in the example file would
help).
Feel free to ask any other thing you need.
Best regards.
Sergio.
On Sat, Mar 7, 2009 at 1:18 PM, James Lamanna jlama...@gmail.com wrote:
Hi,
Does anyone have some good examples of an OpenSIPs
configuration that integrates with Asterisk?
Essentially I
Hi,
I want to use OpenSIPs as the registrar (and NAT handler) for an
Asterisk/Trixbox installation.
I've got things partially working, but I've totally made a mess of my
config (I can post it if you would like).
Some things that I need:
I'm having problems with SIP-SIP calls because I need
On Wed, May 20, 2009 at 7:46 AM, Iñaki Baz Castillo i...@aliax.net wrote:
2009/5/20 James Lamanna jlama...@gmail.com:
Hi,
I want to use OpenSIPs as the registrar (and NAT handler) for an
Asterisk/Trixbox installation.
I've got things partially working, but I've totally made a mess of my
Hi,
I've been looking for a way within OpenSIPS (without needing to write
my own module),
where I can select a rewrite host/port based on the authentication
name (or from uri username) of a request.
I've looked at dispatcher and drouting, but dispatcher calculates a
hash over the username, and
Nevermind.
I figured out how to do this with avpops :)
On Sun, May 24, 2009 at 7:45 PM, James Lamanna jlama...@gmail.com wrote:
Hi,
I've been looking for a way within OpenSIPS (without needing to write
my own module),
where I can select a rewrite host/port based on the authentication
name
Hi,
Has anyone been able to get the BLF to work with a SPA-942 and the
OpenSIPS presence module?
There must be something different from the BLF responses from an
Asterisk server and from OpenSIPS,
because the BLF works great when the phone is monitoring the Asterisk
server directly.
I can see the
On Wed, May 27, 2009 at 1:52 AM, Iñaki Baz Castillo i...@aliax.net wrote:
2009/5/27 James Lamanna jlama...@gmail.com:
Hi,
Has anyone been able to get the BLF to work with a SPA-942 and the
OpenSIPS presence module?
There are many types of presence:
- Presence SIMPLE (user status: online
As an update, removing the call to handle_subscribe() stops the
infinite rebooting as well.
On Fri, Jun 5, 2009 at 6:01 PM, James Lamannajlama...@gmail.com wrote:
Hi,
I have a Linksys 942 (Phone A) where one of the line keys is setup to
do BLF (Ext B).
If I make a call from Phone A to Ext B,
On Fri, Jun 5, 2009 at 6:05 PM, James Lamannajlama...@gmail.com wrote:
As an update, removing the call to handle_subscribe() stops the
infinite rebooting as well.
Leaving in handle_subscribe and removing presence_dialoginfo also
stops the infinite reboot,
however removing presence_dialoginfo
Hi,
So, I've got presence mostly working now, however I have the
interesting problem that when someone hangs up,
the BLF on my Linksys SPA-942 still remains red. Do I need to do some
special BYE handling to make this clear?
Thanks.
-- James
___
Users
On Sat, Jun 6, 2009 at 12:21 PM, Iñaki Baz Castilloi...@aliax.net wrote:
El Sábado, 6 de Junio de 2009, James Lamanna escribió:
Hi,
So, I've got presence mostly working now, however I have the
interesting problem that when someone hangs up,
the BLF on my Linksys SPA-942 still remains red. Do
El S?bado, 6 de Junio de 2009, James Lamanna escribi?:
On Sat, Jun 6, 2009 at 12:21 PM, I?aki Baz Castilloi...@aliax.net wrote:
El S?bado, 6 de Junio de 2009, James Lamanna escribi?:
Hi,
So, I've got presence mostly working now, however I have the
interesting problem that when someone
I am using 1.5.2
--James
On Jul 28, 2009, at 1:48, Anca Vamanu a...@opensips.org wrote:
Hi James,
What OpenSIPS version are you using?
Anca
James Lamanna wrote:
Hi,
I have some SPA942 and 962 phones that I'm trying to get BLF to work
properly with.
I've found it works correctly most
a...@opensips.org wrote:
Hi James,
What OpenSIPS version are you using?
