Re: [OpenSIPS-Users] Mediaproxy
On Feb 16, 2009, at 4:14 AM, ram wrote: Hi Same thing i have complained to maintainer its been 10days no mail from him we need to just wait until we hear from them Ram On Sat, Feb 14, 2009 at 10:46 PM, NYam nya...@yahoo.com wrote: Dear Sir/Madam, I have following issue and need to address to you. I have operating system-debian R7, which installed kamailio 1.4. When i tried to install mediaproxy-2.3.2.tar.gz, it does not match because python should be equal or higher than 2.5. A In the documentation is stated that you need Python 2.4. Where do you see that it needs at least Python 2.5? Adrian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Users Digest, Vol 7, Issue 77
kishore kumar kishoreinbangal...@gmail.com wrote: Hi , I installed the software dependencies regarding the openxcap but when I am trying to run the openxcap server it is throwing the following error. /opt/python2.5/bin/openxcap start Traceback (most recent call last): File /opt/python2.5/bin/openxcap, line 41, in module from xcap.logutil import start_log File /opt/python2.5/lib/python2.5/site-packages/xcap/logutil.py, line 4, in module from twisted.web2 import responsecode ImportError: No module named web2 I installed twisted-web2 also Please help me in sorting out this error. I am eagerly waiting for reply Thanks, Kishore. Why do you start openxcap like that? 'openxcap' script does not have start command, it's /etc/init.d/openxcap that does. Regarding your question, if you have installed twisted-web2 successfully, this should work: $ python2.5 -c import twisted.web2 Until that works, it's a problem with twisted-web2 installation but not with openxcap. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] R: Milliseconds in the accounting table
Hi Bogdan, I modified the script as you said but in the DB there is always the same problem. I used an avp variable to store the time value in the onreply_route function, now I invoke the t_on_reply()function before the INVITE t_relay() function. When the server received the 200 OK (onreply_route function), I see the correct time value in the log file, but the stored value on the DB is different. It is the time taken at the INVITE received time...I Think... Any suggestions?? Thanks a lot MD -Messaggio originale- Da: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Inviato: sabato 14 febbraio 2009 22:22 A: Mauro Davi' Cc: users@lists.opensips.org Oggetto: Re: [OpenSIPS-Users] Milliseconds in the accounting table Hi Mauro, The time for accounting is when the reply is received - for the acc'ed INVITEs, it is the time for the 200 OK reply. So, what you can do is to use onreply_route to store in a $var(x) the string with the time representation you want and set $var(x) as extra acc; Regards, Bogdan Mauro Davi' wrote: Hi, I'm always a newbye so be patient. I need to trace in the accounting table the start/stop dialog time in milliseconds. I don't know if this is the correct way, but I modified the cfgutils. Now I can write $time(msec) and I receive the millisecs... So I store this information in an avp variable that I store on the DB in the acc table via the multi_leg_info parameter. Obviusly I store the entire date time in the form $time(year)/$time(mon)/$time(mday) $time(hour):$time(min):$time(sec).$time(msec). Problem: The date time stored for the INVITE message, with this method differ to the time writed in the time field on the same acc table... I think that the time saved in the time field is that of the received ACK message... On the otherhand the time saved with the multi_leg_info is the time of the first INVITE... I need the correct date time with millisecs of a dialog start/stop, so the time field in the accounting table isn't good enough... Any suggestion to bypass this problem? Thanks in advance MD ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Installing a Particular module using source
hi all, I have installed opensips-1.4.4 using debian packages. I have install presence and xmlrpc packages for presence support. I need RLs support too. which debian package should install to get rls support. Or can i install only rls module using source of opensips. If yes how can i do that -- Thanking You, Ashwini BR Naidu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] how to do opensips' register replication
Hello Tony, I didn't get why do you want to rewrite REALM and To/From header
filed, you must have same REALM on both opensips.
I send you the way to share registration on both server. Put same code
inside REGISTER method and replace LOCAL_IP, PAIR_IP with your real
information
When one OpenSIPS receive a REGISTER save it in memory and database and then
send it to the other pair OpenSIPS.
...
