Couple of suggestions: 1. On the subject of DHCP. On the router interface to which phones are connected, you should be able to specify ip helper address. This is the address of dhcp server. At least, this is the way it is done on cisco routers. 2. On the subject of firewall, I found that turning alg and sip helpers off is not enough. You also, need to configure pure NAT (no PAT). 3. On the subject of call dying after 15 seconds, I would check on sipx server, if the registration took place. I found many times that this was a symptom of unregistered phone.
On Thu, Apr 26, 2012 at 3:18 PM, Simon Brûlé <[email protected]> wrote: > Thank you very much for your help and I will see what I can do to make it > work. > > > 2012/4/26 Michael Picher <[email protected]> >> >> quite possibly... >> >> given the proper alignment of the starts and your reading of dhcp >> materials... >> >> >> On Thu, Apr 26, 2012 at 2:51 PM, Simon Brûlé <[email protected]> >> wrote: >>> >>> Sorry i forgot to mention that i wanted to move my server into the DMZ on >>> the adresse 192.168.180.xxx. Would this work? >>> >>> >>> 2012/4/26 Michael Picher <[email protected]> >>>> >>>> Well, the asterisk server is not on 192.168.175.x... it's on >>>> 192.168.180.x. so i would say no, this will not work. >>>> >>>> why don't you just manually configure the ip of the phone and dns to >>>> point to openuc... >>>> >>>> mike >>>> >>>> >>>> >>>> On Thu, Apr 26, 2012 at 2:02 PM, Simon Brûlé >>>> <[email protected]> wrote: >>>>> >>>>> Since the asterisk server is already giving DHCP on the switch with the >>>>> phone could I change the dhcpd.conf to give information that would >>>>> redirect >>>>> to my sipxecs server, example change those setting: >>>>> >>>>> option tftp-server-name "Asterisk server IP" >>>>> option sip-servers-name "Asterisk server IP" >>>>> >>>>> to >>>>> >>>>> option tftp-server-name "SipXecs server IP" >>>>> option sip-servers-name "SipXecs server IP" >>>>> >>>>> Is the asterisk server going to be able to redirect all the logging >>>>> request to my sipxecs server? >>>>> >>>>> 2012/4/26 Michael Picher <[email protected]> >>>>>> >>>>>> well then just setup another dhcp server on that leg and hand out >>>>>> similar information to the dhcp that is on the sipxecs server... hint, >>>>>> look >>>>>> at /etc/dhcpd.conf >>>>>> >>>>>> >>>>>> On Thu, Apr 26, 2012 at 1:39 PM, Simon Brûlé >>>>>> <[email protected]> wrote: >>>>>>> >>>>>>> I read a lot on the DHCP Relay thing and I though of a setup that I >>>>>>> think could solve the problem I am having, the thing is I can't test it >>>>>>> since I need to modify the running config. >>>>>>> >>>>>>> If you check the picture of the network I sent the asterisk server >>>>>>> got 2 interface 1 in the 192.168.180.xxx and one in the 192.168.62.xxx. >>>>>>> The >>>>>>> 180.xxx is the DMZ and the 62.xxx is the asterisk dhcp that is giving >>>>>>> it to >>>>>>> the phone. I though I could put the SipXecs server in the DMZ so the >>>>>>> computer could have access to it as easily as they have access to the >>>>>>> asterisk server right now and use the DHCP Relay on the asterisk server >>>>>>> so >>>>>>> the phone in the 192.168.62.xxx would get their dhcp request relay to >>>>>>> the >>>>>>> sipxecs server in the 192.168.180.xxx. >>>>>>> >>>>>>> The only thing is i am not sure it will work and i can't test it. >>>>>>> >>>>>>> 2012/4/26 Michael Picher <[email protected]> >>>>>>>> >>>>>>>> sure... this really isn't a voip phone thing, it's a basic network >>>>>>>> infrastructure thing... >>>>>>>> >>>>>>>> it's ok, i've seen people who've been doing this for years not know >>>>>>>> it too... >>>>>>>> >>>>>>>> >>>>>>>> On Thu, Apr 26, 2012 at 8:49 AM, Simon Brûlé >>>>>>>> <[email protected]> wrote: >>>>>>>>> >>>>>>>>> Thank you for your quick answer and I am sorry for the basic >>>>>>>>> question but I am only a student and have near to zero experience >>>>>>>>> with VOIP >>>>>>>>> Phone. >>>>>>>>> >>>>>>>>> >>>>>>>>> 2012/4/26 Michael Picher <[email protected]> >>>>>>>>>> >>>>>>>>>> I'd suggest you read up on DHCP a bit... >>>>>>>>>> >>>>>>>>>> DHCP has a feature call DHCP relay you might be interested in. >>>>>>>>>> DHCP is of course a layer 2 protocol and needs to be configured at >>>>>>>>>> the >>>>>>>>>> network level (i.e., in a switch, router or firewall). DHCP >>>>>>>>>> requests are >>>>>>>>>> relayed to a DHCP server which must also be configured >>>>>>>>>> to serve up addresses >>>>>>>>>> for this other IP range. >>>>>>>>>> >>>>>>>>>> This is of course not a sipXecs issue, just a simple networking >>>>>>>>>> issue. This is pretty basic stuff from a networking perspective. >>>>>>>>>> >>>>>>>>>> http://en.wikipedia.org/wiki/Dhcpd >>>>>>>>>> >>>>>>>>>> Happy reading. >>>>>>>>>> >>>>>>>>>> Mike >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Wed, Apr 25, 2012 at 4:51 PM, Simon Brûlé >>>>>>>>>> <[email protected]> wrote: >>>>>>>>>>> >>>>>>>>>>> I