Has anyone managed to run SERweb with PHP5 or is that asking too much from
it?
Regards
Paul
-----Original Message-----
From: Paul Cowie
Sent: 16 September 2004 09:36
To: serusers(a)lists.iptel.org
Subject: Problems running SERweb
Hi I am having problems running SERWEB - I have read and made the changes to
the various config files and .php pages as per the SER HowTo Document - and
adjusted the correct file locations.
The problem I have seems to be with the PHPLLB stuff in that when I run a
web page all I get is text from part of the .php page - I can see that when
I run the page it is not distinguishing between '>' to denote the start of a
code block and various locations where '>' is used in the code i.e.
The index.php page
<?
require "prepend.php";
require "../../../pplib/ooforms.inc";
put_headers();
page_open (array("sess" => seems to be seen as a code block - what I
actually see on the web page is the remaining code and HTML in the page.
I know everything on the PHP side works as I have other pages that don't use
the PHPLLB
Is there something I have not done?
BTW PHP5 -Apache2
Thanks in advance
Paul
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Hello List.
My test settings are as follows -
[UA-1]-------[ser + mediaproxy 1]--------------[ser + mediaproxy 2]-----------------[UA-2]
I have the following setup in both the ser.cfg files (ser 1 and ser 2) -
-----------------------------------------
if (method == "BYE" || method == "CANCEL") {
end_media_session();
}
-----------------------------------------
However it appears that when UA2 ends the call, SER 2 calls end_media_session() - but SER 1 doesn't. When UA1 ends the call, the opposite happens. Shouldn't both the SERs be calling the end_media_session()?
Thanks for any feedback.
Regards,
Dhiraj
Hi I am having problems running SERWEB - I have read and made the changes to
the various config files and .php pages as per the SER HowTo Document - and
adjusted the correct file locations.
The problem I have seems to be with the PHPLLB stuff in that when I run a
web page all I get is text from part of the .php page - I can see that when
I run the page it is not distinguishing between '>' to denote the start of a
code block and various locations where '>' is used in the code i.e.
The index.php page
<?
require "prepend.php";
require "../../../pplib/ooforms.inc";
put_headers();
page_open (array("sess" => seems to be seen as a code block - what I
actually see on the web page is the remaining code and HTML in the page.
I know everything on the PHP side works as I have other pages that don't use
the PHPLLB
Is there something I have not done?
BTW PHP5 -Apache2
Thanks in advance
Paul
**************************************************************************
Disclaimer
This email transmission is confidential and intended solely for the person
or organisation to whom it is addressed. If you are not the intended
recipient, you must not copy, distribute or disseminate the information, or
take any action in reliance of it. Any views expressed in this message are
those of the individual sender, except where the sender specifically states
them to be the views of any organisation or employer. If you have received
this message in error, do not open any attachment but please notify the
sender (above) and delete this message from your system. Please rely on your
own virus check. Although all outbound mail is checked for viruses, no
responsibility is taken by the sender for any damage rising out of any bug
or virus infection.
**************************************************************************
I have compiled SER 0.8.14 from tar and did not get mysql.so. I did
"make all", expecting to get the mysql module, but it isnt there. Any
hints? The compile went fine with no errors, and SER starts up and
runs fine, until I uncomment the mysql.so module. Any help will be
appreciated. Thank you
Has anyone done interop with a provider that requires From:, To: and URI to
be in e.164? The problem we are having is that if our customers dial a
ten-digit number, the To: is written by the UA as NXXXXXXXXX(a)site.com but we
have no way to rewrite that as +1NPANXXXXXX(a)site.com before we send the
message out. All the mailing list messages I have found suggest that it's a
bad idea to rewrite To:. It looks like we could do it with a b2bua, but I
want to avoid adding that extra complexity for what seems (on the surface)
to be a simple problem. Anybody been through this? Thanks.
-Keith
Greetings All,
I have a question regarding the acc table in the ser database. That column
is getting populated with the called number and not the number (username) of
the party making the call. I.E. if user 1234 calls user 5678 the username
column in the acc table will get populated with 5678 for all transactions
regarding the call.
I'm thinking this is not correct. Can anyone verify this? If it is wrong,
any ideas as to why this is happening?
Cheers,
C.
SER is simply a proxy - it does not handle media in the same way that
asterisk does. When you have both clients registered with SER, once the
initial call set up has been completed, no further traffic runs through
SER. Search this list for explanations as to why RTP traffic doesn't
really run through NAT without a helping hand.
If you want to be able to make calls without any special client/NAT
router settings, check out RTPproxy/NAThelper and Mediaproxy - they do
the RTP proxy bit that Asterisk has built in.
Hope this helps.
Dave
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Morten Kuehl
Sent: 13 September 2004 22:35
To: serusers(a)lists.iptel.org
Subject: [Serusers] NAT with SER and ASTERISK, strange behaviour
Hi folks,
I just spent the evening with trying to find a logical solution to a NAT
problem but I had no success. I decided therefore to go to bed and let
the
more intelligent guys have a guess ;):
I do have the following setup:
Cisco 7960G with public IP
Xlite behind router with NAT
Ser server with a public ip
Asterisk with a public ip
Xlite finds out that it is behind a symetric firewall while starting and
sends the external ip of the nat device in its sip messages.
I want to call the xlite client behind NAT from the Cisco phone. It
works
when the cisco phone and the xlite client are registered with the
asterisk
server. I have set nat=yes and canreinvite=no in the account settings
for
the xlite client in the sip.conf of the asterisk server. The Cisco phone
has a normal account. I had a look at the sip messages and as
configured,
the Asterisk server works as a rtp proxy for the media stream. Audio
works
in both ways.
When I use ser and register both clients with ser and start a call,
audio
works only outbound from xlite to the cisco phone but not inbound from
the
cisco phone to xlite through the nat device.
I digged the sip messages several times and did not see any big
difference
between the two sip conversations, besides having the asterisk as an rtp
proxy in the middle in the first attempt. Neither did I see an approach
of
asterisk to make xlite pinhole the nat device as describe for example in
the sip cookbook.
Unfortunately I am stuck at this stage and cannot find a logical
explanation for the described behaviour. If anyone can give me a hint or
needs further information to assist, please let me know!
Cheers
Morten
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I'm Trying to set up ser in my homenetwork. I followed the ser howto,
did all the dns entres and stuff. Now when I connect I always get an
"Too many Hops" error. Buh my sip client and the sip server are in the
same network so there are no hops at all.
the secitons with the headers in the sip.cfg file is unchanged like
this:
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
Did anyone experience the same problem?
---
Lukas Bolliger <bolukas(a)ee.ethz.ch>