# Register method
if ( !(src_ip==PAIR_IP src_port==5060) ) {
# Register is for me
if (!save(location)) {
sl_reply_error();
exit;
}
add_sock_hdr(Local-Sock);
force_send_socket( LOCAL_IP);
t_replicate(sip:PAIR_IP:5060); # send to the other sip server
} else {
# it's a replicated REGISTER
# set the flag for retrieving sock_info from header
# save contact, but only in cache
save(location,0x01);
};
Best,
Gustavo
On Mon, Feb 16, 2009 at 6:06 AM, Tony Liao tonyl...@yuehetone.com wrote:
hi all,
we have two opensips sever as proxy and register server.for handling
the user registered in another proxy server,we want to do the register
replication as bellow:
uac register in opensipsA( OSA),OSA should rewrite the uri
(OSA_IP-OSB_IP),Authorization relm(OSA_IP-OSB_IP) and To/From header
filed. is anyone have codes and success to excute?
thanks.
**
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--
Gustavo Mistrinelli
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Re: [OpenSIPS-Users] OpenXCAP-functionality
Hi, Neither the latest openxcap nor the latest python-xcaplib have class named HTTPConnectionWrapper, so simply upgrading both should fix this. JayaPrakash jp.man...@gmail.com wrote: Hi All, OpenXCAP is installed in my Debian 4.1.1 machine. XCAP Server, Openser-mi-porxy and Soap-simple-proxy are running. However, when the functionality of OpenXCAP is tested with testsuite, it has given the following error. # ./test.py Traceback (most recent call last): File ./test.py, line 7, in module from common import * File /usr/local/lib/python2.5/site-packages/xcap/test/common.py, line 47, in module class HTTPConnectionWrapper(xcaplib.client.HTTPConnectionWrapper): AttributeError: 'module' object has no attribute '*HTTPConnectionWrapper*'. Will you please suggest me, how to fix this issue. Thanks Regards, JayaPrakash.Manchu. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Adtran VQM data into Opensips/CDRTool
Hello, We run a number of Adtran TA900-series gateways in our network. Recent software versions have the ability to do voice quality measurements, and output that data at the end of a call through a PUBLISH event. Normally one would configure these devices to talk to an Adtran n-Command MSP data collector. Because this PUBLISH happens after the call is disconnected, it seems an update would have to occur after the fact, perhaps similar to a Mediaproxy info update. In the example below that Adtran provided to me, the CallID field is unknown¹. I¹m waiting on a response from Adtran engineering to see if that is always the case. If so, it¹s going to be more difficult to match the call. The following is from the perspective of the Adtran unit: Tx: TCP src=10.19.209.11:5060 dst=10.100.13.250:5060 PUBLISH sip:collec...@10.100.13.250 SIP/2.0 From: sip:lbadtn0816ae...@10.19.209.11;tag=4afcca8-0-13c4-3954f-f39d4b80-3954f To: sip:collec...@10.100.13.250 Call-ID: 4afcca8-0-13c4-3954f-e9ebf5f2-39...@10.19.209.11 CSeq: 1 PUBLISH Via: SIP/2.0/TCP 10.19.209.11:5060;branch=z9hG4bK-3954f-dff3d63-6155485b Event: vq-rtcpxr Subscription-State: active Content-Type: application/vq-rtcpxr Content-Length: 1263 VQSessionReport LocalMetrics: Timestamps:START=2008-12-18T17:55:01Z STOP=2008-12-18T17:55:10Z SessionDesc:PT=0 PC=1 CallID:unknown LocalAddr:IP=10.19.209.55 PORT=3004 SSRC=22b4624f RemoteAddr:IP=10.19.209.49 PORT=1 SSRC=22b4624f JitterBuffer: JBRSYNC=5 JBP=463 JBPOO=0 JBPD=0 JBPE=37 JBPL=425 JBRC=28 JBENVD=1 JBENVP=0 JBENVPMIN=0 JBENVPM=4 JBENVN=0 JBENVNMIN=0 JBENVNM=0 JBLT=50.0 JBLTP=100.00 JBLUT=11 JBL=11 JBLPJ=2.0 JBET=10.0 JBETP=100.00 JBE=451 JBEPJ=0.0 JBT=1 JBCDMIN=10 JBCDN=50 JBCDM=100 JBDINC=0 JBDDEC=3 JBD=47 JBDI=35 JBDMIN=35 JBDM=50 PacketLoss:NLR=0.00 JDR=0.00 LR=0.00 JL=0 JD=0 JOD=0 JUD=0 BurstGapLoss:BLD=0.00 BD=0 GLD=0.00 GD=8640 BPD=0 BC=0 BE=0 GPD=432 GC=1 GE=0 Delay:RTD=1 ESD=65 OWD=63 IAJ=451 RTDI=1 RTDM=1 ESDMIN=55 ESDM=70 OWDI=58 OWDM=65 LD=0 LDMIN=0 LDM=2 PPDV=0.3 PDV=0 PDVM=2 PDVMN=0.0 PDVMNI=0.4 PDVMNM=0.0 PDVMNAM=2.0 Signal: QualityEst:RLQ=93 RCQ=92 MOSLQ=4.20 MOSCQ=4.18 BRLQ=92 GRLQ=92 RN=93 RG107=92 MOSPQ=4.45 MOSN=4.20 QL=0 MetricsVersion:MT=ADTRAN MV=01.00 DeviceSerialNum:LBADTN0816AE392 CCMID:39 DSCP:46 LocalCaller:true LocalURI:8249 LocalIfc:4 RemoteURI:8884238726 RemoteIfc:0 Degradation:DL=0.00 DDISC=0.00 DV=0.00 DR=0.00 DD=0.00 DSL=0.00 DNL=0.00 DEL=1.42 Data:TI=35000 RI=100 BR=64000 Rx: TCP src=10.100.13.250:5060 dst=10.19.209.11:5060 SIP/2.0 200 Ok Content-Length: 0 From: sip:lbadtn0816ae...