joined to this post a cleared version of the Network that i >>>>>>>>>>> made earlier. On this one everything on the right is the company >>>>>>>>>>> phone >>>>>>>>>>> network as it is at the moment and everything is working fine. >>>>>>>>>>> Everything on >>>>>>>>>>> the left is what I am working on. Under the router everything is >>>>>>>>>>> perfect, >>>>>>>>>>> they can call each other and I made a Trunk Sip with the Asterisk >>>>>>>>>>> server of >>>>>>>>>>> the company so I can call on my cellphone and other numbers. The >>>>>>>>>>> problem is >>>>>>>>>>> the computer in the middle. He can register to the Existing >>>>>>>>>>> Asterisk server >>>>>>>>>>> and everything is working fine, but when the computer register on >>>>>>>>>>> my sipxecs >>>>>>>>>>> server he can register he can receive call from my phone but that's >>>>>>>>>>> all. He >>>>>>>>>>> can call my hardphone but there is no sound and after 15 sec the >>>>>>>>>>> call close >>>>>>>>>>> and I tried calling my cellphone and same thing my cellphone >>>>>>>>>>> receive the >>>>>>>>>>> call but no sound and the call drop after 15 sec. I know it's a >>>>>>>>>>> problem with >>>>>>>>>>> the router but I don't know what I can do more then this. The >>>>>>>>>>> server is in >>>>>>>>>>> the router DMZ there is no firewall on the server and on the router. >>>>>>>>>>> >>>>>>>>>>> The highes bar on the picture i joined is the Main Router of the >>>>>>>>>>> company. I put a static route so all the traffic going to >>>>>>>>>>> 192.168.0.0/24 >>>>>>>>>>> would be redirect on the 192.168.175.22. >>>>>>>>>>> >>>>>>>>>>> I tried replacing the router of my subnet on the left by a switch >>>>>>>>>>> but I only had a Layer 2 switch so I couldn't manage anything >>>>>>>>>>> everything was >>>>>>>>>>> working on my subnet on my switch but I could't use the computer at >>>>>>>>>>> all at >>>>>>>>>>> that point since the switch couldn't route the call to the server. >>>>>>>>>>> >>>>>>>>>>> The SipXecs is not suppose to be there it's just for the testing >>>>>>>>>>> purpose, but what I am asking my self right now is where can I put >>>>>>>>>>> the >>>>>>>>>>> server so the company phone (on the right) would get DHCP and >>>>>>>>>>> config from >>>>>>>>>>> the server but the server could have access to the asterisk server >>>>>>>>>>> for the >>>>>>>>>>> Sip Trunk and the network for the employe softphone so they can >>>>>>>>>>> connect. >>>>>>>>>>> >>>>>>>>>>> I know what I am asking is more then just a solution to a problem >>>>>>>>>>> but any help with my setup would really be appreciated. >>>>>>>>>>> >>>>>>>>>>> Thank you. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> 2012/4/25 Simon Brûlé <[email protected]> >>>>>>>>>>>> >>>>>>>>>>>> If instead of the router with the incompatible NAT I put a >>>>>>>>>>>> switch (because ultimately that is the setup I am trying to put in >>>>>>>>>>>> the >>>>>>>>>>>> company) what are the setting that would have to change? >>>>>>>>>>>> >>>>>>>>>>>> Can a Layer 2 switch do the job? >>>>>>>>>>>> >>>>>>>>>>>> 2012/4/25 Tony Graziano <[email protected]> >>>>>>>>>>>>> >>>>>>>>>>>>> I think this is still indicative of your firewall NAT type. I >>>>>>>>>>>>> have always had mixed results with Jits (sip communicator), but I >>>>>>>>>>>>> don't >>>>>>>>>>>>> think that is your issue. >>>>>>>>>>>>> >>>>>>>>>>>>> 2012/4/25 Simon Brûlé <[email protected]> >>>>>>>>>>>>>> >>>>>>>>>>>>>> I discovered that when I activate the Sip ALG on the router >>>>>>>>>>>>>> that my sipXecs server is connected on when I do call from the >>>>>>>>>>>>>> hardphone I >>>>>>>>>>>>>> get audio from the 2 sides instead only from 1 but when i call >>>>>>>>>>>>>> with the >>>>>>>>>>>>>> softphone it still doesn't work. >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> This is because sipx is setup to handle NAT traversal using >>>>>>>>>>>>> media relay in a specific way. >>>>>>>>>>>>> >>>>>>>>>>>>> 1. You need a properly configured firewall where sipx is, which >>>>>>>>>>>>> has been covered with you. >>>>>>>>>>>>> 2. sipx does not support ALG's in any way. >>>>>>>>>>>>> >>>>>>>>>>>>> Alternatively you can get a full featured SBC. >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> 2012/4/24 Simon Brûlé <[email protected]> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> I did a lot of thing to try to solve this problem. Now when >>>>>>>>>>>>>>> the hardphone init the call I have sound on both side but when >>>>>>>>>>>>>>> the softphone >>>>>>>>>>>>>>> init the call I got no sound at all. When i call with the >>>>>>>>>>>>>>> softphone the RTP >>>>>>>>>>>>>>> section in Wireshark are now targeting the good address instead >>>>>>>>>>>>>>> of the phone >>>>>>>>>>>>>>> address that he couldn't reach but the Record route when i call >>>>>>>>>>>>>>> with the >>>>>>>>>>>>>>> softphone didn't change it's still the 192.168.0.1. What can I >>>>>>>>>>>>>>> do to change >>>>>>>>>>>>>>> that? >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> 2012/4/24 Simon Brûlé <[email protected]> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> How can I change that so the ACK would be sent out to >>>>>>>>>>>>>>>> 192.168.175.22 that is the address the softphone can reach? >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> 2012/4/22 Joegen Baclor <[email protected]> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> I've looked at the traces. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> 1. No RTP seen It is because sipX did not proxy RTP this >>>>>>>>>>>>>>>>> is evident by simply looking at the IP in 200 OK SDP. >>>>>>>>>>>>>>>>> 2. Softphone is unable to route mid-dialog requests >>>>>>>>>>>>>>>>> because sipX did not set the record-route to the global IP of >>>>>>>>>>>>>>>>> the NAT >>>>>>>>>>>>>>>>> router. ACK is sent out to 192.168.0.1 which is not routable >>>>>>>>>>>>>>>>> from the >>>>>>>>>>>>>>>>> softphone as far as I could tell. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> This is a classic subnet confusion and sipX thinks that the >>>>>>>>>>>>>>>>> softphones subnet is not Natted. I'll let the subnet gurus >>>>>>>>>>>>>>>>> take it from >>>>>>>>>>>>>>>>> here. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> On 04/20/2012 10:14 PM, Michael Picher wrote: >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> try to capture traffic on the sipxecs server itself to see >>>>>>>>>>>>>>>>> where the RTP stream is headed... >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> this will probably help. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> from your description i can't tell what's going on >>>>>>>>>>>>>>>>> (although i'm sure it's clear in your mind). draw pictures >>>>>>>>>>>>>>>>> and this might >>>>>>>>>>>>>>>>> help. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> mike >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> On Fri, Apr 20, 2012 at 10:04 AM, Simon Brûlé >>>>>>>>>>>>>>>>> <[email protected]> wrote: >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> Those are from a call I made from the Hardphone to the >>>>>>>>>>>>>>>>>> softphone. In the file you can see that when I do the call >>>>>>>>>>>>>>>>>> from the >>>>>>>>>>>>>>>>>> Hardphone to the Softphone my computer receive RTP from the >>>>>>>>>>>>>>>>>> hardphone. When >>>>>>>>>>>>>>>>>> I do a call from the softphone to the hardphone both side >>>>>>>>>>>>>>>>>> receive nothing. >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> In the RTP when I receive something on my computer it's >>>>>>>>>>>>>>>>>> from 192.168.175.22 that's normal it's the WAN adresse of >>>>>>>>>>>>>>>>>> the router where >>>>>>>>>>>>>>>>>> the hardphone are plug into but when I send something it's >>>>>>>>>>>>>>>>>> sending it to the >>>>>>>>>>>>>>>>>> 192.168.0.253 that's the address of the hardphone directly >>>>>>>>>>>>>>>>>> (behind the nat >>>>>>>>>>>>>>>>>> of the E2500) maybe that's why the softphone never reach it. >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> Something weird is that when it's the softphone initiating >>>>>>>>>>>>>>>>>> the call the Hardphone can't reach it. But when the >>>>>>>>>>>>>>>>>> Hardphone initiate it he >>>>>>>>>>>>>>>>>> can transmit RTP to the softphone. >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> To answer your suggestion Michael the Wan address of the >>>>>>>>>>>>>>>>>> E2500 is set to get it from DHCPbut the router that give him >>>>>>>>>>>>>>>>>> DHCP is set to >>>>>>>>>>>>>>>>>> give the 192.168.175.22 only to the Mac address of the E2500 >>>>>>>>>>>>>>>>>> so it's the >>>>>>>>>>>>>>>>>> same thing. >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> 2012/4/20 Simon Brûlé <[email protected]> >>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>> Those are the file you resquested Joegen. One is from the >>>>>>>>>>>>>>>>>>> server when i did the call and the other is from the >>>>>>>>>>>>>>>>>>> Softphone on my >>>>>>>>>>>>>>>>>>> computer. The call was made from the softphone to the >>>>>>>>>>>>>>>>>>> hardphone that is on >>>>>>>>>>>>>>>>>>> the same router(E2500) as the sipxecs server. >>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>> 2012/4/20 Joegen Baclor <[email protected]> >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> I forgive all of you for not noticing that there is no >>>>>>>>>>>>>>>>>>>> ACK for the 200 OK. Forget RTP, it wont be there. :-) >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> Check the jitsi side if it is respecting the >>>>>>>>>>>>>>>>>>>> Record-Route presented in 200 Ok. Take a wireshark >>>>>>>>>>>>>>>>>>>> capture from the PC too >>>>>>>>>>>>>>>>>>>> and send it together with the one from sipx. >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> On 04/20/2012 06:14 AM, Michael Picher wrote: >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> or could be stun too if the media relay's just off in >>>>>>>>>>>>>>>>>>>> left