Check
https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/
https://blog.opensips.org/2016/12/20/mid-registrar-scalable-registration-and-call-forking/
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
https://www.siphub.com
On 5/2
Still looking for possibly a template/example code on this.
I am setting a bounty of $150 for anyone willing to help.
You can reach out to me via E-Mail or phone at 407-999-
Thanks!
On Mon, Mar 13, 2023 at 8:26 PM Dylan Cruz wrote:
> I'd love a sample OpenSIPS Config that would let me acc
I'd love a sample OpenSIPS Config that would let me accomplish using it as
a transparent proxy to Asterisk running on the same system. I found a few
tutorials but found a lot of conflicting information and outdated sources,
Once I have that I will have enough to work on to do what I want...
Basical
Here is a list of changes I found:
1) Must build asterisk with ODBC storage enabled for voicemail because
using file storage will not store messages in the database.
2) Uncomment the lines *'odbcstorage=asterisk'* and
*'odbctable=voicemessages'* in voicemail.conf to enable database storage
for m
Nice !
As a way of helping us (project) back, could synthesize a list with
things that did changed since the tutorial was written ? And I will
re-generate the tutorial, so other people will benefit from it.
Thanks and regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.ope
I also found the correct way to deal with the LIMIT problem. Asterisk has a
built-in way to deal with this. In file* /etc/asterisk/res_odbc.conf*, the
following should be added under [asterisk] :
limit => 5
share_connections => no
Now everything is working well without problems.
Nabeel
Perfect ! is there any left to be solved, or everything works fine ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 14.07.2016 13:33, Nabeel wrote:
Hi Bogdan,
I have been able to solve that problem.
The issue was that I had asterisk compiled
Hi Bogdan,
I have been able to solve that problem.
The issue was that I had asterisk compiled with file storage enabled
instead of ODBC storage. I recompiled asterisk with ODBC storage enabled
and now database storage is working.
Thanks.
Nabeel
On 14 Jul 2016 11:15 a.m., "Bogdan-Andrei Iancu"
w
Hi Nabeel,
1) that limit should not be necessary, as you should have in DB a single
record for each subscriber. If multiple records are returned, it means
your data is not correct.
2) in those lines, the "asterisk" and "asteriskcfg" are the names of the
odbc connection - I pasted an example
Hi Nabeel,
That means the vmusers and vmaliases do work ok, still the VM storage
engine does not. Do you have in voicemail.conf the following:
odbcstorage=asteriskrt
odbctable=voicemessages
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
I have been able to solve the issue of loading numbers without using the
voicemail.conf file.
After adding the line *'voicemail => odbc,asterisk,vmaliases'* to
extconfig.cfg, I removed the suffix " |u " from extensions.conf:
exten => _VMR_.,n,Voicemail(${EXTEN:4}*|u*)
Now all phone numbers in
0. The line block was in the default OpenSIPS config, but I agree that it
is not in the tutorial so should be removed (for voicemail).
1. I think there is a misunderstanding here. 'limit' is not a column; I am
referring to the mysql LIMIT clause:
https://dev.mysql.com/doc/refman/5.5/en/select.h
Thank you Nabeel,
The number you added in voicemail file - does it exist in the
sipuser/subscriber table ??
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 04.07.2016 11:38, Nabeel wrote:
Hi Bogdan,
I just added the column to the view by ad
Hi,
0. The cfg block you mentioned as removed does not exists in the cfg as
per tutorial.
1. the "limit" column does not exist in the sipusers as per tutorial, so
it might have been added in newer asterisk versions; not sure what is
its meaning, but if setting it to 1 makes asterisk happy, i
Hi Bogdan,
I just added the column to the view by adding "NULL AS `callbackextension`"
to the SQL view definition. I haven't linked the column to the subscriber
column, so this may not be the correct definition. However, it got rid of
the error.
About the voicemail.conf file, when I attempted to
Hi,
the voicemail.conf file exists in almost all asterisk versions. But if
you use the odbc storage for voicemail, you do not need this file at all.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02.07.2016 15:41, Nabeel wrote:
In the lates
Hi,
What is the definition you used for this new column ?
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02.07.2016 05:29, Nabeel wrote:
In the last error message,/'//callbackextension = ?' /suggested that
this column is missing from the
Hi,
This kind of ordering is valid in older versions of Asterisk. Maybe not
anymore in the newer versions.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02.07.2016 04:23, Nabeel wrote:
The tutorial contains a mistake where the priority order
Hi Samy,
Point 1 I cant imagine how those lines possibly relate to no media error in
> asterisk, I guess it depends on your config setup.
