Re: [sipx-users] disable shared on interface and cannot call out
On 12/13/2012 12:40 PM, De Soca wrote: Hello, I am running the latest yum updated version of 4.6. Can make and receive calls, voicemail and auto attendant work as expected. We have 2 interfaces connecting to different sub-accounts on voip.ms <http://voip.ms>. In an attempt to get calls made on the respective phones to call out on the correct DID, we set up 2 branches and associated each branch with one interface. Two groups were created and one each assigned to a branch. The users were assigned to one group or the other. No user is assigned to both groups. Everything continued working in this configuration except for the DID the call out is made on. This still continued to be incorrect. Are the users in the correct user group. In the user group have you assigned the caller id. -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] ACD call drop after 60 sec on hold
On 11/29/2012 4:30 PM, Ali Dashti wrote: > Geoff, > However I only have this problem when an inbound call comes from ACD, > otherwise its OK! > In other word when an agent picks up only an ACD call and puts it on > hold for more than 60 sec then on resume the call will drop. In direct > calling this problem doesn't exist! > Could I conclude this is not seesion timers on my or ITSP side? Can you call in and leave a voicemail longer than 60sec? -- Regards ------ Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] voip.ms impression
On 11/20/2012 1:22 PM, Burleigh, Matt wrote: I've recently(2 months) started using voip.ms and my support experience has been similar. Ever since Hurricane Sandy I've had numerous issues. I can usually restart SIP trunking to restore service and I don't always get an alarm from sipx. I've had some recent complaints of busy signals as well... That should go away if you can use IP auth. -- Regards ---------- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] voip.ms impression
On 11/19/2012 7:25 PM, Mark Wood wrote: We have been using voip.ms for 6-8 months so far and I want to get some feedback from users that have been at it for longer. Specifically we (main and subaccounts) experience times where our outbound calls just hang after dialing and sometimes abruptly connect, or sometimes not at all. When a subaccount calls to report problems to us and we check our home page it will show all of our accounts as 'not registered', and then slowly one by one they will show as 'registered'. We had an incident over the weekend with a security office that couldn't receive any inbound calls. We logged in to the voip.ms site to check the registrations and initiate a support ticket and the site again said 'not registered'. The instructions had us do and 'echo test' procedure and the results were the same as when they were routed to the subaccount. The support response 12 hours later was 'works for us' and then 'check your routers and firewalls'. Comments? Who are other good candidates for reselling VoIP like this model? Voip.ms is a great service if you don't need t.38. The only "problem" is that there is something fish going on if you use registration authentication. Time to time, you will loose registration. Fortunately you can configure your account to be IP authentication. http://www.drouillard.biz/blog/ip-authentication-with-sipxecs-and-voip-ms/ Their tech support has been fine with exception to the authentication issue which I believe is a capacity issue on their side. Other ITSP players you should check are http://www.voipinnovations.com/ and Appia if you have a lot of patience. -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Hacked SipXecs 4.4
On 11/16/2012 5:24 PM, Noah Mehl wrote: Shall I make a screencast to explain? No. You cannot cannot to a server port if there is nothing listening on that port. Your sipx server smtp server should only be listening on localhost:smtp not *:smtp Check the output of: lsof -i | grep LISTEN -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Hacked SipXecs 4.4
On 11/16/2012 1:57 PM, Noah Mehl wrote: Does nobody on the list know what SSH port forwarding is? I am running the first two commands from a remote machine (connecting to the sipxecs machine) in separate terminals to forward my local 25 port to the sipxecs box, and the 25 port on the sipxecs box locally. The third command is run locally on the remote machine. This exploit gives the remote machine access to port 25 on the SipXecs box even if all other ports are blocked. This could be used for any port that is blocked by firewall, ids, etc, if the remote machine has ssh access to the sipxecs box. ~Noah Do you understand that if your sipx smtp server is only running on localhost that you will not be able to connect to it via telnet/ssh/whatever? -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Hacked SipXecs 4.4
On 11/16/2012 12:45 PM, Noah Mehl wrote: Tony, I just figured out an exploit in 15 minutes with the help of Google http://www.semicomplete.com/articles/ssh-security/: <http://www.semicomplete.com/articles/ssh-security/:> $sudo ssh -vN -L25:localhost:25 PlcmSpIp@sipxecsip $sudo ssh -vN -R25:localhost:25 PlcmSpIp@sipxecsip $telnet localhost 25 Of course you can telnet to port 25 (smtp) on the server to localhost. You have sendmail running on local host. If your sendmail is configured properly you will not be able to access port 25 for another machine or the real ip address of the server. -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Hacked SipXecs 4.4
On 11/16/2012 10:07 AM, Noah Mehl wrote: > Todd, > > The private subnet is: 172.16.0.0 - 172.31.255.255. That IP is a public IP > address, which is part of AOL in Nevada I think. I actually have over 80 > different public IP address entries in my log using that user to SSH to my > SipXecs box. > > I understand that it's a phone system and not a firewall. However it's a > linux server, and IPtables is the best firewall in world, IMHO. I did have > SSH access open to the world, that was my choice. I have never been bitten > by this before. Either way, you should not be able to execute anything by > SSH'ing with the PlcmSpIp user, whether it's a public IP or not. > > I would recommend all your ssh servers have sshd_config with at least: AllowUsers user1name,user2name PermitRootLogin no I am also a big fan of fail2ban -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Sipx Trunk Gateway Configuration
On 11/6/2012 4:50 PM, Roman Gelfand wrote: > It would be great if it could. For instance, my provider is voip.ms. > They have many servers spread out throughout the country. new york > server became unavailable today. Would have been great if it could > automatically switch. You can do that with 2 voip.ms gateways in sipx. You just add the order of pref. in the dial plan(s). -- Regards ------ Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] High CPU sipXproxy (update #22)
On 11/5/2012 4:50 PM, Tony Graziano wrote: > I am seeing the following message within the rls logs: > > sipxrls:"OsSSLServerSocket SSL_accept SSL handshake error:\n SSL > error: 1 'error:0001:lib(0):func(0):reason(1)'" > sipxrls:"OsSSLServerSocket SSL_accept SSL handshake error:\n SSL > error: 336027900 'error:140760FC:SSL > routines:SSL23_GET_CLIENT_HELLO:unknown protocol'" > sipxrls:"SipPublishContentMgr::getContent no container found for key > 'sip:~~rl~C~~~id~xmpprlsclient... > > (as is relates to the RLS component) > > So I am wondering if someone can explain what the "unknown protocol" > means in this instance. The certificate was created in the exact way > it should have, by the system, one time at startup. I see > presenceserver says disabled but shows "running" in sipxconfig and if > I start manually via sipxproc it stays "running" (no change in > sipxconfig). > > I then tried to disable TLS and that broke nat traversal rules and > failed to start proxy, so that did not help. > > I tried deleting the tmp imdb.* files and restarting presence from > sipxconfig but that did not help. The ownership of the files and sizes > look accurate (they were recreated when I restarted presence manually). > > So this is SOLVED as far as the CPU level is concerned. I found a > device that has not been reconfigured (a valcom paging gateway) that > is essentially trying to register without an account, and the > registrar logs show 50-100 per minute (attempts). The rate limiting iptables rule may have help you here. But not fixed the problem ;-) > > I still think there is an SSL issue. Does anyone have any ideas on how > to figure this out? > > That is what I think also.I have disable 5061 forwarding on the firewall for remote clients way back around #18 (I believe) and have enjoyed a few weeks of quite with #22 now. All local clients are not using ssl. -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] High CPU sipXproxy (update #22)
On 11/5/2012 2:16 PM, Tony Graziano wrote: I am looking at a strange issue with a system which had a drive failure. We replaced the drive and reloaded (did not restore) the system, then updated it to the latest update. We see the proxy staying steady at 10% CPU, with not active calls or transactions. It is a basic system with trunking and 12 phones, there should not be such a load. I have sent the server its profiles. I have restarted the system. There is no memory or swap memory issue. I have reviewed the configuration and all of the speeddials and registrations. The first thing I noticed is that noone was able to place outbound calls easily, then when I started looking into it I checked user speeddials, presense and overall configuration and hardware functionality. I still see no issues except that the sipXproxy is taking up "enormous" CPU time. There are 12 phones and a total of 24 subscriptions. Does anyone else have an install similar and can verify whether they are seeing this or not? -- Actually we see a little bit of a decrease since Oct 1ish. In this one we have 83 active registrations about 20 offsite and 8G memory with and SSD drive. Did you look at the logging levels? You can always grab a snapshot and look at what log file is the biggest for an indication of where the activity is or even a packet capture on the server. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Can't retrieve parked calls on 4.4.0-418
On 10/26/2012 6:12 PM, Alan Worstell wrote: > Hello, > I have set up call parking on 4.4.0-418, and can send calls to be > parked, the caller hears the hold music. However, when I attempt to > answer the call by dialing *4 and then the park extension, the hold > music cuts out for a second for the parked call, and then comes right > back, and the phone I attempt to retrieve the call on just has dead air. > Any recommendations? > > Thanks, > Send profiles to the server and then reboot the server. That worked for us in a few cases. Also, not sure when things went a little astray in the patching, but it appears that there were a few resent versions of 4.4 that had issues. We have been using patch 22 for about a week now and it seems to be working fine. -- Regards ------ Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Hacked SipXecs 4.4
On 10/11/2012 11:48 PM, Noah Mehl wrote: > All, > > I just realized that my emails from my SipXecs 4.4 server were not being > delivered. Upon further investigation, I found that my SipXecs VM had a > sendmail queue with over 13000 messages in it. I'm trying to figure out how > my machine was sending mail, and it doesn't look like the relay is open, but > I found something curious: > > [root@sipx1 log]# cat secure | grep "pam_unix(sshd:session): session opened" > Oct 11 06:09:25 sipx1 sshd[22059]: pam_unix(sshd:session): session opened for > user PlcmSpIp by (uid=0) > Oct 11 18:36:02 sipx1 sshd[29185]: pam_unix(sshd:session): session opened for > user PlcmSpIp by (uid=0) > Oct 11 18:36:16 sipx1 sshd[29192]: pam_unix(sshd:session): session opened for > user PlcmSpIp by (uid=0) > Oct 11 18:36:21 sipx1 sshd[29195]: pam_unix(sshd:session): session opened for > user PlcmSpIp by (uid=0) > Oct 11 20:57:58 sipx1 sshd[30561]: pam_unix(sshd:session): session opened for > user PlcmSpIp by (uid=0) > > Those are what I think to be successful ssh logins with the user PlcmSplp. > Is this user part of the SipXecs install? > In your /etc/ssh/sshd_config you should have at the very least: PermitRootLogin no AllowUsers yoursecretusername MaxAuthTries 3 -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)
