Hi, Brett!
Are you using uac_replace_from function with dialog support or the old
approach with route parameters? Would it be possible to increase
OpenSIPS debugging level and paste some extra information?
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
On 12/15/2011 12:41 AM, Brett
Hi, Can someone explain/help me on this
I really will appreciate
Best Regards
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Hello
Look at your TO header in INVITE message
Opensips changes TO header in INVITE.
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of spady
Sent: Tuesday, December 13, 2011 8:13 PM
To: users@lists.opensips.org
Subject:
Hi,
I'm trying to achieve the following scenario:
SIP Phone - OpenSIPS - SIP Provider
From my SIP Phone i want to dial a number, let opensips do the challenge
authentication to the sip provider and get the call connected.
What i currently have in opensips is an extension '1234' which
Hi Denis, I know. it's wanted.
Is changed to 87019.
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You can use the b2b modules to perform authentication.
You will need to load uac_auth first and provision the credentials in
the credentials modparam:
http://www.opensips.org/html/docs/modules/1.7.x/uac_auth.html#id249026
You need to match the realm from 401 with the realm from modparam.
Hi,
I have noticed that I have a number of calls stuck in the dialog list in
state 5 (dialog ended)?
Apart from restarting the system I cannot see a way of clearing these.
I assume they should clear themselves and most do but for some reason
these are not.
I use database persistent
As I understand you use uac_replace_to() function to change TO header, and as I
can see from debug you have restore_mode parameter
(http://www.opensips.org/html/docs/modules/1.6.x/uac.html#id249076) in UAC
module which was established to value 3 (or not established at all). So
Opensips
Hi Denis, that was it!!! It was setted to auto .
I set it to none and now it works as aspected Perfet.
Thank you very much for your hint ;-)
Best regards
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Can someone help me with this?
I checked again config and seems ok but form CP nothing yet.
Regards
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Brett,
Is the other end an Acme? If so, they need to implement some custom
parameters (which I do not have) to* *honor some parts of section 12 of
RFC3261 in such a way that won't break uac_replace_from(). Let me know if
this is the case and we'll talk more.
Rasvan,
Can you share more about
Hi all,
I'm looking for a SIP/SIMPLE to XMPP gateway solution and my google searches
have brought me here. I have an account with a voip provider that supports
SMS via SIP/SIMPLE MESSAGE (http://tools.ietf.org/html/rfc3428).
Unfortunately, I'm stuck using an old Blackberry which doesn't have any
Out of curiosity, based on the feedback in this bug; is this something that's being fixed? I notice this bug was for 1.6.4, but my experience is in 1.7.1. so I want to make sure if this was fixed, I report a new bug for 1.7.1Hi Bogdan,
This bug fix requires further work in tm module, in
It was not fixed. It is the same bug.
Regards,
Ovidiu Sas
On Thu, Dec 15, 2011 at 3:20 PM, Logan voipmas...@me.com wrote:
Out of curiosity, based on the feedback in this bug; is this something
that's being fixed? I notice this bug was for 1.6.4, but my experience is in
1.7.1. so I want to
ore.
Rasvan,
Can you share more about the "new" way to do it with the dialog module? Is
this available in 1.6?
- Jeff
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Hello,
Jeff, I was wrong, this feature is not available in 1.6, only in 1.7 .
Anyway, according to Brett's traces it seems to be a problem with the
function and I would really appreciate if he could help me debugging this.
Regards,
Ra(zvan
On 12/15/2011 07:27 PM, Jeff Pyle wrote:
Brett,
Is
Why the username field?
Cheers,
Nick.
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Razvan,
I'm using the new dialog based method. If I remember right the old way was
with vsf/vst headers and I'm definitely not using that anymore.
The version I'm using is a svn pull from 1.6.3 (7919), which I think is how
I'm doing the dialog based version. I'm not quite ready to move to 1.7 but
Hello,
I have to delete codec for similar reasons. I'm trying to use
codec_delete_except_re().
Here's the incoming SDP:
m=audio 49152 RTP/AVP 123 122 125 121 124 9 0 8 113 100
a=sendrecv
a=ptime:20
a=rtpmap:123 G7221/32000
a=fmtp:123 bitrate=48000
a=rtpmap:122 G7221/32000
a=fmtp:122
I'm using opensips on a computer with 2 ip addresses one steady one
and one is a floating ip address provided by heartbeat, when heartbeat
is on and I have 2 ip addresses opensips takes a very long time to
start and I get a error in the messages file, the error is
opensips:
Opensips is trying to do a reverse dns on the IP address. You can try
adding a reverse dns record for that ip or try disabling the dns
and/or auto_aliases options.
On Thu, Dec 15, 2011 at 2:52 PM, Schneur Rosenberg
rosenberg11...@gmail.com wrote:
I'm using opensips on a computer with 2 ip
thanks your advice solved the problem
On Fri, Dec 16, 2011 at 1:12 AM, Ryan Bullock rrb3...@gmail.com wrote:
Opensips is trying to do a reverse dns on the IP address. You can try
adding a reverse dns record for that ip or try disabling the dns
and/or auto_aliases options.
On Thu, Dec 15,
Hi, Brett!
I was talking about the DEBUG info (debug core parameter set to 6). If
you can provide the logs for such a corruption on pastebin.com or
something it would be great. Otherwise I will try to reproduce myself
this scenario and see what's happening.
I'm not sure what version are you
Hello Everyone,
We would like to use the DROUTING module to:
* Limit users to specific dialplans
* Perform LCR
Is there any step by step tutorial on how to do this?
Thanks in Advance,
Nick.
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That is the only way you could add a user to a group.
On Dec 15, 2011 3:39 PM, Nick Khamis sym...@gmail.com wrote:
Why the username field?
Cheers,
Nick.
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Thank you so much for your time Duane, and grp in that context makes
perfect sense however,
the username is really throwing me off. What I have now is:
++--+--+-+-+
| id | username | domain | grp | last_modified
Here is an example from my test database
Proxy02:~# opensipsctl db show grp | grep 9012732009
3 9012732009 irock.com int 2010-10-25 13:41:57
285 9012732009 irock.com ld 2011-03-24 19:35:01
284 9012732009 irock.com local 2011-03-24 19:34:49
Is this valid naming for avp's - $avp($var(i))
I am trying to loop through exec_avp results as I will never know how many
rows are being returned. It is in a startup route and I am using a while
statement to loop through. It is not recognizing naming. $avp($avp(i))
always returns NULL while
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