Re: [OpenSIPS-Users] Mediaproxy receives only
Hi Dan -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Dan Pascu Sent: Thursday, 10. September 2009 01:09 To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Mediaproxy receives only What could be a possible cause of mediaproxy (current src) to only receive media from both parties but not to relay it. There are a number of possible causes: 1. Check that there is no iptables rule blocking the traffic 2. Make sure IP forwarding is enabled: /proc/sys/net/ipv4/ip_forward should be 1 3. Make sure rp_filter is disabled: /proc/sys/net/ipv4/conf/all/rp_filter is 0 /proc/sys/net/ipv4/conf/default/rp_filter is 0 /proc/sys/net/ipv4/conf/eth0/rp_filter is 0 (replace eth0 with whatever interface you use) The rp_filter has been set to 1. I have overlooked this. Thanks for the helpful hint! Best regards Sebastian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] mediaproxy relay on none default route interface failed
Dear All, I tried to make the relay to listen to the none default route interface, but failed. Below is my configuration info and my env. Using the latest mediaproxy-2.3.8 on CentOS 5.3. I had media-dispatcher and media-relay running on my server which has multiple NIC interface, two public ips which are belong to two different provider. for example: eth0: 192.168.88.12 eth1: 203.xx.xx.xx eth2: 63.xx.xx.xx eth3: 10.11.12.21 route -n Kernel IP routing table Destination Gateway Genmask Flags Metric RefUse Iface 63.xx.xx.320.0.0.0 255.255.255.240 U 0 00 eth2 203.xx.xx.128 0.0.0.0 255.255.255.192 U 0 00 eth1 10.11.12.0 0.0.0.0 255.255.255.0 U 0 00 eth3 192.168.88.00.0.0.0 255.255.255.0 U 0 00 eth0 169.254.0.0 0.0.0.0 255.255.0.0 U 0 00 eth3 0.0.0.0 203.xx.xx.129 0.0.0.0 UG0 00 eth1 But I notice that the relay are always binding to the one that has default route as said in the document. I can use two machine of course, but is it possible to have two separate relay that will only sent out packet from the interface it's listening to. my config file is: [Relay] dispatchers = 63.xx.xx.xx 203.xx.xx.xx port_range = 3:4 log_level = DEBUG [Dispatcher] socket_path = dispatcher.sock listen_management = 127.0.0.1 log_level = DEBUG relay_timeout = 5 [TLS] certs_path = tls [Database] [Radius] [OpenSIPS] socket_path = '/var/run/opensips.sock' max_connections = 20 I tried to use the media_relay_avp before calling use_media_proxy(); but it seems the relay still listening to the other interface which has the default route. The script of opensips is below: $avp(s:media_relay) = 63.xx.xx.xx; use_media_proxy(); Regards, -- Jinke Jiang the debug log of mediaproxy: Sep 10 18:16:54 ssw1 media-dispatcher[21925]: debug: Issuing update command to relay at 63.xx.xx.xx Sep 10 18:16:54 ssw1 media-relay[21930]: debug: Received new SDP offer Sep 10 18:16:54 ssw1 media-relay[21930]: mediaproxy.mediacontrol.StreamListenerProtocol starting on 30028 Sep 10 18:16:54 ssw1 media-relay[21930]: mediaproxy.mediacontrol.StreamListenerProtocol starting on 30029 Sep 10 18:16:54 ssw1 media-relay[21930]: mediaproxy.mediacontrol.StreamListenerProtocol starting on 30030 Sep 10 18:16:54 ssw1 media-relay[21930]: mediaproxy.mediacontrol.StreamListenerProtocol starting on 30031 Sep 10 18:16:54 ssw1 media-relay[21930]: debug: Added new stream: (audio) 10.10.101.147:20562 (RTP: Unknown, RTCP: Unknown) - 203.xx.xx.xx:30028 - 203.xx.xx.xx:30030 - Unknown (RTP: Unknown, RTCP: Unknown) Sep 10 18:16:54 ssw1 media-relay[21930]: debug: created new session 4022675...@10.10.101.147: 8...@test.com (3297793466) -- 0086135x...@test.com Sep 10 18:16:55 ssw1 media-dispatcher[21925]: debug: Issuing update command to relay at 63.xx.xx.xx Sep 10 18:16:55 ssw1 media-relay[21930]: debug: updating existing session 4022675...@10.10.101.147: 8...@test.com (3297793466) -- 0086135x...