Greetings,
As a developer and consultant who spends considerable time on projects
involving the fusion of Asterisk and products derived from the SER
ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have
found that there is a great volume of interest in this topic on the
mailing
Good work, I am sure this will be endorsed by many and will be useful for
lots of small VoIP user who are ready to expand. Only problem I have seen is
that people who have done (deployed) this type of integration does not share
complete solution mainly because of compititive disadvantage. But
That's not actually true. SER is very much alive and well and under
constant development.
How do I KNOW it's constant development (other than the chatter on the
mailing list)? Because things keep changing in CVS, but there never
seems to be a 'release' version. Just a release candidate. ;)
SIP wrote:
Seriously, though... this seems to be a popular misconception. I hear it
a lot. Where did you come across the information that SER is no longer
developed?
That seems to be a consequence of looking at the releases.
Anyway, I spoke too soon in saying that there's absolutely
Alex Balashov wrote:
SIP wrote:
Seriously, though... this seems to be a popular misconception. I hear it
a lot. Where did you come across the information that SER is no longer
developed?
That seems to be a consequence of looking at the releases.
Anyway, I spoke too soon in
SIP wrote:
I see a lot of parellels there with OpenSIPS and SER. OpenSIPS is a
stable plaform that has dozens of modules and documentation galore on
how to mesh the system with this, that, and the other. SER has
rock-solid, incredibly innovative core code, but prefers to leave the
writing
Alex Balashov schrieb:
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.
Well, i am not getting the correct meaning of 'defunct', but from the
last part of your suggestion i guess you value Kamailio/OpenSIPS more
than SER.
Are there some hard reasion for this.
I
No, the issue isn't my value or preference. The issue is that SER is no
longer maintained or developed and has not been for several years.
Tobias Wolf wrote:
Alex Balashov schrieb:
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.
Well, i am not getting the
Alex Balashov wrote:
No, the issue isn't my value or preference. The issue is that SER is no
longer maintained or developed and has not been for several years.
The above statement is totally false. SER is indeed an ongoing project
which is actively maintained. If you subscribed to the
On 10/18/08 05:28, Grey Man wrote:
As far as I'm aware SER (and it's derivatives) cannot initiate
outbound registraitions. They can do the opposite and act as a SIP
Registrar. For outbound registrations you should be able to use
Asterisk.
Regards,
Greyman.
Yes, I use Asterisk for iax outside
On 10/18/08 10:39, ram wrote:
On Sat, Oct 18, 2008 at 9:20 AM, Joseph [EMAIL PROTECTED] wrote:
I would gladly go with any of the newer packages if I only could.
I'm just working with what I can find in portage; I'm sure it will be
eventually available. It will first show up via overlay.
No, a proxy cannot *initiate* anything.
ram wrote:
On Sat, Oct 18, 2008 at 9:20 AM, Joseph [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
On 10/17/08 23:23, Kristian Kielhofner wrote:
On 10/17/08, Alex Balashov [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Doesn't mean it's not defunct.
Joseph wrote:
I'm using Gentoo and the only package I was able to find in portage was SER;
I could compile manually but it is harder to upgrade and keep track of
dependencies.
--
#Joseph
On 10/17/08 22:42, Alex Balashov wrote:
SER is defunct. Kamailio
Joseph wrote:
Thanks for your help.
How to use UAC Module to register with a provider?
Is there something like STUN for SER?
I don't want to open too many ports on my firewall.
You do not need to open any ports on your firewall if your NAT gateway
does proper translation.
You cannot use
On 10/18/08 13:51, Alex Balashov wrote:
Joseph wrote:
Thanks for your help.
How to use UAC Module to register with a provider?
Is there something like STUN for SER?
I don't want to open too many ports on my firewall.
You do not need to open any ports on your firewall if your NAT gateway
Joseph wrote:
On 10/18/08 13:51, Alex Balashov wrote:
Joseph wrote:
Thanks for your help.
How to use UAC Module to register with a provider?
Is there something like STUN for SER?
I don't want to open too many ports on my firewall.
You do not need to open any ports on your firewall if
On 10/18/08 15:31, Alex Balashov wrote:
Joseph wrote:
On 10/18/08 13:51, Alex Balashov wrote:
Joseph wrote:
Thanks for your help.
How to use UAC Module to register with a provider?
Is there something like STUN for SER?
I don't want to open too many ports on my firewall.
You do not need
Joseph wrote:
Thanks for the info Alex,
Do you have a good links that would help accomplish it?