Anca
James Lamanna wrote:
Hi,
I have some SPA942 and 962 phones that I'm trying to get BLF to work
properly with.
I've found it works correctly most of the time, however on occasion,
the BLF lights will get stuck as RED
On Jul 28, 2009, at 1:48, Anca Vamanu a...@opensips.org wrote:
Hi James,
What OpenSIPS version are you using?
Anca
James Lamanna wrote:
Hi,
I have some SPA942 and 962 phones that I'm trying to get BLF to work
properly with.
I've found it works correctly most of the time, however on occasion
regards,
Anca
James Lamanna wrote:
Hi,
I've managed to get a SIP trace of what happens when the light gets stuck
on.
Apparently opensips sends the terminated state correctly, but then
for some reason immediately follows it up with a confirmed.
Any help would be greatly appreciated because
Hi,
I'm running the svn 1.5 branch of opensips.
I've noticed after some amount of time, usually a day or so, opensips
just completely stops
responding to incoming SIP requests, REGISTER, NOTIFY, etc...
The only way to recover from this is to restart opensips.
In fact it seems to stop doing
Hi,
Again, the OpenSIPS 1.5 branch I have built has stopped responding to
any and all SIP requests.
In fact, it has even stopped recording anything in the log.
This has happened after maybe 2-3 days of very light use.
Only 8 phones are using this OpenSIPS as a UA.
Unfortunately the BLF fixes that
,
Bogdan
-- James
James Lamanna wrote:
Hi,
I'm running the svn 1.5 branch of opensips.
I've noticed after some amount of time, usually a day or so, opensips
just completely stops
responding to incoming SIP requests, REGISTER, NOTIFY, etc...
The only way to recover from this is to restart
Hi,
I'm trying to get presence (BLF) working with some Linksys 942 phones.
I've noticed that I get the error,
handle_subscribe: Missing or unsupported event header field value
I did a trace and the phone is trying to subscribe to the x-spa-cti event.
Is there a way to support/fix this? Is there
.
-- James
On Tue, Mar 30, 2010 at 7:56 PM, James Lamanna jlama...@gmail.com wrote:
Hi,
I'm trying to get presence (BLF) working with some Linksys 942 phones.
I've noticed that I get the error,
handle_subscribe: Missing or unsupported event header field value
I did a trace and the phone
On Wed, Mar 31, 2010 at 10:28 AM, Anca Vamanu a...@opensips.org wrote:
James Lamanna wrote:
Anca Vamanu Wrote:
Andrew, this patch is already in 1.6.2 and trunk.
James, the first thing that you need to check is that you receive
Subscribes from the phones with event 'dialog'. And indeed
On Wed, Mar 31, 2010 at 9:16 PM, James Lamanna jlama...@gmail.com wrote:
On Wed, Mar 31, 2010 at 10:28 AM, Anca Vamanu a...@opensips.org wrote:
James Lamanna wrote:
Anca Vamanu Wrote:
Andrew, this patch is already in 1.6.2 and trunk.
James, the first thing that you need to check is that you
On Wed, Mar 31, 2010 at 9:28 PM, James Lamanna jlama...@gmail.com wrote:
On Wed, Mar 31, 2010 at 9:16 PM, James Lamanna jlama...@gmail.com wrote:
On Wed, Mar 31, 2010 at 10:28 AM, Anca Vamanu a...@opensips.org wrote:
James Lamanna wrote:
Anca Vamanu Wrote:
Andrew, this patch is already
On Thu, Apr 1, 2010 at 12:26 AM, Anca Vamanu a...@opensips.org wrote:
[snip]
Ok I think I got this somewhat working.
I was missing a dialoginfo_set() in another INVITE path.
However, does anyone know how, if you add a new phone, to make the
presence initialize to idle?
The BLF light blinks
Also, I've found a case where the BLF light stays red, even when a
call is hung up.
This seems to happen in the intercom case, where the SIP URI is
sip:u...@ip;intercom=true.
It doesn't happen on every intercom call, but once it does happen, it
is impossible to clear without clearing the
] INFO:presence:send_notify_request: NOTIFY
sip:000...@opensips.ip via sip:000...@phone.nat.ip::6095 on
Regards,
--
Anca Vamanu
www.voice-system.ro
-- James
James Lamanna wrote:
On Thu, Apr 1, 2010 at 12:26 AM, Anca Vamanu a...@opensips.org wrote:
[snip]
Ok I think I got this somewhat working.
