Hi,
I am trying to use dbaccounting with mediaproxy. Undfortunately it doesn't seem to send anything to mysql db.
How can I trace where the problem is ?
My mediaproxy accounting config is :
[Accounting]
dbaccounting = on
[Database]
user = ser
password = heslo
host = 127.0.0.1
database = radius
table = radacct
Thanks,
ilker
<http://387555.sigclick.mailinfo.com/sigclick/060F0F04/0D034500/0403014D/0A1…>
_____________________________________________________________________________________________________________________________________________
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This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence.
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Hi Users,
I'm cracking my head with audio problem..
I installed openser 1.0.1 in Linux + freeRadius + RtpProxy + mysql
When I make a call ( which are behind the nat ) the audio is not hearing
and audio signal of Softphone is blinking up and down..
In Debug output I didn't see any
SIP reply or sip request for ACK..
I thing so that, If client nehind the NAT didn't receive the ACK. this may
be problem...
SIP/2.0 200 Ok
v: SIP/2.0/UDP 192.168.2.21:5060;received=61.17.248.68
;rport=4007;branch=z9hG4bK9615579CC17F47929BB2FEF3F33F4BFC
f: ravi <sip:1003@24.98.157.209>;tag=145161195
t: <sip:104@24.98.157.209>;tag=641313084
m: <sip:104@61.17.248.68:5060>
Record-Route: <sip:192.168.2.4;ftag=145161195;lr=on>
i: 839BBA5C-84B5-4D09-ACF1-B19FA0EB3C59(a)192.168.2.21
CSeq: 54483 INVITE
c: application/sdp
Server: X-Lite release 1105x
Content-Length: 166
v=0
o=104 86794725 86799157 IN IP4 192.168.2.20
s=X-Lite
c=IN IP4 192.168.2.4
t=0 0
m=audio 35034 RTP/AVP 1
a=rtpmap:1 pcmu/8000
a=sendrecv
a=nortpproxy:yes --------------------- > this is problem
openSER server didn't get request ACK I think So..
What may the problem... plz help me
--
Thanks and Regards w ith
cheers
Sunkara Ravi Prakash (Voip
Developer)
Hyperion
Technology
www.hyperion-tech.com
I'm not very familiar with freeradius, but in general there is only one
dictionary, sometimes with some sort of #include-support if you want to
split up the dictionary.
g-)
Kostas Marneris wrote:
> thanks for your help.
> I hadn't understand all the Radius issues..
>
> I should add the following line on FreeRadius Dictionary .
> ATTRIBUTE Sip-Rpid 213 string # Proprietary, auth_radius
>
> Is this the proper way or should I have a separate dictionary file ?
>
>
> thank you.
>
>
> Greger V. Teigre wrote:
>
>> The SIP-Rpid attribute will be returned as part of the RADIUS auth if
>> present (0.9.x). In 0.10.x I believe it is SIP-AVP=rpid:value
>> This is the "saved value".
>> g-)
>>
>> Kostas Marneris wrote:
>>
>>
>>> Hello all,
>>>
>>> I'm trying to get 'caller-id with Remote-Party-ID' work,
>>> but I think I'm missing something..
>>>
>>> What is the "saved value of the SIP URI received from radius" ?
>>> (according to auth/README file).
>>>
>>>
>>> What is supposed to be the RADIUS attribute/value in RADIUS user profile ?
>>>
>>>
>>> Can someone send an example ?
>>>
>>>
>>> thank you.
>>> _______________________________________________
>>> Serusers mailing list
>>> Serusers(a)lists.iptel.org
>>> http://lists.iptel.org/mailman/listinfo/serusers
>>>
>>>
>>>
>>>
>>
>>
>
>
Hello Ilker,
I am writing to address of the sip message to fifo and from the fifo call sipp instead of mysql.
But , I do not now how to send message from sipp when the user is online.
can you please help in sending message from sipp to user agent(to address).
Thanks & Regards
Sriram Srinivas
________________________________
From: İlker Aktuna (Koç.net) [mailto:ilkera@koc.net]
Sent: Monday, June 19, 2006 7:06 PM
To: Srinivas Sriram (WT01 - IP-Multimedia Carrier & Ent Networks)
Subject: RE: [Serusers] How to send instant message programmatically
Hi Sriram,
I am also interested in sending missed calls as IM when the user becomes online.
Do you know how to write missed calls to a mysql table ?
If that's done, we can send messages via sipp when the user is online.
Please share with me your findings...
Regards,
ilker
________________________________
From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of srinivas.sriram(a)wipro.com
Sent: Monday, June 19, 2006 3:37 PM
To: serusers(a)iptel.org
Subject: RE: [Serusers] How to send instant message programmatically
Hi Greger
I have checked with sipp , with that we can send messages. But What is my requirement is i have write the code for constructing SIP "MESSAGE" message for a given sip uri. This I have to call form one of the ser module.