On 10/11/2012 10:37 AM, Henry Kwan wrote: I am a total newbie on SipXecs. I am also green when it comes to the SIP and VoIP PBX scene. Please excuse my seemingly simple question. The problem that I am encountering, essentially, is that external calls cannot be transferred to voice mail when a call is not answered. Internal calls that were not answered were transferred to voice mail without a problem. My setup: - SipXecs 4.4.0 installed from the download ISO and updated to the latest patches with yum. OS is also updated to Centos 5.8, with the latest patches. - Phones are Linksys SPA942 only, no other phones are on the system. Only 3 phones are on the system. - Domain: mydomain.company.com. company.com is registerd but mydomain.company.com is local/internal and the DNS server is the Sipx PBX. - Sipx PBX is the DHCP server, dhcp.conf file is edited to assign a limited range of IP addresses. No other dhcp servers are on the subnet. - The workarounds stated on the sipfoundry wiki for the SPA942 are implemented, i.e.: a. MOH Server:~~mh~@mydomain.company.com <mailto:%7e%7emh...@mydomain.company.com> b. Message Waiting:checked c. Mailbox ID:$USER_ID d. Voice Mail Server:extens...@mydomain.company.com <mailto:extens...@mydomain.company.com>. I have also changed mydomain.company.com to the IP address of the sipx server. - Use internal sipXbridge to connect to my SIP trunk. SIP trunk authenticated successfully and works. - Router used is Linksys WRVS4400N. Port 5080 and 3 to 31000 are forwarded to the SipX PBX. - Aliases are setup for these 3 phones are set for DID. With the above setup, I can dial extensions and have their respective voice mail kick-in when a call is not answered. Dial out and DID work as well. The problem that I am encountering now is that voice mail does not kick-in when an external call is not answered. Voice mail does work for internal calls, though. I've also added domain aliases of the IP address of the PBX and PBX.mydomain.company.com to the setup but that did not help. I also setup one of the phones to call forward to another phone, then voice mail. The call forwart to another extension worked but call forward to voice mail did not. In desperation, I also added an A record for mydomain.company.com in my DNS server but that did not help. Lacking the experience of sipX, VoIP PBX, SIP, and network debug tools, I hope experienced SipXecs users can shed some on my plight. External calls not transferring usually have 2 causes: your ITSP does not support it, the call did not come in on 5080/registration, or a firewall issue. Who is your ITSP? Did you try to forward 5060 udp/tcp also? Is your ITSP sending the calls to your based on your registration is it IP based. If IP the call has to come in on 5080 to be able to transfer. Did you do a "yum update"? Send profiles to the server: System|Servers Reboot -- Regards ------ Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Call transfer
On 10/10/2012 9:07 AM, Veréb Norbert wrote: > Hi! > > I have a problem with the call transfers. > I'm using sipXecs (4.4.0- 2012-09-29). > I can call the AA. (working) > I can call an extension. (working) > I can call an external phone. (working) > When I call the AA and the AA transfer this call an extension I hear this: > "Please hold while i transfer your call", but nothing happen, the call is not > transfer to extension. > > Any idea? > > > Do you have patch #20 installed? If not, you should. Did this work before and now it is not? -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Bug fix release update: sipXecs 4.4.0 update #20
On 10/5/2012 12:12 PM, Douglas Hubler wrote: > Update #20 > == > - ** No security updates in this update ** > - ISO has *not* been rebuilt as decided in release policy. Yum update > after installation is recommended for getting these updates. > - Thank you all for your continued testing and fixes. > > Thank you. Seems to be working great on 4 different systems now. -- Regards ---------- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Weighting on ACD
On 10/3/2012 11:49 AM, Aaron Carlson wrote: > Many thanks for the quick reply. :) > > I am using 4.4, actually. Is there any reason NOT to upgrade? > Reliability and stability may be an issue at the moment. I have not had a chance to look at the old and new ACD in 4.6 yet. -- Regards ------ Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Weighting on ACD
On 10/2/2012 3:04 PM, Aaron Carlson wrote: Hi folks, I'm hoping someone might have a minute to explain some call weighting scenarios to me. I'm trying to set up the following situation : I have an ACD queue with 9 agents in it, plus a manager who should only get a call if all agents are busy. In theory you could have 2 queues with the first circular/longest idle queue having an overflow to the queue with the manager in it. Just test it out in your environment though. In 4.4 there is currently a bug in the ACD overflow if someone is logged in and does not answer an outside/bridged call. The call will stay stuck in the first queue, I'm not certain what hapens if I set it up with circular, rather than longest idle. If I have it set to circular, and agent #4 is next in line, but is currently set to 'do not disturb', I presume it skips to agent #5, but what happens when they come back in? Are they slotted in for the next time it comes around, or does the system retroactively slip them in to take the next call before agent #6? With the ACD you can log in/out by dialing *88 or *86 by default. Also, I want to set things up so that there's one agent that's in the queue, but only gets calls if all other agents are on a call or on DND. Is there a way to do that? If all the calls came from the ACD. The current ACD has no knowledge about calls that came in some other way or the the agent is making a call. Would that be to create a hunt group with that single agent as the target (with a voice mail if they are dnd) or would it be to create a separate queue as the overflow with that single agent as the target of the queue? Can I set it to only overflow there IF that agent is logged in, and otherwise have the call stay in a holding pattern? Hunt groups are pretty good for small groups but lack the "circular" or "longest idle" abilities. It also lack the ability to keep the user on hold until the next available agent. Should I be using hunt groups instead? For a small group I use: * call comes in to a hunt group * if no answer roll call into a AA and give the caller a chance to leave a message * if they don't want to leave a message transfer to an ACD. My apologies for the detailed nature of the question, I'm trying to set this up and am both impressed and a little overwhelmed by the complexity of the platform. -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Lastest Patches and Alias
On 9/29/2012 3:49 PM, George Niculae wrote: > Todd, > > you should yum update sipxregistry and sipxconfig from here for the > moment, they contain changes reverted: > > http://download.sipfoundry.org/pub/sipXecs-stage/4.4.0/CentOS_5/x86_64/ > > George We have an install with #19 and their ACD's are not going to overflow for outside calls when there is at least one agent signed in and does not answer. Inside calls work properly and also when there are no agents signed in. I also tried with the sipxregistry and sixconfig from the above link. -- Regards ---------- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] 4.6 Cluster
On 9/27/2012 10:07 AM, darthzejdr wrote: I'm trying to get clustering to work, but i have some problems. Some calls work, while others don't. I think that the ones that work are people registered on main server. I thinkDoes anyone have any idea what might the problem be, and how to fix it? I have sip registrar and dns working on both servers, and they can ping eachther by name. I also tried disabling the firewall, and i get accept all, but on secondary server it's still running. Tried send profiles, rebooting server, and restarting iptables. Didn't help. From what i managed to pull out, i'm loosing registrations on secondary server, and the server is not forwarding invites to server 1. Without a log or a packet capture we can only guess. My guess is that there are a few bugs in the registration in 4.4 and 4.6 at the moment that they are working on. -- Regards ---------- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] 4.4 version 08-17 getting SipRegistrar:"Response auth hash does not match (bad password?) for remote user
On 9/26/2012 11:44 AM, Tony Graziano wrote: > It finally started working on a Polycom. But I do not get to > redirected IVR when noone answers. So I guess i am looking at an > update and trying again. I still cannot register with a softphone. 3cx was giving us problems with #18 both local and remote. Jitsi works for us. > > On Wed, Sep 26, 2012 at 11:36 AM, Gerald Drouillard > wrote: >> On 9/26/2012 11:14 AM, Tony Graziano wrote: >>> I sent server profiles. I deleted and recreated the user. In the >>> registrar it shows the PassTokenDB='plain text password' >>> authTypeDB='DIGEST' >>> >>> the passtoken IS the real password stored in the system.It does not >>> matter if I use TCP or UDP or seemingly what type of UA I use in this >>> case. >> We were having many auth issues with #18. #19 seems to be better. I >> believe there is a TLS issue still open. We turned off 5061 on our >> firewall to force everyone in on 5060 with #18 and that helped most >> clients. Not sure if it is fixed in #19. >> >> >> -- >> Regards >> -- >> Gerald Drouillard >> Technology Architect >> Drouillard & Associates, Inc. >> http://www.Drouillard.biz >> >> _______ >> sipx-users mailing list >> sipx-users@list.sipfoundry.org >> 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 sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] 4.4 version 08-17 getting SipRegistrar:"Response auth hash does not match (bad password?) for remote user
On 9/26/2012 11:14 AM, Tony Graziano wrote: > I sent server profiles. I deleted and recreated the user. In the > registrar it shows the PassTokenDB='plain text password' > authTypeDB='DIGEST' > > the passtoken IS the real password stored in the system.It does not > matter if I use TCP or UDP or seemingly what type of UA I use in this > case. We were having many auth issues with #18. #19 seems to be better. I believe there is a TLS issue still open. We turned off 5061 on our firewall to force everyone in on 5060 with #18 and that helped most clients. Not sure if it is fixed in #19. -- Regards ------ Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Server with 2 NIC Cards
On 9/25/2012 1:06 PM, Tommy Laino wrote: > > I am installing a SipX 4.4 for a customer of mine. They are > a web design company and insisted they buy their own server. > When looking at the configuration I noticed that the Dell > BOM had the server with dual NIC cards. It is my > understanding that SipX will not work on a server with 2 NIC > cards even if only one is being used. > > Am I correct in my assumption? It will work. Just configure only one of them. -- Regards ------ Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] new patch for XX-10177
On 9/25/2012 1:44 AM, Joegen Baclor wrote: Andrew, any update on this? Update #19 fixed many issues we were having with offsite registration. With #18 we had turned off 5061 and that seemed to fix most clients. We have not turned on 5061 with #19 yet. I would like a day or two of stability before working it back in. -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
[sipx-users] 4.4 ACD
Not sure when this stopped working but in #17 and #18 you can call into a ACD queue with no active agents and the call will go on hold but not transfer to the overflow. The overflow in this case is an AA. Everything works fine if you call the ACD via an internal phone to it's extension. It is just the external calls. Other calls from the sip provider (Voice Innovations) transfer just fine. I have "Activated" the ACD. Sent profiles to the server and restarted with no success. I have a capture if it will help: http://www.ask-services.com/tmp/acd.cap -- Regards ------ Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Multiple remote worker issues?