@10.19.209.11;tag=4afcca8-0-13c4-3954f-f39d4b80-3954f Call-ID: 4afcca8-0-13c4-3954f-e9ebf5f2-39...@10.19.209.11 CSeq: 1 PUBLISH Via: SIP/2.0/TCP 10.19.209.11:5060;branch=z9hG4bK-3954f-dff3d63-6155485b To: sip:collec...@10.100.13.250;tag=482617583 Allow: OPTIONS, PUBLISH Allow-Events: vq-rtcpxr SIP-ETag: 1229622954690.781 Expires: 0 First real question: what¹s the best way to get the ³VQSessionReport² data out of this packet? Some of the data is useful (particularly the MOSPQ value on the QualityEst line), much of it is not. Unfortunately I don¹t know anything about the normal uses for PUBLISH. Assuming one can extract this useful information from this and match it to an existing call in radius and push the useful information into the RTPStatistics field, what would CDRTool do with it? Thanks, Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] routing module
I am using opensips-1.4.3-notls I need to route calls to differents remote gateway based on prefix in the dialed number. Can you tell me if the module carrierreoute, would be the recommended solution ? 1- README-MODULES file does not list carrierroute module 2- carrierroute does not compile error : route_config.c:30:21: error: confuse.h: No such file or directory I would like to know if I am looking for the most strait-forward/supported solution to meet my requirements ? Thank you, Julien From: users-boun...@lists.opensips.org on behalf of Bogdan-Andrei Iancu Sent: Thu 22/01/2009 3:20 PM To: ibrahim tunali Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] drouting module - dr_gateways table attrs field Hi Ibrahim, have you set the attr avp? Something like: modparam(drouting, attrs_avp, '$avp(s:dr_attrs)') Then after do_routing() or use_next_gw(), do : xlog(-gw attr is $avp(s:dr_attrs)\n); the value of the attr is whatever you want - the module does not interpret it - it is just reading it from DB and pass it to the AVP when you use the GW. It is your decision what to put there and how to use the value. Regards, Bogdan ibrahim tunali wrote: Hello, I'm playing with the new module drouting on svn trunk and i need to get which gateway is used on last request. I might be able to get it with attrs_avp and attrs field on dr_gateways table, i guess. I try some values to attrs but opensips crashed. Could you give an example to use attrs_avp and what is the value format of attrs fields. Regards, Ibrahim TUNALI ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] routing module
About the error : route_config.c:30:21: error: confuse.h: No such file or directory I found that there was a module depency clearly stated in the doc : The following libraries or applications must be installed before running OpenSIPS with this module loaded: * libconfuse, a configuration file parser library. For my main question, I will read about the new dynamic routing in 1.5.x But a good advice could be much appreciated ? From: users-boun...@lists.opensips.org on behalf of Julien Chavanton Sent: Mon 16/02/2009 9:10 PM To: users@lists.opensips.org Subject: [OpenSIPS-Users] routing module I am using opensips-1.4.3-notls I need to route calls to differents remote gateway based on prefix in the dialed number. Can you tell me if the module carrierreoute, would be the recommended solution ? 1- README-MODULES file does not list carrierroute module 2- carrierroute does not compile error : route_config.c:30:21: error: confuse.h: No such file or directory I would like to know if I am looking for the most strait-forward/supported solution to meet my requirements ? Thank you, Julien ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] pcapsipdump with no RTP