Total what I am trying is to send a IM message when there is a missed call for a particular SIP uri. In SER there is a code for writing into the data base of the sip uri , when there is a missed call for that Uri. At this point we have to send IM message.
Thank you,
Regards,
Sriram Srinivas.
________________________________
From: Greger V. Teigre [mailto:greger@teigre.com]
Sent: Monday, June 19, 2006 5:35 PM
To: Srinivas Sriram (WT01 - IP-Multimedia Carrier & Ent Networks)
Cc: serusers(a)iptel.org
Subject: Re: [Serusers] How to send instant message programmatically
You are better off generating the message outside SER. SER is not a UAS. Have a look at sipsak or sipp (both found at sourceforge, I think).
g-)
srinivas.sriram(a)wipro.com wrote:
Hello,
I am using SER-0.9.6. How to send instant message programmatically, how to build sip message and which function can by used(like t_uac, t_request).
Can you help where and which to be changed in ser code.
Thank you,
Regards,
Sriram Srinivas.
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________________________________
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_____________________________________________________________________________________________________________________________________________
Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez.
This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence.
_____________________________________________________________________________________________________________________________________________
The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments.
WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email.
www.wipro.com
Hi List
I want to get to c= and m= value after use_media_proxy or force_rtp_proxy
(after real RTP flow between UA and media/rtpproxy)
I want this in order to steal this pair of information, and bypass the
RTPProxy/MediaProxy and use this information for UA to talk with other
application server (eg. Asterisk)
Pls. help
Brgds
Hello, I was wondering, how should I implement this. I would like the user to press the #key first if they would like to dial SIP 2 SIP calls and just leave PSTN calls as it is? How am I suppose to do that?
I have this problem too, which whenever I hangup, the other side doesn't gets disconnected. What should I check for to enable hang up on both sides if one sides suddenly hangs up? Thank you.
---------------------------------
Want to be your own boss? Learn how on Yahoo! Small Business.
Thank you; that was what I'm looking for.
How can I match i.e IP block 10.224.0.0 - 10.255.255.255 ?
Is there an easy way or should I list them with || as in :
(uri=~"^.*@10\.224\..* <mailto:uri=~^.*@10\.224\..*> " || uri=~"^.*@10\.225\..*" || .....)
Thanks,
ilker
________________________________
From: Greger V. Teigre [mailto:greger@teigre.com]
Sent: Wednesday, June 21, 2006 12:50 PM
To: İlker Aktuna (Koç.net)
Cc: serusers(a)iptel.org
Subject: Re: [Serusers] checking source ip and dest ip
Check the request uri after lookup, right before t_relay(). (uri=~"^.*@10\..*" <mailto:^.*@10\..*> ) As long as you don't have nated clients (and thus don't use fix_nated_register()) the request uri will contain the actual IP address.
g-)
İlker Aktuna (Koç.net) wrote:
Hi,
In fact my network is not 10.0.0.0 ; that was just an example (bad example).
The network addresses are specific public network addresses (lets say 3 different C Class network) and has no relation to NAT.
That's why I need to check the IP adrresses (from and to) of the SIP message before deciding if it should use mediaproxy or not.
Any idea how to do that ?
Soruce address can be checked as :
if (src_ip="195.1.1.+") {
}
but that won't be same for destination address as "dst_ip" is the SER server ip.
How can I check the ip address of the client that the SIP message will be forwarded ?
Thanks,
ilker
________________________________
From: Greger V. Teigre [mailto:greger@teigre.com]
Sent: Tuesday, June 20, 2006 10:26 PM
To: İlker Aktuna (Koç.net)
Cc: serusers(a)iptel.org
Subject: Re: [Serusers] checking source ip and dest ip
Yes, and your network is in the RFC1918 (private) address space, so you can use the same logic (even without change as you want to proxy when the address is on your network).
g-)
İlker Aktuna (Koç.net) wrote:
In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not.
what do you mean by RFC1918 addresses ?
Thanks,
ilker
________________________________
From: Greger V. Teigre [mailto:greger@teigre.com]
Sent: Tuesday, June 20, 2006 1:40 PM
To: İlker Aktuna (Koç.net)
Cc: serusers(a)iptel.org
Subject: Re: [Serusers] checking source ip and dest ip
Why not test for RFC1918 addresses? That's what the onsip.org nat-handling scripts do. A NATed destination is determined by a flag in the location record and the src ip is checked with nat_uac_test(). It will give a more robust NAT handling that testing on IP addresses for parts of the RFC1918 addresses.