I was under impression that nathelper is only for incoming connection, not
outgoing.
Sure - it's incoming from the point of view the proxy, if you do:
Asterisk --- proxy w/NAT traversal
On 10/18/08 16:10, Alex Balashov wrote:
Joseph wrote:
Thanks for the info Alex,
Do you have a good links that would help accomplish it?
I was under impression that nathelper is only for incoming connection, not
outgoing.
Sure - it's incoming from the point of view the proxy, if you do:
Joseph wrote:
On 10/18/08 16:10, Alex Balashov wrote:
Joseph wrote:
Thanks for the info Alex,
Do you have a good links that would help accomplish it?
I was under impression that nathelper is only for incoming connection,
not outgoing.
Sure - it's incoming from the point of view the proxy,
On 10/18/08 16:48, Alex Balashov wrote:
[snip]
There is not really a lot of good conceptual introduction to OpenSER,
although Flavio Goncalves' book (Building Scalable Telephony
Applications With OpenSER) may be somewhat of aid. The documentation
primarily serves those that already know what
On Sat, Oct 18, 2008 at 4:48 PM, Alex Balashov [EMAIL PROTECTED]wrote:
Joseph wrote:
On 10/18/08 16:10, Alex Balashov wrote:
Joseph wrote:
Thanks for the info Alex,
Do you have a good links that would help accomplish it?
I was under impression that nathelper is only for incoming
Steve Totaro wrote:
If someone wrote a nice webmin module with all the configuration options
as check boxes and fill in the blanks, that would be very NICE!
The problem with simply doing a GUI frontend to *SER is that it's very
polymorphic far too extensible; there are far too many
On Sat, Oct 18, 2008 at 5:35 PM, Alex Balashov [EMAIL PROTECTED]wrote:
Steve Totaro wrote:
If someone wrote a nice webmin module with all the configuration options
as check boxes and fill in the blanks, that would be very NICE!
The problem with simply doing a GUI frontend to *SER is that
Steve Totaro wrote:
Kind of like SwitchVox, FreePBX, Thirdlane..
I don't know that I'd make that comparison.
I would say that in general, OpenSER is more low-level and amorphous and
multipurpose than Asterisk or any GUI that wraps it.
Asterisk has many applications and uses and niches,
I am running Asterisk and would like to add SER to register my (sip) DID and
connect it to asterisk;
but I'm not sure if this is the correct forum.
I have as DID, sip account with one VoIP provider; currently Im using just
stand alone SIP phone and register with the VoIP provider via:
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.
On Fri, October 17, 2008 9:36 pm, Joseph wrote:
I am running Asterisk and would like to add SER to register my (sip) DID
and connect it to asterisk;
but I'm not sure if this is the correct forum.
I have as DID, sip
On 10/17/08, Alex Balashov [EMAIL PROTECTED] wrote:
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.
Slight clarification: Kamailio (formerly OpenSER) / OpenSIPS is the
thing to do.
--
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
I'm using Gentoo and the only package I was able to find in portage was SER;
I could compile manually but it is harder to upgrade and keep track of
dependencies.
--
#Joseph
On 10/17/08 22:42, Alex Balashov wrote:
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.
On
On 10/17/08 23:23, Kristian Kielhofner wrote:
On 10/17/08, Alex Balashov [EMAIL PROTECTED] wrote:
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.
Slight clarification: Kamailio (formerly OpenSER) / OpenSIPS is the
thing to do.
I would gladly go with any of the
As far as I'm aware SER (and it's derivatives) cannot initiate
outbound registraitions. They can do the opposite and act as a SIP
Registrar. For outbound registrations you should be able to use
Asterisk.
Regards,
Greyman.
___
-- Bandwidth and
On Sat, Oct 18, 2008 at 9:20 AM, Joseph [EMAIL PROTECTED] wrote:
On 10/17/08 23:23, Kristian Kielhofner wrote:
On 10/17/08, Alex Balashov [EMAIL PROTECTED] wrote:
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to
do.
Slight clarification: Kamailio (formerly
Hi,
i use a ser, as proxy sip server(authentication), then a cisco router
as sip2h323 gw(authorization and accounting). i want to start asterisk
as sip statefull b2bua server, any suggestion to howto or documentation
to asterisk integration and b2b use?
ty in advance.