I
On Fri, Apr 2, 2010 at 4:40 PM, James Lamanna jlama...@gmail.com wrote:
On Fri, Apr 2, 2010 at 4:52 AM, Anca Vamanu a...@opensips.org wrote:
Hi James,
I think that the problem is with those Notifies without a body sent by
OpenSIPS when the phone was started. This is normal behavior, correct
Hi,
I'm trying to use OpenSips as a registration server for Asterisk
(assuming we can get presence working ok).
Do I need to setup and use MediaProxy (or similar)? Or is the
nathelper stuff good enough?
I've made test calls from phones behind NAT to opensips to asterisk
and I haven't experienced
on one box.
Asterisk does not seem to be designed to be able to handle
registrations for that many devices,
especially when it is also under load from handling 50+ simultaneous calls.
-- James
On 3 Apr 2010, at 20:31, James Lamanna wrote:
Hi,
I'm trying to use OpenSips as a registration server
On Mon, Apr 5, 2010 at 12:52 AM, Angel Marin an...@anmar.eu.org wrote:
On 03/04/10 01:40, James Lamanna wrote:
On Fri, Apr 2, 2010 at 4:52 AM, Anca Vamanu a...@opensips.org wrote:
Hi James,
I think that the problem is with those Notifies without a body sent by
OpenSIPS when the phone
On Apr 6, 2010, at 3:24, Anca Vamanu a...@opensips.org wrote:
James Lamanna wrote:
On Mon, Apr 5, 2010 at 12:52 AM, Angel Marin an...@anmar.eu.org
wrote:
On 03/04/10 01:40, James Lamanna wrote:
On Fri, Apr 2, 2010 at 4:52 AM, Anca Vamanu a...@opensips.org
wrote:
Hi James,
I
The phones should never receive the Publish message. Please catch a
trace containing this Publish and send it to me.
What do you mean by before? Before updating from svn with my patch?
Before I updated from 1.6.2 to SVN I think - I'll try and double-check.
I have a revised patch that covers
On Wed, Apr 7, 2010 at 2:06 AM, Anca Vamanu a...@opensips.org wrote:
Hi James,
What you see happens because of a improvement that I made in
pua_dialoginfo module. Now the presentity_uri for the callee ( the uri
that will be used as RURI in Publish message) is taken from RURI of
Invite in the
Btw, this happens whenever I try to place a call.
It also crashes my phone, a Cisco 509G.
-- James
On Sun, Oct 3, 2010 at 8:05 AM, James Lamanna jlama...@gmail.com wrote:
Hi,
I just had a situation where the opensips process has totally hung and
now will not respond to SIP traffic
On Sun, Oct 3, 2010 at 8:17 AM, James Lamanna jlama...@gmail.com wrote:
Btw, this happens whenever I try to place a call.
It also crashes my phone, a Cisco 509G.
Also, removing all pua/presence function calls enables calls to be made again.
I assume something changed in 1.6.3 that has made my
On Sun, Oct 3, 2010 at 8:24 AM, James Lamanna jlama...@gmail.com wrote:
On Sun, Oct 3, 2010 at 8:17 AM, James Lamanna jlama...@gmail.com wrote:
Btw, this happens whenever I try to place a call.
It also crashes my phone, a Cisco 509G.
Also, removing all pua/presence function calls enables
-system.ro
On 10/03/2010 06:36 PM, James Lamanna wrote:
On Sun, Oct 3, 2010 at 8:24 AM, James Lamannajlama...@gmail.com wrote:
On Sun, Oct 3, 2010 at 8:17 AM, James Lamannajlama...@gmail.com wrote:
Btw, this happens whenever I try to place a call.
It also crashes my phone, a Cisco 509G.
Also
Yeah,
I've never successfully had presence for BLF work properly for me with
Linksys/Cisco SPA 9xx/5xx phones.
This is the one thing that is blocking me from moving all my phone
registrations to OpenSIPS, so I'm really hoping that we can get it all
working.