@test.com Sep 10 18:16:55 ssw1 media-relay[21930]: debug: Received updated SDP answer Sep 10 18:16:55 ssw1 media-relay[21930]: debug: Got initial answer from callee for stream: (audio) 10.10.101.147:20562 (RTP: Unknown, RTCP: Unknown) - 203.xx.xx.xx:30028 - 203.xx.xx.xx:30030 - 203.xx.xx.15x:18550 (RTP: Unknown, RTCP: Unknown) Sep 10 18:16:56 ssw1 media-relay[21930]: debug: Got traffic information for stream: (audio) 10.10.101.147:20562 (RTP: 119.145.xx.xx:20562, RTCP: Unknown) - 203.xx.xx.xx:30028 - 203.xx.xx.xx:30030 - 203.xx.xx.15x:18550 (RTP: Unknown, RTCP: Unknown) Sep 10 18:16:56 ssw1 media-relay[21930]: debug: Got traffic information for stream: (audio) 10.10.101.147:20562 (RTP: 119.145.xx.xx:20562, RTCP: Unknown) - 203.xx.xx.xx:30028 - 203.xx.xx.xx:30030 - 203.xx.xx.15x:18550 (RTP: 203.xx.xx.15x:18550, RTCP: Unknown) Sep 10 18:16:58 ssw1 media-relay[21930]: debug: Got traffic information for stream: (audio) 10.10.101.147:20562 (RTP: 119.145.xx.xx:20562, RTCP: Unknown) - 203.xx.xx.xx:30028 - 203.xx.xx.xx:30030 - 203.xx.xx.15x:18550 (RTP: 203.xx.xx.15x:18550, RTCP: 203.xx.xx.15x:18551) Sep 10 18:17:03 ssw1 media-dispatcher[21925]: debug: Issuing update command to relay at 63.xx.xx.xx Sep 10 18:17:03 ssw1 media-relay[21930]: debug: updating existing session 4022675...@10.10.101.147: 8...@test.com (3297793466) -- 0086135x...@test.com Sep 10 18:17:03 ssw1 media-relay[21930]: debug: Received updated SDP answer Sep 10 18:17:03 ssw1 media-relay[21930]: debug: Unchanged stream: (audio) 10.10.101.147:20562 (RTP: 119.145.xx.xx:20562, RTCP: Unknown) - 203.xx.xx.xx:30028 - 203.xx.xx.xx:30030 - 203.xx.xx.15x:18550 (RTP: 203.xx.xx.15x:18550, RTCP: 203.xx.xx.15x:18551) Sep 10 18:17:04 ssw1 media-dispatcher[21925]: debug: Issuing remove command to relay at 63.xx.xx.xx Sep 10 18:17:04 ssw1 media-relay[21930]: debug: removing session 4022675...@10.10.101.147: 8...@test.com
[OpenSIPS-Users] RTP traffic opensips
Hi there, I have been configuring OpenSIPS for a while the last 2 days and I have been able to configure a simple server that register clients against a Database. The problem I am facing is that when the call is established there's no Audio. I can see RTP traffic from one client to the SIP Proxy and from the other client to the SIP Proxy. On the server where opensips is running I can see both RTP traffics but nothing goes from OpenSIPS to the clients. The configuration I am using is the the one that comes with the installation, changed to support mysql. I have followed this tutorial http://www.voip-info.org/wiki/view/Opensips+Installation,+How+to. As far as I understand, OpenSIPS should act as SIP proxy so RTP traffic should be between clients. Any help would be apprecieted. Thanks in advance, Jonathan -- Personal webpage - www.jonbaraq.eu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RTP traffic opensips
A SIP proxy does not relay RTP. The only way RTP should ever hit the OpenSIPS host is if you're running MediaProxy, RTPProxy, etc. on the same host. Otherwise, RTP should flow entirely between the two hosts and OpenSIPS has absolutely *zero* to do with it. If it's not working, NAT is the most likely reason. Jonathan González wrote: Hi there, I have been configuring OpenSIPS for a while the last 2 days and I have been able to configure a simple server that register clients against a Database. The problem I am facing is that when the call is established there's no Audio. I can see RTP traffic from one client to the SIP Proxy and from the other client to the SIP Proxy. On the server where opensips is running I can see both RTP traffics but nothing goes from OpenSIPS to the clients. The configuration I am using is the the one that comes with the installation, changed to support mysql. I have followed this tutorial http://www.voip-info.org/wiki/view/Opensips+Installation,+How+to. As far as I understand, OpenSIPS should act as SIP proxy so RTP traffic should be between clients. Any help would be apprecieted. Thanks in advance, Jonathan -- Personal webpage - www.jonbaraq.eu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy relay on none default route interface failed