--
Riccardo Cupardo
Riccardo Cupardo wrote:
Hi,
i use a ser, as proxy sip server(authentication), then a cisco router as
sip2h323 gw(authorization and accounting). i want to start asterisk as
sip statefull b2bua server, any suggestion to howto or documentation to
asterisk integration and b2b use?
Well,
Hi,
I'm trying to have a SER machine send calls to an Asterisk server
working as an IVR. I was able to do this part just fine. Also, when
the caller makes certain options in the IVR, the call is then
transferred to an extension via SER. This part is also just fine.
However, I'm trying to
On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I'm trying to have a SER machine send calls to an Asterisk server
working as an IVR. I was able to do this part just fine. Also, when
the caller makes certain options in the IVR, the call is then
transferred to an extension via
I have ser sitting on my iptables nat box and my asterisk box on the lan
. Ser does forwarding so that any requests (register,invite,ack,...) to
the nat box at 5060 r sent to my asterisk box on the lan .I can register
from outside
to my asterisk box but there is only one way audio , reason being
Have a check through:http://www.voip-info.org/wiki-NAT+and+VOIPhttp://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
RegardsRobOn 03/09/06, Siqhamo Sifo [EMAIL PROTECTED] wrote:
I have ser sitting on my iptables nat boxand my asterisk box on the lan. Ser does forwarding so that any requests
have a look at the nathelper examples in SER distribution. This is from
an rather old installation of mine.
--
# !! Nathelper
# Special handling for NATed clients; first, NAT test is
# executed: it looks for via!=received and RFC1918 addresses
# in Contact (may fail if
Hi,
I'm deploying a SER + Asterisk architecture, where SER is used to manage
acc, users database and sip routing, and Asterisk is used for voicemail
and PSTN gateway.
The system is already able to make and receive calls from the PSTN,
although, only after the call has been established it can
Problem solved.
It was needed to insert the following code in ser.cfg:
-
if (method==CANCEL) {
route(1);
break;
}
-
and also:
Hi all
I have badly NATed Clients proble with one way Voice
After reading some documents people ask me to use STUN Server
But still i have some problem with one way Voice
I have setup like below
iam trying with 2 extensions
1 extention in the same LAN where the * installed
2 extension in
ram wrote:
Hi all
I have badly NATed Clients proble with one way Voice
After reading some documents people ask me to use STUN Server
But still i have some problem with one way Voice
use stun on dinamic ip :)
I have setup like below
iam trying with 2 extensions
1 extention in the same
Hi
thanks for the reply
ya the default is NAT=YES only
if i keep reinvite=no, the my server b/w consuming lot
since i have bottleneck of server bandwidth
ram
On 3/16/06, Andrei Sotirov [EMAIL PROTECTED] wrote:
ram wrote: Hi all I have badly NATed Clients proble with one way Voice
After
hello,
I am trying to pass MWI from Asterisk to SER.my user agents register
with Ser.i am not able to figure out how to do this.
i added the changes for mailbox in sip.conf for ser peer entry.
[ser]
type=friend
mailbox=XYZ
also changes in chan_sip.c for asterisk but not seeing the notify
When using Asterisk and SER together, should SER place calls to the PSTN,
and Asterisk only deal with special features such as voicemail, queues,
autoattendants, etc? Or should SER be used ONLY as a proxy/registrar, and
all calls be routed to Asterisk so that Asterisk places the calls to the
]
Subject: Re: [Asterisk-Users] SER Asterisk combination to get around
NAT
Importance: High
Hello Stuart, we have, and I would be happy to help you setup both
Asterisk and SER on a consultancy basis.
You can find more information about me here:
http://mark.teamcebu.com
Basically, it requires SER
Stuart Hirst ha scritto:
Has anyone successfully used SER and Asterisk together on the same
server to get around NAT traversal issues.
I have looked at many of the NAT traversal topics which either involve
commercial products and significant costs or solutions such as STUN or
proprietary
Has anyone successfully used SER and Asterisk together on the same
server to get around NAT traversal issues.
I have looked at many of the NAT traversal topics which either involve
commercial products and significant costs or solutions such as STUN or
proprietary systems such as xten.
--
No
No !
Asterisk should send the invite request to sip proxy .
Harry
--- Walter Willis [EMAIL PROTECTED] a écrit :
the ser an asterisk run in the same box???
redirect host + port :)
2005/11/4, harry gaillac [EMAIL PROTECTED]:
Hello,
I wish to setup this scheme:
ser-0.9.4
Hello,
I wish to setup this scheme:
ser-0.9.4
asterisk-1.2-bêta
polycom ip300 phones
asterisk 5050--
|firewall+nat|-192.168.
ser 5060---
My ip phones use ser as outbound sip proxy and
asterisk as sip registrar server.