-- James
On Mon, Oct 4, 2010 at 8:49
exactly which is the behavior. Do you see the opensips process ocupying
100% cpu or what happens?
Regards,
--
Anca Vamanu
www.voice-system.ro
On 10/05/2010 04:09 AM, James Lamanna wrote:
Unfortunately, I'm still getting a hang of some sort with 1.6 trunk.
It doesn't happen quite
I removed the modules/xlog directory and rebuilt, and now things seem
to be working better.
I'll let you know if I have any more issues
Thanks for your help.
-- James
On Tue, Oct 5, 2010 at 7:12 AM, James Lamanna jlama...@gmail.com wrote:
Ok it looks like the 1.6 trunk has an error
Hi,
I was wondering if anyone had any experience getting a Cisco 7960
phone to register to opensips when the phone is behind a PIX firewall.
I'm having a hell of a time getting it to register.
I see these messages:
U nat.ip:2260 - opensips.ip:5060
REGISTER sip:opensips.ip SIP/2.0..Via:
))..Content-Length: 0
-- James
Mario
http://advantia.ca
On Mon, Dec 6, 2010 at 10:17 AM, James Lamanna jlama...@gmail.com wrote:
Hi,
I was wondering if anyone had any experience getting a Cisco 7960
phone to register to opensips when the phone is behind a PIX firewall.
I'm having
, opensips is doing the right stuff (doing symmetric
signalling) - there is nothing more you can do here for opensips..Maybe it
is something specific to the NAT device - any possibility to debug/trace on
it ?
Regards,
Bogdan
James Lamanna wrote:
Hi,
I was wondering if anyone had any
since they aren't part of an established connection get dropped.
Maybe going to opensips these phones need sip fixup on, though going
directly to Asterisk, they have been working with sip fixup off...
-- James
On Tue, Dec 7, 2010 at 10:22 AM, James Lamanna jlama...@gmail.com wrote:
Hi Bogdan
at 1:34 PM, James Lamanna jlama...@gmail.com wrote:
On Tue, Dec 7, 2010 at 9:32 AM, Duane Larson duane.lar...@gmail.com
wrote:
From your SIP message
U nat.ip:2370 - opensips.ip:5060 REGISTER sip:opensips.ip
SIP/2.0..Via: SIP/2.0/UDP nat.ip:8427;branch=z9hG4bK79682dfb..
From: sip
to a more recent firmware and seeing if that is helps
with the port numbering issue. Just a guess...
Mario
On Tue, Dec 7, 2010 at 1:14 PM, James Lamanna jlama...@gmail.com wrote:
On Tue, Dec 7, 2010 at 11:42 AM, Duane Larson duane.lar...@gmail.com
wrote:
From your original post before you set up
fix the issue.
Is adding that test bad in any way?
-- James
On Thu, Dec 9, 2010 at 9:04 AM, James Lamanna jlama...@gmail.com wrote:
Here's the SIP traffic from my phone now running v8.9 with nat_enable
= 1 and nat_received_processing = 1.
BTW this phone has no issues registering to asterisk
Hi,
I'm having some issues getting a correct NAT configuration going.
The problem I'm having is
I get a 479 We don't forward to private IP addresses back when
receiving a call to a phone from Asterisk, presumably because the
location table has private IPs in it for some reason.
This seems to be
Does anyone have a working example with fix_nated_register() that they
could post (or email me directly).
I'd really like to see how this is done properly.
Thanks.
-- James
On Mon, Dec 13, 2010 at 5:54 PM, James Lamanna jlama...@gmail.com wrote:
Some other weird stuff that happens if I remove
Hi,
Is it possible to share the same DB tables among several running
OpenSIPs instances?
What I'm trying to do is use OpenSIPs as a registration front-end to Asterisk.
The idea is to have a cluster of registration servers, and then a
cluster of Asterisk servers.
Can an Asterisk server pass a call
Bogdan,
Wow, I didn't know about the live DVD.
Any chance someone could create this as an OpenVZ container in
addition to VMWare?
-- James
On Mon, Jan 10, 2011 at 2:25 AM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Hi Damon,
Well, the answer is simple - download the opensips virtual
Hi,
I'm trying to move to OpenSIPS as being the registrar front end for
cluster of Asterisk boxes.