On Thursday 10 September 2009 11:56:00 Ghaith ALKAYYEM wrote: Hello, I think it's not possible to use two separate relays on the same server, I tried that a lot then I switched to RTPproxy. That's not true, you could run as many Realys as you want on the same server, only have to patch mediaproxy-relay to be able to call it with a diferent .cfg as the default one, have diferent listen ports and no more. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dynamic Routing - routing using source IP address
With Dynamic Routing, how can we route based on source IP address/Gateway ? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy relay on none default route interface failed
Dear Raúl, Thanks for the idea. But I'm wondering if there is another way to do this easily. Regards, Jinke Jiang 2009/9/10 Raúl Alexis Betancor Santana r...@dimension-virtual.com: On Thursday 10 September 2009 11:56:00 Ghaith ALKAYYEM wrote: Hello, I think it's not possible to use two separate relays on the same server, I tried that a lot then I switched to RTPproxy. That's not true, you could run as many Realys as you want on the same server, only have to patch mediaproxy-relay to be able to call it with a diferent .cfg as the default one, have diferent listen ports and no more. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ 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] Regarding the durationPeriodRated in CDRTool
hi all, Is there anyway 1) that i can mention different durationPeriodRated for different destination. 2) Can we round off the call rate to the next min after i the person has spoken for certain seconds ex: User A calls User B for 1 min and 12 secs. can we charge him for 2 mins. If yes how can we do it in CDRTool -- Thanking You, Ashwini BR Naidu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy relay on none default route interface failed
On Thursday 10 September 2009 15:02:21 Dan Pascu wrote: You don't need to patch anything. Just unpack mediaproxy in as many different directories as you need, run ./build_inplace and modify each config.ini in those directories as needed. Then run mediaproxy from those directories and each of them will use the local config.ini from its own directory. So ... killing flies with bazookas ... So ... when having to update, need to update n instances, instead of only one .. So ... wasting hard disk (I know it's cheap ... but wasting anyway) ... I think it's easier to patch media-relay to be able to pass the config file as a param, as on many other programs around the world .. Alternatively, if you want to use a system wide installation, you can copy the binaries from /usr/bin to a number of different directories and add a config.ini in each directory. Then run those binaries from those directories instead of /usr/bin/ and each binary will use the config.ini file in its own directory to overwrite settings from the global /etc/mediaproxy/config.ini. ... no coments Mediaproxy uses 2 configuration files. The global one resides in /etc/ mediaproxy/config.ini. On top of that if a config.ini is present in the same directory as the binary (media-relay media-dispatcher) that one will be used to overwrite the settings from the global one having priority over it. I have no tested that ... but anyway it requires me to been duplicating installations ... I still think that a simple param like -c configfile is better. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dynamic Routing - routing using source IP address
Usually you set the group based on something else. I use memcache for this.. so I toss the IP against memcache, which returns a group id, then I use that in the do_routing() cmd. Of course, this requires pre-populating the cache src_ip = group_id The newly released startup_route can do that for you now. :D -Brett On Thu, Sep 10, 2009 at 8:00 AM, Julien Chavanton j...@atlastelecom.comwrote: With Dynamic Routing, how can we route based on source IP address/Gateway ? ___ 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] mediaproxy relay on none default route interface failed
