Hello,
On 20/06/14 06:16, David Wilson wrote:
Hi All,
I'm trying to add a permanent usrloc entry via kamctl ul add.
This works, but the created entry has a q value of 1.0 which is higher than I
need.
Is there a way to either:
1. Specify a q value when using kamctl ul add, or
apparently the
Hello,
why is R sending the 408? You should catch it there in a failure route,
or where do you execute m_store() inside R config?
Cheers,
Daniel
On 20/06/14 07:20, Allen Zhang wrote:
Hi all,
I have an edge proxy (E) and a registrar (R) behind it.
In R, if a MESSAGE failed to deliver for a
Hi all,
I have an edge proxy (E) and a registrar (R) behind it.
In R, if a MESSAGE failed to deliver for any reason, R stores the MESSAGE in
msilo.
If the MESSAGE timed out, R sends a 408 time out first and then send a 202
Accepted after the MESSAGE is stored.
The problem is, E happily forwards
kamctl is a basic script to get users familiarized with kamailio tables.
You will need to use SQL command to configure specific fields.
Alternatively you can install siremis or enable the xhttp_pi module.
Regards,
Ovidiu Sas
On Jun 20, 2014 12:16 AM, "David Wilson" wrote:
> Hi All,
>
> I'm tryin
Hi All,
I'm trying to add a permanent usrloc entry via kamctl ul add.
This works, but the created entry has a q value of 1.0 which is higher than I
need.
Is there a way to either:
1. Specify a q value when using kamctl ul add, or
2. Edit the q value of an existing record by using a kamctl co
Shedding some more light on the situation, here are some further facts I
could figure out:
1) I did the same test without the proxy in between (UAC <--> UAS) purely
in SIPp, everything was fine, no dead call error
2) With the Kamailio proxy in between, I can see that there is a problem in
call te
Thanks very much Daniel,
This is solving a part of my issue for now.
Here is the routing section of my configuration file (it may be good to
share it once complete as a minimally working proxy):
route{
> if (!mf_process_maxfwd_header("10")) {
> sl_reply("483","Too Many Ho
Hello,
in the source code, the sql scripts are in:
utils/kamctl/mysql/
With v4.1.x, you should be able to create tables individually by:
kamctl add-tables ID
where you take the ID from the sql create scripts, by removing the
suffix '-create.sql', like:
kamctl add-tables standard
kamctl add
Hello,
It's unfortunate that error codes are not included with management commands.
For example, if you tape " kamctl diaplan rm rule 2 999 " instead of "
kamctl diaplan rmrule 2 999 " then all the entries in dialplan table will
be deleted.
In fat the system perform "kamctl diaplan rm"
It's ve
Hello and good evening all!
Am trying to build a Kamailio installation in my lab and have hit the first
hurdle; database build. I have a three node MariaDB cluster in place and
attempting to get kamdbctl create to set up the tables. What seems apparent is
that command requires the database to
Hi,
I am trying to establish a chat session(MSRP) with two sipclients endpoints
by registering to kamailio server(4.0.0), but I am getting 500 Internal
server error(Reason: SIP ;text="media stream failed to start" ;cause=500).
What might be the problem?? Does it mean that kamailio can not handle ms
On Thursday 19 June 2014 10:42:49 Keith wrote:
> Thanks Daniel, however don't I need to send a register to get the 401 back?
> If so how do I do this bit?
Your message sounded like need to authenticate outbound calls, registrations
will direct inbound calls to your machine. Never tried it, but fo
I will need to take a look at this after hours. I did installed siremis
4.0 and then 4.1 on the localhost before moving the DB. I also did copy
the sql backups to the new server.
Thank you
On Thu, Jun 19, 2014 at 7:48 AM, Joel White wrote:
> Yeah, this is the message I get in syslog.
>
>
> J
Hi Mark,
Yes you need to define the sip proxy as USECALLMANAGER instead of the IP itself.
If you want to send me to xml file I can take a look, as I have configs for a
few different phones.
Thanks
Jon
From: m...@darkorigins.com
Date: Thu, 19 Jun 2014 12:24:04 +0100
To: sr-users@lists.sip-router.o
Yeah, this is the message I get in syslog.
Jun 19 07:41:37 VoIP-Platform-Primary /usr/local/sbin/kamailio[1682]:
ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on
query: The user specified as a definer ('kamailio'@'localhost') does not
exist
Jun 19 07:41:37 VoIP-Platform-
Hi Jon
Thanks for that. Trying to translate between the txt version and the xml
version, most of the settings seem to be about right.
I think the problem might be with the way we’re referring to the sip server in
the xml. The previous advice I got was to do it using the call manager section
Hi Mark,
Yes we had exactly the same issue with a 7940.
It is with the .cnf xml file.
You need to make sure that the line name is populated;
line1_name: "<>"line1_authname: "<>"line1_password:
"supersecret"
See an example file attached, hopefully will give you an idea.
Thanks
Jon
From: m...@da
Hi Jon,
In terms or software version for the Cisco, we use SIP45.9-3-1SR4-1S, this
being for 7945,7965,7940,7941,7961.
All seem to work ok with kamailio version 4.x
Jon
From: john.mur...@skyracktelecom.com
To: sr-users@lists.sip-router.org
Date: Thu, 19 Jun 2014 10:58:10 +0100
Subject: Re: [SR-
Hi Jon
I’m working with a variety of routers here at the moment (testing / lab
environment :-) ranging from Mikrotik, Draytek to Firebricks.
So far;
- Phone on test is a 7970
- Variety of routers here at the moment (testing / lab environment :-) ranging
from Mikrotik, Draytek to Firebricks.
Jon hi,
Which software version/image do you find works best for this?
We found some later versions to have introduced more problems.
Regards
John
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Jonathan Hunter
Sent: 19 June 20
Hi Mark,
Sure of course, they are some what painful to get working due to their
asymmetric NAT behaviour.
What handset models are you working with, and what firewall devices do you have
on site, as I have them working with a Cisco ASA on the network edge.
Then I can give you some more details on
Hi Jon
It sounds like you’re a few steps in front of me. I’m just starting to try and
get a working set of config files etc for provisioning to the phones. Which at
the moment is going through the motions of registering but not actually sending
anything.
Would you mind sharing your experienc
Hi All,
As you guys might remember I have been doing alot of work with these legacy
handsets recently, in particular the Cisco IP phones 7945G and 7965G.
They work well now with kamailio, using UDP or TCP as the transport protocol.
I am now looking to implement SIP over TLS with them, and wondered
Hi,
Thanks Daniel, however don't I need to send a register to get the 401 back?
If so how do I do this bit?
Thanks
Keith
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