Re: [sipx-users] Help with SUBSCRIBE for MWI

From: sipx-users-boun...@list.sipfoundry.org [sipx-users-boun...@list.sipfoundry.org] On Behalf Of David Becker [david.bec...@itison-ikt.de] Hm, closer inspection reveals that the Siemens phone doesn't send an algorithm=md5 parameter. Also the contact add

Re: [sipx-users] SIPX Appliance

On 11/8/2010 2:30 PM, Luis Fernando Jaramillo wrote: > HI everybody. > > There are any appliance with Sipxecs embedded? > > None that I have heard of and defintiely none that I have seen from sipfoundry. I load them on a device from here: http://www.logicsupply.com/ No moving parts, flash based dr

[sipx-users] SIPX Appliance

HI everybody. There are any appliance with Sipxecs embedded? -- Luis Fernando ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/

Re: [sipx-users] Route by Caller-id ... freeswitch

you would forward from the phantom user to a sip uri in freeswitch with a port of 15060 (fs). Im not sure it wouldn't loop though, it's really an internal FS dialplan. An independent gateway with the proper logic could also act as an intermediary. Whether that be a patton device or a standalone FS

Re: [sipx-users] Soft phone

Was a bit busy privately, happens as well every now and then, but see below Paul Tony Graziano wrote > I've come to the same solution, that counterpath could do a better > job before releasing in making sure things work as expected. I won't > go down that road here... Valid point, the applica

Re: [sipx-users] Help with SUBSCRIBE for MWI

Hm, closer inspection reveals that the Siemens phone doesn't send an algorithm=md5 parameter. Also the contact address is doubled, is that problematic? Am 20.10.2010 10:53, schrieb David Becker: >I found this in the sipstatus.log: > "2010-10-20T08:46:36.635044Z":756:AUTH:DEBUG:test.voip.ikt-

Re: [sipx-users] Adding second route

> You can also leave the server untouched and add a route on your (SIP/RTP) > default-gateway (192.168.1.6): >  > ip route 172.16.30.0 255.255.255.0 192.168.1.1. > This solution has the advantage that the SipX box can be > replaced/rebuild/upgraded etc without having to think about the routing. G

Re: [sipx-users] Adding second route

You can also leave the server untouched and add a route on your (SIP/RTP) default-gateway (192.168.1.6): ip route 172.16.30.0 255.255.255.0 192.168.1.1. All your SIP/RTP traffic will flow as usuall. All the Web/Gui traffic will flow via 192.168.1.6 to 192.168.1.1 If you also send ICMP-redire

Re: [sipx-users] Route by Caller-id ... freeswitch

Rather than using an unmanaged gateway i was think of using a phantom user to forward the call to the custom freeswitch profile. then freeswitch will transfer it to a different user extensionso no loop if that works. I'll take a crack at it. if it works well it would be a great feature for

Re: [sipx-users] Adding second route

> This is just plain routing on linux and does not really have anything to do > with sipx. I posted this on sipx list because this is where the sipx experts are who would know if there might be problems in doing this, specifically with this software. _

Re: [sipx-users] Adding second route

Correct. But the reverse must also be true. The destination host must have a route back if it normally follows a different route, however a traceroute will show you the path it takes. This is just plain routing on linux and does not really have anything to do with sipx. ===

Re: [sipx-users] Adding second route

> When you use this command: > route add -net 172.16.30.0 netmask 255.255.255.0 gw 192.168.1.1 >  > All Traffic sent by your sipx that is destined to the network 172.16.30.0 > is sent over the gateway 192.168.1.1 all other traffic is not affected at > all. To confirm then, traffic originated from

Re: [sipx-users] Localization by Group?

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <54100> Message-ID: Given Doug's resply above Should I raise a JIRA improvement request to get this in the Web UI? Thanks Abdul

Re: [sipx-users] Route by Caller-id ... freeswitch

Matt, as far as i understand, you can just use caller_id_number instead of destination_number. See http://wiki.freeswitch.org/wiki/Dialplan_XML#Condition. You could create separate FS profile, on sipx route a call to it (using unmanaged g

Re: [sipx-users] Adding second route

Sorry I have been very busa at the end of last week. Yes indeed. That is what I meant. When you use this command: route add -net 172.16.30.0 netmask 255.255.255.0 gw 192.168.1.1 All Traffic sent by your sipx that is destined to the network 172.16.30.0 is sent over the gateway 192.168.1.1 all oth