Hi, I have been looking at pcapsipdump which creates a separate pcap file for each call passing through your opensips server, so you can go back and debug problems after they have happened. However, if you are trying to use this on a media server, it dumps RTP and does not seem to have an option to turn it off? Is anyone using this tool and has a way of not dumping RTP? My thinking is to dump these files for a couple of days then delete them and continually cycle these files only dumping sip messages. Cheers Robert ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] routing module
Hi Julien, Julien Chavanton wrote: I am using opensips-1.4.3-notls I need to route calls to differents remote gateway based on prefix in the dialed number. Can you tell me if the module carrierreoute, would be the recommended solution ? yes, carrierroute and lcr modules are the one for doing prefix based routing. 1- README-MODULES file does not list carrierroute module probably not updated :), but check : http://www.opensips.org/index.php?n=Resources.DocsModules14 2- carrierroute does not compile error : route_config.c:30:21: error: confuse.h: No such file or directory you need to install libconfuse package. See: http://www.opensips.org/html/docs/modules/1.4.x/carrierroute.html#id227214 Regards, bogdan I would like to know if I am looking for the most strait-forward/supported solution to meet my requirements ? Thank you, Julien *From:* users-boun...@lists.opensips.org on behalf of Bogdan-Andrei Iancu *Sent:* Thu 22/01/2009 3:20 PM *To:* ibrahim tunali *Cc:* users@lists.opensips.org *Subject:* Re: [OpenSIPS-Users] drouting module - dr_gateways table attrs field Hi Ibrahim, have you set the attr avp? Something like: modparam(drouting, attrs_avp, '$avp(s:dr_attrs)') Then after do_routing() or use_next_gw(), do : xlog(-gw attr is $avp(s:dr_attrs)\n); the value of the attr is whatever you want - the module does not interpret it - it is just reading it from DB and pass it to the AVP when you use the GW. It is your decision what to put there and how to use the value. Regards, Bogdan ibrahim tunali wrote: Hello, I'm playing with the new module drouting on svn trunk and i need to get which gateway is used on last request. I might be able to get it with attrs_avp and attrs field on dr_gateways table, i guess. I try some values to attrs but opensips crashed. Could you give an example to use attrs_avp and what is the value format of attrs fields. Regards, Ibrahim TUNALI ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] routing module
Hi Julien, Personally, I would recommend the Dynamic Routing module, mainly because of performance issues. see: http://www.opensips.org/html/docs/modules/devel/drouting.html#id227252 Regards, Bogdan Julien Chavanton wrote: For my main question, I will read about the new dynamic routing in 1.5.x But a good advice could be much appreciated ? *From:* users-boun...@lists.opensips.org on behalf of Julien Chavanton *Sent:* Mon 16/02/2009 9:10 PM *To:* users@lists.opensips.org *Subject:* [OpenSIPS-Users] routing module I am using opensips-1.4.3-notls I need to route calls to differents remote gateway based on prefix in the dialed number. Can you tell me if the module carrierreoute, would be the recommended solution ? 1- README-MODULES file does not list carrierroute module 2- carrierroute does not compile error : route_config.c:30:21: error: confuse.h: No such file or directory I would like to know if I am looking for the most strait-forward/supported solution to meet my requirements ? Thank you, Julien ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT
Hi Julian, You can still handle the NAT wih COMEDIA even for T.38, but you have to handle the re-INVITE also . In your scenario, who is generating the re-INVITE? Regards, Bogdan Julian Yap wrote: The full story is that I was looking to get T.38 working behind NAT. Unfortunately, no matter what I tried, it wouldn't work behind NAT. I had the initial INVITE (G.711) working fine but when there was the T.38 re-INVITE, the RTP media would connect up fine but just wouldn't negotiate properly with T.38. Very strange as it worked fine with the UA behind a public IP. In the end, I implemented RTPProxy and T.38 works fine behind NAT. - Julian On Sun, Feb 15, 2009 at 1:25 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Julian, That is cool - in this way you save a lot of bandwidth and processing power with media relaying. Regards, Bogdan Julian Yap wrote: Hi all, I eventually played around with the Audiocodes box and enabled some settings so it worked with Comedia support. Thanks, Julian On 2/10/09, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: HI Julian, If it has, you can actually force it by adding direction=active into SDP as indication. See fix_nated_sdp(1) : http://www.opensips.org/html/docs/modules/1.4.x/nathelper.html#id270439 Regards, Bogdan Julian Yap wrote: Thanks all. I'll check to see if the AudioCodes gateway does have comedia support. That clarifies some half baked NAT/RTP knowledge in my head. - Julian On 2/10/09, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Olle, Johansson Olle E wrote: 10 feb 2009 kl. 12.25 skrev Iñaki Baz Castillo: 2009/2/10 julianok...@gmail.com: You don't know if RtpProxy should be running, does it mean you are trying to use it or not? I don't want to spend time inspecting what you want to do by reading your config, sorry. Yeah, I'm trying not to run RTPProxy. After more testing, I'm thinking I may need to. You cannot decide if you need RtpProxy or not based on testing, it's pure theory: A RTP proxy is NOT needed when (assuming the proxy has in the public internet): - Both caller and callee have public IP or use STUN. - Both caller and callee are in the *SAME* private LAN. - The caller is in a private LAN and the callee has public IP and supports Comedia mode (typical in some media servers and gateways). - The callee is in a private LAN and the caller has public IP and supports Comedia mode. A RTP proxy is needed when: - Caller is in private LAN (with no STUN) and callee in public internet (and not supporting Comedia). - Caller and callee are in different private LAN's with no STUN. I would like to add that it's the device that can't receive audio that needs the RTP proxy to get incoming audio. If both devices are on private IP's, there's going to be two RTP proxys involved if they're on different SIP networks. Each SIP service needs an RTP proxy for supporting their local users. To simplify: - If my user is on a private IP and sends an INVITE, add RTP proxy handling to the INVITE - If my user receives a call and sends a 200 OK, add RTP proxy handling to the 200 OK This logic is simple but not efficientTheoretically, if a call has already a leg in public net, there is not need for a media relay for traversing the nat. The only requirement is that all the devices to support symmetric media (comedia support). So, after the caller proxy forced RTPproxy, the callee should not do the same because the SDP already have a public IP, the nat traversal works even if the callee is behind a nat. Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT
In an example scenario, the re-INVITE is handled by the end device. So: PSTN Fax -- GW -- OpenSIPS -- UA (ATA attached to Fax machine) UA answers the call and then sends the re-INVITE which is correct as that is the terminating side. I read this RFC http://tools.ietf.org/html/draft-mule-sip-t38callflows-02 which was quite handy. :P The re-INVITE get accepted and RTP communication starts... However, for some reason, the T.38 part fails. In theory it should work but doesn't for me. Perhaps it's something wrong with my config at the time and the handling of the re-INVITE and NAT. Or perhaps it was some obscure issue with the GW and T.38 communications and timing, etc... Eventually I re-implemented it all with RTPProxy and that worked for me first time, inbound and outbound. Perhaps if someone has a clean working config with re-INVITE without using RTPProxy or MediaProxy, I can try that. Seems like all the example configs out there are used with a RTP proxy. - Julian On Mon, Feb 16, 2009 at 1:04 