On 8/8/2012 4:05 PM, Nathaniel Watkins wrote: > I've never had a reason to attempt multiple remote workers from a single > remote location. Will this work, or can there only be 1 remote phone on the > far end over a NAT'ed connection? > > This message and any files transmitted with it are intended only for the > individual(s) or entity named. If you are not the intended individual(s) or > entity named you are hereby notified that any disclosure, copying, > distribution or reliance upon its contents is strictly prohibited. If you > have received this in error, please notify the sender, delete the original, > and destroy all copies. Email transmissions cannot be guaranteed to be secure > or error-free as information could be intercepted, corrupted, lost, > destroyed, arrive late or incomplete, or contain viruses. Garrett County > Government therefore does not accept any liability for any errors or > omissions in the contents of this message, which arise as a result of email > transmission. > > > Garrett County Government, > 203 South Fourth Street, Courthouse, Oakland, Maryland 21550 > www.garrettcounty.org > ___ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ We have better luck with difficult locations using tcp instead of udp. -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Date and Time menu issue in 4.4
On 7/13/2012 2:15 PM, Kurt Albershardt wrote: On Jul 13, 2012, at 9:55 , Kurt Albershardt wrote: Everything appears to be working in the GUI, with the exception of the "Date and Time" menu item, which produces the familiar An internal error has occurred. Click here <https://sipx.domain.com:8443/sipxconfig/restart.svc> to continue. Installing ntp and rebooting the server did not change this behavior. All of my phones (which I did finally manage to get registered) have USA-5 timezone set, and while I can manually update them I suspect that menu might allow me to set a system timezone? Is your ntp server a linux server... * some older versions of ubuntu have the ntp.conf file in /var/lib/ntp/ntp.conf.dhcp, you may want to delete that one and use the one in /etc Polycom will not sync the time if on the linux server you run: ntpdate -q localhost and you get back: "no server suitable for synchronization found" if so then: * rm /var/lib/ntp/ntp.conf.dhcp * service ntp stop * ntpdate us.pool.ntp.org * edit /etc/ntp.conf * service ntp start you can tell if the ntp server is connected to other ntp servers with: ntpq -p After you make the changes, ntp server may take a few minutes before it is synchronized. * -- Regards ------ Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Resend Acknowledgement
On 7/9/2012 12:53 PM, Bryan Anderson wrote: > Here is a trace from when I was talking with one of the users. This > was the first trace I noticed it on. This was obtained using > sipx-dialog-count and sipx-trace. > > I have seen 5 other traces that match this behavior and all the calls > cut at ~24sec. The carrier is NexVortex. When I brought this them > they pulled their traces and we com paired. They never got the ACK. > My first instinct says Comcast but not necessarily, and I am not sure > where to test it. Once thought was between two of our SipXecs System > on different internet service providers. > > -Bryan Anderson > Sounds like you are have an ALG/firewall issue on one of your routers. -- Regards ------ Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] analog adapter recommendation for a fax machine
On 6/15/2012 5:02 PM, gabriel wrote: > hello sipx-users, > > I am running a brand new sipx install (4.4.0) with polycom 670 phones and > a few sip trunks. I want to connect an analog fax machine to the system and > was wondering which analog adapters you guys recommend. > I have used cisco ata186 before and wasn't very happy with them (dtmf > issues among other things - those where connected to a different phx not > sipx) > > We have had good success with Linksys ata2102. Here is a how to: http://www.drouillard.biz/blog/sipxecs-and-linksys-ata3102-ata2102/ -- Regards ------ Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Invalid 200OK to reinvite?
On 6/15/2012 12:22 PM, Matt White wrote: As I stare at my traces I'm seeing an issue with sipxbridge via headers. The PBX is behind a NAT and has it configured. Calls to the SIP trunk for external calls are fine. The gateway that is configured between the PBX and the SeimensPBX are not separated by nat. The IP address is separated by only a few digits. The gateway is set to NOT use the public ip for call setup. However, the via header from sipxbridge to the siemens has the public ip in the VIA header. Then when the 200ok comes back the seiemens appends a "received=privateip" to the via. I've confirmed the local subnets are correct. I even removed them all and out in the entire 10.0.0.0/8. Same problem. Any thoughts on why sipxbridge is throwing the public ip in the via? I suspect this might be my issue with the transfer as sipxbridge should be using the via header to validate the response right? -M ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ So the IP address of the SeimensPBX is found in local subnets? Do you have the Siemens setup as a gateway? Send server profile and reboot sipx? -- Regards ---------- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] 4.4 remote users now working with udp on some devices after update
On 6/1/2012 9:13 AM, Michael Picher wrote: > i assume you mean 'not' instead of 'now' in the subject. > > there may be an issue there with patch 16, we're not sure yet. if you > have any captures / thoughts they might be helpful. > > also, see if sending server profile helps. > > Sending the server profile and restarting fixed it. -- Regards ------ Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] 4.4 remote users now working with udp on some devices after update
On 6/1/2012 9:13 AM, Michael Picher wrote: > i assume you mean 'not' instead of 'now' in the subject. > > there may be an issue there with patch 16, we're not sure yet. if you > have any captures / thoughts they might be helpful. We seem to be having the same problem with a remote branch that was working and now is not after a phone reboot. Setting to TCP or UDP did not help. Even tried the "TCP Fast Failover" that seems to been added recently. > > also, see if sending server profile helps. I'll try that this weekend. -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Unmanaged services plan for 4.6
On 5/29/2012 11:05 AM, Dave May wrote: This is the way we configure it as well. The parent domain is managed by corporate DNS servers, which delegate a sub domain to the sipXecs/openUC config server to manage. We have other servers in the sipXecs domain though, which has made managing DNS more difficult than it needs to be. I know we can manually edit the cfengine files in order to have the best of both worlds -- sipXecs managed DNS and custom records. But, would it nice if these extra records could be managed in the user interface. Does anybody know if webmin has a cfengine plugin which is compatible with the 4.5.2 design? Not sure what/how you want to manage your DNS but cfengine is there: http://www.webmin.com/standard.html IMO, if you are to do this correctly, you would want to use master/slave DNS zones. -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Unmanaged services plan for 4.6
On 5/24/2012 9:10 AM, Douglas Hubler wrote: > DNS, IP tables, NTP and DHCP are among the few services that some > folks configure separately on sipxecs 4.4 or older systems. Starting > with the 4.6 release these services are integrated in a much tighter > way. In order not to conflict with any custom configuration methods, > these select services now have a "Unmanaged" setting you can set which > allows you to configure the services yourselves. George and I > realized that for each service, an unmanaged state can have different > consequences depending on what the service does or how it's > configured. > > So in short, there is no common specification for how unmanaged > services are dealt with, so George and I urge you to test out 4.6 and > see if you can still configure the systems as you once did. Don't > worry, there will *always* be a way to hack want you want together in > 4.6 because all the rules are now in editable text files, but the goal > is to lower the level of hacking you would have to do to make system > easier to setup out of the box and easier to maintain thru a system's > major and minor upgrades. > ___ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ I think you can get as crazy as you want with this. I believe the ultimate in interfaces for managing the services on the server is webmin. It is very rare that you cannot do anything you want within webmin. I have to admit that a lot of "newbies" to sipx have trouble setting up the DNS. We get a lot of people with phone backgrounds coming to sipx when a "network/server" background would be the best. UC device dns settings come from the dhcp server on the network or can be static. You just cannot take over being the DHCP server without going in and shutting it off on whatever else is on the network. I can see a lot of people bringing down their network without knowing what they are doing. I think the current 4.4 way is the correct way of doing it. The page that goes out and looks at the current settings on the network is good. If you don't know how to change your current DNS or DHCP server then you should get assistance from somebody that does before installing a phone server. It is only going to get harder when their firewall is not working either. I think we have to be careful on who we are targeting on the "install" phase. There is no harm in the server being configure to be a DNS and ntp server for the UC devices. But that can be misleading as those settings can be superseded by the "real" DHCP server on the network. Don't forget all the option settings in dhcp like: sip-servers-name boot-server "Dynamic DNS reverse domain" and "Dynamic DNS domain name" could also come into play for both DNS and dhcp. -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] T.38 Recommendations
On 5/23/2012 7:59 AM, Mike Pinkerton wrote: > We need to implement faxing. At the moment, I am most concerned > about inbound faxes. My understanding is that sipXecs can do a T.38 > to e-mail conversion and mail inbound faxes to users' mailboxes. Is > that correct? > > Does anyone have a SIP trunk provider with good T.38 support that he > or she would recommend? If so, are there any peculiar config > settings required to work with that provider? > > Thanks. > VoipInnovations.com: They can setup a test account for a free trial. If you decide to go with them, they will auto-recharge at $100 to bring up to a $200 balance. IMO, voip.ms has a better interface but voip.ms doesn't have t38. Per minute rates are very cheap. Callcentric.com: has issues with >5min faxes outbound using hylafax and t38modem. Not sure if inbound has the same problem. Appiaif you have the time and like "packages". -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Anyone using appia IP auth, port 5080
On 4/27/2012 1:17 PM, m...@grounded.net wrote:
So quick question in terms of using appia and registration.
ITSP Identifier Registration Status
sip.appiaservices.com [314925] AUTHENTICATION_FAILED
I've created the gateway, added the reg information but cannot authenticate.