In point 1 I was referring to this error:
WARNING[17112] res_odbc.c: SetConnectAttr (Txn isolation) returned an
> error: HY000: [MySQL][ODBC 5.2(w) Driver]Yo
Hi Nabeel,
Point 1 I cant imagine how those lines possibly relate to no media error in
asterisk, I guess it depends on your config setup.
The logical answer to your point 2 would be Asterisk realtime. However this
is not going to be as staraight forward as making asterisk use subscriber
table for
The last error message has been solved by removing the following lines from
opensips.cfg:
if (!db_does_uri_exist()) {
>send_reply("420","Bad Extension");
>exit;
>}
>
>t_newtran();
>t_reply("480",
In the latest version of Asterisk, there is a new file voicemail.conf which
must be configured correctly for voicemail, but the tutorial does not
mention this file at all. Please let me know how to configure this file for
integration with OpenSIPS.
Nabeel
__
In the last error message,* '**callbackextension = ?' *suggested that this
column is missing from the sipusers mysql view. So I added this column to
the view and now that error has been resolved. Only the following error
remains now:
[Jul 2 03:25:48] WARNING[19330][C-0005]: app.c:1633
> __ast
The issue in my last Email has solved the error about missing extension.
Now the following errors remain:
[Jul 2 02:29:18] WARNING[18226][C-0001]: res_config_odbc.c:117
> custom_prepare: SQL Prepare failed![SELECT * FROM sipusers WHERE host = ?
> AND callbackextension = ? AND port = ?]
> [Jul
The tutorial contains a mistake where the priority ordering in
extensions.conf should start with 1, not n:
; Voicemail
> exten => _VMR_.,1,Ringing
> exten => _VMR_.,n,Wait(1)
> exten => _VMR_.,n,Answer
> exten => _VMR_.,n,Wait(1)
> exten => _VMR_.,n,Voicemail(${EXTEN:4}|u)
> exten => _VMR_.,n,Hang
Hi,
Adding 'limit 1' or 'limit 5' to the supusers mysql view resolves part of
the error, but I don't understand why that is and whether this is correct
for the setup. Maybe something to do with connection pooling?
Now the following errors remain:
[Jun 30 01:07:53] NOTICE[17067][C-] chan_
Hi Nabeel,
The "sipusers" mysql view (as per
http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8#toc7
) has both the name and host fields - not sure why that query may fail.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-soluti
Hi Bogdan,
I was able to install the latest versions of Asterisk (13.1) and Opensips
(2.3) according to the tutorial, but when attempting to leave a voicemail I
get the following errors:
> [Jun 30 01:07:53] NOTICE[17067][C-] chan_sip.c: Call from
> '+447867958678' (162.249.6.206:12221) t
Hi Nabeel,
We will update the tutorial for 2.2, but it should still match. Give it
a try and if you hit issues, just let me know.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.06.2016 10:18, Nabeel wrote:
Hi,
I will be following this t
Hi,
I will be following this tutorial to integrate OpenSIPS and Asterisk:
http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8
The tutorial mentions the use of OpenSIPS version 1.8 and Asterisk version
1.8. I would like to know if I can use the latest versions of OpenS
Apologies, I just re-read the start of the tutorial that confirms that
registration should be carried out by openSIPS rather than asterisk,
which is there to support media services...
I've clearly spent too long looking at this! ;)
On 04/14/2013 08:50 PM, sjs205 wrote:
But should the registra
But should the registration be carried out by asterisk or opensips in
the tutorial environment?
On 04/14/2013 08:31 PM, Nick Khamis wrote:
On 4/14/13, Olle E. Johansson wrote:
13 apr 2013 kl. 21:43 skrev Nick Khamis :
Make sure that you have host=dynamic on both the general level (i.e.,
si
On 4/14/13, Olle E. Johansson wrote:
>
> 13 apr 2013 kl. 21:43 skrev Nick Khamis :
>
>> Make sure that you have host=dynamic on both the general level (i.e.,
>> sip.conf) and at the
>> peer level (i.e., extensions, sip_peers in the database etc...)
>
> host=dynamic has no effect whatsoever in the
Hello Olle,
I had previously tried this, and it seemed to help somewhat, although I
then have to create the following fields too otherwise Asterisk
complains about not being able to update them:
alter table subscriber add column `fullcontact` int(35) DEFAULT NULL;
alter table subscriber add co
13 apr 2013 kl. 22:08 skrev sjs205 :
> Hello N,
>
> Thanks for getting back to me on this. This is one of the issues with this
> tutorial, one can not set the asterisk sip_peers to dynamic since the
> tutorial creates a view from the 'subscriber' table and uses the 'domain'
> field as the 'ho
13 apr 2013 kl. 21:43 skrev Nick Khamis :
> Make sure that you have host=dynamic on both the general level (i.e.,
> sip.conf) and at the
> peer level (i.e., extensions, sip_peers in the database etc...)
host=dynamic has no effect whatsoever in the general section of sip.conf.
