Normally I run this against a carrier rate sheet, using the description.
Both Twilio and Flowroute have decent download-able sheets with prefix
<-> country/mobile description.
--fred
On 01/18/2017 09:21 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> slightly off-topic, but
What happens when you try:
modparam("sipcapture", "hep_capture_on", 1)
On 01/13/2017 10:33 AM, JR Richardson wrote:
> Iptables is not blocking, but it was worth a check.
>
> Thanks.
>
> JR
>
>
> I assume you have ruled out firewall? It's something that can nab even
> experienced people:
>
>
> listen=udp:MY_IP_ADDR:5060 advertise MY_PUBLICIP_ADDR:5060
That statement does not exist anywhere in the files you sent.
--fred
On 12/29/2016 11:19 AM, Pranathi Venkatayogi wrote:
> Yes. I defined advertised address and even used listen with advertise as
> below. Still Kamailio does
Thank you for the post-- definitely appreciate you sharing it on this list.
--fred
On 12/8/16 6:02 PM, Matthew Jordan wrote:
Hey all -
The Asterisk project just released a security advisory for a security
vulnerability in which Asterisk using chan_sip with a proxy can allow for
On 10/06/2016 05:43 AM, Daniel-Constantin Mierla wrote:
> On 05/10/16 16:35, Fred Posner wrote:
>> On 10/05/2016 10:25 AM, Daniel-Constantin Mierla wrote:
>>> Hello,
>>>
>>> writing here to decide on a topic opened by pull request 779:
>>>
>>
me to use additional
authentication methods.
I believe Polycom still max's out at 32.
--fred
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http://lists.kamailio.org/cgi-bin/mailman/listinfo/business
You also may want to check out the business directory:
https://www.kamailio.org/w/business-directory/
There are some products that involve Kamailio at it's core, such as
Canonical SIP Routing Platform (CSRP), Enswitch, Sip:Wise, 2600
On 07/11/2016 11:40 AM, Jay Li wrote:
> Fred,
>
> Thanks a lot your detailed explanation. About the media server addition
> to Kamailio, do you have any suggestions I should look into besides
> Jitsi and FreeSWITCH? Thanks.
>
> Regards,
> Jay
You could look into Asteri
on video conferencing. So, you will need this being done
either by a separate media server or endpoint capable of doing this.
There are some products like Jitsi Video Bridge and FreeSWITCH that
support video conferencing "out of the box." You can combine these with
Kamailio as well to handle a
ibutes and they are not removed.
>>
>> Why SDPOPS does not remove these attributes?
>
> Probably because there's a problem rewriting parts of the SDP body more
> than once. But if you don't want ICE attributes in the output SDP, you
> can use the rtpengine fla
If it's just 2 servers, consider as Juha said, corosync/pacemaker with drbd.
Fred Posner
direct: +1 (224) 334-FRED (3733)
> On Jun 5, 2016, at 5:26 PM, Moacir Ferreira
> wrote:
>
> Hi,
>
> Sorry... I should have mentioned before. You guys are thinking on the
> s
ooks/4.4.x/transformations#to-body_transformations
--
Fred Posner
@fredposner
The Palner Group, Inc.
http://www.palner.com
direct/sms: +1 (503) 914-0999
direct/sms: +1 (224) 334-FRED
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ould this be wanted?
I ask, as if you were using a srv record as the result of a load balance
lookup, wouldn't the point be to be able to quickly change location of
the domain in case of an outage/issue?
Otherwise, I'm not positive the benefit of doing a srv lookup for this
scenari
Have you considered either dispatcher or just using a failure route?
-- Fred
> On May 6, 2016, at 7:15 AM, Alberto Sagredo
> wrote:
>
> Hi
>
> I have it working but i have re-read documentation and do not see how to do
> what i need.
>
> I explain it :)
&
ds... it just continues on the mailing lists. =)
Best regards,
Fred Posner
http://www.palner.com
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ort() accept PV
> arguments, or are they pre-PV "core function folk traditions" in the
> same way as rewritehostport() and force_send_socket()?