Have emailed support who is looking into it as I try but another thing is
unclear.
In the Configuration section, I have tried leaving the port to 0 which is port
5080 trying by default and have tried port 5060 but neither works. In this
thread, it was mentioned that registering using default port 5080 should be
fine so wondering if I've overlooked something.
It is normal to have issues on the first attempt with them. Here are my
notes for appia install:
*Sipx setting Registration Interval: 180 (not 600 - not sure about
this one though)*
**
*Usually first password doesn't work. Tell them to Reload trunk group.*
*When ordering specify 10 digit calling - no +1{phonenumber}*
--
Regards
------
Gerald Drouillard
Technology Architect
Drouillard& Associates, Inc.
http://www.Drouillard.biz
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] UPDATE Re: Anyone using appia IP auth, port 5080
On 4/27/2012 12:36 PM, Tony Graziano wrote: On Fri, Apr 27, 2012 at 12:13 PM, m...@grounded.net <mailto:m...@grounded.net> <mailto:m...@grounded.net>> wrote: So, got a conference call from sales and the vise president of Appia this morning, wanting to make sure I had proper information. He explained that he wanted to make sure that if there is any confusion, that I get first hand information concerning any possible hardware being offered on port 5080. He explained that Tony has provided all of the information about his use of port 5080 because as far as he knows, Tony's testing was a bust, nothing ever worked right and that was that. (I'm not saying that Tony, he did, just relaying) I beg to differ. The only thing that has been a bust was the fact their sales channel guy could never keep appointments with me, etc. BOTH worked, and I relayed this to them, but they have a horrible helpdesk/communications platform AND they insist on wasting my time to schedule installations. They are clearly old school telecom guys and have to have a flippin' excel spreadsheet for WTF I dunno. It's moronic. +1 -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Anyone using appia IP auth, port 5080
On 4/27/2012 8:19 AM, Tony Graziano wrote: In my testing with voipinnovations I found t.38 not very functional with sipx. Have you tested it? If so, how recently? Using it right now with a location that does about 150 faxes a day. Outbound is with t38modem/hylafax right now. Inbound is 4.4 sipx. We are getting better inbound results with sipx than ringcentral numbers which have been struggling in the last couple of weeks. There seems to be only 1 or 2 numbers out of hundreds with whom there are problems faxing out to which is on my list of things to look at today. There may have been problems with those numbers before the switch. On Fri, Apr 27, 2012 at 6:58 AM, Gerald Drouillard mailto:gerryl...@drouillard.ca>> wrote: On 4/26/2012 6:45 PM, m...@grounded.net <mailto:m...@grounded.net> wrote: Is anyone else using appia on port 5080 with IP auth? We signed up with them a few weeks ago but have been talking with them for a couple of months. For that amount of time, they have been telling us they do not provide port 5080 services and are only in a testing phase at this time. I have pushed and pushed this subject with them and Sean has put me in touch with support and development who tells me they don't even have a release date at this time. If it's working for you, it means you're probably on some switch that is prior to their buying up the other company. This input will help me to push them a little in getting port 5080 working for everyone. We use registration auth and it has been working very well. 3 things I don't like about Appia: * You have to buy a calling plan * No voip.ms <http://voip.ms> like interface for your account * IMO, The setup process/project is unnecessary On the other hand they have t.38 and their voip service has been rock solid. If this is a big account then you should consider http://voipinnovations.com/ . There is a $200/month minimum. They do t.38 and their price per min is very low. -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org <mailto:sipx-users@list.sipfoundry.org> List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net <mailto:tgrazi...@voice.myitdepartment.net> 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: helpd...@voice.myitdepartment.net <mailto:helpd...@voice.myitdepartment.net> Helpdesk Customers: http://myhelp.myitdepartment.net <http://myhelp.myitdepartment.net> Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org 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 sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Anyone using appia IP auth, port 5080
On 4/26/2012 6:45 PM, m...@grounded.net wrote: Is anyone else using appia on port 5080 with IP auth? We signed up with them a few weeks ago but have been talking with them for a couple of months. For that amount of time, they have been telling us they do not provide port 5080 services and are only in a testing phase at this time. I have pushed and pushed this subject with them and Sean has put me in touch with support and development who tells me they don't even have a release date at this time. If it's working for you, it means you're probably on some switch that is prior to their buying up the other company. This input will help me to push them a little in getting port 5080 working for everyone. We use registration auth and it has been working very well. 3 things I don't like about Appia: * You have to buy a calling plan * No voip.ms like interface for your account * IMO, The setup process/project is unnecessary On the other hand they have t.38 and their voip service has been rock solid. If this is a big account then you should consider http://voipinnovations.com/ . There is a $200/month minimum. They do t.38 and their price per min is very low. -- Regards ---------- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] pfsense port forwarding
On 4/26/2012 5:01 PM, m...@grounded.net wrote: > Anyone know of a document showing how to configure pfsense (2.0) to forward > port 5060 to port 5080 for ITSP use on sipx. > > I can't seem to get this to work and am not sure why. Since port 5060 is used > by remotes and I need to catch ITSP traffic, I created a separate rule for a > second port 5060 which allows only the ITSP to have access. I'm not sure that > can work however but either way, I then forward that port 5060 to port 5080. > > I'm trying to allow Appia to work with sipx but they don't provide incoming > on port 5080 so until they do, I need to forward and have not been able to > figure out how to make this work. > > Thanks. > > Mike > If you use the registration method with Appia then you will get the calls on 5080. At least that is how we have one account setup. Not sure if they allow 5080 on IP auth. -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Sending faxes
On 4/13/2012 8:40 PM, Joegen Baclor wrote:
It should be easy enough to enable this feature in FS by enabling a
custom dialplan to route back to the proxy for faxses. In the web
interface, In the web interface, one can introduce a page that would
do the following.
1. Browse for a file (PDF)
2. Convert PDF to tiff using ImageMagic [ convert -density 204x98
-units PixelsPerInch -resize 1728x1186\! -monochrome -compress Fax
txfax.pdf txfax.tiff]
3. Send an ESL command to FS to transmit the file [originate
sofia/gateway// &txfax(/path_to_fax_file)]
I did a little experimenting and have been able to get trxfax working
from a bash shell on the phone server:
File: fs_cli_txfax.sh
CLI=/opt/freeswitch/bin/fs_cli
TIF=/install/fax/txfax-sample.tiff
DEST=1551212
DOMAIN=example.com
HEADER=Your Company Name
IDENT=Your Name
IM='convert -density 204x98 -units PixesPerInch -resize 1728x1186\! -monochrome -compress Fax
"%1" "%2"'
#$CLI -x "sofia status"
#$CLI -x "sofia status profile $DOMAIN"
$CLI -x "originate
{ignore_early_media=true,fax_header='$HEADER',fax_ident='$IDENT',absolute_codec_string='PCMU,PCMA',fax_enable_t38=true,fax_use_ecm=false,fax_enable_t38_request=true,proxy_media=false,bypass_media=false,fax_disable-v17=true}sofia/gateway/$DOMAIN/$DEST&txfax($TIF)"
$DOMAIN
Now I just have to figure out if I want to do all the work to migrate the
t38modem/hylafax server.
--
Regards
--
Gerald Drouillard
Technology Architect
Drouillard& Associates, Inc.
http://www.Drouillard.biz
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Jitsi provisionning
On 4/17/2012 5:49 PM, Cyril Constantin wrote: > Hi Guys, > > I just would like to know if there is any plan to integrate Jitsi into > provisioning phone? > > They now have a stable release since beginning of April. > > http://jitsi.org/ > > Thanks a lot for your feedback. > We have been using this for a few days now and have been impressed. We had a site that could not connect with 3cx (probably because of alg in some upstream router) and Jitsi worked instantly. Call quality is better than 3cx, due to the many encodings available. -- Regards ---------- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] voip.ms
On 4/23/2012 9:45 AM, Kumaran wrote: > Hi All, > Whether Voip.ms supports t.38 codec so that I can assign a DID number > to user fax extension? > > Regards, > Kumaran T > ___ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ For low cost pay per usage plans: I had a some success with callcentric.com. See: http://www.drouillard.biz/blog/sipxecs-hylafax-and-t38modem/ Some that I haven't tried yet that may work: http://www.voicepulse.com/ http://www.t38faxing.com/ For large installs where you have over $200/month in services you may want to consider: http://www.voipinnovations.com/ -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Trunk to Trunk Transfer
On 4/20/2012 1:08 PM, Tommy Laino wrote: > Content-Type: text/plain; >charset="utf-8" > Content-Transfer-Encoding: 8bit > Organization: SipXecs Forum > X-FUDforum: 08063afcdd00a6e76393c5b9527381e8<67729> > Message-ID:<10891.4f919...@forum.sipfoundry.org> > > > > I have 3 IP trunks on my test system. I am trying to have an > option from the auto attendant transfer to a cell phone. If > i do it from a local or remote polycom phone it works fine. > Once an external call comes into a trunk and it chooses the > option the caller gets disconnected. Anything that I might > be missing. Your provider may not support hair pin transfers. Who is your ITSP? -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Nat Problem
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 <mailto:gerryl...@drouillard.ca>>
On 4/19/2012 2:58 PM, Simon Brûlé wrote:
I added 192.168.175.0/24 <http://192.168.175.0/24> to the
intranet subnet and I still have the same problem.
2012/4/19 Gerald Drouillard mailto:gerryl...@drouillard.ca>>
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 <http://192.168.0.0/24>.
You may need to add 192.168.175.0/24
<http://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.
