Hi all
I wanted to know that which soft phone will work with SER.
waiting for reply
Thanks
Mukesh
---------------------------------
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Hello Users,
I'm tring to configuring the load balance ,
Initial i have 2 sip proxies and centralized database.
for 2 sip proxies database is common , ie. centralized.
Let Assume the proxy1 is a.b.c.b , other proxy2 is 1.2.3.4
and proxy domain name is sip.sample.com , is common
if the proxy1 is not started or some failure case , it has goes to the
second proxy2..
but this is not happening at all.
i follow this url for DNS_setting -- >
http://www.voip-info.org/wiki/view/DNS+SRV
all the request going to proxy1 only. but not the proxy2....
please any can guide me regarding this...
--
Thanks &Regards
Ravi Prakash Sunkara
Hi All
I would like to be able to have a difference Mediaproxy group selected
per user account on inbound DID calls.
Our termination provider sends the calls to our openSER with a URI of
something <number>@<IP_address>. '08723451234(a)202.xxx.xxx.xxx', we use
the alias_db_lookup("dbaliases") function to resolve the phone number to
a user account on the DID calls.
I would think that some type of rewrite of the '@202.xxx.xxx.xxx' would
need to be done but I'm sure this would cause other issues.
Any ideas would be great :)
Thanks
Mike
Hi,
I have SIP UA's register to my OpenSER server and they
can call each other. This has been working for some time
now without any real issues and I'm very happy.
Now I want to be able to have 1 destination be available
on 2 SIP UA's. What I think the best way to do is the
following:
- create two accounts
- make the detination point to one UA A and create an
"alias" for the destination at UA B (for instance by
prefixing A)
- On an incoming call detect that there is the
possibility to reach the destination at A and B
- Fork the incoming call to A and B so that when one
arm of the fork accepts the call (by going to ringing,
or immediately answering, etc).
My questions are:
- How do I check if I can start a fork?
- How do I fork and have both arms follow the logic I am
currently using?
--
Andreas Sikkema
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Hash: SHA1
Hi,
I played around with CPL-C Module. I loaded this CPL successfully via
MI-FIFO-Module into database:
<?xml version="1.0" encoding="UTF-8"?>
<cpl>
<outgoing>
<address-switch field="destination" subfield="user">
<address is="v111">
<reject status="603" reason="fhhhghh"/>
</address>
</address-switch>
</outgoing>
</cpl>
When a call is placed, the script above is executed and throws an error:
0(8741) DEBUG:cpl-c:get_orig_user: trying to get user from From
0(8741) DEBUG:get_user_script: fetching script for user <v101>
0(8741) DEBUG:get_user_script: we got the script len=62
0(8741) DEBUG:cpl_run_script: processing CPL node
0(8741) DEBUG:cpl_run_script: processing address-switch node
0(8741) DEBUG:run_address_switch: testing ADDRESS branch attr_name=0
attr_val=[v111](4)..
0(8741) DEBUG:run_address_switch: extracted uri is <sip:v111@domain>
0(8741) DEBUG:run_address_switch: extracted val. is <v111>
0(8741) DEBUG:run_address_switch: matching on ADDRESS node (IS)
0(8741) DEBUG:cpl_run_script: processing reject node
0(8741) ERROR:cpl_c:cpl_run.c:553: attribute is an empty string
0(8741) ERROR:cpl_c:cpl_run_script: script error
When I change that reject line to this:
<reject status="reject"/>
then it works fine. Obviously openser (1.2.0) has some problems with
parsing reject's reason attribute ...
Is there a fix around or did I do a mistake somewhere?
regards
helmut
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Hello,every one
I configure snmpstats modules in openser1.2.0 and the net-snmp version is
5.4.But
when I¡¡use openser -c command its result as follows:
ERROR: load_module: could not open module
</usr/local/lib64/openser/modules/snmpstats.so>:
/usr/local/lib64/openser/modules/snmpstats.so: undefined symbol: real_init_smux
Who can give me some suggestion? Thank you very much.
Hi all -
Not really sure where to post this question as I am just starting to
research this issue.
We'd like an all VOIP solution where we have a telephone number that
terminates at our server.
At our server, we'd like to:
1. get their phone number via caller ID. look up data with the caller id.
2. generate a wave file based on the data returned & play it to the
user over the established voip link.
How is this done? Totally new to the game here.
I've read about DID origination, SIP channels, SIP peers... its all
quite confusing!
I'm a java developer, so something that works via Java would be great,
but scripting in various languages is no big deal. The application
doesn't need to do much more than query a database & choose a WAV file
to play back based on that info.
Thank you--
Matt Pease
ParkingHero, Inc.
On 06/05/2007 05:17 PM, raviprakash sunkara wrote:
> Hi Alexander
> Can use fix_nated_sdp("flag"); and also force_rtp_proxy("flags");
I'm already using force_rtp_proxy() (without arguments) and it already
rewrites the "c=" address in the sdp information but it uses the wrong
interface address. (The openser-server/rtpproxy has two IP adresses in
the same subnet.)
Since I know the address which is used by the clients or contacting the
server, I could hardcode the address as the second parameter to
force_rtp_proxy to force the use of this address but I was looking for a
cleaner solution.
I'd like to use the address where the SIP request came in for both RTP
streams. Maybe I can use the pseudo-variable $Ri (Received IP address)
as the second parameter to force_rtp_proxy? This should force the use of
the correct IP address in the "c=" field and will hopefully persuade
the client to use this IP address as the destination address for its RTP
data. But how do I tell rtpproxy to send its RTP data from this address?
