Displaying 20 results from an estimated 300 matches similar to: "opensips and asterisk canreinvite"
2005 May 09
1
Asterisk + SER and NAT
Hi,
We are testing a SIP solution * + ser solution for a large implementation.
All the clients are nated.
When a client is dialing outside the domain (to a FWD sip account for
example) all is perfect ! ;-)
But ,when a call is done to a sip account, the client is ringing, then the
caller can hear the nated client very well, but the nated client does'nt
hear anything. RTP issue no ?
I've
2007 May 12
3
Asterisk High-Capacity Stability
Thanks Alex, some great ideas.
I think, however, I'm leaning towards Asterisk at this point- since I have
quite a bit of experience there, and very little with SER. At this point,
I'm wondering from a dimensioning standpoint, what kind of capacity my
machine will have (Dual Core Xeon 2.4GHz 4GB RAM). As I said, I don't plan
to do any transcoding. I read the voip-info page on
2014 Feb 01
1
larger than minimun MTU, forwarding via other node
First off, I would like to express my appreciation for the tinc software,
it has been such a great vpn solution for what i need, its amazing.
I am setting up another node on the vpn. "KVM" is my public facing node,
"MacbookAir" is my workstation, "NewNode" is my node i have recently
configured and the one with the issue presumably. NewNode and MacbookAir
are on the
2004 Jan 20
1
Toll-Free Gateway Beta Test: freenum.org
The freenum.org beta continues to roll forward. If you have an Asterisk or SER SIP gateway/proxy, please see if you can make some sense of the examples below and install them in your system. Your users will hopefully be able to dial toll free numbers in various nations just like they dial regular numbers in those same country codes.
I'd like to get some additional people trying to make
2010 Oct 27
0
Send INVITES and REFERs from OpenSIPS to Asterisk with multiple Contexts
I currently have OpenSIPS set up with users and most of my call handling.
OpenSIPS won't be able to handle things like Call Park, Hunt Groups, ACD,
etc. So I want to send these types of requests to Asterisk. I also want to
set Asterisk up as Multi Tenant. So my question is
How can I send requests to Asterisk and have them funnel into the specific
context for that specific Tenant? So if
2012 Jan 09
1
Asterisk as register server through OpenSIPS
Hi all,
I've been trying to register a SIP user agent to an Asterisk server using
OpenSIPS as SIP router. The functionality is working fine. However,
Asterisk uses the IP address of the OpenSIPS server as the peer IP address.
How can I use the original IP address of the peer without changing the
peer's nat=yes?
I appreciate any kind of help. Thanks!
Regards,
Ronald
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2014 Oct 15
0
OpenSIPS Summit Oct 21st before Astricon
Hello Everyone!
We wanted to let everyone coming to Astricon know that we will be
holding an OpenSIPS Summit on Tuesday Oct 21st, 2014 at the Suncoast
Casino & Spa.
Suncoast is about 10 minutes away from Red Rock and we will be provide
shuttle service to and from the Summit. For those of you that had to
book at Suncoast it should be really easy to find us!
Here are some things you can
2020 Jan 29
0
Invitation for OpenSIPS Summit 2020 Call for Paper
Hello fellows VOIPer,
If you want to share with the rest of the VoIP & RTC community some
news, interesting or breaking through ideas, or even more, some
experience you had in terms of designing, integrating or operating
various solutions or platform based on Open Source Softwares, then you
should consider submitting a paper for the OpenSIPS Summit 2020 in May,
Amsterdam.
2006 Jan 18
1
Problem in rpc_api_pipe related to the \spoolss pipe
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
i
I have a problem with a samba-3.0.21a (as a PDC), when I use rpcclient
to set the driver for a printer. I receive the following error (log
level 12 for relevant parts rpc*, printerdrivers,tdb attached):
rpc_api_pipe: Remote machine pdc pipe \spoolss fnum 0x76dereturned
critical error. Error was Call returned zero bytes (EOF)
prs_mem_get: reading
2009 Mar 20
1
Asterisk + OpenSIPs Integration - Rewrite URI on Trunk Numbers of a SIP Trunk
Hello All,
I have a little complicated question about the Dial command.
I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers.
Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers:
For each peer on Asterisk, "Addr->IP" is IP of the Proxy and "Reg. Contact" is the IP
2013 Mar 10
1
Register Free Opensips/Asterisk Integration
Hello Everyone,
I have gone through a few really good tutorials from the OpenSIPS
site, Asterisk resources etc.. The unanswered question (and final
piece of our puzzle) is if it's possible to have a register free
environment in an OpenSIPS/Asterisk integration. Most approaches have
OpenSIPS relay the UA's REGISTER request to Asterisk which has
"host=dynamic" set for the
2016 Jul 05
2
OpenSIPS or Kamailio based fronting for Asterisk?
Hello,
I am beginning to front my Asterisk cluster with OpenSIPS/Kamailio and so
far my biggest issue is the complete lack of quick-start-like documentation
for either. Is there any place I can get a very simple HA configuration
(telling me where the config files are, for starters, is a good thing) for
OpenSIPS or Kamailio with the following features:
(a) Support an arbitrarily large number of
2011 Mar 04
3
OT: OpenSIPS vs Kamailio -- which do you use and why?
I'm starting a new project similar to a previous project where I used
OpenSER to front a bunch of Asterisk servers.
Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely
candidates.
I'm leaning towards OpenSIPS because it's in EPEL so I can install it with
yum. Also, because I think the name sounds more 'professional' when
discussing architecture with clients :)
2008 Dec 13
3
SER, OpenSER, Kamailio, OpenSIPS -- what are you using?
One of the above is frequently used to front-end Asterisk.
I used OpenSER to front-end a farm of Asterisk servers and was very happy
with it. The ability to take a box out of service or to route a specific
DNIS to a box for testing rocks.
Since OpenSER has died (I don't care about the
politics/personalities/trademarks), Kamailio and OpenSIPS have risen from
the ashes. What are you using?
2015 Jan 21
1
PJ SIP realtime with Kamailio / opensips
Hi all,
I saw Matt Jordan's recent Kamailio world talk and was interested in the
idea he proposed of stripping out authentication and registration from
asterisk and letting Kamailio handle it.
All of the tutorials I've seen (e.g. on asipto) show Kamailio forwarding
registrations to asterisk.
In order to do what Matt suggested would I be correct in assuming I would
have to use the
2013 Apr 09
1
[OpenSIPS-Users] 404 When BYE initiated by external callee
On Tue, Apr 9, 2013 at 1:22 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>wrote:
> **
> Hi Nick,
>
> The BYE is not properly formed and rejected by script - in the 200 OK of
> the INVITE, you can see that your opensips is doing Record-Routing, but the
> BYE does not contain the corresponding Route hdr, so SIP routing is
> impossible.
>
> Regards,
>
>
2007 May 03
1
Dovecot SASL for postfix: Client host rejected: when relaying using POP mail client
Hi everyone
I have setup an ISP style mail system using postfix, dovecot, squirrelmail
and mysql on ubuntu server 7.04. Everything seems to be working fine with
sending recieving emails but relaying on submission service (port 587). I
want to allow any client to be able to relay as long as they AUTH using
dovecot SASL and are valid users regardless of what IP/host they come in
from. I have a
2009 Mar 20
3
OpenSIPS on CentOS
Hello,
I've been looking into OpenSIPS to see if it's a worthwhile addition to our setup. We're currently running a cluster, using Heartbeat, between two servers. It works well but I'm interested in seeing if we can improve it. My manager heavily uses RPM's for installations rather than source, particularly using yum to update. I'm trying to actually install OpenSips via
2009 Feb 05
0
Pattom M-ATA, T.38 and Asterisk 1.4. Canreinvite=yes ? [SOLVED]
2009/2/5 Olivier <oza-4h07 at myamail.com>
> Hi,
>
> Here http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 is a
> table listing ATA/Gateways combinations.
> Could anyone successfully set a Patton M-ATA to work with another one,
> using Asterisk 1.4 ?
>
> Is reinvite (canreinvite=yes) necessary or not ?
>
> Regards
>
>
Replying to myself, I
2007 Jun 08
0
Asterisk, NAT and canreinvite=yes
Hi,
I can not get this working:
Asterisk on public IP.
Two SIP phones behind NAT - in the same LAN.
I works perfectly (two way sound) when each peer (friend) can not
reinvite - audio stream goes through Asterisk.
The problem pops up when I define canreinvite=yes on each peer
definision so I suppose to stream audio directly between phones (in the
same local LAN).
Right after called party