field. >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> i would setup for static outside address if you haven't. >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> mike >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> On Thu, Apr 19, 2012 at 4:59 PM, Tony Graziano >>>>>>>>>>>>>>>>>>>> <[email protected]> wrote: >>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>> Most likely if that is the router sitting in front of >>>>>>>>>>>>>>>>>>>>> sipx, yes. It's a prerequisite for media relay to >>>>>>>>>>>>>>>>>>>>> function. >>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>> 1. Server behind nat has to be enabled. >>>>>>>>>>>>>>>>>>>>> 2. Support remote workers needs to be enabled (sip >>>>>>>>>>>>>>>>>>>>> trunking does not need to be enabled for remote workers, >>>>>>>>>>>>>>>>>>>>> media relay does). >>>>>>>>>>>>>>>>>>>>> 3. NAT for server needs to have access to a public IP >>>>>>>>>>>>>>>>>>>>> address by getting it via STUN server or manually >>>>>>>>>>>>>>>>>>>>> entering it. >>>>>>>>>>>>>>>>>>>>> 4. Firewall in front of sipx needs to be able to do >>>>>>>>>>>>>>>>>>>>> symmetrical outbound NAT (AON or FULL CONE NATE, at least >>>>>>>>>>>>>>>>>>>>> for the outbound >>>>>>>>>>>>>>>>>>>>> NAT of the sipx internal address) and have any SIP helper >>>>>>>>>>>>>>>>>>>>> turned off. >>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>> If the two routers are adjacent and you can route >>>>>>>>>>>>>>>>>>>>> (without NAT) you would simply do so and add the PC >>>>>>>>>>>>>>>>>>>>> subnet to the intranets >>>>>>>>>>>>>>>>>>>>> page in sipx ONLY IF it does not have to pass through NAT >>>>>>>>>>>>>>>>>>>>> (i.e route or site >>>>>>>>>>>>>>>>>>>>> to site vpn, etc.). >>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>> Your router is changing the ports and hence sipx is >>>>>>>>>>>>>>>>>>>>> expecting the audio to come back on a port that isnt >>>>>>>>>>>>>>>>>>>>> being sent by the >>>>>>>>>>>>>>>>>>>>> router. >>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>> 2012/4/19 Simon Brûlé <[email protected]> >>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>> So your saying the problem may come from the router I >>>>>>>>>>>>>>>>>>>>>> have (Linksys E2500) because it's not doing the >>>>>>>>>>>>>>>>>>>>>> symmetrical Nat so the RTP >>>>>>>>>>>>>>>>>>>>>> is getting lost? >>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>> 2012/4/19 Tony Graziano <[email protected]> >>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>> If there is NAT between your PC and the sipx server >>>>>>>>>>>>>>>>>>>>>>> then: >>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>> 1. The local firewall to your PC needs to have any >>>>>>>>>>>>>>>>>>>>>>> SIP helper or Application Layer Gateway "turned off". >>>>>>>>>>>>>>>>>>>>>>> 2. At your firewall where sipx is the SIP helper or >>>>>>>>>>>>>>>>>>>>>>> Application Layer Gateway " needs to be turned off AND >>>>>>>>>>>>>>>>>>>>>>> the NAT type for the >>>>>>>>>>>>>>>>>>>>>>> outbound NAT from the sipx server needs to be >>>>>>>>>>>>>>>>>>>>>>> symmterical. Home brew and >>>>>>>>>>>>>>>>>>>>>>> residential routers usually will not do this. >>>>>>>>>>>>>>>>>>>>>>> 3. If you have an entry in your intranet subnets that >>>>>>>>>>>>>>>>>>>>>>> includes the PC network number it must be listed ONLY >>>>>>>>>>>>>>>>>>>>>>> IF IF DOES NOT GO >>>>>>>>>>>>>>>>>>>>>>> THROUGH NAT. >>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>> One of these three (or a combination of them) is >>>>>>>>>>>>>>>>>>>>>>> keeping RTP from flowing. >>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>> 2012/4/19 Simon Brûlé <[email protected]> >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>> I used the following command on the SipXecs >>>>>>>>>>>>>>>>>>>>>>>> server tcpdump -n -s 0 -i any -w filename.cap and then >>>>>>>>>>>>>>>>>>>>>>>> I transfer it on my >>>>>>>>>>>>>>>>>>>>>>>> computer so I could open it with my Wireshark and have >>>>>>>>>>>>>>>>>>>>>>>> a look at it. >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>> I joined the file to this e-mail so you could take a >>>>>>>>>>>>>>>>>>>>>>>> look and tell me what you think. This one is from a >>>>>>>>>>>>>>>>>>>>>>>> call I did from the >>>>>>>>>>>>>>>>>>>>>>>> softphone to the Hardphone where this one bugged like >>>>>>>>>>>>>>>>>>>>>>>> I described earlier. >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>> Thanks. >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>> 2012/4/19 Simon Brûlé <[email protected]> >>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>> As i dig more and more in Wireshark i came to the >>>>>>>>>>>>>>>>>>>>>>>>> conclusion that the Wireshark information that I just >>>>>>>>>>>>>>>>>>>>>>>>> sent you is pretty >>>>>>>>>>>>>>>>>>>>>>>>> much useless as I now see it. I will keep looking for >>>>>>>>>>>>>>>>>>>>>>>>> some piece of >>>>>>>>>>>>>>>>>>>>>>>>> information that could