hi all,
Which is the kernel version of the amazon instance? There was some minnor
net issues with incorrect checksum that affected 2.6.18. You can disable the
checksum with ethtool
ethtool -K [intercace] tx off
Try above command and check whether the debug message disappear.
Hope it helps,
Samue
Hi,
Rls-services document is working now!
Thanks a lot. The problem was the URI where the document was stored. It
must contain the same userID and domain as in the To: header in the Sip
subscribe message.
Regards,
Andrés.
El 25/01/11 15:45, Daniel-Constantin Mierla escribió:
Hello,
hav
Hi,
Yes, the mysql debug was very useful. It helps me to find that the
username and domain in the xcap url where the document is stored, must
be the same as the to header in the subscribe, and also in the service uri.
Thanks a lot!
Regards,
Andrés.
El 25/01/11 20:57, Klaus Darilion escri
Hello my friends,
I have a requeriment, which indicates that i have to record every SIP
conversation between peers (also for callings to the PSTN); the
recording server will be built for our company following this
requeriments (also requested for the client):
My doubt is: How can i handle sip con
Danny Dias wrote:
> Hello my friends,
>
> I have a requeriment, which indicates that i have to record every SIP
> conversation between peers (also for callings to the PSTN); the
> recording server will be built for our company following this
> requeriments (also requested for the client):
>
> My do
Whoops! some SIP traffic IS peer-to-peer.
Jeremya wrote:
> Danny Dias wrote:
>
>> Hello my friends,
>>
>> I have a requeriment, which indicates that i have to record every SIP
>> conversation between peers (also for callings to the PSTN); the
>> recording server will be built for our company fo
Media NEVER goes through a Proxy core...the question is, how should i
record conversations when the calls are all passing through a sip
proxy? some lights will be enough for me :)
2011/1/26 Jeremya :
> Whoops! some SIP traffic IS peer-to-peer.
>
> Jeremya wrote:
>
> Danny Dias wrote:
>
>
> Hello m
Actually - being pedantic - some proxy cores have a co-located media
server. e.g. rtpproxy.
The problem is getting ALL SIP traffic to run through the same SIP proxy
and/or proxies. It usually happens but needs careful attention.
As regards simple accounting, the Kamailio/SER system has full call
On 01/26/2011 03:51 PM, Danny Dias wrote:
Media NEVER goes through a Proxy core...the question is, how should i
record conversations when the calls are all passing through a sip
proxy? some lights will be enough for me :)
Hello,
Use Record-Route headers to force in-dialog requests to have
Danny Dias escribió:
Hello my friends,
I have a requeriment, which indicates that i have to record every SIP
conversation between peers (also for callings to the PSTN); the
recording server will be built for our company following this
requeriments (also requested for the client):
My doubt is:
Someone correct me if I'm wrong, but I've seen enough examples of
out-of-dialog requests (e.g. BYE) not using the record route to wonder
if this is in fact required for a new dialog.
I've managed this by setting outbound proxy, but a general rule would help.
marius zbihlei wrote:
> On 01/26/2011
Thanks Jeremya, but it's a requeriment from the client to record the calls
through an external server and not with rtpproxys, my question is how the
media should be handled in order to record the conversations if the server
is external?
Signaling: Phone_A <---> Proxy <---> Phone_B
Media: Phone_A
On 01/26/2011 04:07 PM, Jeremya wrote:
Someone correct me if I'm wrong, but I've seen enough examples of
out-of-dialog requests (e.g. BYE) not using the record route to wonder
if this is in fact required for a new dialog.
Hello
You seem to misunderstand some notions. First of all, RR will
Danny Dias escribió:
Thanks Jeremya, but it's a requeriment from the client to record the calls
through an external server and not with rtpproxys, my question is how the
media should be handled in order to record the conversations if the server
is external?
Signaling: Phone_A <---> Proxy <--->
On Wednesday 19 January 2011, Henning Westerholt wrote:
> in about two and a half week the annual open source developer conference
> FOSDEM (http://fosdem.org/2011/) takes place in Brussels. As in the last
> two years we would like to meet here for a social event, probably a dinner
> on Saturday ev
the SIP proxy server does not see media indeed, you better look at a media
server
such as SEMS. (iptel.org/sems)
jiri
On 1/26/11 2:41 PM, Danny Dias wrote:
Hello my friends,
I have a requeriment, which indicates that i have to record every SIP
conversation between peers (also for callings to
Many thanks Jaremya,
The main problem is that both terminals, SHALL (required and must not be
changed, because of standards of EUROCAE ED-137 Part3) initiate a session
with the recorder server (a commercial one, can't use Asterisk for my
disgrace) sending INVITE and receiving the subsequent respon
OOps, made a mistake on tipping.take a look down please...
2011/1/26 Danny Dias
> Many thanks Jaremya,
>
> The main problem is that both terminals, SHALL (required and must not be
> changed, because of standards of EUROCAE ED-137 Part3) initiate a session
> with the recorder server (a commer
As said, SER is not the best vehicle for that as it in its SIP proxy definition
does not process media. SEMS does this (recording, mixing, storing) quite
decently. I'm just wondering what is the reason that Asterisk can't be used --
perhaps SEMS would fail that criteria as well.
-jiri
On 1/2
Hi All,
is there any limitation on the size of kamailo.cfg file. When i tried to add
few avp_subst, then the kamailio starts showing up the error out of package
memory on receiving SIP message. If i remove those lines then everything
works fine.. avp_subst is simple one. Our kamailio cfg file is
I forgot to mention the kamailio information
kamailio -V
version: kamailio 3.1.0 (i386/linux) 21a375
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, US
Danny Dias escribió:
Many thanks Jaremya,
The main problem is that both terminals, SHALL (required and must not be
changed, because of standards of EUROCAE ED-137 Part3) initiate a session
with the recorder server (a commercial one, can't use Asterisk for my
disgrace) sending INVITE and receiv
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