The problem has already been solved, why not reuse? ;-)
g-)
İlker Aktuna (Koç.net) wrote:
Hi,
I need to check source ip of a SIP message and also destination IP to which the message will be sent.
i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0
is it possible ?
Thanks,
ilker
C "-//W3C//DTD HTML 4.0 Transitional//EN">
In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not.
what do you mean by RFC1918 addresses ?
Thanks,
ilker
________________________________
From: Greger V. Teigre [mailto:greger@teigre.com]
Sent: Tuesday, June 20, 2006 1:40 PM
To: İlker Aktuna (Koç.net)
Cc: serusers(a)iptel.org
Subject: Re: [Serusers] checking source ip and dest ip
Why not test for RFC1918 addresses? That's what the onsip.org nat-handling scripts do. A NATed destination is determined by a flag in the location record and the src ip is checked with nat_uac_test(). It will give a more robust NAT handling that testing on IP addresses for parts of the RFC1918 addresses.
The problem has already been solved, why not reuse? ;-)
g-)
İlker Aktuna (Koç.net) wrote:
Hi,
I need to check source ip of a SIP message and also destination IP to which the message will be sent.
i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0
is it possible ?
Thanks,
ilker
C "-//W3C//DTD HTML 4.0 Transitional//EN">
Hi,
In fact my network is not 10.0.0.0 ; that was just an example (bad example).
The network addresses are specific public network addresses (lets say 3 different C Class network) and has no relation to NAT.
That's why I need to check the IP adrresses (from and to) of the SIP message before deciding if it should use mediaproxy or not.
Any idea how to do that ?
Soruce address can be checked as :
if (src_ip="195.1.1.+") {
}
but that won't be same for destination address as "dst_ip" is the SER server ip.
How can I check the ip address of the client that the SIP message will be forwarded ?
Thanks,
ilker
________________________________
From: Greger V. Teigre [mailto:greger@teigre.com]
Sent: Tuesday, June 20, 2006 10:26 PM
To: İlker Aktuna (Koç.net)
Cc: serusers(a)iptel.org
Subject: Re: [Serusers] checking source ip and dest ip
Yes, and your network is in the RFC1918 (private) address space, so you can use the same logic (even without change as you want to proxy when the address is on your network).
g-)
İlker Aktuna (Koç.net) wrote:
In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not.
what do you mean by RFC1918 addresses ?
Thanks,
ilker
________________________________
From: Greger V. Teigre [mailto:greger@teigre.com]
Sent: Tuesday, June 20, 2006 1:40 PM
To: İlker Aktuna (Koç.net)
Cc: serusers(a)iptel.org
Subject: Re: [Serusers] checking source ip and dest ip
Why not test for RFC1918 addresses? That's what the onsip.org nat-handling scripts do. A NATed destination is determined by a flag in the location record and the src ip is checked with nat_uac_test(). It will give a more robust NAT handling that testing on IP addresses for parts of the RFC1918 addresses.
The problem has already been solved, why not reuse? ;-)
g-)
İlker Aktuna (Koç.net) wrote:
Hi,
I need to check source ip of a SIP message and also destination IP to which the message will be sent.
i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0
is it possible ?
Thanks,
ilker
C "-//W3C//DTD HTML 4.0 Transitional//EN">
In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not.
what do you mean by RFC1918 addresses ?
Thanks,
ilker
<http://387555.sigclick.mailinfo.com/sigclick/070C0F04/02074D00/0507084D/051…>
_____________________________________________________________________________________________________________________________________________
Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez.
This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence.
_____________________________________________________________________________________________________________________________________________
hi user...
What means " t_newtrans :- transaction already in process " and " I'm
terribly sorry, server error occurred ".
How to resovle this....
Please help me...
--
Thanks and Regards with cheers
Sunkara Ravi Prakash (Voip Developer)
Hyperion Technology
www.hyperion-tech.com
<b>
Hello to everybody,
I would like to say.... Happy Birhtday Openser!!!! :O)
Sometimes I got answers with delay, sometimes I spent nights trying to understand how to configure a module, or even a single statement, somrtimes I needed to read ten times the same document because it was unclear related to my knolwledge and still I want to thank the community for this great product.
Every day I try to read all the emails coming from the community, and every day I learn a little bit more (threads, ideas, suggestions,solutions...).
By the way, there is any open source for a sip softphone in windows (minisip even doesnt start on my laptop with windows 2000) :o))
thanks
Alessandro
Can someone please clarify whether the statement in
Subject is correct? Does mediaproxy not simply take a
RTP stream and "proxy" it in both directions without
really caring about codecs? Codec negotiation is done
between endpoints and mediaproxy does not come in the
picture.
If the above is true, does it mean that if two SIP
user agents support video and we use mediaproxy, it
won't be an issue?
Please educate me the what if any relationship
mediaproxy has with codecs...
Dave
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