27;ll post back here if I find out what the
> problem was, in case someone is having similar issues.
>
> Thanks again,
> Olli
>
>
>
> 2014-04-22 21:06 GMT+03:00 Pedro Niño :
>
>> Don't forget to include peer type (friend), and The callbacknumber In The
>>
a bit and ask in the asterisk list on
> more tricks on asterisk side. I'll post back here if I find out what the
> problem was, in case someone is having similar issues.
>
> Thanks again,
> Olli
>
>
>
> 2014-04-22 21:06 GMT+03:00 Pedro Niño :
>
>> Don't forge
Don't forget to include peer type (friend), and The callbacknumber In The
table.
It happened to me and asterisk/kamailio behavior was wayyy to weird until
made sure both parameters were there.
-
In this setup I have SIP peers in an asterisk table added like this:
INSERT INTO sippeers (name,
Getting?
El abr 7, 2014 1:21 PM, "Slava Bendersky"
escribió:
> Hello Pedro,
> I just come back on line.
> If i remove this line I start getting
>
>
> ------
> *From: *"Pedro Niño"
> *To: *"Kamailio (SER) - Users Maili
rs-boun...@lists.sip-router.org [mailto:
> sr-users-boun...@lists.sip-router.org] *De la part de* Pedro Niño
> *Envoyé :* mercredi 2 avril 2014 02:47
> *À :* Kamailio (SER) - Users Mailing List
> *Objet :* Re: [SR-Users] Crash on REGISTER
>
>
>
> Looks like its crashing becaus
Looks like its crashing because a special character trying to be parsed,
maybe a character used for password?
Not sure, but try to use simple password (just for test) without special
characters and check if having crash again.
BTW, Daniel is busy, expect answer but be polite and have some patienc
d_reply("200", "OK"); }
-
El abr 1, 2014 7:58 PM, "Pedro Niño" escribió:
> Sorry, I was out for a while. Still have this issue?
>
> From what I am seeing, asterisk is expecting for the password. Is the
> voicemail configured ? Check username and passw
SIP dialog
> 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' Method: INVITE
> Reliably Transmitting (no NAT) to 10.237.236.207:5060:
> OPTIONS sip:10.237.236.207 SIP/2.0
> Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK51a7f1ef
> Max-Forwards: 70
> From: "asterisk"
it?
>
> Rizwan Khan
>
>
>
>
> On Mon, Mar 31, 2014 at 5:52 AM, Pedro Niño wrote:
>
>> As Alex said, I/O and calls per second (CPS) is Dependant on what type of
>> design you are using.
>>
>> As a tip, mysql is good for small to medium sizes, but begin to chok
uot; enabled.
> A 484 is used for overlap dialing. The server says "I need more digits to
> complete this call".
>
> /O
>
> On 31 Mar 2014, at 02:30, Pedro Niño wrote:
>
> I think this is the correct behavior, as asterisk server is complaining
> about the addr
2014/03/30 22:32:47.140301 192.168.10.120:5060 -> 192.168.10.120:5062
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK6f8354a0.
> From: "asterisk" ;tag=as2cbae229.
> To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.93f4.
> Call-ID: 456f80c06d85d9b34027ccc5
sponse with a 200 yourself, eg:
route[REQINIT]{
...
if($si=="192.168.182.24" && is_method("OPTIONS"))
{
sl_send_reply("200","Up and running");
exit;
}
It's up to you to decide to which OPTIONS
98f5f3a7e58.b6d2
> Call-ID: 4dc3968e64c3c16c4ad4f2407f2af...@networklab.loc
> CSeq: 102 OPTIONS
> Contact: ;expires=3600
> Server: kamailio (4.1.2 (x86_64/linux))
> Content-Length: 0
>
>
> if ($rU==$null)
> {
> # request with no Username
As Alex said, I/O and calls per second (CPS) is Dependant on what type of
design you are using.
As a tip, mysql is good for small to medium sizes, but begin to choke at
1000 of concurrent connections. but in a scalable size like you are
planning, I would recommend to make a design with resilience
I think this is the correct behavior, as asterisk server is complaining
about the address/request not containing all the necesary data to process
the message
Can you please elaborate with a bit more of detail? Also can use tools
like sngrep, tcpdump (or wireshark) to have a better view of the co
27;OK' .
Will keep trying, don't like to leave that back door open at the phones...
On Thu, Mar 13, 2014 at 8:26 AM, Pedro Niño wrote:
> The other (ugly) option, is to remove the auth from the phone, for the Sip
> Provisioning, but that would leave and open door to a reboot attack
Sorry for intruding, but this is puzzling me a lot.
I followed the guide to make Kamailio work with asterisk and realtime,
Kamailio version 4 and asterisk 11.
(http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
)
up until now its working, and now want to get the phones (
Hi, how are you all
I have a lab setup for using Kamailio and asterisk, with realtime, and
currently I am facing 2 problems.
The first one, is when trying to get the phones to reboot. Asterisk send a
notify with the message, reboot. But the phone answers with a 401, and asks
for a digest.
In a n
Try to move to a wired solution, or if you are in a crowed wireless site,
try switching the WiFi frequencies. Moving from 2.4 to 5 GHz can help .
I would place my bets on the wireless section. Maybe QoS, if your router
can handle?
El feb 2, 2014 8:48 AM, "Ravi" escribió:
> Dear Brian,
>
> Thank
19 matches
Mail list logo
