Hi everyone again,
I have some troubles with serweb installation . Help is needed ;)
After entering admin username and password I see blank screen in admins
page Actually I don't see mysql query for a password in a server. And I
think this is the problem. I have filled mysql configuration in
config/config_data_layer.php .
Can anyone tell me then all files in /config dir are loaded?
this is serweb log:
Apr 27 15:23:01 serweb [info] Useing file from default domain for
filename: styles.css
Apr 27 15:23:01 serweb [info] Useing file from default domain for
filename: prolog.html
Apr 27 15:23:01 serweb [info] Useing file from default domain for
filename: separator.html
Apr 27 15:23:01 serweb [info] Useing file from default domain for
filename: epilog.html
Apr 27 15:23:11 serweb [debug] User login: values from login form:
username: admin(a)domain.ltd, password: somepass
Apr 27 15:23:11 serweb [debug] User login: checking password of user
with username: admin, domain: domain.ltd
Thanks,
Mindaugas
Hi all,
I installed the SER and Asterisk on two different boxes on LAN.. I configured the SER and Asterisk in a way thay that the user agents communicate between them.
Now, the problem is in the second method , if i use the register command in the Asterisk (sip.conf) , the user registers in the SER and we can view that user in the location table in SER,but at the same time the UA's send busy tone and i gets the messege in the asterisk proxy authentication required.
I use this procedure for registeration
1) In the general section of sip.conf
register => user[:secret[:authuser]]@host[:port][/extension]
2)sip.conf
[mysipprovider-out]
type=peer
secret=password
username=2345
host=sipserver.mysipprovider.com
fromuser=2345
fromdomain=fwd.pulver.com
nat=yes
;context=from-mysipprovider ;
3)In extensions.conf
exten => _9.,1,Dial(SIP/${EXTEN:1}@mysipprovider-out,30,r)
Can anybody tell me what i am doing wring with this process...
best regards,
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Dear All
I am trying to figure out how to use Music on hold
(using asterisk may be) for my SER
Any ideas how to do it?
Dashy
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Hi Bogdan
The first SIP message is from call that WORKS perfectly: the "c header" has a valid IP that handle the media. Below it, we have the message with wrong "c header": there is an invalid IP. This way, the media isnt sent to the media proxy.
I did this dumps in the SIP PROXY. The only way to correct the situation is rebooting the openser. By the way: I dont change anything in the script to make it works.
Best regards
Bruno Machado
U IP_My_Proxy:5060 -> 200.171.160.13:5060
INVITE sip:11000606@200.171.160.13:5060 SIP/2.0
Record-Route: <sip:IP_My_Proxy;ftag=ad50ff6e6fb26e6e;lr=on>
Via: SIP/2.0/UDP IP_My_Proxy;branch=z9hG4bKa079.9071d6a5.0
Via: SIP/2.0/UDP 192.168.88.101;rport=57157;received=200.216.162.18;branch=z9hG4bK9771e2044f41cee3
From: "Alexandre BT100" <sip:24005110@myproxy>;tag=ad50ff6e6fb26e6e
To: <sip:11000606@myproxy>
Contact:<sip:24005110@200.216.162.18:57157>
Supported: replaces
Proxy-Authorization: Digest username="24005110", realm="myproxy",
algorithm=MD5, uri="sip:11000606@myproxy",
nonce="44510d68180175d68510d001330893f695eadd50", response="ed83c6657bc59f00635361215d1b8b93"..Call-ID:
db5d8eb678f6c018(a)192.168.88.101
CSeq: 55645 INVITE
User-Agent: Grandstream BT110 1.0.8.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 429
P-hint: usrloc applied
P-hint: nat-p2p
v=0
o=24005110 8000 8001 IN IP4 192.168.88.101
s=SIP Call
c=IN IP4 myproxy
t=0 0
m=audio 64344 RTP/AVP 18 4 0 8 2 99 9
101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=20
a=rtpmap:9 G722/16000
a=ptime:20
a=rtpmap:101
telephone-event/8000
a=fmtp:1010-11
a=direction:active
a=nortpproxy:yes
U IP_My_Proxy:5060 -> 200.171.160.13:5060
INVITE sip:11000606@200.171.160.13:5060 SIP/2.0
Record-Route: <sip:IP_My_Proxy;ftag=da3a34829a170ace;lr=on>
Via: SIP/2.0/UDP IP_My_Proxy;branch=z9hG4bKd6cc.be1fc1f4.0
Via: SIP/2.0/UDP 192.168.88.101;rport=57157;received=200.216.162.18;branch=z9hG4bK019a1f171ca1bf14
From: "Alexandre BT100" <sip:24005110@myproxy>;tag=da3a34829a170ace
To: <sip:11000606@myproxy>
Contact: <sip:24005110@200.216.162.18:57157>
Supported: replaces
Proxy-Authorization: Digest username="24005110", realm="myproxy",
algorithm=MD5, uri="sip:11000606@myproxy",
nonce="44510c217bc74fe6e203d22c12c3863536776e75", response="c403be69f25add894daeccd2a29bf0e4"
Call-ID: 57efde4d3bb5a88f(a)192.168.88.101
CSeq: 5727 INVITE
User-Agent: Grandstream
BT110 1.0.8.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 410
P-hint: usrloc applied
P-hint: nat-p2p
v=0
o=24005110 8000 8001 IN IP4 192.168.88.101
s=SIP Call
c=IN IP4 192.168.88.101
t=0 0
m=audio 5004 RTP/AVP 18 4 0 8 2 99 9 101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=20
a=rtpmap:9 G722/16000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=direction:active
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Hi,
Just for the (pipermail) record: How I got OpenSER+SEMS to work on FreeBSD.