Ser Forward REGISTER requests to asterisk however
the ser an asterisk run in the same box???
redirect host + port :)
2005/11/4, harry gaillac [EMAIL PROTECTED]:
Hello,I wish to setup this scheme:ser-0.9.4asterisk-1.2-bêtapolycom ip300 phonesasterisk 5050-- |firewall+nat|-192.168.ser 5060---My ip phones use ser as outbound sip proxy and
Hello Walter,
The ser an asterisk run in the same box.
What do you mean redirect host + port :)
Sip agents send sip requests to ser (outbound proxy)
and this one to asterisk !
sip agents are both registered on ser and asterisk.
Please to explain me how asterisk redirect the
requests.
Regards
you could wait infinitely or try users list..On 11/4/05, harry gaillac [EMAIL PROTECTED] wrote:
Hello Walter,The ser an asterisk run in the same box.What do you mean redirect host + port :)
Sip agents send sip requests to ser (outbound proxy)and this one to asterisk !sip agents are both registered
my bad you are.. lol didnt realize..
On 11/4/05, Jimmy Smith [EMAIL PROTECTED] wrote:
you could wait infinitely or try users list..On 11/4/05, harry gaillac
[EMAIL PROTECTED] wrote:
Hello Walter,The ser an asterisk run in the same box.What do you mean redirect host + port :)
Sip agents send sip
Hello,
I set SER as sip proxy and ASTERISK as voicemail
server (ARA) and serweb as TUI (telephone user
interface) .
Serweb
|
Ua---ser---asterisk voicemail
| |
Mysql DB
I add user agents with address sip:[EMAIL PROTECTED] +
aliases
Hello,
Thanks for help it's ok with static file
voicemail.conf
However something is wrong with ARA .
app_voicemail search entries in voicemail.conf ?!
I set apps/Makefile for USE_ODBC_STORAGE.
Regards
Harry
//
Connected to Asterisk CVS-HEAD
Hello,
I try set Ua---SERAsterisk (voicemail/ARA)
|
Ua
ser stable
asterisk cvs head
I read
http://mail.iptel.org/pipermail/serusers/2005-February/015997.html
to forward unavailable or busy sip agents to asterisk
voicemail in failure route.
How may I configure
You'll want some rules in your sip.conf to handle the connection from
SER. A
starting point might be:
[ser ip addr:ser port ?= 5060]
type=peer
context=my sip context name
tos=lowdelay; tos delay
allow=ulaw ; dtmfmode=inband only works with
I am fairly new to Asterisk / VOIP and have been playing around with it
for long enough to have a whole lot of questions so far without answers.
Presently Im running Asterisk (v.1.0.7) on a Debian Sarge installation
with 2 soft phones (for testing purposes). A live deployment will
Title: SER Asterisk SIP =513 Message Too Big
Using Asterisk 1.0.9
When I try to make an outgoing call with SIP I get the message 513 Message too big back from SER. Any ideas what I am doing wrong?
Debug below.
SER and Asterisk are running on the same Server
SER is on port 5060
hello
I am using ser with asterisk
asterisk on 5070 (on back end)
ser on 5060 (on front end)
i am getting all requests at asterisk.
i tried by changing asterisk port
bindport=5090
but still getting all requests from sjphone at
asterisk.
can any one tell what is the reason
regrads
Kamran
On 19/05/05, Kamran Ahmad [EMAIL PROTECTED] wrote:
hello
I am using ser with asterisk
asterisk on 5070 (on back end)
ser on 5060 (on front end)
i am getting all requests at asterisk.
i tried by changing asterisk port
bindport=5090
but still getting all requests from sjphone at
I have been trying to setup Asterisk in combination with SER on the
same box as a PBX with SIP clients. I would like to have it available
for both external and internal users so I have the box setup with
external and internal IP address. I am running into all kinds of
troubles with this
I'm working with SER + Asterisk. I was told that to have SER push calls to
multiple Asterisk servers, I can use the LCR Module, I'll just give all
the Asterisk servers the same weight/price, and SER will randomly send
outbound requests to each Asterisk server. It's not truly equally
balanced, so
I have searched high and low for these, but to no avail, nothing useful
back from google, nothing I could find on this mailing list, or
voip-user.org.