>
I have a main listen=udp:192.168.25.31 advertise PUBLIC:5060
and then when needed...
set_advertised_address("192.168.25.31&q
this problem once, and it resolved once I switched. Switching back
hasn't seen the issue return.
http://www.fredposner.com/1680/kamailio-4-2-3-update-from-git/
Fred Posner
The Palner Group, Inc.
+1-224-334-FRED (3733) direct
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rnal address... but that being said, if you're using public
IP for everything on the LAN, look into advertised_address:
https://www.kamailio.org/wiki/cookbooks/4.3.x/core#advertised_address
This being said, I'm confused by your scenario.
Fred Posner
The Palner Group, In
cally opposed to a B2BUA in Kamailio
> to the threshold of physical violence.
>
> -- Alex
>
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-224-334-FRED (3733) direct
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Hi,
the X-Lite is a normal Sip Client .. for IMS registration you need an IMS
client .. such as Boghe based upon doubango.
You are only able to use normal username/password register while using X-Lite
Cheers,
Fred
Am 18.02.2016 um 11:53 schrieb
sainath.ellend...@wipro.com
Hi Jason,
thank you for your answer. But could you please explain how the UE is
identified?
Is it the contact header? Or some other stuff .. I wasn’t able to find any
information.
And yes, we plan to use the Rx as well. I am can make some traces and logs at
Monday.
Thank you
Fred
Am
Again, thank you!
?
Von: sr-users im Auftrag von Aaron
Hamstra
Gesendet: Donnerstag, 28. Januar 2016 21:14
An: Kamailio (SER) - Users Mailing List
Betreff: Re: [SR-Users] P-CSCF
Fred,
I think you would need to use rtpengine instead of rtpproxy.
https
pected.
ERROR: rtpengine [rtpengine.c:1622]: rtpp_test(): proxy responded with invalid
response
Any advise would be nice.
Thank you
Fred
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Hello Daniel,Just got this email from
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb,my
requirement is that I am looking at creating an asterisk server to work as
SBC.wondering if you can help me on this one.
thanks and regards,Fred
can run sql queries as well as call stored procedures.
You could easily run the query, store in xavp, free results and the use
the xavps later on (such as in failed route, etc).
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-224-334-3733 (direct)
0x6235BD69.asc
Description: a
facing and fix it accordingly.
>
> Regards,
> Ovidiu Sas
>
> On Oct 8, 2015 09:04, "Dirk Teurlings - SIGNET B.V."
> mailto:dteurli...@signet.nl>> wrote:
>
> On 30-09-15 13:29, Fred Posner wrote:
>
>
> Without a version of rtpproxy using
On 09/30/2015 07:15 AM, Dirk Teurlings - SIGNET B.V. wrote:
> On 30-09-15 12:23, Fred Posner wrote:
>>
>> Are you using -A flag in rttproxy?
>>
>
> Unfortunately we're running a version of RTPPROXY at the moment that
> doesn't have this flag. I'm con
27;ll need to set rtpproxy (override) to
use NAT/LAN IP, and use the public IP for the other side.
Something like...
if(dst_ip==ASTERISKIP) {
rtpproxy_manage("co","PRIVATEIP");
} else {
rtpproxy_manage("c
t README displayed as markdown file by github.com on
> the website of kamailio project repository, without renaming it to
> README.md.
>
> Cheers,
> Daniel
>
I was incorrect.
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
__
rticles/github-flavored-markdown/
And of course a list of markups listed with
https://github.com/github/markup.
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
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On 08/20/2015 08:34 AM, Alex Balashov wrote:
> Fred,
>
> Throwing in B2BUAs is a common recipe for interop problems; one side
> is liberal in what it accepts, the other conservative in what it
> emits. In theory.
>
> For lightweight, signalling-only duty, allow me to sugg
On 08/20/2015 08:25 AM, Fred Posner wrote:
[SNIP]
>
> Without topology hiding, the Mitel will ignore Record-Route headers. If
> using topology hiding, they will send an in-dialog invite to the sdp c=
> address...
>
EDIT: on the o= information... I need more sl
ave in a more respectful manner.