OpenSER is not causing any problems in combination with SEMS-svn anymore.
No need to use any specific version, just the CVS-head from openser.org.
It now works completely, an IVR app is reading out .wav files and
processing the DTMF music that I play for it.
In openser.cfg I used
...
unix_sock="/tmp/openser.sock"
...
if(!t_write_unix("/tmp/sems.sock", "ivr")){
t_reply("500", "error contacting sems");
}
...
In sems.conf I used
...
socket_name=/tmp/sems.sock
reply_socket_name=/tmp/sems.reply.sock
ser_socket_name=/tmp/openser.sock
send_method=socket
...
I started OpenSER with
openserctl start
I started SEMS with
/path/to/sbin/sems -d vr0 -f /path/to/etc/iptel/sems/sems.conf
The name vr0 happens to be the ethernet interface that I am using, on Linux
you'd mostly use eth0 for the same setup.
All this runs on FreeBSD with adaptions that were checked into SVN, rev.39.
Greetings,
-Rick
The problem appears when I tried to do that . The error I mentioned
below was encountered, session.py didn't show any session establish if
this error appears on the syslog.
Any idea ?
-----Original Message-----
From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro]
Sent: Wednesday, April 26, 2006 3:41 PM
To: Sam Lee
Cc: users(a)openser.org
Subject: Re: [Users] Force RTP stream to go through mediaproxy
Hi,
most probably, in your script, you use mediaproxy only if NAT was
detected (and a flag was set). So, you do not have to test the flag (to
use mediaproxy only for nated calls), but call mediaproxy for all calls.
regards,
bogdan
Sam Lee wrote:
>Can I don't bother about the NAT test and push everything thru
>mediaproxy ?
>I can't quite understand why are there different NAT_Test , and the
>example configs uses almost all of them...
>
>Regards,
>Sam
>
>-----Original Message-----
>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro]
>Sent: Wednesday, April 12, 2006 5:04 PM
>To: Sam Lee
>Cc: users(a)openser.org
>Subject: Re: [Users] Force RTP stream to go through mediaproxy
>
>Hi,
>
>maybe you do not perform the proper nat tests. see:
> http://openser.org/docs/modules/1.0.x/mediaproxy.html#AEN113
>
>regards,
>bogdan
>
>
>Sam Lee wrote:
>
>
>
>>Hi all,
>>
>>I got the openser and mediaproxy up working and fine.
>>The mediaproxy is able to work with certain NATED configurations, but
>>not all.
>>When i tried to check what is the problem , those NATED configurations
>>
>>
>
>
>
>>that were not working was found not to have made use of the
>>mediaproxy, which causes the problem.
>>
>>What i am trying to do now, but without much success, is to force all
>>the RTP media to go through the mediaproxy. Is there any way i can
>>force all the RTP media to go through the mediaproxy without all the
>>client_nat_test stuff...?
>>
>>Any hints will be much appreciated !
>>
>>Regards,
>>Sam
>>
>>----------------------------------------------------------------------
>>-
>>-
>>
>>_______________________________________________
>>Users mailing list
>>Users(a)openser.org
>>http://openser.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>>
>
>
>
Hi,
I am new user in ser, I still have a problem with SERWEB. I installed the serweb but it is not working in the browser. Can anybody tell me how to configure the serweb.
serweb location folder is : /var/www/html/serweb/
when i keep any .php or .html file in /var/www/html/ it woks , but the serweb is not accessible with command http://localhost/serweb/admin/index.php.
If anybody have the Idea about SERWEB configuration,plz let me know....
thanx
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Hi Everybody.
Just a small conformation about OpenSER and Softphone..
I'd installed openser and after so that install the sip softphone(X-Lite).
After the settings in X-lite (sip-proxy..domain ), then i make a call ...
But shows " Logging failed contact network admin " what's means this ?.
I think I need to write a script ( modules ) in openser.cfg file for
Register and authentication .
how to make a calls and what are the steps should take in openser.cfgfile...?
please help in this regarding and please excuse " in My English "..
Thanks and Regards....
Sunkara Ravi Prakash....:-)
Hello,every one:
I use ser-0.9.6.I have database authorize.I have accounts like
10000,10001 and so on.I want to send a real phone number (user difined) to
pstn.This means when user 1000 call a pstn number he can send his real
moblie phone number to the caller.(The caller can see this number).
What should i do.
Thanks a lot.
i just got it!
bash-3.00$ serctl alias add 2002 sip:bob@mylab.com
this allows calls from skinny ip phones (sccp) handled by a ccm to exit it's mtp/sip trunk then hit the sip proxy/registrar and have a way of resolution from number to name......the line 2 on my sip phone called "bob" rang
aaron
________________________________
From: serusers-bounces(a)lists.iptel.org on behalf of Gould, Aaron
Sent: Thu 4/27/2006 10:17 AM
To: serusers(a)lists.iptel.org
Subject: [Serusers] sccp to sip calling and vice versa
does anyone understand how i would dial from a sccp phone using letters so that i could dial a sip aor url/uri ?
i have a ccm 4.0(2)a configured with a sip trunk and mtp so that i can dial from sccp phones joined up to the ccm to sip phones that are joined up to the sip proxy/registrar that i have the ccm sip trunk pointing to
but i can only do this if the username portion of the sip aor uri is numbers........like .... sip:2001@mylab.com then from the sccp phone i dialed 2001 and the sip phone rings.
but what if my sip phone address is .... sip:bob@mylab.com ...how then do i dial the digits "bob" from the sccp phone?
aaron
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