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Julian, You can still handle the NAT wih COMEDIA even for T.38, but you have to handle the re-INVITE also . In your scenario, who is generating the re-INVITE? Regards, Bogdan Julian Yap wrote: The full story is that I was looking to get T.38 working behind NAT. Unfortunately, no matter what I tried, it wouldn't work behind NAT. I had the initial INVITE (G.711) working fine but when there was the T.38 re-INVITE, the RTP media would connect up fine but just wouldn't negotiate properly with T.38. Very strange as it worked fine with the UA behind a public IP. In the end, I implemented RTPProxy and T.38 works fine behind NAT. - Julian On Sun, Feb 15, 2009 at 1:25 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Julian, That is cool - in this way you save a lot of bandwidth and processing power with media relaying. Regards, Bogdan Julian Yap wrote: Hi all, I eventually played around with the Audiocodes box and enabled some settings so it worked with Comedia support. Thanks, Julian On 2/10/09, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: HI Julian, If it has, you can actually force it by adding direction=active into SDP as indication. See fix_nated_sdp(1) : http://www.opensips.org/html/docs/modules/1.4.x/nathelper.html#id270439 Regards, Bogdan Julian Yap wrote: Thanks all. I'll check to see if the AudioCodes gateway does have comedia support. That clarifies some half baked NAT/RTP knowledge in my head. - Julian On 2/10/09, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Olle, Johansson Olle E wrote: 10 feb 2009 kl. 12.25 skrev Iñaki Baz Castillo: 2009/2/10 julianok...@gmail.com: You don't know if RtpProxy should be running, does it mean you are trying to use it or not? I don't want to spend time inspecting what you want to do by reading your config, sorry. Yeah, I'm trying not to run RTPProxy. After more testing, I'm thinking I may need to. You cannot decide if you need RtpProxy or not based on testing, it's pure theory: A RTP proxy is NOT needed when (assuming the proxy has in the public internet): - Both caller and callee have public IP or use STUN. - Both caller and callee are in the *SAME* private LAN. - The caller is in a private LAN and the callee has public IP and supports Comedia mode (typical in some media servers and gateways). - The callee is in a private LAN and the caller has public IP and supports Comedia mode. A RTP proxy is needed when: - Caller is in private LAN (with no STUN) and callee in public internet (and not supporting Comedia). - Caller and callee are in different private LAN's with no STUN. I would like to add that it's the device that can't receive audio that needs the RTP proxy to get incoming audio. If both devices are on private IP's, there's going to be two RTP proxys involved if they're on different SIP networks. Each SIP service needs an RTP proxy for supporting their local users. To simplify: - If my user is on a private IP and sends an INVITE, add RTP proxy handling to the INVITE - If my user receives a call and sends a 200 OK, add RTP proxy handling to the 200 OK This logic is simple but not efficientTheoretically, if a call has already a leg in public net, there is not need for a media relay for traversing the nat. The only requirement is that all the devices to support symmetric media (comedia support). So, after the caller proxy forced RTPproxy, the callee should not do the same because the SDP already have a public IP, the nat traversal works even if the callee is behind a nat. Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___
[OpenSIPS-Users] Help help help....