Does anyone have any good urls and or pointers which will assist in
configuring SIP Express Router and Asterisk talking to each other on the
same
: [Asterisk-Users] ser - asterisk configs anyone?
I have searched high and low for these, but to no avail, nothing useful
back from google, nothing I could find on this mailing list, or
voip-user.org.
Does anyone have any good urls and or pointers which will assist in
configuring SIP Express
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
hi,
we have the ser sip-proxy for registration and we forwarding
the call to our cisco gateway and it works.
but now we will forwarding the calls to the asterisk and
the asterisk shoud forward the calls to our gw (via sip not h323).
how must i
Hi List,
Can I use asterisk to enable call conferencing? I'm using ser for the UA's to
register, can I do something like if they dial a certain digits, it will
forward it asterisk and use asterisks meetme feature? can i do meetme using
only sip?
Sorry for my terms, hope you understand my
:[EMAIL PROTECTED]
Sent: Donnerstag, 31. März 2005 16:07
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] ser, asterisk and conferencing
Hi List,
Can I use asterisk to enable call conferencing? I'm using ser for the UA's
to
register, can I do
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ron
Sent: Thursday, March 31, 2005 9:07 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] ser, asterisk and conferencing
Hi List,
Can I use asterisk to enable call conferencing? I'm
be
helpful for you.
Regards
Cameron
- Original Message -
From: hans [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 01, 2005 1:14 AM
Subject: [Asterisk-Users] ser - asterisk -cisco gateway
-BEGIN PGP
Hi there,
I'm using ser and asterisktogether. Asterisk
for voice mail etc and ser forregistration of the users
usig database.I can restrict forwarding
callsfrom another sip proxy to ser(using proxy_authorize) but how
can I restrict access to asterisk ... Now everyone can forward calls to
Pavel Siderov - Hostmates wrote:
I can restrict forwarding calls from another sip
proxy to ser (using proxy_authorize) but how can I restrict access
to asterisk ... Now everyone can forward calls to my asterisk and
can place pstn calls.
Use iptables on the asterisk machine to only allow
Discussion
asterisk-users@lists.digium.com
Sent: Thursday, March 17, 2005 10:40 AM
Subject: RE: [Asterisk-Users] ser+asterisk - security
Pavel Siderov - Hostmates wrote:
I can restrict forwarding calls from another sip
proxy to ser (using proxy_authorize) but how can I restrict access
to asterisk
[EMAIL PROTECTED] wrote:
it's impossible to use iptables due to the reason that audio
flows through asterisk and users won't be able to communicate w/ *...
I was thinking of just the SIP port. I am assuming that asterisk
protects its RTP ports from processing traffic from a third party.
--
Hello all! I googled lists.digium.com and ser mailing list, but did
not find any working configuration of asterisk used as voicemail for
SER. This is my config
if (uri==myself) {
if (method==REGISTER) {
save(location);
log
If I use rewritehostport instead of forward, the call does not reach asterisk:
failure_route[1] {
revert_uri();
rewritehostport(69.70.x.x:5060);
t_relay()
break();
SER log:
Your failure route should read:
failure_route[1] {
revert_uri();
Keith Burns wrote:
Hi,
I am looking for SER/Asterisk consultants in Denver, please contact me at
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On Friday 18 February 2005 13:44, Michael Welter wrote:
Keith Burns wrote:
Hi,
I am looking for SER/Asterisk consultants in Denver, please contact me at
[EMAIL PROTECTED]
[... quoted signature deleted ...]
Hello Keith,
My name is Michael Welter, and I have been installing Asterisk
: Re: [Asterisk-Users] SER/Asterisk consultants in Denver
Keith Burns wrote:
Hi,
I am looking for SER/Asterisk consultants in Denver, please contact me
at
[EMAIL PROTECTED
Hi,
I am looking for SER/Asterisk consultants in Denver, please contact me at
[EMAIL PROTECTED]
attachment: winmail.dat___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hi all,
I have SER and Asterisk set up together with ser handling user
registrations and asterisk providing voicemail services. When I ring
a phone and it doesnt answer after a designated amount of time, the
request is forwarded to asterisk, and I can leave a message.
Now, this may seem a
The sipsak way simply lites the MWI (or not) to indicate a message is
waiting. You need to provide instructions in extensions.conf that route
the call into voicemailmain. I use
exten = 68007,1,VoicemailMain
exten = 68007,2,Hangup
-Steve
Aisling O'Driscoll wrote:
Hi all,
I have SER and Asterisk
Hi everybody!
this is third day I'm supposed to work on some telecomunications solution.