help. >>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>> Thanks. >>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>> 2012/4/19 Simon Brûlé <[email protected]> >>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>> My Computer is connected in the Lan of the company >>>>>>>>>>>>>>>>>>>>>>>>>> and my E2500 is connected in this Lan too. My Server >>>>>>>>>>>>>>>>>>>>>>>>>> SipXecs and my >>>>>>>>>>>>>>>>>>>>>>>>>> hardphone are on the E2500. >>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>> So there is an other router between my computer >>>>>>>>>>>>>>>>>>>>>>>>>> and the router that have my Server connected on it. >>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>> For the Wireshark part when I answer the phonecall >>>>>>>>>>>>>>>>>>>>>>>>>> I do from the softphone to my hardĥone those request >>>>>>>>>>>>>>>>>>>>>>>>>> are coming in until i >>>>>>>>>>>>>>>>>>>>>>>>>> close the call. >>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>> 109 19.596310 192.168.175.22 192.168.175.136 >>>>>>>>>>>>>>>>>>>>>>>>>> SIP/SDP 1466 Status: 200 OK, with session description >>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>> Status-Code: 200 >>>>>>>>>>>>>>>>>>>>>>>>>> [resent packet : True] >>>>>>>>>>>>>>>>>>>>>>>>>> [Suspected resend of frame:104] >>>>>>>>>>>>>>>>>>>>>>>>>> [Request Frame : 57] >>>>>>>>>>>>>>>>>>>>>>>>>> [Response Time (ms): 10950] >>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>> followed by this one: >>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>> 110 19.598495 192.168.175.136 192.168.0.1 SIP 714 >>>>>>>>>>>>>>>>>>>>>>>>>> Request: ACK >>>>>>>>>>>>>>>>>>>>>>>>>> sip:[email protected]:5060;transport=tcp >>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>> Request-Line: ACK >>>>>>>>>>>>>>>>>>>>>>>>>> sip:[email protected]:5060;transport=tcp SIP/2.0 >>>>>>>>>>>>>>>>>>>>>>>>>> Method: ACK >>>>>>>>>>>>>>>>>>>>>>>>>> Request-URI: >>>>>>>>>>>>>>>>>>>>>>>>>> sip:[email protected]:5060;transport=tcp >>>>>>>>>>>>>>>>>>>>>>>>>> [Resent Packet: False] >>>>>>>>>>>>>>>>>>>>>>>>>> [Request Frame: 105] >>>>>>>>>>>>>>>>>>>>>>>>>> [Response Time (ms): 512] >>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>> All those test have been done on Wireshark on the >>>>>>>>>>>>>>>>>>>>>>>>>> Computer with the Softphone on it. And the >>>>>>>>>>>>>>>>>>>>>>>>>> 192.168.0.253 that you see is the >>>>>>>>>>>>>>>>>>>>>>>>>> hardphone IP adresse. >>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>> 2012/4/19 Gerald Drouillard >>>>>>>>>>>>>>>>>>>>>>>>>> <[email protected]> >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> On 4/19/2012 3:25 PM, Simon Brûlé wrote: >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> How can I do a capture with wireshark on the >>>>>>>>>>>>>>>>>>>>>>>>>>> SipXecs server? >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> If you google a little you will find it. >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> About the ALG you think that the other Router >>>>>>>>>>>>>>>>>>>>>>>>>>> that give the DHCP to my Laptop and the Wan adresse >>>>>>>>>>>>>>>>>>>>>>>>>>> of my router would have >>>>>>>>>>>>>>>>>>>>>>>>>>> the Sip ALG activate? >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> That would be the only thing inbetween your >>>>>>>>>>>>>>>>>>>>>>>>>>> softphone and the sipx server... right? >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> http://screenshots.portforward.com/Cisco/Linksys_E2500/Management.htm >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> 2012/4/19 Gerald Drouillard >>>>>>>>>>>>>>>>>>>>>>>>>>> <[email protected]> >>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>> On 4/19/2012 2:58 PM, Simon Brûlé wrote: >>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>> I added 192.168.175.0/24 to the intranet subnet >>>>>>>>>>>>>>>>>>>>>>>>>>>> and I still have the same problem. >>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>> 2012/4/19 Gerald Drouillard >>>>>>>>>>>>>>>>>>>>>>>>>>>> <[email protected]> >>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>> On 4/19/2012 2:37 PM, Simon Brûlé wrote: >>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>> Hi, I know I already posted something very >>>>>>>>>>>>>>>>>>>>>>>>>>>>> similiar to this problem but I haven't found a >>>>>>>>>>>>>>>>>>>>>>>>>>>>> solution to it so here i am >>>>>>>>>>>>>>>>>>>>>>>>>>>>> reposting my problem but with more precision this >>>>>>>>>>>>>>>>>>>>>>>>>>>>> time. >>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>> I have a softphone (Jitis) on a Ubuntu 11.10 >>>>>>>>>>>>>>>>>>>>>>>>>>>>> installation connected to the network of the >>>>>>>>>>>>>>>>>>>>>>>>>>>>> company. >>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>> I have a router Linksys E2500 connected to the >>>>>>>>>>>>>>>>>>>>>>>>>>>>> same network. The laptop have the adresse >>>>>>>>>>>>>>>>>>>>>>>>>>>>> 192.168.175.136 giving by dhcp and >>>>>>>>>>>>>>>>>>>>>>>>>>>>> the router have