You need it for typ
Hello N,
Thanks for getting back to me on this. This is one of the issues with
this tutorial, one can not set the asterisk sip_peers to dynamic since
the tutorial creates a view from the 'subscriber' table and uses the
'domain' field as the 'host', 'default', 'ipaddr', 'fromdomain' and
'regse
Make sure that you have host=dynamic on both the general level (i.e.,
sip.conf) and at the
peer level (i.e., extensions, sip_peers in the database etc...)
N.
On 4/13/13, sjs205 wrote:
> Hello all,
>
> I'm going round and round in circles trying to integrate openSIPS and
> asterisk using the tut
Hello all,
I'm going round and round in circles trying to integrate openSIPS and
asterisk using the tutorial found at:
http://www.opensips.org/Resources/DocsTutAsterisk18
I have managed to get openSIPS installed and starting without errors
using the configuration scripts included in the abov
HI Qasim,
Thanks for the Replay
Can you post one sample config file so that i can go head with that
Thanks
Jagadish
On 29 March 2013 06:04, qasimak...@gmail.com wrote:
> If you are using Asterisk then you dont need media proxy as asterisk can
> handle NAT and Media issues. You just need to f
If you are using Asterisk then you dont need media proxy as asterisk can
handle NAT and Media issues. You just need to forward SIP messages to
asterisk. Use OpenSIPs in LoadBalancer/Dispatcher scenerio.
Regards,
Qasim
On Thu, Mar 28, 2013 at 12:07 AM, Jagadish Thoutam
wrote:
> HI All,
>
> I am
HI All,
I am New Here, i am getting Confusion while i am useing Openisps with My
asterisk Cluster My Implimentation Plan is Like this
(NAT)Opensips
1--- |
Asterisk1
Inbound
n...@lists.opensips.org] On Behalf Of David J.
> Sent: 04 May 2010 18:00
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] opensips and asterisk
>
> Sorry, The way I recommend doing this was assuming the user on the
> Asterisk box needed to be publicly reachable from an
04 May 2010 18:00
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] opensips and asterisk
Sorry, The way I recommend doing this was assuming the user on the
Asterisk box needed to be publicly reachable from anywhere.
I think that approach makes sense when using DID's and inboun
Sorry, The way I recommend doing this was assuming the user on the
Asterisk box needed to be publicly reachable from anywhere.
I think that approach makes sense when using DID's and inbound routing
that does need authentication.
On 5/4/10 12:55 PM, Olle E. Johansson wrote:
> 4 maj 2010 kl. 18
4 maj 2010 kl. 18.30 skrev Brett Nemeroff:
> Carmelo,
> If you have an SIP peer that matches the host and port of the opensips
> server.. ie:
> [opensips]
> type=friend
> host= port= (can be omitted if port 5060)
>
> Then it'll match that.. typically if it's coming from opensips you'll want to
Carmelo,
If you have an SIP peer that matches the host and port of the opensips
server.. ie:
[opensips]
type=friend
host= (can be omitted if port 5060)
Then it'll match that.. typically if it's coming from opensips you'll want
to add:
insecure=invite
so that opensips won't be challenged to authen
Check in the SIP.conf where you send all unauthenticated calls.
On 5/4/10 11:45 AM, wüber wrote:
> The problem seems to be not only in the extensions.conf file, but also in the
> sip.conf file.
> I still get this forbidden message!
>
___
Users ma
The problem seems to be not only in the extensions.conf file, but also in the
sip.conf file.
I still get this forbidden message!
--
View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/opensips-and-asterisk-tp4962200p5003971.html
Sent from the OpenSIPS - Users mail
You need to add a route in your extensions.conf in the context where you
send all un-authenticated calls.
Maybe its your default context?
[default]
exten = 1001,1,Dial(SIP/1001,10,tTr);
On 5/4/10 9:02 AM, wüber wrote:
> Hi Bogdan,
>
> connecting Opensips with Asterisk I can see that if a cl
Hi Bogdan,
connecting Opensips with Asterisk I can see that if a client registered on
Opensips server tries to make a call to a client in Asterisk domain, after
the INVITE, it receives a "forbidden" message from asterisk. I have set the
forwarding functionality in Opensips (rewriteuri function) a
Hi Carmelo,
routing between different SIP domains in typically done via DNS
(resolving the DNS part of the RURI).