I've also thrown in either FreeSWITCH or Asterisk as a media server for
those end-points to just "handle" it; which may end up being a decent
solution for the trouble-making end-points.
--fred
___
SIP Ex
On 08/18/2015 09:39 AM, Jean-Marie Baran wrote:
> Should I understand that the router should send the packet back to
> Kamailio which then send them to the SIP server ?
RTP packets will flow from client to client or with rtpproxy, from
client <-> rtpproxy <-&g
up issue on the Mitel. I
don't work with Mitel enough to discuss in "Mitel speak" what to change.
Apparently RTP ports is a different language to the people I am working
with. =)
--fred
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On 08/18/2015 09:06 AM, Jean-Marie Baran wrote:
>
> The question mark denotes the fact that I lose trace of RTP packets
> here. Any idea why the packets are not relayed ?
What ports do you have opened and forwarded on your firewall/router
I rarely integrate these with Kamailio and am having some "resistance"
in assisting. I was wondering if anyone would be willing to share any
configuration recommendations for Mitel when using Kamailio as a SIP
trunk (outbound/inbound).
Sincerely,
Fred Posner
The Palner Group,
should also be in the advertised statement within kamailio.
--fred
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On 08/13/2015 03:30 PM, Alex Balashov wrote:
> I see a dialog established followed by a reinvite. Which frame # do you
> take to be the "on-hold invite"?
>
I was taking frame 8 as a new invite. I'll see what I'm doing wrong.
--fred
__
On 08/13/2015 02:42 PM, Alex Balashov wrote:
> On 08/13/2015 02:34 PM, Fred Posner wrote:
>
>> Sadly, no. Straight up INVITE.
>
> Wait, what? That doesn't make any sense. Can you provide a full
> signalling capture?
>
Here's an example exported from the pcap
certain if my lack of proxied media would be a factor.
--fred
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Hello all,
Always fun integrating with proprietary pbx's... is there a way that is
"generally accepted" as handling hold invites from devices such as Mitel
or Toshiba?
We're not proxying the media, so these devices are trying to send an
Invite to the endpoint when calls are p
Is there a reason why you cannot disable SIP ALG?
--fred
On 07/30/2015 03:12 PM, Joachim Büchse wrote:
> I’m coming back to this very old question as we have still not resolved
> this issue with Juniper.
>
> Are you aware of any RFC section that mandates, that the VIA headers
>
dispatcher will update the
load state for the node.
Fred Posner
f...@palner.com
http://palner.com
On 07/23/2015 03:48 AM, Alberto Sagredo wrote:
> Hi
>
> Hi have read documentation but it seems dispatcher does not keep how
> many calls has been dispatched or currently are in any o
INVITE
if (!pl_check("$au", "traildrop", "$var(limita)")) {
pl_drop();
exit;
}
}
if(is_method("REGISTER")) {
# Checking limit on REGISTER
if (!pl_check("$au", "traildrop", "$var(limitb)")) {
pl_drop(
On 07/08/2015 12:31 AM, solution wrote:
> but while i manually killing that application fail over switching is not
> happening.
Are there any errors?
--fred
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sr
r/commit/0ead7ab31bf0df6eb313dc3f9a4979c2a45cce8b),
> but that code is very different, so I'm wondering if a regression has
> crept in? I'm using Kam 4.1.6.
>
> Thanks,
> Kyle
>
Kyle,
How are you using allow_address?
--fred
_
On 06/22/2015 11:41 AM, kai.ohnac...@cbc.de wrote:
> In the txt file you can find the ngrep traffic.
>
> Cheers,
> Kai
>
> -
>
> Perhaps you have an ngrep of the sip traffic?
>
> --fred
>
Is .3 the u
2 localhost blox-0-9-6-beta[1975]:
> ERROR:core:parse_cseq: no method found
>
> Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]: ERROR:core:parse_cseq:
> bad cseq
>
> Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]:
> ERROR:core:get_hdr_field: bad cseq
>
> Perhaps someon
using either Dialog module or HTABLE instead?
http://kamailio.org/docs/modules/stable/modules/dialog.html
http://kamailio.org/docs/modules/stable/modules/htable.html
I would start with htable...