I have have manage to install opensips and including cdrtool, freeradius, and mediaproxy. I am encountering problems. Below is the error log. any information would be highly appreciated... thanx... Feb 17 08:40:02 ws16 cdrtool[4048]: Checking user quotas for data source OpenSIPS Proxy/Registrar Feb 17 08:40:02 ws16 cdrtool[4048]: Using database queries to block accounts Feb 17 08:40:02 ws16 cdrtool[4048]: Normalize lock id 88 aquired for opensips_radius:radacct Feb 17 08:40:02 ws16 cdrtool[4048]: Init quota of data source opensips_radius for all accounts Feb 17 08:40:02 ws16 cdrtool[4048]: Database error: Invalid SQL: select UserName,#012count(*) as calls,#012sum(AcctSessionTime) as duration,#012sum(Price) as cost,#012sum(AcctInputOctets + AcctOutputOctets)/2 as traffic#012from radacct#012where AcctStartTime = '2009-02-01 00:00'#012and Normalized = '1'#012 #012#012group by UserName#012 Feb 17 08:40:02 ws16 cdrtool[4048]: 58 Feb 17 08:40:02 ws16 cdrtool[4048]: Unlock opensips_radius:radacct Feb 17 08:40:02 ws16 cdrtool[4050]: Normalize datasource opensips_radius, database DB_radius, table radacct Feb 17 08:40:02 ws16 cdrtool[4050]: Normalize lock id 90 aquired for opensips_radius:radacct Feb 17 08:40:02 ws16 cdrtool[4050]: Database error: Invalid SQL: select *, UNIX_TIMESTAMP(AcctStartTime) as timestamp#012from radacct where (1=1) and Normalized = '0' and AcctStopTime != '-00-00 00:00:00' and (ConnectInfo_stop is not NULL or MediaInfo is NULL or MediaInfo != '' or (UNIX_TIMESTAMP(NOW()) - UNIX_TIMESTAMP(AcctStopTime) 20)) Feb 17 08:40:02 ws16 cdrtool[4050]: 58 Feb 17 08:40:02 ws16 cdrtool[4050]: Unlock opensips_radius:radacct Feb 17 08:40:02 ws16 media-dispatcher[3306]: [ManagementControlProtocol,9,127.0.0.1] Connection to Management interface client lost: Connection was closed cleanly. -- View this message in context: http://n2.nabble.com/Help-help-help-tp2338335p2338335.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Openxcap error for Opensips 1.4.4
Thanx Denis Bilenko, openxcap is running now... Denis Bilenko wrote: bay2x1 r...@racequeen.ph wrote: I am a newbie on this technology and I have manage to install opensips 1.4.4 but I am having problems with openxcap configuration. I have encountered this error on the error log. Feb 13 15:12:12 ws16 openxcap[10596]: [-] Log opened. Feb 13 15:12:12 ws16 openxcap[10596]: [-] Starting OpenXCAP 1.0.7 Feb 13 15:12:12 ws16 openxcap[10596]: [-] Supported Root URIs: https://xcap.example.com/xcap-root Feb 13 15:12:12 ws16 openxcap[10596]: [-] warning: Certificate file 'tls/server.crt' could not be loaded: File 'tls/server.crt' does not exist Feb 13 15:12:12 ws16 openxcap[10596]: [-] warning: Private key file 'tls/server.key' could not be loaded: File 'tls/server.key' does not exist Feb 13 15:12:12 ws16 openxcap[10596]: [-] fatal error: the TLS certificates or the private key could not be loaded Any information regarding this problem would be greatly appreciated. Thanks Did you generate a certificate and a private key for your server? Refer to GNUTLS documentation for information on how to do that. Once you have them, simply put them under /etc/openxcap/tls/server.crt and /etc/openxcap/tls/server.key names (provided that your config is under /etc/openxcap/ directory) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- View this message in context: http://n2.nabble.com/Openxcap-error-for-Opensips-1.4.4-tp2319950p2338508.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Missing call control module!
I have installed call control in fedora 10. As I was integrating its configuration to the opensips.cfg file I notice that the call_control.so is missing on the modules directory. The callcontrol daemon is already running on my machine. -- View this message in context: http://n2.nabble.com/Missing-call-control-module%21-tp2338529p2338529.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mediaproxy
On Monday 16 February 2009, Gabriel Bermudez wrote: In the documentation is stated that you need Python 2.4. Where do you see that it needs at least Python 2.5? Adrian When I tried to install mediaproxy on Ubuntu 8.04LTS using the ag-project's debian repositories it did complain about python2.5, thats why I upgraded to Ubuntu 8.10 We do not provide ubuntu packages. Here is a little more info about the package r...@ripley:~# aptitude show mediaproxy-common Package: mediaproxy-common New: yes State: installed Automatically installed: yes Version: 2.3.2 Priority: opcional Section: net Maintainer: Dan Pascu d...