I've got same trouble. And fix it like you.
On Tue, 08 Jun 2010 09:34:13 +0600, Matthew Lehner wrote:
> I am setting up opensips to act as a proxy between a SIP trunk
> provider and more than one asterisk server. I am using alias_db to
> determine which asterisk server a particular DID/user shou
yes,you are missing module,you can check what is module of this
fuction in www.opensips.org->document->cookbook->fuction.
2010/6/7 Premalatha Kuppan :
> Hi,
>
> Can someone tell me, how to recover from this error,
>
> Jun 7 06:17:52 [10803] DBG:core:find_cmd_export_t: found (2)
> in module sl [//
Did you add "Domain Module" in loadmodule section?
On Mon, 07 Jun 2010 16:39:52 +0600, Premalatha Kuppan
wrote:
> Hi,
>
> Can someone tell me, how to recover from this error,
>
> Jun 7 06:17:52 [10803] DBG:core:find_cmd_export_t: found (2)
> in module sl [//lib64/opensips/modules/]
> Jun 7 06
Hi All,
I've picked up an issue with Asterisk not adding a Via header when calls
are passed to it from OpenSIPS. Now this doesn't seem to affect the
following call flow:
UA --> OpenSIPS --> Asterisk > Callee
However, when the below call flow happens, the callee side answers the
Thanks a lot.
I have one question. If i route the call to asterisk for IVR ( in my case
ivr is to authenticate the user to access the system), who will have the
control, meaning who will maintian all the transactions and dailog. Is it
opensips/Asterisk ?
Thanks,
Prem
On Tue, Jun 8, 2010 at 9:50
Ok thanks i will try with sequantial and multiple forking have you some
simple examples with sequantial and multi ringing operation in the
configuration file?
i read the fonction serialize_branches(clear) but it's not easy to use in my
opinion
Thanks,
Mehdi BOUDOU
Hi,
I basically use the load_balancer module to dispatch to different
asterisk servers
on the main route block after handling auth, registrations, etc
if(db_is_user_in("Request-URI", "ccivr")) {
xlog("The call will be redirect to calling card server");
route(3);
}
# route fo
Hi BR !
Thanks for your updates.
Can you please point me to appropriate file to be modified and verified.
Thanks,
Rvrao
On Tue, Jun 8, 2010 at 7:43 AM, Max Mühlbronner wrote:
> Hi,
>
> I would try to use something like this, although i have not tested it
> yet and have not found any similar i
Hi,
I would try to use something like this, although i have not tested it
yet and have not found any similar info. But i think force_send_socket
could take an ipv6 socket as an argument? So if it is ipv6 you would
send out via the correct socket? Maybe i am wrong, i can´t find any
documentatio
After upgrading Opensips from 1.5.3 to 1.6.2 we get the following errors:
ERROR:load_balancer:do_load_balance: failed to create dialog
Googling this error showed no results. Any hints where this comes from?
here's the (anonymized) config file:
debug=1
memlog=1
fork=yes
children=2
log_stderror
Hi All !
I was running Opensipserver version 1.4.2 past 2 years without any problem.
The topology is
UE1(v6)
>(v6)SIPProxy(v4)->(v4)OpenSer(v4)--->(v4)SIPProxy(v6)->(v6)UE2
The problem started with UE orginating from IPV6 address and trying to
reg
I forgot the link
i did this work some time back
Lost the link, google it
Opensips+asterisk+a2b
Ram
On Mon, Jun 7, 2010 at 4:07 PM, Premalatha Kuppan
wrote:
> Can you pleae guide me how to do this ?
>
>
> On Mon, Jun 7, 2010 at 4:04 PM, Douglas Lane wrote:
>
>> Hi Premalatha,
>>
>> Perhaps h
i'm a new one in opensips.This is my project:
-PSTN
192.168.8.140123.30.102.16 123.30.101.3
192.168.147.2 192.168.128.26
Now,when I calling from UserA to PSTN,SDP field in Invite mess
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