--fred
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Kamailio is released under GNU Public License v2 (GPLv2), which does
provide for commercial use. It's a very common licensing structure and
of course I am not a lawyer and this should not be taken as legal advise.
You can/should use the licensing information provided to have your
lawyer advise yo
Even tel2sip shows it as a valid uri.
http://kamailio.org/docs/modules/4.2.x/modules/siputils.html#siputils.f.tel2sip
I again apologize.
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
On 06/12/2015 02:52 PM, Alex Balashov wrote:
>
> On 06/12/2015
I stand corrected.
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
On 06/12/2015 02:41 PM, Alex Balashov wrote:
> Another example:
>
> http://www.dialogic.com/webhelp/IMG1010/10.3.3ER2/WebHelp/Description/Interworking/SIP_Carrier_Identification
My understanding of 4694 was that for SIP, it still references 3261,
which was :USERNAME:PASSWORD@HOSTPORT;PARAMS, no?
I thought the 4694 use of parameter was only applicable in a tel: or
similar field.
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
On
sed
on parameters starting with a ; and ending with the last @.
The sed for it would be:
sed 's/^sip:\(.*\);.*@/\1@/'
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
On 06/12/2015 12:38 PM, Alex Balashov wrote:
> Could, but that would be a complica
You could always to a search replace for the uri based on {uri.params}
(replacing with nothing).
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
On 06/12/2015 12:24 PM, Alex Balashov wrote:
> Sure, but that presumes that I know the position of
t version...
http://www.fredposner.com/1457/kamailio-behind-nat/
The only changes I would make would be to use the stable version of
Kamailio (or 4.3 at this point)... and install rtpproxy 2 from git.
Fred Posner
The Palner Group, Inc.
http://www.palner
Have tested it on virtual and physical. Works well, no need to patch for VM or
advertised address. Had no complaints from users with 1.2 and none since 2.0
Installed from git for testing.
-- Fred
> On May 22, 2015, at 4:37 AM, Klaus Darilion
> wrote:
>
> Hi!
>
> I j
I sent an answer to this same question from you yesterday.
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
On 05/06/2015 10:44 AM, Yogendra Gupta wrote:
> Hello,
>
> I have setup kamailio at AWS server and it is connecting at private IP
> n
6) on 54.149.7.246: Cannot assign requested address
>
>
>
> Let me know where I need to fixed it
>
>
>
> Thanks
>
In your LISTEN statement, you probably have an IP that is not on the system.
Try something like:
listen=udp:X.X.X.X:5060 advertise Y.Y.Y.Y:5060
wher
, channel went silent.
I've added it back, partly out of spite, but mostly for the "proper"
handling.
--fred
On 03/27/2015 06:47 AM, Alex Balashov wrote:
> I suppose this speaks to it:
>
> https://tools.ietf.org/html/rfc6665#appendix-B.19
>
Yes, record route is being generated for all SUBSCRIBE, NOTIFY, and just
in case REFER, INFO, PUBLISH.
--fred
On 03/27/2015 06:38 AM, Daniel-Constantin Mierla wrote:
> 202 is ok, so freeswitch has created the subscription dialog and should
> send notify requests with event dialog.
>
&
For both UDP and TCP I receive a 202.
--fred
On 03/26/2015 06:45 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> actually there is nothing wrong with the NOTIFY via TCP. But it is not
> for Event: dialog (which is for BLF), it is for Event: refer (which is
> for other purposes,
ent-Length: 16.
.
SIP/2.0 200 OK.
My thoughts is that this is on FreeSWITCH, just not sending a NOTIFY
with anything of value. That being said, I'd love to know if (a) anyone
else has had this issue or (b) if someone has a different theory.
Thanks!
--
Fred Posner
The Palner Group, Inc.
135
> sip:0049...@sip.domain.tld:: 1
> Entry:: 205
> sip:0049...@sip.domain.tld:: 1
> FIFO command was:
> :sht_dump:kamailio_receiver_20144
> foo
>
>
> any idea what is wrong?