Thanks for the detail instruction. I just use a symlink into the directory, it's working properly now. Just like below: /usr/local/relay1/media-relay - /usr/bin/media-relay /usr/local/relay2/media-relay - /usr/bin/media-relay Regards, Jinke Jiang On Thu, Sep 10, 2009 at 10:02 PM, Dan Pascu d...@ag-projects.com wrote: On 10 Sep 2009, at 15:47, Raúl Alexis Betancor Santana wrote: On Thursday 10 September 2009 11:56:00 Ghaith ALKAYYEM wrote: Hello, I think it's not possible to use two separate relays on the same server, I tried that a lot then I switched to RTPproxy. That's not true, you could run as many Realys as you want on the same server, only have to patch mediaproxy-relay to be able to call it with a diferent .cfg as the default one, have diferent listen ports and no more. You don't need to patch anything. Just unpack mediaproxy in as many different directories as you need, run ./build_inplace and modify each config.ini in those directories as needed. Then run mediaproxy from those directories and each of them will use the local config.ini from its own directory. Alternatively, if you want to use a system wide installation, you can copy the binaries from /usr/bin to a number of different directories and add a config.ini in each directory. Then run those binaries from those directories instead of /usr/bin/ and each binary will use the config.ini file in its own directory to overwrite settings from the global /etc/mediaproxy/config.ini. Mediaproxy uses 2 configuration files. The global one resides in /etc/ mediaproxy/config.ini. On top of that if a config.ini is present in the same directory as the binary (media-relay media-dispatcher) that one will be used to overwrite the settings from the global one having priority over it. -- Dan ___ 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] mediaproxy relay on none default route interface failed
Does the dispatcher work on the same machine also? Could you provide me with more details about running these two instances? Regards. On Thu, 2009-09-10 at 23:39 +0800, Jiang Jinke wrote: Thanks for the detail instruction. I just use a symlink into the directory, it's working properly now. Just like below: /usr/local/relay1/media-relay - /usr/bin/media-relay /usr/local/relay2/media-relay - /usr/bin/media-relay Regards, Jinke Jiang On Thu, Sep 10, 2009 at 10:02 PM, Dan Pascu d...@ag-projects.com wrote: On 10 Sep 2009, at 15:47, Raúl Alexis Betancor Santana wrote: On Thursday 10 September 2009 11:56:00 Ghaith ALKAYYEM wrote: Hello, I think it's not possible to use two separate relays on the same server, I tried that a lot then I switched to RTPproxy. That's not true, you could run as many Realys as you want on the same server, only have to patch mediaproxy-relay to be able to call it with a diferent .cfg as the default one, have diferent listen ports and no more. You don't need to patch anything. Just unpack mediaproxy in as many different directories as you need, run ./build_inplace and modify each config.ini in those directories as needed. Then run mediaproxy from those directories and each of them will use the local config.ini from its own directory. Alternatively, if you want to use a system wide installation, you can copy the binaries from /usr/bin to a number of different directories and add a config.ini in each directory. Then run those binaries from those directories instead of /usr/bin/ and each binary will use the config.ini file in its own directory to overwrite settings from the global /etc/mediaproxy/config.ini. Mediaproxy uses 2 configuration files. The global one resides in /etc/ mediaproxy/config.ini. On top of that if a config.ini is present in the same directory as the binary (media-relay media-dispatcher) that one will be used to overwrite the settings from the global one having priority over it. -- Dan ___ 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] Dynamic Routing - routing using source IPaddress
Thank you, I will look further after using group, I am also facig some
requirements to append prefix, when a call comes in from a defined Gateway.
Here is a recap, of what I understand after reading testing Dynamic Routing :
When a call is comming from a given Gateway we can not use Dynamic Routing to
append/strip prefixes, DR module only work on destination Gateway.
Therefore I selected to use Dial Plan for append/strip prefix on inbound calls.