@ag-projects.com Uncompressed Size: 373k Depends: python ( 2.6), python (= 2.5), python-support (= 0.7.1), libc6 (= 2.4), libnetfilter-conntrack1 (= 0.0.89), python-application (= 1.0.9), python-cjson, python-gnutls, python-twisted-core (= 2.5.0), python-twisted-names, python-zopeinterface Recommends: python-pyrad (= 1.1), python-sqlobject At least the package does needs python =2.5 The package we provide is build and distributed for debian lenny and sid. You cannot use use it with whatever debian based distribution you want, nor can you use it with an older version of debian. You can only use it with lenny and sid as well as an ubuntu version that is equivalent with one of the two (i.e. has the same package set for the dependencies as debian lenny or sid). -- Dan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS libmysqlclient.so segfault after MySQL restart
Hi Bogdan, I have submitted the bug on the tracker. Thanks, Om. On Tue, Feb 17, 2009 at 3:50 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Om, Thanks for the report - I may have an idea what is the problem; I will try to reproduce the crash in the following days. Could you please open a bug report on the tracker - just not to forget about it :D (see http://www.opensips.org/index.php?n=Development.Tracker) Regards, Bogdan Om Bikram Thapa wrote: Hi, OpenSIPS is dying with libmysqlclient.so Segmentation Fault after MySQL server restart. The log shows opensips[16769]: segfault at 4c8 ip 7f36728bf283 sp 7fff7ba11a90 error 4 in libmysqlclient.so.15.0.0[7f3672862000+1bf000] and gdb bt shows: - #0 0x7f36728bf283 in mysql_stmt_result_metadata () from /usr/lib/libmysqlclient.so.15 #1 0x7f3672c707b7 in db_mysql_do_prepared_query (conn=0x78b278, query=value optimized out, v=0x7fff7ba11d20, n=1, uv=0x0, un=0) at dbase.c:363 #2 0x7f3672c72e1f in db_mysql_query (_h=0x78b278, _k=value optimized out, _op=value optimized out, _v=0x7fff7ba11d20, _c=value optimized out, _n=1, _nc=2, _o=0x0, _r=0x7fff7ba11dc0) at dbase.c:591 #3 0x7f366f3f0398 in authorize (_m=0x78fe98, _realm=value optimized out, _table=value optimized out, _hftype=value optimized out) at authorize.c:107 #4 0x0040f869 in do_action (a=0x780a48, msg=0x78fe98) at action.c:961 #5 0x0040e7e5 in run_action_list (a=value optimized out, msg=0x78fe98) at action.c:139 #6 0x0046f194 in eval_elem (e=0x780b18, msg=0x78fe98, val=0x0) at route.c:1189 #7 0x004708ed in eval_expr (e=0x120b1e0, msg=0x78fe98, val=0x0) at route.c:1486 #8 0x0047089c in eval_expr (e=0x780b60, msg=0x78fe98, val=0x0) at route.c:1502 #9 0x004708c5 in eval_expr (e=0x780ba8, msg=0x78fe98, val=0x0) at route.c:1507 #10 0x0040f91c in do_action (a=0x780f40, msg=0x78fe98) at action.c:688 #11 0x0040e7e5 in run_action_list (a=value optimized out, msg=0x78fe98) at action.c:139 #12 0x004114d3 in do_action (a=0x781eb0, msg=0x78fe98) at action.c:705 #13 0x0040e7e5 in run_action_list (a=value optimized out, msg=0x78fe98) at action.c:139 #14 0x00410e07 in do_action (a=0x77ee88, msg=0x78fe98) at action.c:119 #15 0x0040e7e5 in run_action_list (a=value optimized out, msg=0x78fe98) at action.c:139 #16 0x004114d3 in do_action (a=0x77f028, msg=0x78fe98) at action.c:705 #17 0x0040e7e5 in run_action_list (a=value optimized out, msg=0x78fe98) at action.c:139 #18 0x004125fe in run_top_route (a=0x777e78, msg=0x78fe98) at action.c:119 #19 0x0045e898 in receive_msg ( buf=0x749180 REGISTER sip:x SIP/2.0\r\nCSeq: 2 REGISTER\r\nVia: SIP/2.0/UDP x.x.x.x:5061;branch=z9hG4bK60f1b2dd-57fa-dd11-9401-0015c5404858;rport\r\nUser-Agent: Ekiga/2.0.12\r\nAuthorization: Dige..., len=749, rcv_info=0x7fff7ba13530) at receive.c:165 #20 0x0049d3a6 in udp_rcv_loop () at udp_server.c:449 #21 0x004291fb in main (argc=value optimized out, argv=0x7fff7ba13718) at main.c:778 Server runs fine after restart until MySQL is restarted again. In my lab setup, OpenSIPS is getting killed triggered by registration request every morning (probably due to MySQL being restarted daily with logrotate). The server is the latest trunk on Debian lenny/AMD64. Thanks, Om. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