We have SIP Express Router to maintain and redirect incomming calls to
asterisk.
The problem is that we (i mean my company) have to run some prepaid
solution with asterisk.
I'm wondering if modified prepaid
Hi all
I am in this strange situation: we had ser configured to relay calls to
numbers to asterisk extensions and all used to work nicely, with both ser and
asterisk running on the same machine with public ip (ser on port 5060 and *
on 5061). We had to move temporarily our server to another
Hi All !
First I was having trouble using attended call transfer using asterisk but
thatnks to you guys I was able to make it work by adding 't' in options
and applying the patch. Now I am using SER along with asterisk. SER works
as SIP proxy and Asterisk performs all the necessary PBX
Hi,
since a while I try get Asterisk and SER work together. But until now I
have no success.
I want to use Asterisk as Gateway to the old telephone world.
Is there somebody who can give me a small example of the ser.cfg and the
Asterisk config files.
This will be very nice.
Thanks
Bastian
Hello.
I trying to use SER with Asterisk together. I have a question
regarding the RTP path. If i make a call from one of my endpoints
registered in SER Server, and that call in particular is forwarded to
Asterisk and then to a PSTN-GW, Does the media goes through Asterisk?? is
there a
Hi
We have a phone system consisting primarily of SER and Asterisk, and are
having trouble transferring inbound calls from the PSTN.
We believe the problem is basically that because our phones register
with SER, the Asterisk box never sees the call from the original callee
to the new callee.
i.e.
Hi there,
I've seen people using SER with Asterisk. I took a look at SER
website, and I didn't see the point in using it, since Asterisk
already handles SIP very well (apparently, at least).
But, as I'm starting, and some of you (more experienced) use it, I
know that there's something there...
hi list,
i want to use the astersik in conjunction with
the ser
so i followed the instructions provided on the
voip-info.org site
but when calling from one user to another it gives me
problem in the asterisk cli that
failed user authentication
my aim of doing this is to use the
Welesley Sibelson Dias wrote:
Hi All.
I'am using Asterisk with SER. I can make call between two internal VoIP
gateways or from na internal to external VoIP gateway. But when I get a
external call, this call hang ups 5 seconds after and I reveive the
following messages
*CLI -- Executing
Geert Nijpels wrote:
-- Attempting native bridge of SIP/16008-3d17 and SIP/16007-8c24 Mar
I know of a GrandStream bug which generates a wrong ack to the 200 OK
asterisk sends on connecting. SER drops this ack and asterisk drops the
call, as it should. This is fixed in latest firmware image.
Hi All.
I'am using Asterisk with SER. I can make call between two internal VoIP
gateways or from na internal to external VoIP gateway. But when I get a
external call, this call hang ups 5 seconds after and I reveive the
following messages
*CLI -- Executing Dial(SIP/16008-3d17,
But now i'm stumbling on another problem.. Asterisk seems to want
to send the SIP udp packets directly to the SIP clients.
In the case of a SIP user/client behind a NAT, this obviously doesn't
work.
Have you tried reinvite=no in your [ser] section of sip.conf?
P
On Sat, 2004-01-17 at 01:33, [EMAIL PROTECTED] wrote:
Thanks guys, thought SER had to 'register' to be able to use
any Asterisk contexts.
But just defining a new entry in the sip.conf with just context ip worked!
But now i'm stumbling on another problem.. Asterisk seems to want
to send the
[EMAIL PROTECTED] wrote:
I'm trying to bundle the powers of Asterisk and SER.
Asterisk for pabx functionalities and termination to landline/PSTN, and
SER as SIP Gateway/Proxy.
With my current configuration the SIP user just adds 0 as a prefix to a
number, and the call will go out to PSTN over
Yes, you can keep non-authorized SIP callers from accessing the
PSTN by setting up the .conf file correctly as below
but you can also
run a fire wall on the box that Asterisk runs on. Firewall off
SIP ports except for if they come from your SER server.
This will work even if Asterisk is broken
Thanks guys, thought SER had to 'register' to be able to use
any Asterisk contexts.
But just defining a new entry in the sip.conf with just context ip worked!
But now i'm stumbling on another problem.. Asterisk seems to want
to send the SIP udp packets directly to the SIP clients.
In the case of
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