the adresse 192.168.175.22 giving >>>>>>>>>>>>>>>>>>>>>>>>>>>>> by dhcp too. >>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>> On that router I have my SipXecs server and 2 >>>>>>>>>>>>>>>>>>>>>>>>>>>>> hardphones connected. My SipXecs server have the >>>>>>>>>>>>>>>>>>>>>>>>>>>>> adresse 192.168.0.1, the >>>>>>>>>>>>>>>>>>>>>>>>>>>>> internal adresse of the router is 192.168.0.2 and >>>>>>>>>>>>>>>>>>>>>>>>>>>>> the 2 hardphones have dhcp >>>>>>>>>>>>>>>>>>>>>>>>>>>>> adresse given by the SipXecs server. >>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>> The problem is the following : >>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>> When I call with the softphone that is >>>>>>>>>>>>>>>>>>>>>>>>>>>>> registered on the SipXecs server to a hardphone >>>>>>>>>>>>>>>>>>>>>>>>>>>>> that is registered on the >>>>>>>>>>>>>>>>>>>>>>>>>>>>> server too the call get there but there is no >>>>>>>>>>>>>>>>>>>>>>>>>>>>> sound on either side and the >>>>>>>>>>>>>>>>>>>>>>>>>>>>> hardphone is still flashing like the call is >>>>>>>>>>>>>>>>>>>>>>>>>>>>> still coming and i didn't >>>>>>>>>>>>>>>>>>>>>>>>>>>>> answer it. By the way the phone is a Polycom 321. >>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>> When i call from the Hardphone to the softphone >>>>>>>>>>>>>>>>>>>>>>>>>>>>> everything is fine except that the softphone >>>>>>>>>>>>>>>>>>>>>>>>>>>>> can't do any sound but he can >>>>>>>>>>>>>>>>>>>>>>>>>>>>> hear the hardphone. >>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>> The firewall on the SipXecs server is disabled, >>>>>>>>>>>>>>>>>>>>>>>>>>>>> the firewall on the router is disabled too, the >>>>>>>>>>>>>>>>>>>>>>>>>>>>> SipXecs server is in the DMZ >>>>>>>>>>>>>>>>>>>>>>>>>>>>> of the router, Sip ALG is disabled on the router >>>>>>>>>>>>>>>>>>>>>>>>>>>>> too. >>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>> On the SipXecs server System --> Internet >>>>>>>>>>>>>>>>>>>>>>>>>>>>> calling I have the Nat traversal enabled and the >>>>>>>>>>>>>>>>>>>>>>>>>>>>> Server behind nat. The >>>>>>>>>>>>>>>>>>>>>>>>>>>>> intranet domain is the default one and for the >>>>>>>>>>>>>>>>>>>>>>>>>>>>> intranet i put the >>>>>>>>>>>>>>>>>>>>>>>>>>>>> 192.168.0.0/24. >>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>> You may need to add 192.168.175.0/24 also if it >>>>>>>>>>>>>>>>>>>>>>>>>>>>> is local. >>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>> I have seen polycom phones act like this >>>>>>>>>>>>>>>>>>>>>>>>>>>> before. In my case: >>>>>>>>>>>>>>>>>>>>>>>>>>>> The user portion of a SIP dialog MUST match the >>>>>>>>>>>>>>>>>>>>>>>>>>>> ACK and if it does not match exactly the phone >>>>>>>>>>>>>>>>>>>>>>>>>>>> will ignore it. Without a >>>>>>>>>>>>>>>>>>>>>>>>>>>> valid ACK the phone won’t start sending RTP and >>>>>>>>>>>>>>>>>>>>>>>>>>>> the UI won’t show the call >>>>>>>>>>>>>>>>>>>>>>>>>>>> as answered. You may want to do a capture on the >>>>>>>>>>>>>>>>>>>>>>>>>>>> sipx server and look at >>>>>>>>>>>>>>>>>>>>>>>>>>>> the results with wireshark. >>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>> Sounds like you may still have ALG at the >>>>>>>>>>>>>>>>>>>>>>>>>>>> gateway on the 192.168.175.0 network. >>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>> -- >>>>>>>>>>>>>>>>>>>>>>>>>>>> Regards >>>>>>>>>>>>>>>>>>>>>>>>>>>> -------------------------------------- >>>>>>>>>>>>>>>>>>>>>>>>>>>> Gerald Drouillard >>>>>>>>>>>>>>>>>>>>>>>>>>>> Technology Architect >>>>>>>>>>>>>>>>>>>>>>>>>>>> Drouillard & Associates, Inc. >>>>>>>>>>>>>>>>>>>>>>>>>>>> http://www.Drouillard.biz >>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>>>>>>>>>>>>> sipx-users mailing list >>>>>>>>>>>>>>>>>>>>>>>>>>>> [email protected] >>>>>>>>>>>>>>>>>>>>>>>>>>>> List Archive: >>>>>>>>>>>>>>>>>>>>>>>>>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>>>>>>>>>>>> sipx-users mailing list >>>>>>>>>>>>>>>>>>>>>>>>>>> [email protected] >>>>>>>>>>>>>>>>>>>>>>>>>>> List Archive: >>>>>>>>>>>>>>>>>>>>>>>>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> -- >>>>>>>>>>>>>>>>>>>>>>>>>>> Regards >>>>>>>>>>>>>>>>>>>>>>>>>>> -------------------------------------- >>>>>>>>>>>>>>>>>>>>>>>>>>> Gerald Drouillard >>>>>>>>>>>>>>>>>>>>>>>>>>> Technology Architect >>>>>>>>>>>>>>>>>>>>>>>>>>> Drouillard & Associates, Inc. >>>>>>>>>>>>>>>>>>>>>>>>>>> http://www.Drouillard.biz >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>>>>>>>>>>>> sipx-users mailing list >>>>>>>>>>>>>>>>>>>>>>>>>>> [email protected] >>>>>>>>>>>>>>>>>>>>>>>>>>> List Archive: >>>>>>>>>>>>>>>>>>>>>>>>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>>>>>>>>> sipx-users mailing list >>>>>>>>>>>>>>>>>>>>>>>> [email protected] >>>>>>>>>>>>>>>>>>>>>>>> List Archive: >>>>>>>>>>>>>>>>>>>>>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>> -- >>>>>>>>>>>>>>>>>>>>>>> ~~~~~~~~~~~~~~~~~~ >>>>>>>>>>>>>>>>>>>>>>> Tony Graziano, Manager >>>>>>>>>>>>>>>>>>>>>>> Telephone: 