Thanks,
--leo
> On 6/5/07, Alexander Bergolth <leo(a)strike.wu-wien.ac.at> wrote:
>>
>> Hi!
>>
>> I'm experiencing problems using openser and rtpproxy on a multihomed box
>> (IP addresses are 137.208.89.45 and 137.208.89.65).
>>
>> The client (firewalled) contacts the server via the first IP address
>> (137.208.89.45) but rtpproxy sometimes uses it's second IP
>> (137.208.89.65) in the "c=" SDP-information. (See below.)
>> However it _sends_ RTP packets from the _first_ IP address. Those
>> packets will never pass through the firewall, of course, since the
>> clients outgoing RTP traffic is sent to another IP address than the
>> incoming traffic is hitting the firewall from.
>>
>> Is there a way to tell rtpproxy to use the same IP-address both for the
>> c= SDP header and as source address?
>>
>> -------------------- 8< --------------------
>> ##
>> T 2007/06/05 15:57:36.041537 137.208.89.45:5060 ->
>> 137.208.90.164:3099[AP]
>> SIP/2.0 183 Session Progress.
>> Via: SIP/2.0/TCP
>> 137.208.90.164:56136
>> ;branch=z9hG4bK-d87543-a107a77a6311422d-1--d87543-;rport=3099.
>> From: "Alexander Bergolth"<sip:30001@wu-wien.ac.at>;tag=5b6d7b31.
>> To: "996050"<sip:996050@wu-wien.ac.at>;tag=as2e40807f.
>> Call-ID: MTVjYzlhMDdjMjAwODExZDEyNzk5M2IwN2UxNjhlYTc..
>> CSeq: 2 INVITE.
>> User-Agent: Asterisk PBX.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
>> Supported: replaces.
>> Contact: <sip:996050@137.208.89.135>.
>> Content-Type: application/sdp.
>> Content-Length: 283.
>> .
>> v=0.
>> o=root 6978 6978 IN IP4 137.208.89.135.
>> s=session.
>> c=IN IP4 137.208.89.65.
>> t=0 0.
>> m=audio 35064 RTP/AVP 0 8 101.
>> a=rtpmap:0 PCMU/8000.
>> a=rtpmap:8 PCMA/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=silenceSupp:off - - - -.
>> a=ptime:20.
>> a=sendrecv.
>> a=nortpproxy:yes.
>> -------------------- 8< --------------------
>>
>> -------------------- 8< --------------------
>> 15:57:35.814136 IP 137.208.90.164.3099 > 137.208.89.45.5060: P
>> 4236850032:4236851080(1048) ack 224540451 win 64256
>> 15:57:35.814147 IP 137.208.89.45.5060 > 137.208.90.164.3099: . ack 1048
>> win 2003
>> 15:57:35.814351 IP 137.208.89.45.5060 > 137.208.90.164.3099: P
>> 1:550(549) ack 1048 win 2003
>> 15:57:35.814430 IP 137.208.89.45.5060 > 137.208.90.164.3099: P
>> 550:1264(714) ack 1048 win 2003
>> 15:57:35.814850 IP 137.208.90.164.3099 > 137.208.89.45.5060: . ack 1264
>> win 64177
>> 15:57:35.915829 IP 137.208.90.164.3099 > 137.208.89.45.5060: P
>> 1048:1418(370) ack 1264 win 64177
>> 15:57:35.955948 IP 137.208.89.45.5060 > 137.208.90.164.3099: . ack 1418
>> win 2003
>> 15:57:36.017309 IP 137.208.90.164.3099 > 137.208.89.45.5060: P
>> 1418:2672(1254) ack 1264 win 64177
>> 15:57:36.017320 IP 137.208.89.45.5060 > 137.208.90.164.3099: . ack 2672
>> win 2003
>> 15:57:36.017397 IP 137.208.89.45.5060 > 137.208.90.164.3099: P
>> 1264:1813(549) ack 2672 win 2003
>> 15:57:36.018604 IP 137.208.89.45.5060 > 137.208.90.164.3099: P
>> 1813:2388(575) ack 2672 win 2003
>> 15:57:36.019170 IP 137.208.90.164.3099 > 137.208.89.45.5060: . ack 2388
>> win 64256
>> 15:57:36.041454 IP 137.208.89.45.35064 > 137.208.90.164.10676: UDP,
>> length 172
>> 15:57:36.041537 IP 137.208.89.45.5060 > 137.208.90.164.3099: P
>> 2388:3200(812) ack 2672 win 2003
>> 15:57:36.046899 IP 137.208.90.164.10677 > 137.208.89.65.35065: UDP,
>> length 132
>> 15:57:36.061155 IP 137.208.89.45.35064 > 137.208.90.164.10676: UDP,
>> length 172
>> 15:57:36.080128 IP 137.208.90.164.10676 > 137.208.89.65.35064: UDP,
>> length 172
>> 15:57:36.081157 IP 137.208.89.45.35064 > 137.208.90.164.10676: UDP,
>> length 172
>> [...]
>> -------------------- 8< --------------------
>>
>> Thanks in advance,
>> --leo
>>
>> P.S.: I'm using openser-1.2 and rtpproxy-cvs20070222.
>>
>> --
>> e-mail ::: Alexander.Bergolth (at) wu-wien.ac.at
>> fax ::: +43-1-31336-906050
>> location ::: Computer Center | Vienna University of Economics | Austria
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users(a)openser.org
>> http://openser.org/cgi-bin/mailman/listinfo/users
>>
>
>
>
--
e-mail ::: Alexander.Bergolth (at) wu-wien.ac.at
fax ::: +43-1-31336-906050
location ::: Computer Center | Vienna University of Economics | Austria