> and is there a documentation how to use "kamcrl mi"
there is no way to verify if this is really
>> the latest release. No ssl, no dnssec, no signed checksums. These should be
>> considered also.
>
> I would love seeing signatures on releases. I think there's a key for the RPM
> packages somewhere.
>
> /O
+1
Tom,
This list is for users, but we do have a list just for business requests:
http://lists.kamailio.org/cgi-bin/mailman/listinfo/business
We also have a business directory:
http://www.kamailio.org/w/business-directory/
You'll have great luck with both of the above.
Fred Posner
The P
Do you have any of the sip traffic?
Also, are you using FreeSWITCH for the media of WSS?
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
On 02/05/2015 09:38 PM, Don Fanning wrote:
Hello All,
I currently am running Kamailio in a WSS configuration with
-by-step instructions for installing via git,
which is the recommended method.
http://www.kamailio.org/wiki/install/4.2.x/git
You can copy / paste most of these commands to do a fairly quick install.
Regards,
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999
Hello Kai,
Many examples of this question exist on the mail list if you search
through it.
Also, I believe this is an example of what you might want to look for:
https://github.com/xlab1/sipfe_kamailio/blob/master/kamailio.cfg
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web
erisk based hardware, enthusiasts, or support vendors.
Again, this might be a great way for you to contribute.
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)
___
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With no private repositories, gitlab is free which is nice.
---Fred
> On Nov 9, 2014, at 1:20 PM, Jan Janak wrote:
>
>> On Thu, Nov 6, 2014 at 11:40 AM, Jan Janak wrote:
>> If you prefer to keep a self-hosted git repository, I think we should at
>> least move
I'm certain that LOD would be willing to sponsor the server for git /
tracker and I'd offer to handle the sysadmin of the server.
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)
On 11/05/2014 09:25 AM, Daniel-Constantin Mi
You will need to install mysql if you would like to use a mysql
database. It is not required that you use mysql. Other databases are
supported as well as a database not being a requirement for the software.
Fred Posner
On 10/24/2014 08:57 PM, Mahmoud Ramadan Ali wrote:
No ! i do not have
Do you have mysql installed?
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)
On 10/24/2014 08:52 PM, Mahmoud Ramadan Ali wrote:
Hiii everyone,
I can not create kamailio database and get this error message...
any ideas ?
Thanks in
If you want to call a user on Kamailio from Asterisk...
example...
exten => s,1,Verbose(4,calling user on kamailio)
same => n,Dial(SIP/USERNAME@KAMAILIO,time,options)
same => n,--after dial logic --
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (d
the chain of systems on your network (ie the
asterisk boxes). Sometimes the use of TOPOH helps to integrate with the
carriers who have chosen their own "interpretations" of RFC for "security."
And there's more...
The bottom line, is that the devil is in the details.
$avp(s:user) = $rU;
};
if (is_avp_set("$avp(s:user)")){
sip_trace();
#setflag(22);
};
};
Has anyone a tip for me how I can get rid of the empty traced_user lines?
Regards
Fred
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es listed here. It is your
sole decision to do business with any of the entities listed here and
all commercial relations and liabilities are only between you and your
business partner, without any involvement of the two open source
projects."
On that page, you can find a gre
c would show if you have something perhaps
removing this information before sending to the client.
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)
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Hello Yuriy,
> If I write at kamailio.cfg:
> alias=sip.myserver.com
>
> I see error at log - bad_uri sip.myserver.com
try adding the port...
alias=sip.myserver.com:5060
Also, since you're behind NAT make sure you also advertise the address
with advertised_address="si
I think my head isn't fully woken up yet -- sorry about that.
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)
On 07/03/2014 07:48 AM, Olle E. Johansson wrote:
> I am looking for calls setups per second - not concurrent calls.
e to combine the lookup over multiple media servers and
kamailio servers. The lookup checks the db so any modifications occur in
real-time.
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)
On 07/03/2014 07:37 AM, Olle E. Johansson wro
do you have an ngrep of the sip traffic? This can happen if the sip/rtp
cannot connect (perhaps blocked by the dsl router)
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)
On 07/01/2014 01:12 PM, Carlos Rangel wrote:
> Hello L
t; menu item and a second user called
"SystemAdmin" which only sees the "Dispatcher_List" menu item.