Outbound number manipulation prefix append/strip : module (Dynamic Routing)
STRIP : striped from the RURI when the call is going to the target gateway
PRI Prefix : assigned to the RURI when the call is going to the target gateway
Inbound number manipulation prefix append/strip : module (Dial Plan)
--
Prefix : a lookup in the dialplan table for the source IP address will return a
prefix to append if any
Example:
# find if there is a prefix for traffic from the IP address
dp_translate(0, $src_ip/$avp(s:prefix));
if($avp(s:prefix) method==INVITE !has_totag()){
xlog(prefix for IP[$src_ip]=[$avp(s:prefix)]\n);
$rU = $avp(s:prefix) + $rU ;
}
else{
xlog(no prefix for IP[$src_ip]\n);
}
From: users-boun...@lists.opensips.org on behalf of Brett Nemeroff
Sent: Thu 10/09/2009 4:18 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Dynamic Routing - routing using source IPaddress
Usually you set the group based on something else. I use memcache for this.. so
I toss the IP against memcache, which returns a group id, then I use that in
the do_routing() cmd.
Of course, this requires pre-populating the cache src_ip = group_id
The newly released startup_route can do that for you now. :D
-Brett
On Thu, Sep 10, 2009 at 8:00 AM, Julien Chavanton j...@atlastelecom.com wrote:
With Dynamic Routing, how can we route based on source IP
address/Gateway ?
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Re: [OpenSIPS-Users] mediaproxy relay on none default route interface failed
On Thursday 10 September 2009 22:25:39 Dan Pascu wrote: On 10 Sep 2009, at 18:00, Raúl Alexis Betancor Santana wrote: So ... killing flies with bazookas ... So ... when having to update, need to update n instances, instead of only one .. So ... wasting hard disk (I know it's cheap ... but wasting anyway) ... I'm pretty sure you felt smart and clever while wording your indignant reply. Indeed, you absolutely had to put me into my rightful place and I deserved every dot and comma in it. What was in my mind trying to suggest some alternatives that work with the software as it is and do not require patching or a new version? But for a strange reason I cannot pinpoint, I do not feel motivated by your rhetoric. No sir. Shamefully I must admit that it made me think that I can spend my time better elsewhere. Imagine that good sir. I'm so ashamed. It also made me have unclean thoughts about you. It made me think that maybe you have a size problem and you're compensating. Can you imagine that? So, good sir, maybe you should fire more of your aggressive indignation towards me to break this unclean thinking in my mind and finally make me see the light. Forget about asking nicely or, God forbid, contributing back. That's for pussies and it only weakens one's character. Every time we have talked about this ... you always react the same ... as If I was fighting against you ... I don't know from where you get that feeling from my words. I'm only expressing my opinion, your are free to ignore it if you don't like it. Yes you gave a solucion and I arge that a better solucion (and from a desing point of view and from the KISS principles, easier) it's so simple add a -c config param. Do you want the patch? ... they are just 4 lines, I did it for 2.3.4 version of mediaproxy, also for callcontrol and other very good software you have developed but that lacks THAT SIMPLE option, to be able to set the config file to use from the command line ... as 99% of other software allow you. I still think that a simple param like -c configfile is better. That may be, but I made a principle of not helping people that hate me or abuse me verbally. I don't think I have abused you ... if you think soo it's your problem. I only express my opinion and comment other possible solutions. Besides, free in free software doesn't stand for let's use that software that is free of charge and then feel free to abuse the developer. It stands for fell free to improve it and contribute back. Wait, scratch that. From you I wouldn't even want patches. The contribution you made is already more than enough... Dan, sincerily .. you don't like to hear opinions that are agains yours ... and that's not good. If you don't want my patches, you are free to ignore them. Not always a patch is needed to improve a software or to contribute to it, most of time another point of view about a problem it's enought. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Voicemail doesn't work with Mediaproxy
I have been trying to fix this for the last 4 days. I have searched the
mailing list with the following keywords (mediaproxy rewritehostport one-way
audio, etc) and I find some posts that explain that my 200 ok messages
need say that the connection information needs to point to my mediaproxy.
I am trying to use fix_nated_sdp(3, ip of mediaproxy0), but I am sure I
am placing it and using it in the wrong spot in my script.