--
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--
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http://www.Drouillard.biz
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--
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http://www.Drouillard.biz
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Re: [sipx-users] Nat Problem
On 4/19/2012 2:58 PM, Simon Brûlé wrote: I added 192.168.175.0/24 <http://192.168.175.0/24> to the intranet subnet and I still have the same problem. 2012/4/19 Gerald Drouillard <mailto:gerryl...@drouillard.ca>> 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 <http://192.168.0.0/24>. You may need to add 192.168.175.0/24 <http://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 sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Send external calls to VM
On 4/19/2012 2:34 PM, Sven Evensen wrote: > A customer wants all external calls before 8am to go straight to VM > while the internal calls should ring. Is this possible today somehow > or does anyone know if this is in the roadmap? They are on 4.4 sipx > > Have the external calls go to a phantom user with forwarding rules/schedule. -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Nat Problem
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 <http://192.168.0.0/24>. You may need to add 192.168.175.0/24 also if it is local. From what i saw on forums a no sound problem is often related to a nat problem on the rtp port. If there is any log file that could help you help me just let me know i will do a copy paste here. Jitsi 1.0.0 Ubuntu 11.10 SipXecs 4.4 Polycom 321 BootRom 4.3.1 and Software 3.3.4 ___ sipx-users mailing list sipx-users@list.sipfoundry.org 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 sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Jitsi provisionning
On 4/17/2012 5:49 PM, Cyril Constantin wrote: > Hi Guys, > > I just would like to know if there is any plan to integrate Jitsi into > provisioning phone? > > They now have a stable release since beginning of April. > > http://jitsi.org/ > > Thanks a lot for your feedback. > > Thanks for the link. I am trying it out now and so far I am impressed. -- Regards ---------- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Sending faxes
On 4/16/2012 10:49 AM, m...@grounded.net wrote: > > I thought we were talking about a real integration and not using hylafax on > the same server? Mind you, I did make a mention that it would be ok to have > to use a separate server which could be part of the sipx install. > > No big deal, if not many are interested, then it's a moot point. I did post a link on how to use t38modem, hylafax (for sending) and sipx together. In this case the hylafax server was a separate machine, but it doesn't have to be if you can get a version t38modem working on your sipx server and a t.38 ITSP. http://www.drouillard.biz/blog/sipxecs-hylafax-and-t38modem/ Most of my effort was looking for a t.38 ITSP that would work with sipx and t38modem. Just wanted to point out that hylafax does have email to fax gateways, linux command line sending, and print drivers for outbound. Although it does seem like a lot of extra baggage to have to install hylafax, it does provide a solution for high outbound fax sites. IMO, if you are not a big outbound fax site that doesn't need command line sending and print drivers, then you would be better off using something like ringcentral or maybe your ITSP has a email to fax gateway. -- Regards ---------- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Sending faxes
On 4/13/2012 8:20 PM, m...@grounded.net wrote: > I believe I saw a thread a while back where someone was asking about sending > faxes. Some searching shows that some have asked but that there are no plans. > > Is this still the case or are others interested in this? Even a shared > outgoing account as a 'group' would be so very welcome and would instantly > eliminate our having to use additional hylafax/avantfax servers just for this > function. It would be way nicer to be able to tell potential customers that > everything can be done from the one system. > > We recently had an install that was a heavy hylafax user with usb modems. We are now using sipx for receiving faxes and hylafax for sending. http://www.drouillard.biz/blog/sipxecs-hylafax-and-t38modem/ -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Proposed Firewall Config
On 3/28/2012 8:39 AM, Douglas Hubler wrote: > In 4.6 we're using iptables to restrict access to services. This is > different than 4.4 where we had either clunky, home grown > authorization schemes (shared secret based) or no protection at all > (not security risk, just DoS or Buffer overflow vulnerabilities) > > Goals: > - Default rules out of box will fit most use cases > - Provide some level of customization for the most common tasks > - If configuration doesn't meet demands allow user to take over > firewall config manually for each server > - Plugins can contribute to the default rules > - If firewall is handled by separate system allow user to disable > firewall config completely > > I stumbled across this today which seems to be the most extensive sip/iptables filter: http://etel.wiki.oreilly.com/wiki/index.php/SIP_DoS/DDoS_Mitigation -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Proxy to Bridge Invite Failing
On 3/27/2012 6:12 PM, Josh Kennedy wrote: > > I'm getting an intermittent outbound issue and it appears to > be a problem between the proxy and the bridge during an > invite. Below is an excerpt from the trace. The first invite > was from a successful call, the second from a failed call. > Both were to the same number within a minute of each other. > Is this a new install? A capture at the firewall usually tells all. Probably some kind of ALG / sip helper. -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] voip.ms config
On 3/27/2012 12:03 PM, Stiles Watson wrote: This is where one swallows one's pride The way I was entering data caused the drop-down to not be displayed. To keep this short: 1. When you first select Add new gateway>Sip Trunk, the template drop down is not visible. I was not aware this was the case until yesterday. I just thought it was not there. 2. The template drop-down is only displayed after you enter a name for the gateway and then select the default SBC. 3. If you ever click the Apply button before both the name and SBC are entered, the drop down is never displayed. This is why I never saw the template drop-down. Now, having said all of that, I deleted my existing voip.ms gateway and created a new one using the template drop-down. However, this did not fix my problem and everything is as it was before. I still can not retrieve a call from hold or cancel a transfer. I have verified in my voip.ms account that it is registered with the public IP and port 5080. So it looks like we are back to a firewall problem, correct? Yes. What kind of firewall do you have? -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Good Contact at CounterPath
On 3/26/2012 2:48 PM, Todd Hodgen wrote: Can you explain the case that is not working. Call in from DID, transfer from 3cx softphone to 3cx softphone seems to work. Call from 3cx to Polycom seems to work. Are you saying a call from one 3cx to another, and then transfer to a third doesn't work? If that is the case, not sure I have a use case for that with any customers. That is correct. -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Good Contact at CounterPath
On 3/26/2012 2:08 PM, Todd Hodgen wrote: SO, I found they had a bug reported on this that was fixed in their V10, Service Pack 3. What it doesn't state was if this bug was related to their PBX software, or the 3CX Softphone. You might want to try downloading the latest Softphone to see if it works for you now. I'm running version 5.x, the version now being downloaded is version 6.x I am running 6.0.2.0943.0 which is the newest I believe. -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Good Contact at CounterPath
On 3/26/2012 1:41 PM, Todd Hodgen wrote: Internal Blind transfer seems to work fine for me. It works ok for us if one of the endpoints is a polycom phone or via a DID. 3cx to 3cx transfer does not work. -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Good Contact at CounterPath
On 3/26/2012 12:43 PM, Todd Hodgen wrote: My frustration with them, as a reseller, they don't support resellers. They used to have a discount program. Now, they sell to you like any enduser, but they give you a commission, once you reach $500 in Commission. Let's see, $40 a copy, $4 commission, when you sell 125 copies you get paid. Seems they like selling direct. Their name never comes up in conversation with my customers unless I am desperate for a softclient. On the other hand, the Free 3CX softclient seems to work well, as does the Voice Operator Panel products. We like 3cx also. One small issue with 3cx is you cannot transfers internal calls. External work fine. You can have multiple accounts active. *From:*sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Robert Schroeder *Sent:* Monday, March 26, 2012 9:28 AM *To:* sipx-users@list.sipfoundry.org *Subject:* Re: [sipx-users] Good Contact at CounterPath CounterPath offers a good product however when it comes to support and sales they are not that great. I am being really nice because my experience with sales has not been great. Support, we'll let us just say I am frustrated as the normal response is no -- we do not support that, no -- that is a server function or no -- that is a stupid request! LOL -- Many Tears on this one. Oh the shame of it all!!! You can try this information. Tiffany Zinck Sales Operations Manager CounterPath Corporation T 604.320.3344 F 801.640.0011 Good Luck, *Robert Schroeder* IT Manager Information Systems Member First Mortgage *From:*sipx-users-boun...@list.sipfoundry.org <mailto:sipx-users-boun...@list.sipfoundry.org> [mailto:sipx-users-boun...@list.sipfoundry.org] <mailto:[mailto:sipx-users-boun...@list.sipfoundry.org]> *On Behalf Of *Becker, Jesse *Sent:* Monday, March 26, 2012 10:44 AM *To:* Discussion list for users of sipXecs software *Subject:* [sipx-users] Good Contact at CounterPath All, Does anyone have a good contact at Counterpath? I can only get to their general sales mailbox. I have left messages and no one has called me back (maybe they don't like me?) I am trying to see if there are educational discounts for Bria or if we get the same prices listed on their web page. Thanks, Jes -- Jesse Becker Technical Manager Office of Information Technology Network+ | Linux+ Certified Professional DATATEL+SGHE @ SUNY Ulster 491 Cottekill Road, Stone Ridge, NY 12484 Tel 845-687-5064 | Fax 845-687-5105 beck...@sunyulster.edu <mailto:beck...@sunyulster.edu> | www.sunyulster.edu <http://www.sunyulster.edu/> Open or check the status of a ticket by visiting Helpdesk Online <https://helpdesk.sunyulster.edu/> Look up answers to frequently asked questions by visiting the Knowledge Base <https://kb.sunyulster.edu/> NOTICE: This electronic mail message and any content within it are intended exclusively for the individual(s) or entities to which it is addressed. The message, together with any attachments and all other content, may contain confidential and/or privileged information. Any unauthorized review, use, print, save, copy, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies. ___ sipx-users mailing list sipx-users@list.sipfoundry.org 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 sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Can not transfer from Auto Attendant
On 3/26/2012 11:00 AM, Stiles Watson wrote: Anyone else have an idea why I lose audio when retrieving a call from hold or canceling a transfer? Sounds like the call did not come in on port 5080? -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] sipXecs 4.4 - Phones Won't Register - 401 Unauthorized
On 3/26/2012 10:12 AM, W. E. W. Russell wrote:
All,
Hello, my name is William Russell and I have MAJOR issue with my
sipXecs. Currently, my office completely out of phone service.
Over the weekend, I moved our office to a new IP scheme. I've been
able to get everything back up and running with the exception of our
phone system.
I have attached snapshot of the logs. I hope someone can help me out
ASAP. I've been up all night trying to get this fixed, but I can't
seem to get it to register. NONE of the phones register, so when you
look at the logs you will see MANY 401 unauthorized from all the
phones. This leads me to believe it is something in sipXecs or the
related network elements that is causing this phone to not
get authorized.
We are using Polycom VVX 1500 phones, sipXecs 4.4, on RHEL 5. All of
the configuration tests pass with flying colors. I even see my SIP
trunks registering with my ITSP, but I can't get my phones to register
locally. I've been the through the sipregistrar.log file, but I didn't
see any error or issues. In fact, it looks like it says that
everything is VALID and should be authorized.
