Thanks Bogdan for your reply. Yes Destination is supporting TCP; iam using
PJSIP. Registration is successful.
On Thu, Jul 22, 2010 at 8:19 PM, Bogdan-Andrei Iancu wrote:
> Hi Premalatha,
>
> That means opensips was not able to open a TCP connection to the
> destiation address. I see this is an I
Hi Bogdan,
Thanks for your reply. Is the logic correct ?
After IVR, i want asterisk to transfer call to opensips and opensips to
handle the call further. But, i still wonder how opensips would get the call
details of orignator, do i need to manipulate nything.
Also, after googling found Opensips
Marcio,
This looks like a bug in Cisco ATA186. Cisco A received 200 OK (packet
no. 32):
Record-Route:
Contact:
And sent ACK sip:10...@187.13.212.160 SIP/2.0
Route:
But it must put remote target URI learned from Contact to Request-URI:
ACK sip:10...@187.13.212.160:5081
According to RFC 3261
Hello
I need count for number of active calls on kamailio server. I have done
following configuration in my kamailio.cfg file
...
loadmodule "dialog.so"
modparam("dialog","profiles_no_value","domain")
modparam("dialog", "dlg_flag", 4)
route[0] {
...
if(is_method("I
Hi all,
I am two days working over a mistery on an ACK from a 200 that was missing.
I was using 2 cisco ATA 186 both behind NAT, and both on other ports than
5060 (one on 5080 and other on 5081).
When calling eachother the ACK was sent from cisco A to OpenSIPS (1.6.2)
which processed the
Sorry I forgot to mention I upgraded to OpenSIPS 1.6.2 with Mediaproxy 2.4.3.
Hope someone can give me a hint thank you
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I upgraded OpenSIPS 1.3 with Mediaprxoy 1 to OpenSIPS 1.6 with Mediaproxy 2.
ATA->OpenSIPS is working well. ATA->Tunnel Server->OpenSIPS get conntrack
timeout via engage_media_proxy and use_media_proxy even call is connected
with 2 way audio. I spend two days looking into this and still can't figu
Hello Bogdan,
thanks for this info.
Concerning Question 2 I try to find an example with opensipsctl to
disable a destination address (destination address are in the database):
opensipsctl fifo ds_set_state i 1sip:127.0.0.1:5080 # set inactive
Or am I doing wrong something here? Does this also
the 200 OK replies are captured by siptrace via a TM callback - you are
not doing the tracing from script, so the transmissions will be absorbed
by TM.
Regards,
Bogdan
Julien Chavanton wrote:
> The "200 OK" retransmistion are forwarded but sip_trace() is not
> inserting them in the DB.
>
>
Hi,
again, you cannot control the behaviour (in regards to timeout) of other
proxies - it is a matter of local policy.if you think I'm missing
the point, send a callflow wiht explanations.
Regards,
Bogdan
mehdi boudou wrote:
> Hello Bogdan,
>
> I don't formulated well my question, how can
Hi Bogdan,
What i understood in what you said is that i have to call back a route
before proxying in CPL adding modparam("cpl-c","proxy_route",1),in this
route I add myself in record-route and because i'm in record route CPL will
send the Invite to myself and i can trigg B2BUA behaviour to modify
Hi Mehdi,
A proxy must not change the Cseq for a call. In parallel ringing
(multi-ringing), you have multiple branches of the same calls and all of
them have the same CSEQ - id the S-cscf does not like it, you should fix
it as it is standard SIP.
Regards,
Bogdan
mehdi boudou wrote:
> Hello Bo
Hi Peter,
Peter P GMX wrote:
> Hello,
>
> we want to use the dispatcher module for a very simple load balancing
> with hash over callid or round robin (we have also tried the
> load-balancer module but have problems when we want to use it for
> generating 302 redirects).
>
> I have read that the d
Excellent, thanks.
On Thu, Jul 22, 2010 at 10:51 AM, Bogdan-Andrei Iancu <
bog...@voice-system.ro> wrote:
> Hi Bobby,
>
> compile out the EXTRA_DEBUG option - this option includes some too
> aggressive checks used for debugging purposes only.
>
> Regards,
> Bogdan
>
> Bobby Smith wrote:
> > Same
Hi Bobby,
compile out the EXTRA_DEBUG option - this option includes some too
aggressive checks used for debugging purposes only.
Regards,
Bogdan
Bobby Smith wrote:
> Same thing just happened about 2 hours ago. Can't figure out exactly
> what scenario caused it, but I do have another core. I
Hi Premalatha,
That means opensips was not able to open a TCP connection to the
destiation address. I see this is an INVITE, so the question is - is the
destination party supporting TCP ?
Regards.
Bogdan
Premalatha Kuppan wrote:
> Hi,
>
> Iam using opensips with TLS enabled. Iam trying to make
Hi Premalatha,
have you checked the opensips logs for error? can you post a call flow
showing exactly which step fails ?
Regards,
Bogdan
Premalatha Kuppan wrote:
> Hi,
>
> I posted before my query; but no repsonse :(
>
> Can some1 helps, whether this logic works,
>
> Iam using opensips 1.6.2(T
Hi Mehdi,
not reallyCPL module is desigened to work in proxy mode, not in
b2bua mode.
What you can do is to use b2bua, spiral to call back to proxy and run
cpl script there.
Regards,
Bogdan
mehdi boudou wrote:
> Hello,
>
>
> to be more accurate with my older post: is it possible to combin
Hi Saúl,
On Jul 21, 2010, at 5:14 PM, Saúl Ibarra Corretgé wrote:
> Hi Jeff,
>
> On 21/07/10 21:50, Jeff Pyle wrote:
>> Hello,
>>
>> I am using Mediaproxy 2.3.8 with Opensips 1.6 r6702. I use the
>> engage_media_proxy() function. Most things work pretty well except for the
>> following scen
Hi,
I posted before my query; but no repsonse :(
Can some1 helps, whether this logic works,
Iam using opensips 1.6.2(TLS) and Asterisk(1.4.3.1).
1. all users registering @ opensips
2. All calls to opensips forwarded to sasterisk for IVR; thorugh IVR input
destination is known.
3. Since asterisk
Hello,
to be more accurate with my older post: is it possible to combine B2BUA
logic with Cpl module ?
Thanks,
Regards.
Mehdi BOUDOU
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