One of the services we currently offer is DND w/ BLF.
I'm trying to figure out how to implement this with OpenSIPS.
From what I've read, DND can be implemented through ACLs, correct?
However, is
Hi Anca,
On Fri, Jan 14, 2011 at 6:19 AM, Anca Vamanu a...@opensips.org wrote:
Hi James,
On 01/13/2011 06:43 PM, James Lamanna wrote:
Hi,
I'm trying to move to OpenSIPS as being the registrar front end for
cluster of Asterisk boxes.
One of the services we currently offer is DND w/ BLF
Hi Jeff Bogdan,
I'm looking into a setup very similar to this as well, essentially I
want to have a cluster of OpenSIPS servers for registration and then a
cluster of Asterisk Boxes for all the dialplan handling.
I have the unfortunate problem that all of my clients are going to be
behind NAT.
My
appreciated. Presence/BLF for my phones works great when
everything is on the same opensips instance, but my goal here is not
to have single instances managing a specific group of phones.
Thanks.
-- James
On Sun, Jan 30, 2011 at 10:11 AM, James Lamanna jlama...@gmail.com wrote:
Hi,
I'm trying
.
Regards,
Henk
On 29-01-11 22:13, James Lamanna wrote:
Hi Jeff Bogdan,
I'm looking into a setup very similar to this as well, essentially I
want to have a cluster of OpenSIPS servers for registration and then a
cluster of Asterisk Boxes for all the dialplan handling.
I have
Hi,
I'm trying to redirect a REGISTER request to a different proxy, mostly
for load balancing purposes.
The UAC is behind NAT, so in order to properly communicate directly
with the next proxy, the UAC must send a new REGISTER request to the
new proxy.
I've tried sending back a 302 Moved
. You can do this by playing with the contact when forwarding the REGISTER
from P1 to P2.
Regards,
Bogdan
James Lamanna wrote:
Hi,
I'm trying to redirect a REGISTER request to a different proxy, mostly
for load balancing purposes.
The UAC is behind NAT, so in order to properly communicate
Hi,
I've been investigating a problem where I was noticing that nathelper
was not pinging all the contacts it should be pinging.
I've narrowed down the problem to this code in nh_timer() in nathelper.c:
rval = ul.get_all_ucontacts(buf, cblen, (ping_nated_only?ul.nat_flag:0),
((unsigned
Hi,
I've noticed after a period of time, Nathelper will stop sending pings to
some contacts.
I've verified that the contact is still registered (it is even in the
location table) but the ping process appears to skip some contacts for
unknown reasons.
Could someone please look into this? I have
the fix_nated_registered() function
Regards,
Bogdan
On 07/01/2011 05:58 PM, James Lamanna wrote:
Hi Bogdan,
Unfortunately I've found that it doesn't fix the entire problem.
I have a contact now that is online, that still isn't getting pinged for
some reason.
I think there's something subtle
the '??z'
shouldn't be printed...
-- James
On Fri, Jul 8, 2011 at 6:12 PM, James Lamanna jlama...@gmail.com wrote:
Hi Bogdan,
I've been monitoring it (I wrote myself a script that checks the AORs
in memory against syslog).
I haven't seen any issues lately.
The trailing garbage is weird as well
Does anyone have experience getting BLF working properly with a Snom 821?
I'm using 1.6.4 and BLF works great with Linksys/Cisco phones, but
with Snom 821s I have
the problem where the light will stay on, even after the monitored
extension has hung up.
BLF to the Snom works properly when getting
/modules/1.6.x/presence.html#id248928
Regards,
Anca Vamanu
On Mon, Jul 25, 2011 at 12:52 AM, James Lamanna jlama...@gmail.com wrote:
Hi,
I'm trying to get BLF working on a Polycom 550 with Opensips 1.6.4 and
I find that the BLF light never shuts off
when the monitored user hangs up.
If you
Hi everyone,
I've noticed that I'm having issues with successive subscribes from
Cisco phones with firmware 7.4.8.