Any help - ANY new direction would be extremely helpful. I'm simply at
a loss especially since it was working fine previously - a change in
IP scheme shouldn't cause this problem if all the configuration tests
passed. That indicates to me that the new IP addresses have taken hold
well and the routing is operating correctly.
Thank you very much in advance!
What change did you make to the IP scheme?
Did you update the sip server with a new IP/subnet/gateway?
Is the SIP server your dhcp server?
Any vlans?
In the sipx web admin screen did you:
* Update the server IP address under System|Servers|{your
server}|Configure
* Update Intranet Domains in System|Internet Calling
* Send profiles to the server?
* Send profiles to the phones?
--
Regards
------
Gerald Drouillard
Technology Architect
Drouillard& Associates, Inc.
http://www.Drouillard.biz
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Re: [sipx-users] VoIP.ms Setup Experience
On 3/23/2012 5:49 PM, Chris Rawlings wrote: > if you honestly want flawless service.. i recommend getting an Ingate > SIParator SBC with the Remote NAT Traversal Module, SIP Trunking > Module, and enough CAL's to cover all call flows... i have found that > VoIP.ms with a SIParator provides FLAWLESS service to SipX. while > having to register to VoIP.ms becuase they do not support IP based > authentication on port 5080 has a few issues sometimes. > > i have right now an issue with Appia where we sometimes get a 481 > Unknown Dialog from the carrier when we park a call.. this in turn > dropps the call.. we are currently waiting for a fix.. but as a temp > fix we have forwarded our number with Appia to a temp number at > VoIP.ms and everything is working 100%. > > its honestly too bad we can not just port our number to VoIP.ms > problem would have already been solved > > You can use IP auth with voip.ms http://www.drouillard.biz/blog/ip-authentication-with-sipxecs-and-voip-ms/ -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] TLS Integration (P.S) I know I posted it on the forum and it's like double post but I didn't read the how to post on the forum before I did it so I didn't knew about the mail list)
On 3/23/2012 2:11 PM, Josh Patten wrote: > VPN or possibly a session border controller with TLS capabilities. I have not had any success with TLS getting through firewalls or access points that have ALG or Verizon MiFi. -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] supporting/proxying https://server/ going to admin ui
On 3/15/2012 4:11 PM, Douglas Hubler wrote:
I had to unraveled apache config in 4.6 when we introduced cfengine.
For admin interface I decided to go for simple apache config instead
of static html landing page w/meta tag. I came up with an apache
config. This redirects http requests to admin ui to https and
redirects "/" to "/sipxconfig". This works, i just want to pass this
thru and apache gurus for validation.
RewriteEngine On
RewriteCond %{SERVER_PORT} !^443$
RewriteRule ^/sipxconfig/(.*)$ https://%{SERVER_NAME}/sipxconfig/$1 [L,R]
RewriteRule ^/+$ https://%{SERVER_NAME}/sipxconfig/ [L,R]
ProxyPass/sipxconfig http://127.0.0.1:12000/sipxconfig
ProxyPassReverse /sipxconfig http://127.0.0.1:12000/sipxconfig
Try this:
RewriteEngine on
RewriteCond %{HTTPS} !=on
RewriteRule ^.*$ https://%{HTTP_HOST}%{REQUEST_URI} [L,R]
#if proxy to a different box using https enable the next 2 lines
#SetEnv proxy-sendcl 1
#SetEnv force-proxy-request-1.0 1
ProxyPasshttp://127.0.0.1:12000/sipxconfig
ProxyPassReverse http://127.0.0.1:12000/sipxconfig
P.S. I took care of this by removing "base" tag altogether
http://thread.gmane.org/gmane.comp.voip.sipx.devel/6362
Damian never said why tag was needed at all. I read up on the base
tag and i don't think we need it. Basic test verifies we do not need
it.
P.P.S. We using the apache server now that's managed by the OS
instead of launching an apache instance with a separate config file.
P.P.P.S This means we can remove https listener on sipxconfig or any
other internal service if we want and proxy them all thru single
apache server. I'll wait on this one. But it will definitely make
installing web certs easier if nothing else.
Happy days!
--
Regards
------
Gerald Drouillard
Technology Architect
Drouillard& Associates, Inc.
http://www.Drouillard.biz
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Re: [sipx-users] Cannot activate ACD server
On 3/8/2012 10:13 AM, Elwin Formsma wrote: Thanks Gerald Ive now completely removed the second interface. Sipxecs restarted profiles send still the same problem… Do you have ACD checked under server roles? Maybe uncheck and check again -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Cannot activate ACD server
On 3/8/2012 9:57 AM, Elwin Formsma wrote: Ive disabled the seconds interface. I still cannot activate the ACD server. The service has started and everything looks fine. Problems occur when i try to activate the server. Any other logs i can poste? I believe I had a similar problem last week. Try sending profiles to the server. I eventually got it to start working without have to dig into the log files ;-). -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Off topic: Anyone using appia?
On 2/14/2012 4:44 PM, m...@grounded.net wrote: > Kinda confused here > > This was the reply I got today asking for an update since this seems to be > going on and on. This is what I got. > I guess they don't know what people are using with their services but they > did know sipx. I thought someone said they were receiving services on port > 5080 just fine? > > Reply; > > We do not currently have any sipx customers on the platform and have not > had any customers receiving traffic on ports other than 5060 so we have > not run into this issue before. The configuration says that we should > be sending to you on port 5080 but this is not happening. I am working > with the vendor of the specific piece of equipment that is having the > issue to resolve the issue. Sounds like you are trying to do IP auth? Have your sipx gateway register with their server if they cannot send calls to 5080. That is how we have it configured. > My Question;. > >> This is starting to get rather silly. I just tested and the calls are >> still coming in to port 5060. What is the status and why is it that >> other sipx users don't have the same problems? > > ___ > sipx-users mailing list > sipx-users@list.sipfoundry.org > 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 sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Off topic: Anyone using appia?
On 2/11/2012 1:22 PM, m...@grounded.net wrote: > On Sat, 11 Feb 2012 13:04:08 -0500, Tony Graziano wrote: >> They support (we have setup) both IP ACL trunks and registration trunks. >> Both work for us. >> >> You should ask your reseller to assist. It's not a difficult configuration. > As I said, my vendor did work with them, we got into a conference call. There > is nothing wrong with the hardware or the sipx server. Appia themselves said > they needed to add programming each time we tried testing something. > Unless you got lucky and got on a different switch or something, my > experience with them so far is not at all like yours has been. > > The problems are not at our end. > Never had these problems. Maybe because we went on the t.38 servers. I also have to say that I didn't care for the "project management" style of startup. It should be routine, and faster. -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Off topic: Anyone using appia?
On 2/10/2012 5:19 PM, m...@grounded.net wrote:
Is anyone on the list using appia?
Yes. We have one account with them. They have t.38 if you ask for it.
I am not a big fan of fix price "plan" pricing. I miss having a voip.ms
interface, and to be able to self serve, like if you have to call
forward from the ITSP side.
Their phone service has been rock solid though and the setup was a breeze.
They don't yet have a GUI but talk about it being done soon.
Just changing ports takes having to communicate with support.
So far, every time we come to test something, we're told they need to program
that functionality in. We were trying to test a sip to analog gateway to use
with pbx's but nothing worked. They found some problems which took a week to
resolve saying that they didn't cover that functionality.
Now with sipx, they can't send to me on port 5080 yet I thought someone here
mentioned using them and they work fine.
You don't have to do IP auth with them. They do registration also.
Here are my notes for appia:
*Sipx setting Registration Interval: 180 (not 600 - not sure about this
one though)
Usually first password doesn't work. They have to reload trunk group.
When ordering specify 10 digit calling - no +1{phonenumber}* (if you
want voip.ms style)
--
Regards
------
Gerald Drouillard
Technology Architect
Drouillard& Associates, Inc.
http://www.Drouillard.biz
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Re: [sipx-users] Server Recommendations
On 2/9/2012 10:27 AM, Tommy Laino wrote: I am going to be deploying my first SipX after much work and testing on my lab system. Just curious what most of the experts are using for servers on their deployments. I am going to have a very simple setup to start. 20 local sets and 2 remote users using NAT and SIP trunking from the cable company. Very please with one of last servers we made. We were looking for low power high performance. At the time we couldn't find a board over 4G that is fanless. This system is rated at 200W. From Crucial.com Qty: 1 CT2032751 Part Number: CT2KIT51272BA1067 Price: $95.99 Description: 8GB Kit (4GBx2), 240-pin DIMM Upgrade for a Supermicro SuperServer 5017C-LF System If you think the server is going to be lightly used you could go down to 4. I believe concurrent voicemail access is the deciding factor. ---2--- Qty: 1 CT128M4SSD2BAA Part Number: CT128M4SSD2BAA Price: $214.99 Description: 128GB Crucial m4 2.5" SSD w/ 3.5" Adapter Bracket You could probably go down to 64G or 32G for 20 users. Depending on voicemail use/retention. You don't need the bracket version as stated above, but at the time there didn't seem to be a difference in price. From newegg.com Below you will also need a drive cradle for the ssd drive or any 2.5"drive: ($9.99) SRV ACC SUPERMICRO|MCP-220-00051-0N 16-101-383 SERVER_BB SUPERMICRO|SYS-5017C-LF R 1 $359.99 $359.99 19-115-094 CPU INTEL|CORE I3 2120T 2.6G 3M R 1 $134.99 $134.9 -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] FW: Alarm SPX00016: The ITSP Account could not be reached
On 2/9/2012 8:02 AM, Philippe Laurent wrote: > Thanks Gerald! Working on the set up today after end-of-business > day. I'm assuming that for IP Auth that I will have to port forward > 5080 to 5060? > > No. 5080 to 5080 -- Regards ---------- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] FW: Alarm SPX00016: The ITSP Account could not be reached
On 2/9/2012 7:43 AM, Tony Graziano wrote: > If your firewall is configured correctly, why would NAT be set to YES. > Normally it would be set to NO if sipXbridge is in use and advertising > a public IP address (behind NAT). It has always been suggested to have > the provider DISABLE or TURN OFF NAT when using sipxbridge. In the > instances where it does not work for someone, I have always found the > outbound NAT type and/or STATIC port nat were not set properly before > creating the NAT entries, creating a dependency on NAT=YES at the > ITSP. This could potentially lead to the alarms? I never have them, so > I'm just saying... I hear you, and technically you are correct. I have tried everything. In my experience there was a case where 2 separate accounts/locations could not register. The accounts had worked perfectly for months. Nothing changed. After much testing and tweaking, simply switching to NAT=yes made the accounts come back up. IMO something had changed at voip.ms. Anyway it works. And I am sure it works both ways, I just lean to setting it to Yes now. -- Regards ---------- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] FW: Alarm SPX00016: The ITSP Account could not be reached
On 2/9/2012 7:50 AM, Tony Graziano wrote: > for those of you who experience this problem, what is your > registration interval setting? With some ITSP's who have given me this > headache, I found lowering my setting (sometimes 180) really reduced > the problem (voxitas, etc.). It didn't change the frequency of the problem. It appears to affect just how quickly the registration recovers. IMO the registration system on voip.ms can get overloaded. I had captures showing voip.ms rejecting the registration. -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] FW: Alarm SPX00016: The ITSP Account could not be reached
On 2/8/2012 10:48 PM, Philippe Laurent wrote:
Gerald -
After repeated hours and iterations, and I'm at the point where I know
I'm missing something.
