search for: siproxd

Displaying 20 results from an estimated 27 matches for "siproxd".

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2005 Jan 24
1
Asterisk -> static nat -> laptop w/siproxd -> cisco 7960
Ok, I have a 7960 that's plugged into my laptop. my home network is wireless so I don't have a switch anywhere to plug the phone into directly. I'm running siproxd on my OS X laptop and I can make outbound calls from the 7960 fine (I guess I don't have the phone configured to register inbound calls via SIP), but the phone isn't registering to the asterisk box via siproxd and I can't figure out why.. [ Asterisk ] -- Net -- [ Router w/static na...
2004 Apr 27
0
Issues with Asterisk & siproxd
I'm running Asterisk on an external static IP address, siproxd on a different server with its own external static IP address, and communicating using a Grandstream behind a NAT firewall configured to register with Asterisk using siproxd as the outbound proxy. Now I'm aware that siproxd is not intended to be used as an outbound proxy but rather as a SIP re...
2015 Apr 30
2
OpenVPN Clients Intermittently Cannot Call In
...other Yealink model) on > > 192.168.32.0/24 which have an OpenVPN client configured on them that > > connects back to the LAN network through a pfSense gateway with OpenVPN > > configured on it. > > I faced problems with pfsense -no VPN involved- and finally installed > siproxd on it. Also set the firewall mode to conservative. Daniel, Thanks for the information. Do you have an example or documentation on the siproxd configuration that you used? Thanks, Andrew
2006 Jan 10
0
outboundproxy issue
...ain, if I dial from within to the asterisk box via a local sip client I get audio. I might be on the wrong track with the outboundproxy, but since I'm not setting nat or stun in the other sip clients and they can make and receive calls, I can't see what else it could be. I also read about siproxd, and it said in it's docs that (at that time) asterisk didn't support outboundproxy and siproxd could be used as a transparent proxy. Could siproxd be used behind the firewall as I am? I'm running asterisk on my local box not on the firewall I'm behind. If someone has experience...
2015 May 05
2
OpenVPN Clients Intermittently Cannot Call In
....32.0/24 which have an OpenVPN client configured on them that > >>> connects back to the LAN network through a pfSense gateway with OpenVPN > >>> configured on it. > >> > >> I faced problems with pfsense -no VPN involved- and finally installed > >> siproxd on it. Also set the firewall mode to conservative. > > > > Daniel, > > > > Thanks for the information. Do you have an example or documentation on the > > siproxd configuration that you used? > > No, just follow the basis of the parameters given by the package. I...
2015 May 01
0
OpenVPN Clients Intermittently Cannot Call In
...) on >>> 192.168.32.0/24 which have an OpenVPN client configured on them that >>> connects back to the LAN network through a pfSense gateway with OpenVPN >>> configured on it. >> >> I faced problems with pfsense -no VPN involved- and finally installed >> siproxd on it. Also set the firewall mode to conservative. > > Daniel, > > Thanks for the information. Do you have an example or documentation on the > siproxd configuration that you used? No, just follow the basis of the parameters given by the package. If I remember, SIP use the proxy si...
2015 May 05
0
OpenVPN Clients Intermittently Cannot Call In
...ave an OpenVPN client configured on >>>>> them that connects back to the LAN network through a >>>>> pfSense gateway with OpenVPN configured on it. >>>> >>>> I faced problems with pfsense -no VPN involved- and finally >>>> installed siproxd on it. Also set the firewall mode to >>>> conservative. >>> >>> Daniel, >>> >>> Thanks for the information. Do you have an example or >>> documentation on the siproxd configuration that you used? >> >> No, just follow the basis...
2015 Apr 30
2
OpenVPN Clients Intermittently Cannot Call In
Hello, I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP phones, which appear to be working correctly. I have a few external phones (Yealink SIP-T32G or other Yealink model) on 192.168.32.0/24 which have an OpenVPN client configured on them that connects back to the LAN network through a
2003 May 07
2
SIPPROXD for SIP thru NAT
Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: siproxd.url Type: application/octet-stream Size: 82 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030507/bddd870b/siproxd.obj
2004 Jun 01
2
Router, Firewall, SIP Rewriter, and GnuGK
Hi I am running firewall/router "brew" made of RedHat, Shorewall, Siproxd and GnuGK on a box that connects through PPPoE to Internet. I run Asterisk on another box behind of it and it seem to work fine for me. I am thinking of replacing the router box because hardware is getting flaky. I do not want to go through pain of assembling all this stuff together again. Doe...
2004 Aug 09
1
How do folks handle NAT routing?