Here's a SIP trace of a 2nd subscribe (when the first one is about to expire)
SUBSCRIBE sip:regdev:5060 SIP/2.0..Via: SIP/2.0/UDP
192.168.2.3:7813;branch=z9hG4bK-9160ffcb..From:
Hi,
I'd like to know if it is possible to have multiple presence servers.
The idea is to distribute SUBSCRIBE messages around so that each
presence server is aware of all subscriptions.
However, it seems as though there is a problem with the to_tag when
doing this since it is used as a matching
Hi Anca, Bogdan, and list,
I've been banging my head against this for some time now, so I'm
wondering what I'm trying to do is possible.
My goal is to create an Opensips cluster that provides the following:
1) Registration for phones behind NAT
2) BLF/Presence information (through Event: dialog)
.
the presence state must stay on a single server, so all PUBLISHs for entity
A and all SUBSCRIBES to entity A must go to the same presence server.
Regards,
Bogdan
On 08/05/2011 11:31 AM, James Lamanna wrote:
Hi,
I'd like to know if it is possible to have multiple presence servers.
The idea
Hi,
Is there a good way to work around a broken NAT?
I have a customer who is receiving one-way inbound audio (outbound is fine).
Here's what a REGISTER packet looks like:
U public.ip:9241 - opensips.ip:5060
REGISTER sip:registrar SIP/2.0..Via: SIP/2.0/UDP
Any ideas with this one?
BTW the phones work fine when registered directly to Asterisk, which
tells me there must be a way to make this work.
-- James
On Wed, Aug 24, 2011 at 7:33 AM, James Lamanna jlama...@gmail.com wrote:
Hi,
Is there a good way to work around a broken NAT?
I have
Hi,
I'm experiencing a crash after about 5 minutes on a system (same
config works fine on other systems).
This is a system running under OpenVZ.
Here's the tail end of syslog:
Nov 27 18:04:18 opensips2 /usr/local/sbin/opensips[7280]:
DBG:tm:matching_3261: RFC3261 transaction matched,
, James Lamanna wrote:
On Mon, Nov 28, 2011 at 12:00 AM, Bogdan-Andrei Iancu
bog...@opensips.org wrote:
Hi James,
The relevant part is:
Nov 27 18:04:22 opensips2
/usr/local/sbin/opensips[7275]:INFO:core:handle_sigs: child process 7281
exited by a signal 11
Nov 27 18:04:22 opensips2
/usr
, len = 0}
(gdb) print p-from_tag
$7 = {s = 0x0, len = 0}
-- James
On Tue, Nov 29, 2011 at 2:16 PM, James Lamanna jlama...@gmail.com wrote:
Here's probably a more telling BT (Debug=3)
#0 0x2aba8e563f33 in get_dialog (dialog=0x7fff59e5b5d0,
hash_code=value optimized out) at hash.c:480
480
Hi Bogdan,
Any ideas here?
Thanks.
-- James
On Tue, Nov 29, 2011 at 3:57 PM, James Lamanna jlama...@gmail.com wrote:
Realized I got variables from the wrong frame.
(gdb) print p
$1 = (ua_pres_t *) 0x2aba902bfb00
(gdb) print *p
$2 = {hash_index = 305, local_index = 0, id = {s
Hi,
I have 2 opensips servers that I'm trying to use a shared database with.
The reason for this is to simplify BLF handling and allow phones to
register to either server, increasing redundancy.
However, all of my UAs are behind NAT which presents a problem in the
following case:
Phone P
Hi,
I've noticed lately that a server of mine is getting repeatedly hit by
an attacker trying to make international calls.
The scary part is that the attacker seems to be able to register
correctly on different extensions, even though each extension has a
different, random password.
I'm not sure
to make VoIP forensic on it .
thanks
Aws
Msc VoIP security
2012/2/1 James Lamanna jlama...@gmail.com
Hi,
I've noticed lately that a server of mine is getting repeatedly hit by
an attacker trying to make international calls.
The scary part is that the attacker seems to be able to register
are missing
something in there and it is allowing someone to register even thought the
password is wrong.
Definitely an issue with your script. Somewhere in there you are rejecting
credentials but carrying on anyway...
On , James Lamanna jlama...@gmail.com wrote:
Hi,
I know
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