I'll swallow my pride and say that I don't quite get how to do what
you've proposed (IP Auth with Voip.ms + SIP URI + what you've done on
the firewall to make it happen). I looked back at the list, and
although I see you mention this in a previous voip.ms <http://voip.ms>
discussion, I guess I missed the part about how it works.
Can you describe for me (us?) what you did where (sipx, firewall,
voip.ms <http://voip.ms>) to make IP Auth to work? My clients have
become very accustomed to voip.ms <http://voip.ms> rates, and if I can
get their on-again off-again on-again service to behave a bit better
(what has changed??), then it's a win-win.
Many thanks in advance for your time and efforts.
Philippe
In your voip.ms account:
DID Numbers
SIP URI's
Create a new SIP URI
The SIP URI can be {DID}@yourhostname.com:5080 or
{DID}@yourIPaddress:5080
Manage DID(s)
Edit each DID and Change Routing from SIP/AIX to SIP URI and
pick the SIP URI you created above.
At this point your inbound calls are not dependent on your
registration status. Most people say the inbound calls
actually connect (first ring) faster than through registration.
Sub Accounts
You may have been registering with the main account in which
case you will not have any sub accounts. You will have to
create one.
If you already have sub accounts you have the choice to change
the existing sub account or create a new one.
Create Sub Account
Authentication type: Static IP
IP Address: Your static Public IP
Username: whatever you want (I don't think it matters)
Device Type: Asterisk,
NAT: yes (unless your server is configured with a public
static IP)
Manage Sub Accounts
Edit the account you want to switch. See "Create Sub Account"
In your Sipxecs Server
Devices
Gateways
Edit Your Voip.ms account
Under "ITSP Account"
Show Advanced Settings
Uncheck "Register on initialization"
Sipx will now say it has to restart some things go ahead and do that.
Sit back and enjoy your registration free connection.
--
Regards
--
Gerald Drouillard
Technology Architect
Drouillard& Associates, Inc.
http://www.Drouillard.biz
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Re: [sipx-users] FW: Alarm SPX00016: The ITSP Account could not be reached
On 2/7/2012 2:12 AM, Tony Graziano wrote:
> realize you cannot do call transfers with the above mentioned method,
> so your mileage may vary.
Yes you can! You have the have the DID configured to come in on 5080.
Example voip.ms SIP URI:
{DID}@pbx1.example.com:5080
or
{DID}@youripaddress:5080
>
>
>
> On Mon, Feb 6, 2012 at 9:31 PM, Gerald Drouillard
> wrote:
>> On 2/6/2012 3:10 PM, Nathaniel Watkins wrote:
>>> Has anyone else seen the below behavor? I seem to get this about once a
>>> day - one minute later I get a message that the ITSP account has recovered.
>>>
>>> I was quick on the draw today - so as soon as the first email came in - I
>>> dialed my cell phone as a long distance call - sure enough, it was routed
>>> thru the PRI (which is my secondary gateway in sipXecs). A minute later
>>> when the email came thru that everything was working - I called it again
>>> and it went thru voip.ms
>>>
>>> I'm guessing that calls are dropped when this happens - although, I've not
>>> gotten any calls about it...
>>>
>>> I suppose I can constantly ping newyork.voip.ms and see if I'm losing
>>> connectivity from my pc (although, it uses a different router).
>>>
>>> Thoughts?
>> This problem started for me about 3 months ago on all my voip.ms
>> clients. Many chats, pings and traces with voip.ms. Changed voip.ms
>> servers many times, tweaked settings but we would still loose registration.
>>
>> DO YOURSELF A FAVOR... switch your voip.ms to IP auth and inbound calls
>> via sip uri. See my posts not to long ago about this.
>>
>>>
>>> -Original Message-
>>> From: sipXecs Alarm Notification Service
>>> [mailto:postmas...@sipx.garrettcounty.org]
>>> Sent: Monday, February 06, 2012 3:04 PM
>>> To: ITStaff
>>> Subject: Alarm SPX00016: The ITSP Account could not be reached
>>>
>>> Message from sipXecs
>>> Alarm: SPX00016
>>> Reported on: sipx.garrettcounty.org
>>> Reported at: 2012-02-06T20:03:39.760207Z
>>> Severity: CRIT
>>> Alarm Text: An attempt to signal the ITSP 'newyork.voip.ms' timed out.
>>> Suggested Resolution: Check your ITSP Account Domain, Proxy and Registrar
>>> settings and restart the SIP Trunking service.
>>>
>>> This message and any files transmitted with it are intended only for the
>>> individual(s) or entity named. If you are not the intended individual(s) or
>>> entity named you are hereby notified that any disclosure, copying,
>>> distribution or reliance upon its contents is strictly prohibited. If you
>>> have received this in error, please notify the sender, delete the original,
>>> and destroy all copies. Email transmissions cannot be guaranteed to be
>>> secure or error-free as information could be intercepted, corrupted, lost,
>>> destroyed, arrive late or incomplete, or contain viruses. Garrett County
>>> Government therefore does not accept any liability for any errors or
>>> omissions in the contents of this message, which arise as a result of email
>>> transmission.
>>>
>>>
>>>Garrett County Government,
>>> 203 South Fourth Street, Courthouse, Oakland, Maryland 21550
>>> www.garrettcounty.org
>>> _______
>>> sipx-users mailing list
>>> sipx-users@list.sipfoundry.org
>>> 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
>> sipx-users@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
Regards
--
Gerald Drouillard
Technology Architect
Drouillard& Associates, Inc.
http://www.Drouillard.biz
___
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List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] FW: Alarm SPX00016: The ITSP Account could not be reached
On 2/6/2012 3:10 PM, Nathaniel Watkins wrote: > Has anyone else seen the below behavor? I seem to get this about once a day > - one minute later I get a message that the ITSP account has recovered. > > I was quick on the draw today - so as soon as the first email came in - I > dialed my cell phone as a long distance call - sure enough, it was routed > thru the PRI (which is my secondary gateway in sipXecs). A minute later when > the email came thru that everything was working - I called it again and it > went thru voip.ms > > I'm guessing that calls are dropped when this happens - although, I've not > gotten any calls about it... > > I suppose I can constantly ping newyork.voip.ms and see if I'm losing > connectivity from my pc (although, it uses a different router). > > Thoughts? This problem started for me about 3 months ago on all my voip.ms clients. Many chats, pings and traces with voip.ms. Changed voip.ms servers many times, tweaked settings but we would still loose registration. DO YOURSELF A FAVOR... switch your voip.ms to IP auth and inbound calls via sip uri. See my posts not to long ago about this. > > > -Original Message- > From: sipXecs Alarm Notification Service > [mailto:postmas...@sipx.garrettcounty.org] > Sent: Monday, February 06, 2012 3:04 PM > To: ITStaff > Subject: Alarm SPX00016: The ITSP Account could not be reached > > Message from sipXecs > Alarm: SPX00016 > Reported on: sipx.garrettcounty.org > Reported at: 2012-02-06T20:03:39.760207Z > Severity: CRIT > Alarm Text: An attempt to signal the ITSP 'newyork.voip.ms' timed out. > Suggested Resolution: Check your ITSP Account Domain, Proxy and Registrar > settings and restart the SIP Trunking service. > > This message and any files transmitted with it are intended only for the > individual(s) or entity named. If you are not the intended individual(s) or > entity named you are hereby notified that any disclosure, copying, > distribution or reliance upon its contents is strictly prohibited. If you > have received this in error, please notify the sender, delete the original, > and destroy all copies. Email transmissions cannot be guaranteed to be secure > or error-free as information could be intercepted, corrupted, lost, > destroyed, arrive late or incomplete, or contain viruses. Garrett County > Government therefore does not accept any liability for any errors or > omissions in the contents of this message, which arise as a result of email > transmission. > > > Garrett County Government, > 203 South Fourth Street, Courthouse, Oakland, Maryland 21550 > www.garrettcounty.org > ___ > sipx-users mailing list > sipx-users@list.sipfoundry.org > 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 sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Sip Vicious and Remote Workers
On 2/5/2012 8:41 AM, Michael Picher wrote: Gerald, Your gz file doesn't seem to be in the same place... I see I had posted a couple of links: http://www.drouillard.biz/fail2ban.tar.gz or http://www.drouillard.biz/sipx_fail2ban.tar.gz They both will work now. Thanks, Mike On Sun, Feb 5, 2012 at 8:23 AM, Gerald Drouillard mailto:gerryl...@drouillard.ca>> wrote: On 2/5/2012 12:20 AM, Tony Graziano wrote: > > Fail2ban requires the firewall use iptables I think. > > You can and should run it on the sipx server. -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org <mailto:sipx-users@list.sipfoundry.org> 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 <http://www.ezuce.com> Hope to see you at the sipX CoLab! http://www.sipfoundry.org/sipx-colab A gathering for - open source users, eZuce customers & eZuce partners Get the inside track on 4.6 and a glimpse at the future of sipXecs! ___ sipx-users mailing list sipx-users@list.sipfoundry.org 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 sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Sip Vicious and Remote Workers
On 2/5/2012 12:20 AM, Tony Graziano wrote: > > Fail2ban requires the firewall use iptables I think. > > You can and should run it on the sipx server. -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Sip Vicious and Remote Workers
On 2/4/2012 11:41 PM, Gerardo Barajas wrote: > Hi members of the list. > ¿Is Fail2ban useful in this situation?? Yes Search the list and you will see how. -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] MOH and Call Park (4.4 latest)
On 2/1/2012 10:58 AM, Tony Graziano wrote: > I have uploaded music on hold to a system for both MOH and Park. I > have set the user to use "Use System Configuration". > > When a call is transferred they hear the MOH. When the user presses > the HOLD button, no MOH is heard. When a call is parked they hear the > correct MOH, but on timeout and during the "transfer back" process > they hear the system (shipping) default MOH which is not what I think > they should hear (I think they should hear the chosen MOH system file > that was uploaded, and would normally hear during a transfer). > > Can anyone else confirm this behavior? > Are you sure your MOH is encoded properly? -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Voip.ms 403 Forbidden Errors after registered for several hours