So here is the problem
When a user calls someone and they are busy or not there the caller will be
directed to voicemail by using the rewritehostport(ip of voicemail)
command. The caller is nated behind a router. The caller doesn't hear the
voicemail voice prompts, but the caller IS able to leave a voicemail and be
heard. So there is one-way audio going on where the caller can't hear, but
voicemail can. When I do a sniff I see that this is because the caller's
RTP is going directly to the Voicemail Servers Public IP address and the
Voicemails RTP is going to the Mediaproxy.
Here is the code in question
failure_route[1] {
##--
##-- If cancelled, exit.
##--
if (t_was_cancelled()) {
exit;
};
##--
##-- If busy send to the e-mail server, prefix the b
##-- character to indicate busy.
##--
if (t_check_status(486)) {
revert_uri();
append_hf(P-App-Name: voicemail\r\nP-App-Param:
Email-Address=blah.blahmail.com;Mode=both;usr=$tU;d...@$td\r\n);
rewritehostport(VoicemailIP);
append_branch();
route(1);
exit;
};
##--
##-- If timeout (408) or unavailable temporarily (480),
##-- prefix the uri with the ucharacter to indicate
##-- unanswered and send to the e-mail
##-- sever
##--
if (t_check_status(408) || t_check_status(480)) {
revert_uri();
append_hf(P-App-Name: voicemail\r\nP-App-Param:
Email-Address=blah.blahmail.com;Mode=both;usr=$tU;d...@$td\r\n);
rewritehostport(VoicemailIP);
append_branch();
route(1);
exit;
};
}
onreply_route[1] {
#
#-- On-replay block routing --
#
if ((isbflagset(6) || isbflagset(7))
(status=~(180)|(183)|2[0-9][0-9])) {
xlog(L_INFO, - Inside OnReply Route 1 with bflag
being 6 or 7 and a bunch other stuff);
if (search(^Content-Type:[ ]*application/sdp)) {
xlog(L_INFO, - Inside OnReply Route 1 with
content type equaling sdp);
append_hf(P-hint: onreply_route|usemediaproxy
\r\n);
use_media_proxy();
};
};
if (client_nat_test(1)) {
xlog(L_INFO, - Inside OnReply Route 1 and client nat
equaled 1);
append_hf(P-hint: Onreply-route - fixcontact \r\n);
fix_contact();
};
exit;
}
So how do I solve this issue?
--
View this message in context:
http://n2.nabble.com/Voicemail-doesn-t-work-with-Mediaproxy-tp3618910p3618910.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
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Re: [OpenSIPS-Users] Expire call with redirecthostport()
I want to control the maximum duration of the entire call. Thanks! -Thiago Rondon Bogdan-Andrei Iancu escreveu: Hi Thiago, by this expire_call_in_seconds(300); you want to control the maximum duration of the entire call ? or only the setup time of the call ? Regards, Bogdan Thiago Rondon wrote: Hi, Today I use redirecthostport() to forward calls for another provider, can I make a static limit for all calls that I send to ? Like: expire_call_in_seconds(300); # redirecthostport( host:5060); Thanks! -Thiago Rondon ___ 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] mediaproxy relay on none default route interface failed
El Jueves, 10 de Septiembre de 2009, Dan Pascu escribió: That's not true, you could run as many Realys as you want on the same server, only have to patch mediaproxy-relay to be able to call it with a diferent .cfg as the default one, have diferent listen ports and no more. You don't need to patch anything. Just unpack mediaproxy in as many different directories as you need, run ./build_inplace and modify each config.ini in those directories as needed. Then run mediaproxy from those directories and each of them will use the local config.ini from its own directory. Is this really preferable to a single and so common -c CONFIG_FILE commandline option? A -c option would break nothing since the absence of it means default config file. Also a -c option allows deb usual installation and upgrade (the only requeriment would be create N init scripts having each one a different -c config file and PID file). Perhaps you suggest the above solution as a feasible workaround without patching the code. I expect it because I cannot imagine how managing N full installations of N mediaproxies can be better than a simple -c option involving a single installation of the software. Of course this is just my opinion. Take is as a suggestion and not as a demand. Regards. -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Voicemail doesn't work with Mediaproxy
I am running OpenSIPS version 1.5.1
I am sure I found the append_branch command from someone's code I found
online.
