-SRV?
According to Aastra's documentation, DNS-SRV is supported.
-- Robert
Tony Graziano wrote:
> Create an "a" record for the domain that points to sipx.
>
> Tony Graziano, Manager
> Telephone: 434.984.8430
Josh,
I only have one that I am testing with. I don't have a Polycom yet. I
will be standardizing my deployments on Polycom 450/550/650s, I've
already made my mind up on that. I just happen to have this Aastra 57i
kicking around here that I used in mid-2008 to test Asterisk with.
See my follow
Create an "a" record for the domain that points to sipx.
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: 434.984.8427
Helpdesk Contr
I was never able to get my 53i to register with sipX. Technically it's
possible but I never delved more deeply into it because I only have one
of them and I didn't feel the time I would spend trying to get it
working was worth it.
How many 57i's do you have? They aren't known to be particularly
Turned on logging in BIND to see what the Aastra was querying for:
She boots, queries NTP, finds the TFTP host, and then tries to look for
an A record instead of the SRV records:
Nov 27 18:36:02 dns named[1080]: client 172.16.0.137#1024: query:
pool.ntp.org IN A +
Nov 27 18:36:05 dns named[10
Folks,
(Be advised that I may be misunderstanding how this is supposed to
work... I am new, previously come from the Asterisk world where
everything was "pixie dust"...)
I've setup DNS and DHCP according to
http://sipx-wiki.calivia.com/images/0/0b/SipXecsDNSConcepts.pdf and the
Aastra 57i han
The weird thing is that these issues only crop up when I'm using
sipXbridge. If I'm trunking through Asterisk I never have call
reliability issues. If I am trunking through Asterisk with sipXbridge
in the middle it is unreliable. If I am trunking directly to my PRI
gateway with sipXbridge in th
On Fri, 2009-11-27 at 12:49 +0300, Nikolay Kondratyev wrote:
> Hi all,
>
>
>
> I have two sipx systems connected via sipxbridge.
You don't really need sipXbridge to connect to sipXecs systems. Instead
of configuring this as a SIP Trunk, just configure the dial plan using
the Site-to-Site rul
I saw in the logs from his earlier postings "two" numbers. It would receive
one, then "dial" another.
My question is whether any of these numbers have a RFC or some type of
forwarding on them from the TELCO. If this is the case, have the provided
issue temporary numbers to test with instead of wha
The "Terminal Adapter" in this case will be a Channel Service Unit or CSU.
They are required on the line, many people don't use them, which should not
be the case. They ensure 1's density on the T-1, so you avoid issues on the
span itself from timing slips, etc. It also provides a loopback point
Hi all,
I have two sipx systems connected via sipxbridge.
All looks to work ok, but.
When a user calls from sipx1 to sipx2 and sipx2 user tries to make blind
transfer to another sipx1 user it does not work.
My analysis is as following: sipxbridge2 converts Refer into Invite, and
sends it to
Yes, they do go back to zero after some minutes.
Regarding Dale's comments: I cannot see any problem with the SIP
scenarios between the load test agents (call sources and destinations)
and SipX (which does not mean that there is not any problem :-)).
The result means to me that SipX under load re
m...@grounded.net wrote:
> I'm still looking for thoughts on this from folks who have installed enough
> of these
> things to know how I might be able to handle this.
>
> The telco did come in, did show me that 612-xxx- is coming into the
> t-berd. I blamed
> the vegastream at the time, now
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