OpenSIPS supports this by default - if you check the default config file
that comes with OpenSIPS, you will find the section when calls targeting
other domains are routed - also ch
Hello,
I'd like to have Opensips and Asterisk on two different networks (and
different sip domains), and make each ua in Opensips domain reachable
from the ua in Asterisk domain and vice versa.
how should I configure Opensips (and Asterisk, if anybody knows)?
thanks for your support!
__
Hi,
could you please draw a small flow of the call, just to understand it...
Like
UA(104) (INVITE)---> Asterisk -> proxy1 ->etc
For INVITE, 3xx , etc
Regards,
Bogdan
Jennifer-4 wrote:
> Hi!
>
> I´m using two Opensips as proxys, and they also take decisions about
> redirections of d
Hi!
I´m using two Opensips as proxys, and they also take decisions about
redirections of different calls.
All messages pass through Asterisk.
In a call from 104 to 100, the first thing I do is send the INVITE message
to Asterisk. Later, after Opensips receives the new INVITE (from Asterisk),
it
Hi Bogdan , something stranger happens when I put the debug in 6 I
don't see that it shows me the opensips log
tail -f /var/log/openser.log
twoxserver /sbin/opensips[3744]: INFO:core:sig_usr: signal 15 received
twoxserver /sbin/opensips[3733]: INFO:core:sig_usr: signal 15 received
twoxserver /sbi
El Miércoles, 29 de Abril de 2009, Iñaki Baz Castillo escribió:
> El Miércoles, 29 de Abril de 2009, troxlinux escribió:
> > Hi Bogdan , I don't have any alias en mi opensips.cfg , the only thing
> > that I have is that when they make calls to the pstn they leave to
> > that ip port
> >
> > route[4
El Miércoles, 29 de Abril de 2009, troxlinux escribió:
> Hi Bogdan , I don't have any alias en mi opensips.cfg , the only thing
> that I have is that when they make calls to the pstn they leave to
> that ip port
>
> route[4] {
> rewritehostport("192.168.10.3:5070");
> route(1);
> }
I see...Could you please get the opensips logs in full debug (debug=6)
for the ACK processing? I can take a look to see what exactly is going on.
Regards,
Bogdan
troxlinux wrote:
> 2009/4/28 Bogdan-Andrei Iancu :
>
>> It seams you have an ACK routing problem. The caller (.30:5064) correctly
>
2009/4/28 Bogdan-Andrei Iancu :
> It seams you have an ACK routing problem. The caller (.30:5064) correctly
> sends ACK with:
> ACK sip:*...@192.168.10.3:5070 SIP/2.0
> Route:
>
> but opensips (.3:5060),sends it out as:
> ACK sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3 SIP/2.0
>
It seams you have an ACK routing problem. The caller (.30:5064)
correctly sends ACK with:
ACK sip:*...@192.168.10.3:5070 SIP/2.0
Route:
but opensips (.3:5060),sends it out as:
ACK sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3 SIP/2.0
this means that OSIPS tinks that 192.16
it is strange, I thought that it was asterisk the problem, but I
upgrade to a version that I consider stable 1.4.24
2009/4/28 Bogdan-Andrei Iancu :
> Hi,
>
> Get a ngrep capture of the SIP traffic between * and OSIPS . Typically a
> retransmission is triggered by a lack of response from the othe
excuseme , I didn't remember that there was a list
2009/4/27 Alex Balashov :
> You may wish to consider posting this to the SER-Asterisk-Interwork list.
>
regardss
--
rickygm
http://gnuforever.homelinux.com
___
Users mailing list
Users@lists.opensip
Hi,
Get a ngrep capture of the SIP traffic between * and OSIPS . Typically a
retransmission is triggered by a lack of response from the other party,
but to see what response is lacking, you need to see the ngrep capture
of the SIP traffic.
Regards,
Bogdan
troxlinux wrote:
> Hi list , I have s
You may wish to consider posting this to the SER-Asterisk-Interwork list.
troxlinux wrote:
> Hi list , I have some days fighting with asterisk and opensips to
> solve this problem, when I use asterisk to listen my voicemail and to
> call to the pstn, asterisk shows me this error message:
>
> WA
Hi list , I have some days fighting with asterisk and opensips to
solve this problem, when I use asterisk to listen my voicemail and to
call to the pstn, asterisk shows me this error message:
WARNING[3196]: chan_sip.c:1976 retrans_pkt: Maximum retries exceeded
on transmission d5a57aa528f5c...@192
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