I did have voicemail working to where there was two-way audio. I thought I
had solved everything, but then I came back from the weekend holiday, tried
to make a call between two natted devices and noticed that I had broken
that. I forget what I did to make voicemail work, but I will try and find
out again. I will do like you say and make sure I am placing
use_media_proxy in all the proper routes. I will post once I figure it
out. I hate it when people on this mailing list figure it out and all the
post is Fixed it without mentioning what they did.
On Thu, Sep 10, 2009 at 5:35 PM, Thomas Gelf tho...@gelf.net wrote:
osiris123d wrote:
I am trying to use fix_nated_sdp(3, ip of mediaproxy0), but I am sure
I
am placing it and using it in the wrong spot in my script.
Don't do that. If you are using fix_nated_sdp together with mediaproxy,
you will for sure mess up something. Calling use_media_proxy in your
INVITE route, your reply route and also for your in-dialog ReINVITES
will automagically fix your SDP.
failure_route[1] {
...
append_branch();
What OpenSIPS version are you using? In recent versions append_branch
is not needed here.
Best regards,
Thomas Gelf
--
mail: tho...@gelf.net
web: http://thomas.gelf.net/
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Today's Topics:
1. Re: Dynamic Routing - routing using sourceIPaddress
(Julien Chavanton)
2. Re: mediaproxy relay on none default routeinterface failed
(Dan Pascu)
3. Re: mediaproxy relay on none default routeinterface failed
(Ra?l Alexis Betancor Santana)
4. Voicemail doesn't work with Mediaproxy (osiris123d)
5. Re: Voicemail doesn't work with Mediaproxy (Thomas Gelf)
6. Re: Expire call with redirecthostport() (Thiago Rondon)
7. Re: mediaproxy relay on none default routeinterface failed
(I?aki Baz Castillo)
--
Message: 1
Date: Thu, 10 Sep 2009 18:15:24 +0100
From: Julien Chavanton j...@atlastelecom.com
Subject: Re: [OpenSIPS-Users] Dynamic Routing - routing using source
IPaddress
To: OpenSIPS users mailling list users@lists.opensips.org,
OpenSIPS users mailling list users@lists.opensips.org
Message-ID:
3a55bd693d48034787fa018dfa2403ee42a...@d3bmvc3j.atlas.private
Content-Type: text/plain; charset=iso-8859-1
Thank you, I will look further after using group, I am also facig some
requirements to append prefix, when a call comes in from a defined Gateway.
Here is a recap, of what I understand after reading testing Dynamic Routing :
When a call is comming from a given Gateway we can not use Dynamic Routing to
append/strip prefixes, DR module only work on destination Gateway.
Therefore I selected to use Dial Plan for append/strip prefix on inbound calls.
Outbound number manipulation prefix append/strip : module (Dynamic Routing)
STRIP : striped from the RURI when the call is going to the target gateway
PRI Prefix : assigned to the RURI when the call is going to the target gateway
Inbound number manipulation prefix append/strip : module (Dial Plan)
--
Prefix : a lookup in the dialplan table for the source IP address will return a
prefix to append if any
Example:
# find if there is a prefix for traffic from the IP address
dp_translate(0, $src_ip/$avp(s:prefix));
if($avp(s:prefix) method==INVITE !has_totag()){
xlog(prefix for IP[$src_ip]=[$avp(s:prefix)]\n);
$rU = $avp(s:prefix) + $rU ;
}
else{
xlog(no prefix for IP[$src_ip]\n);
}
From: users-boun...@lists.opensips.org on behalf of Brett Nemeroff
Sent: Thu 10/09/2009 4:18 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Dynamic Routing - routing using source IPaddress
Usually you set the group based on something else. I use memcache for this.. so
I toss the IP against memcache, which returns a group id, then I use that in
the do_routing() cmd.
Of course, this requires pre-populating the cache src_ip = group_id
The newly released startup_route can do that for you now. :D
-Brett
On Thu, Sep 10, 2009 at 8:00 AM, Julien Chavanton j...@atlastelecom.com wrote:
With Dynamic Routing, how can we route based on source IP
address/Gateway ?
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