I tried different things with new roles, groups and adjustments in the
menu administration but didn't manage to get the desired result.
Does anyone have a hin
For the is_user_in... are you loading the group module?
For avp_write, that function hasn't existed in some time. You can use
logic such as:
$avp(s:fwd_blind) = $ru;
Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)
Good. Fast. Cheap. <- Pick two.
On 05/22/20
ed to set it within the original invite and then "do
something" with it in the event_route[dialog:failed]?
Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)
Good. Fast. Cheap. <- Pick two.
___
SIP Express Router
On 04/28/2014 06:54 PM, Alex Balashov wrote:
I don't think that will work, because no dialog is created by the 302 redirect.
11.3. event_route[dialog:failed]
Executed when dialog is not completed (+300 reply to INVITE).
--fred
___
SIP Ex
f you call set_dlg_profile() at the initial invite
and then so something with this in event_route[dialog:failed], does it
still error?
Fred Posner
The Palner Group, Inc.
f...@palner.com
@fredposner
Good. Fast. Cheap. <- pick two
___
SIP Express Router (SER)
Have you tried something like...
if (sql_xquery("ca", "SELECT * FROM gateways", "gateways") == 1) {
#do stuff
} else {
#dang nabbit
}
Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)
On 04/06/2014 02:37 PM, Alex Balashov
On 04/05/2014 09:01 PM, Alex Balashov wrote:
Does that work for SELECT queries? The documentation says it's only for
INSERT, UPDATE and DELETE.
It did not during my test this afternoon.
--fred
On 6 April 2014 02:14:24 CEST, Kelvin Chua wrote:
dunno if this helps but i use $sq
> I don't think so. As I understood the documentation, at least, $dbr
> doesn't get populated in this case; the rows just go straight to an
> xavp list. I suppose I should verify that.
>
Looks like you're right.
Tested various methods.
Fred Posner
The Palner Group
being said...
Wouldn't this still work for you:
if($dbr(gateways=>rows)>0) { }
Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)
On 04/05/2014 11:32 AM, Alex Balashov wrote:
Hi,
When using sql_xquery() like this:
sql_xquery("ca", &quo
It looks like you may be running Kamailio behind NAT as well, no?
Can you provide any traffic on the connections that fail?
Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)
On 04/03/2014 08:44 AM, Ravi wrote:
Dear Kamailio'ns,
I am awaiting somebody's s
Just to add, besides the uac having some of the best example names...
the callerid you mentioned is most likely set on your phone config;
which kamailio is just passing along.
Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)
On 03/19/2014 07:18 PM, Alex Balashov
On 2/27/14, 1:21 PM, Michelle Jun wrote:
m=audio 21064 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 101.
m=video 23134 RTP/AVP 105 99.
It does look like it's within the range. I would generally ensure that
your firewall is forwarding the ports.
--
Fred Posner | The Palner Group, Inc.
On 02/26/2014 09:11 PM, Michelle Jun wrote:
hi Fred
yes, i forwarded both TCP/UDP 5060 dan 2-3 like in your blog
but still having the issue
thanks
The rtp forwarding should be just udp. For the sip, that's up to how
you're making the connections.
Did you specify a rang
showing only "connecting"
(audio/video)
any idea what did i do wrong?
here is the /var/log./messages
thank you
If you're natted, make sure you have your firewall forwarded for the
ports you've selected for rtp and sip.
Fred Posner
The Palner Group, Inc.
503-914-0999 (
it be OK?
I've never had an upstream provider communicate with me on private nat.
> 6.Is there any better documentation that we should be using to
> make this easier, or should I just man up and try harder?
Man up. =)
Practive makes perfect.
--
Fred Posner
The Palner Group, Inc
On 2/20/14, 6:25 PM, Francesco Maria Magnini wrote:
Fred,
in you ACME replacement, kamailio doesn’t rewrite headers for handling
RTP/SIGNALING and stay in the middle?
For nat it did. For others the media server did.
You can easily force all connections to use rtpproxy to do what you ask.
We
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