search for: outboundproxyport

Hi, I specific outboundproxyport=9097 in version 1.4.4, but it doesn't work. It still connects sip port 5060. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070625/3fbf28e8/attachment.htm

...e following lines. <backupProxy>192.168.20.2</backupProxy> <backupProxyPort>5060</backupProxyPort> <emergencyProxy>192.168.20.2</emergencyProxy> <emergencyProxyPort>5060</emergencyProxyPort> <outboundProxy>192.168.20.2</outboundProxy> <outboundProxyPort>5060</outboundProxyPort> and changed the registerWithProxy back to true as follows: <registerWithProxy>true</registerWithProxy> The phone no longer got stuck with "Registering", the red X is gone, and I can make and receive calls. I'm not sure if there are othe...

Looks like outboundproxyport doesn't support in 1.4.4 If you set the port, then it conflit with the one in "To URI" with host. I saw the code for outboundproxyport from the source, but is it a bug? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/a...

...;sipProfile> <sipProxies> <backupProxy>x.x.x.x</backupProxy> <backupProxyPort>5060</backupProxyPort> <emergencyProxy>x.x.x.x</emergencyProxy> <emergencyProxyPort>5060</emergencyProxyPort> <outboundProxy>z.z.z.z</outboundProxy> <outboundProxyPort>5060</outboundProxyPort> <registerWithProxy>true</registerWithProxy> </sipProxies> <sipCallFeatures> <cnfJoinEnabled>true</cnfJoinEnabled> <callForwardURI>x-cisco-serviceuri-cfwdall</callForwardURI> <callPickupURI>x-cisco-serviceuri-pic...

Hi, ? We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)). ? Recently we have bought a cisco 7942G IP phone. It currently has SIP 42.9-0-2SR1S firmware loaded on it. We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk. ? Do we need to

Anyone have the 7970 xml config for sip yet? Aaron

...oxies> <backupProxy></backupProxy> <backupProxyPort></backupProxyPort> <emergencyProxy></emergencyProxy> <emergencyProxyPort></emergencyProxyPort> <outboundProxy></outboundProxy> <outboundProxyPort></outboundProxyPort> <registerWithProxy>true</registerWithProxy> </sipProxies> <sipCallFeatures> <cnfJoinEnabled>true</cnfJoinEnabled> <callForwardURI>x--serviceuri-cfwdall</callForwardURI> <...

Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people

Hi there! I am having problems registering my 7970 hardphone with Asterisk 1.4(with FreePBX interface). I had an earlier post about trying to get it to work first with a 7970 emulator (Cisco IP Communicator) on the Asterisk Forum : http://forums.digium.com/viewtopic.php?t=19160 Instead I decided to try the real phone instead, and was able to advance further. The firmware was able to install

...at lately, so if you can test this and confirm wheather it works for you or not, I'll be grateful. If I get positive reports, we'll try to add this to chan_sip in CVS. It works like this: * Configure outboundproxy in the general section of sip.conf outboundproxy = <hostname or IP> outboundproxyport = <port #> (defaults to 5060) All SIP communication are now sent to the proxy IP If you configure localnet= networks, these are excluded, so only outbound traffic goes to the outbound proxy. If this works, we might try to add support for peer-specific outbound proxies to be able to handle...

...lines, audio works both ways. But calling in from my land line to the asterisk box via vbuzzer, I get no audio either way. The local sip client rings and when I answer the call, I see asterisk sending/receiving rtp packets. I couldn't find much information on asterisk's outboundproxy and outboundproxyport variables. They were in chan_sip2 last year and then merged with chan_sip. At that time, there were a glofal var, but now I think they can be a peer. I then tried just having the asterisk server answer incoming calls from vbuzzer, and I see it saying it's playing monkeys, but no audio. Agai...

Hi, I've been trying to get asterisk to use an outbound sip proxy. Putting the outboundproxyhost directive in the [general] section of sip.conf, but it doesn't seem to work. My expectation would be that by setting outboundproxy and outboundproxyport in that location, then all dial commans (or at least dial commands of the form Dial(SIP/asdf@asdf.com) or Dial(SIP/asdf@99.99.99.99)) would automatically be forwarded to the proxy. However, this is not happening. In other words, the INVITE packet generated by a Dial() are going straight to the des...

Hi All, If i have this kind of setup, what do i need to make it's load balance. [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | --------------------------------------------------------------------- | mysql cluster |

Hi, I have asterisk register two users (client-1, client-2) with a SIP proxy. I have the same two SIP client registered with asterisk. Now my dial plan setup is such that any call from client-1/client-2 is forwarded to the SIP proxy and the SIP proxy then takes the routing decision. Calls coming from SIP proxy will dial out the respective user. Asterisk is required to stay in the signaling as