434.984.8430 >>>>>>>>>>>>>>>>>>>>>>> sip: [email protected] >>>>>>>>>>>>>>>>>>>>>>> Fax: 434.465.6833 >>>>>>>>>>>>>>>>>>>>>>> ~~~~~~~~~~~~~~~~~~ >>>>>>>>>>>>>>>>>>>>>>> Linked-In Profile: >>>>>>>>>>>>>>>>>>>>>>> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>>>>>>>>>>>>>>>>>>>>>> Ask about our Internet Fax services! >>>>>>>>>>>>>>>>>>>>>>> ~~~~~~~~~~~~~~~~~~ >>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>>>>>>>>>>>>>>>>>>>>> Telephone: 434.984.8426 >>>>>>>>>>>>>>>>>>>>>>> sip: [email protected] >>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>> Helpdesk Customers: http://myhelp.myitdepartment.net >>>>>>>>>>>>>>>>>>>>>>> Blog: http://blog.myitdepartment.net >>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>>>>>>>> sipx-users mailing list >>>>>>>>>>>>>>>>>>>>>>> [email protected] >>>>>>>>>>>>>>>>>>>>>>> List Archive: >>>>>>>>>>>>>>>>>>>>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>>>>>>> sipx-users mailing list >>>>>>>>>>>>>>>>>>>>>> [email protected] >>>>>>>>>>>>>>>>>>>>>> List Archive: >>>>>>>>>>>>>>>>>>>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>> -- >>>>>>>>>>>>>>>>>>>>> ~~~~~~~~~~~~~~~~~~ >>>>>>>>>>>>>>>>>>>>> Tony Graziano, Manager >>>>>>>>>>>>>>>>>>>>> Telephone: 434.984.8430 >>>>>>>>>>>>>>>>>>>>> sip: [email protected] >>>>>>>>>>>>>>>>>>>>> Fax: 434.465.6833 >>>>>>>>>>>>>>>>>>>>> ~~~~~~~~~~~~~~~~~~ >>>>>>>>>>>>>>>>>>>>> Linked-In Profile: >>>>>>>>>>>>>>>>>>>>> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>>>>>>>>>>>>>>>>>>>> Ask about our Internet Fax services! >>>>>>>>>>>>>>>>>>>>> ~~~~~~~~~~~~~~~~~~ >>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>>>>>>>>>>>>>>>>>>> Telephone: 434.984.8426 >>>>>>>>>>>>>>>>>>>>> sip: [email protected] >>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>> Helpdesk Customers: http://myhelp.myitdepartment.net >>>>>>>>>>>>>>>>>>>>> Blog: http://blog.myitdepartment.net >>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>>>>>> sipx-users mailing list >>>>>>>>>>>>>>>>>>>>> [email protected] >>>>>>>>>>>>>>>>>>>>> List Archive: >>>>>>>>>>>>>>>>>>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> -- >>>>>>>>>>>>>>>>>>>> Michael Picher, Director of Technical Services >>>>>>>>>>>>>>>>>>>> eZuce, Inc. >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> 300 Brickstone Square >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> Suite 201 >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> Andover, MA. 01810 >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> O.978-296-1005 X2015 >>>>>>>>>>>>>>>>>>>> M.207-956-0262 >>>>>>>>>>>>>>>>>>>> @mpicher <http://twitter.com/mpicher> >>>>>>>>>>>>>>>>>>>> www.ezuce.com >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> ------------------------------------------------------------------------------------------------------------ >>>>>>>>>>>>>>>>>>>> There are 10 kinds of people in the world, those who >>>>>>>>>>>>>>>>>>>> understand binary and those who don't. >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>>>>> sipx-users mailing list >>>>>>>>>>>>>>>>>>>> [email protected] >>>>>>>>>>>>>>>>>>>> List Archive: >>>>>>>>>>>>>>>>>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>>>>> sipx-users mailing list >>>>>>>>>>>>>>>>>>>> [email protected] >>>>>>>>>>>>>>>>>>>> List Archive: >>>>>>>>>>>>>>>>>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>>> sipx-users mailing list >>>>>>>>>>>>>>>>>> [email protected] >>>>>>>>>>>>>>>>>> List Archive: >>>>>>>>>>>>>>>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> -- >>>>>>>>>>>>>>>>> Michael Picher, Director of Technical Services >>>>>>>>>>>>>>>>> eZuce, Inc. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> 300 Brickstone Square >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Suite 201 >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Andover, MA. 01810 >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> O.978-296-1005 X2015 >>>>>>>>>>>>>>>>> M.207-956-0262 >>>>>>>>>>>>>>>>> @mpicher <http://twitter.com/mpicher> >>>>>>>>>>>>>>>>> www.ezuce.com >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> ------------------------------------------------------------------------------------------------------------ >>>>>>>>>>>>>>>>> There are 10 kinds of people in the world, those who >>>>>>>>>>>>>>>>> understand binary and those who don't. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>> sipx-users mailing list >>>>>>>>>>>>>>>>> [email protected] >>>>>>>>>>>>>>>>> List Archive: >>>>>>>>>>>>>>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>> sipx-users mailing list >>>>>>>>>>>>>>>>> [email protected] >>>>>>>>>>>>>>>>> List Archive: >>>>>>>>>>>>>>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>> sipx-users mailing list >>>>>>>>>>>>>> [email protected] >>>>>>>>>>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> -- >>>>>>>>>>>>> ~~~~~~~~~~~~~~~~~~ >>>>>>>>>>>>> Tony Graziano, Manager >>>>>>>>>>>>> Telephone: 434.984.8430 >>>>>>>>>>>>> sip: [email protected] >>>>>>>>>>>>> Fax: 434.465.6833 >>>>>>>>>>>>> ~~~~~~~~~~~~~~~~~~ >>>>>>>>>>>>> Linked-In Profile: >>>>>>>>>>>>> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>>>>>>>>>>>> Ask about our Internet Fax services! >>>>>>>>>>>>> ~~~~~~~~~~~~~~~~~~ >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> Please get a properly configured firewall in front of sipx and >>>>>>>>>>>>> confirm its operation. >>>>>>>>>>>>> >>>>>>>>>>>>> Thanks. >>>>>>>>>>>>> >>>>>>>>>>>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>>>>>>>>>>> Telephone: 434.984.8426 >>>>>>>>>>>>> sip: [email protected] >>>>>>>>>>>>> >>>>>>>>>>>>> Helpdesk Customers: http://myhelp.myitdepartment.net >>>>>>>>>>>>> Blog: http://blog.myitdepartment.net >>>>>>>>>>>>> >>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>> sipx-users mailing list >>>>>>>>>>>>> [email protected] >>>>>>>>>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _______________________________________________ >>>>>>>>>>> sipx-users mailing list >>>>>>>>>>> [email protected] >>>>>>>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> -- >>>>>>>>>> Michael Picher, Director of Technical Services >>>>>>>>>> eZuce, Inc. >>>>>>>>>> >>>>>>>>>> 300 Brickstone Square >>>>>>>>>> >>>>>>>>>> Suite 201 >>>>>>>>>> >>>>>>>>>> Andover, MA. 01810 >>>>>>>>>> >>>>>>>>>> O.978-296-1005 X2015 >>>>>>>>>> M.207-956-0262 >>>>>>>>>> @mpicher <http://twitter.com/mpicher> >>>>>>>>>> www.ezuce.com >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> ------------------------------------------------------------------------------------------------------------ >>>>>>>>>> There are 10 kinds of people in the world, those who understand >>>>>>>>>> binary and those who don't. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> sipx-users mailing list >>>>>>>>>> [email protected] >>>>>>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> sipx-users mailing list >>>>>>>>> [email protected] >>>>>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Michael Picher, Director of Technical Services >>>>>>>> eZuce, Inc. >>>>>>>> >>>>>>>> 300 Brickstone Square >>>>>>>> >>>>>>>> Suite 201 >>>>>>>> >>>>>>>> Andover, MA. 01810 >>>>>>>> >>>>>>>> O.978-296-1005 X2015 >>>>>>>> M.207-956-0262 >>>>>>>> @mpicher <http://twitter.com/mpicher> >>>>>>>> www.ezuce.com >>>>>>>> >>>>>>>> >>>>>>>> ------------------------------------------------------------------------------------------------------------ >>>>>>>> There are 10 kinds of people in the world, those who understand >>>>>>>> binary and those who don't. >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> sipx-users mailing list >>>>>>>> [email protected] >>>>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> sipx-users mailing list >>>>>>> [email protected] >>>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Michael Picher, Director of Technical Services >>>>>> eZuce, Inc. >>>>>> >>>>>> 300 Brickstone Square >>>>>> >>>>>> Suite 201 >>>>>> >>>>>> Andover, MA. 01810 >>>>>> >>>>>> O.978-296-1005 X2015 >>>>>> M.207-956-0262 >>>>>> @mpicher <http://twitter.com/mpicher> >>>>>> www.ezuce.com >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------------------------------------------ >>>>>> There are 10 kinds of people in the world, those who understand binary >>>>>> and those who don't. >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> sipx-users mailing list >>>>>> [email protected] >>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> sipx-users mailing list >>>>> [email protected] >>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> >>>> >>>> >>>> >>>> -- >>>> Michael Picher, Director of Technical Services >>>> eZuce, Inc. >>>> >>>> 300 Brickstone Square >>>> >>>> Suite 201 >>>> >>>> Andover, MA. 01810 >>>> >>>> O.978-296-1005 X2015 >>>> M.207-956-0262 >>>> @mpicher <http://twitter.com/mpicher> >>>> www.ezuce.com >>>> >>>> >>>> ------------------------------------------------------------------------------------------------------------ >>>> There are 10 kinds of people in the world, those who understand binary >>>> and those who don't. >>>> >>>> >>>> _______________________________________________ >>>> sipx-users mailing list >>>> [email protected] >>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >>> >>> >>> _______________________________________________ >>> sipx-users mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >> >> >> >> -- >> Michael Picher, Director of Technical Services >> eZuce, Inc. >> >> 300 Brickstone Square >> >> Suite 201 >> >> Andover, MA. 01810 >> >> O.978-296-1005 X2015 >> M.207-956-0262 >> @mpicher <http://twitter.com/mpicher> >> www.ezuce.com >> >> >> ------------------------------------------------------------------------------------------------------------ >> There are 10 kinds of people in the world, those who understand binary and >> those who don't. >> >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