I'm interested to hear how folks are handling NAT SIP routing issues "in the wild" for commercial use. Are you using a commerical SIP-aware NAT router solution? If so, what? Are you using a software SIP-proxy like SER or siproxd? If so, which? Do you set everything to "canreinvite=no" in sip.conf? Any comments about real-world implementations would be welcome. Thanks -- David Gurr Congruity Ltd. Hemel Hempstead, UK
2009 Jan 31
1
where to find STUN Server howto
Hi people! Do you guys know where to find a STUN Server Howto?! Why?! We all know, to get Asterisk behind an NAT Router to run, is a bit tricky, and you might have to fire a lot of holes in your firewall. However, I would appreciate it very much if somebody could give me great links of how to set up a STUN Server. Tamer
2003 Jun 30
2
A solution for SIP and NAT
Hi all. I have come to the conclusion that there just isn't anything out there for allowing SIP and NAT to work together nicely. This is rather amazing considering that as far back as March 2000 there are documents describing how to do it. So I've started a really simple SIP and RTP proxy project, SaRP, on sourceforge.net. Yesterday we uploaded 0.2 of the perl based release. This is the
2010 Dec 11
2
Why does "sip show peers" show my router/gateway address as the client IP address?
Hi Everyone, I am using pfSense to do firewall and NAT on an Asterisk server. I have ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local IP 192.168.5.5. However, when a user from outside using Linksys WRP400 ata connects to the Asterisk server and registers I see them as 192.168.1.1 in the "sip show peers" command. In face, all many different of the Linksys WRP400
2009 Jan 19
3
[somewhat OT] seeking ideas/input for my thesis
...at off-topic. At the moment I am studying informatics in the seventh semester and I need to start thinking about my thesis. As I am very interested in VoIP technologies I thought about picking this as my main topic. So far I have only little experience in this area. I have been fiddling around with siproxd and pfSense and have red the one or the other packet dump containing SIP and RTP traffic, had a look into codecs, STUN, etc... but very cursorily, and that's the reason why I am quite unsure on which track to go. I think I am quite familiar with many network protocols and devices... so here com...
2011 Jan 11
2
Do I need a sip proxy?
...houldn't really happen and router should take of this by sending it to the server that has the established connection through that port. *My equipment:* Asterisk 1.6x Pfsense 1.2.3 Dumb Switch *My questions:* A- What is the rational behind this? B- Do I need a sip proxy server? Something like Siproxd, OpenSIPs, or Kamailio? C- Which one of the above is the easiest to get running given I never tried any of those. D- If I am doing an SIP proxy server then it might have to also be redundant. What options do I have in that and which of above or any other suggested package might be great for future...
2003 Aug 12
1
Working with FWD, IPTel, SIPPhone?
...Username Perceived Refresh State 12.37.165.130:4569 username 66.x.x.x:4569 60 Registered Unfortunately(?), any calls through IAX2 never seem to go through. While I'd like to eventually setup an outbound NAT proxy, I've had a difficult time decyphering how to configure SER, siproxd, or PartySIP to register to external SIP providers like FWD, IPTel, and SIPPhone. I'm guessing this is what the additional sections in sip.conf are for? sip.conf ;; Free World Dialup Proxy [fwd.pulver.com] type=friend host=fwd.pulver.com fromuser=48702 fromdomain=fwd.pulver.com...
2015 Apr 30
0
OpenVPN Clients Intermittently Cannot Call In
...e a few external phones (Yealink SIP-T32G or other Yealink model) on 192.168.32.0/24 which have an OpenVPN client configured on them that connects back to the LAN network through a pfSense gateway with OpenVPN configured on it. I faced problems with pfsense -no VPN involved- and finally installed siproxd on it. Also set the firewall mode to conservative. [...] -- Daniel
2003 Sep 03
1
Traversing the NAT
...us SIP users have problems with NAT boxes from hell... :-) There's some components that needs to be documented, like * What's STUNs role and how do we implement it alongside with NAT (maybe Vovida.org stun server) * What is the function of NAT=yes in sip.conf ;-) * Do I have any use of SIProxd in handling this? * Where do I place my asterisk - outside the firewall giving inside clients a problem or inside the firewall, making it a problem for Asterisk to connect to outside SIP servers, like fwd.pulver.com * What's the role of "Outbound SIP server"-support in clients?...
2003 Nov 21
0
One way sound
Hi, I'm having trouble with asterisk: I can't hear both way of a call. here is my current architecture: grandstream -> siproxd -> asterisk -> pstn As I'm just testing for the moment, evrything is on my LAN. I know that there is no need to have a proxy here. But Later, the asterisk will be on a public IP outside of my LAn, so I'm practising... And there is no direct communication between asterisk and my phone...