On 1/25/2012 10:54 AM, Tony Graziano wrote: > I mean real failover (POTS line, cellphone, different branch or hosted > voicemail, since that is all available. That is what I mean also. In the voip.ms interface you still have all the "Routing if Destination Unreachable" options. Maybe you are talking about outbound from the sipx server? I would imaging sipx would bounce to the next gateway if voip.ms was unreachable. -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Voip.ms 403 Forbidden Errors after registered for several hours
On 1/25/2012 10:20 AM, Tony Graziano wrote: > except that, in a registered mode and the server goes away for some > reason, there is an automated failover use case the provider offers. That still applies if the call cannot be connected via URI also. -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Voip.ms 403 Forbidden Errors after registered for several hours
On 1/25/2012 8:51 AM, Nathaniel Watkins wrote: > I'm guessing that you wouldn't need to authenticate inbound calls at all - as > they are simply being forwarded to a sip URI. Exactly. Technically you could have a rule at your firewall to only allow 5080 to the voip.ms server if you wanted. IMO, the registration method is a waste of everyone's resources if you have a static IP. -- Regards ---------- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Voip.ms 403 Forbidden Errors after registered for several hours
On 1/24/2012 10:30 PM, Tim Ingalls wrote:
> Hi. I wonder if anyone has seen this. I have a voip.ms account with a
> few sub-accounts. I am using sipxecs 4.4. I have the voip.ms NAT
> setting for the sub-accounts set for "yes" and I have ports 5060 and
> 5080 forwarded to the same (symmetrical) ports on my sipXecs server.
> All of the sipxbridge gateway settings for this SIP trunk are the
> defaults except for the choice of registration server.
>
> What happens is that after several hours, usually overnight, the
> voip.ms portal still shows that I am registered, but I cannot pass any
> calls. I get a 403 Forbidden message back and hear a fast busy. I have
> tried lots of different settings, but the only thing that seems to
> solve things is to switch the voip.ms portal's NAT setting to "no,"
> wait a minute, switch it back to "yes," and then restart my sipXecs
> services to re-register. I have to do that every day.
>
> If I put the the voip.ms NAT setting to "no" I cannot register.
>
> On the Internet Calling > NAT traversal page I have both check-boxes
> checked. What is frustrating is that I cannot change something and
> immediately test it. I have to wait overnight to see if I have the
> same problem in the morning. I am attaching two sipviewer trace files.
> One is of a call to the 4443 echo test that fails, and the other is of
> the same call that succeeds after toggling the NAT from yes to no to yes.
>
> Does anyone have a clue on this one?
Actually, we have been struggling with intermittent loss of registration
issues with all our voip.ms accounts. We have tried different POP's,
all kinds of different registration settings. We have sent them packet
captures showing the problem but still no progress.
Just yesterday we figured out, you don't have to use registration if you
have a static IP. Everyone says you cannot use IP auth with voip.ms
because they cannot send inbound calls via port 5080. The answer is to
setup you DID's to use SIP URI for inbound and use IP auth for
outbound. Your SIP URI would look like {DID}@pbx1.example.com:5080.
If you don't add the 5080 port then you will not be able to transfer the
call.
--
Regards
--
Gerald Drouillard
Technology Architect
Drouillard& Associates, Inc.
http://www.Drouillard.biz
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Re: [sipx-users] Softphone registration issue with Verizon?
On 1/20/2012 10:45 AM, Michael Picher wrote: > Verizon may be blocking... try a VPN connection on your device back > to your firewall? Blocking or ALG. I know the 4G MiFi's have ALG. Cannot even get around it using TLS. Someone recommended using something like: http://www.cradlepoint.com/news-and-events/news/cradlepoint-announces-support-for-verizon-wireless-4g-lte-network to get around the ALG. Not sure if it works though. > > either way, i'd complain to verizon. :-) and to anyone that makes a router or access point that doesn't have a way to shut off ALG! -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Remote Softphones and ALG
On 1/12/2012 4:43 PM, Michael Picher wrote: > > Tls is currently broken afaik > That is what I thought. Thanks. It would be nice to get it working "if" it allows the client to sneak around their ALG routers. -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Remote Softphones and ALG
On 1/12/2012 10:13 AM, Tony Graziano wrote: > VPN is really the best way short of using an SBC that will handle the > ALG on the sipx side of things. A nice option would be to try the snom > openvpn client using one of the vpn compaitble phones too. > > 3cx tunnel is a 3cx option on 3cx servers. I mentioned that in order draw conversation to TLS. Has anybody had any luck getting softphones connection to sipx using TLS. Looks like it is port 5061. -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
[sipx-users] Remote Softphones and ALG
I was wondering if anybody has had any experience in getting around remote locations (home offices) that have routers with ALG on. We have a install that will have many "home/remote offices" and road warriors. The ALG stuff is everywhere. In some case there is no way to turn off the alg like with Verizon MiFi 4G. I am thinking TLS may do it? Seems like VPN's would be a lot of work. I have tried different ports with not luck. Using x-lite and 3cx as the clients. I noticed 3cx has a "tunnel" feature. -- Regards ---------- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] How to troubleshoot phone registrations
On 12/31/2011 3:05 PM, Scott Howell wrote: > > OK, I resent the profiles to the server as described. I > restarted the required services after this was done. I have > reset my phone and it still did not register after rebooting > the phone. I waited just a bit longer and reset the phone > again and it registered fine. Just to be clear nothing has > changed from last night. Everything was working perfectly > and when I get up this morning they aren't registered. The > send profiles worked apparently, but how do I prevent this? We/you are still looking for the problem. You have to find the problem before it can be fixed/prevented. Your problem is not common, or at least I have not seen it. > > > Back to the sleeping issue . . . Is this normal behavior? > If so how do I prevent the server from sleeping if this is > the case. It was thrown out as a possible cause so quickly > that it seems this is something that people have had > experience with. Have you found evidence that your "server" went to sleep? Is it a laptop? -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] How to troubleshoot phone registrations
On 12/31/2011 2:38 PM, Scott Howell wrote: > > The file var/log/messages only had one line "Dec 31 04:02:33 > sipx syslogd 1.4.1: restart.". Does this mean the system > restarted or just the syslogd service? I would think the > later. correct >Also, send profiles does not work the job simply > fails. In my experience (limited at that) if the phones > aren't registered this wouldn't work. I would assume this > is the desired behavior. It would say "failed" if the phone was not registered or it could not reboot the phone, but the new config files are ready for the phone(s) next time they reboot. -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Voicemails emailed out but still in system??
On 12/31/2011 3:09 PM, Todd Hodgen wrote: One nice thing is that there is typically a ton of voicemail space on the system, since it uses all of the excess disk space. In many systems, where you are having to by storage size, and limit the space that people have, you have to manage that space very closely. Even a simple move the mail to the delete folder after forwarding would be great, as there is a method of removing deleted messages automatically. That script supplied by sipx is already in the cron.daily -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] How to troubleshoot phone registrations
On 12/31/2011 1:52 PM, Scott Howell wrote: > Sorry I should have been a little more specific. The phones > are both local and they were configured by sipX. Also, when > rebooting the phones they will not always re-register the > first time. On a few instances I have to reboot the phone > at least twice and they will register again. So, I expect > that you may be on to something Gerald with the server > sleeping. When I went to the GUI of sipX this morning the > first time it took quite a while to come up and there was > alot of disk activity which usually isn't the case. What > logs should I look at to investigate this possibility? The > phones are running bootrom 4.1.3.0052. > ___ The lack of any log activity for hours would be a clue. /var/log/messages may be a place to look. Also somebody mentioned to "send profiles" to the phone. That is always a good starting point when troubleshooting. -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Voicemails emailed out but still in system??
On 12/31/2011 12:18 PM, Chris Wiegand wrote:
So we switched successfully to sipXecs, which is stable and people are
pretty impressed, but one issue we've run into is that although users
can get their voicemails emailed to them, it's still in the system as
well, so they have to still login to the VM system (or visit the
website) and remove their voicemails from there, making the emails a
little redundant. Our old system, sucky though it was, had an option
to remove them when it emailed them to the user -- anything I'm doing
wrong here? I can't find any checkbox to enable that functionality. If
it's not there, is there any plan to add it to 4.6 or a future
release? Or, is there a way for me to remove them after a period of
time / in batches?
This script moves inbox messages to the deleted folder if they are order
than 90 days (there is a var to change the number of days it you want).
Here is a little thing we drop into /etc/cron.daily:
#!/bin/sh
# voicemail_inbox_clean.sh: moves voicemail messages older than n days
to the deleted folder.
# where a day is defined as a 24 hour period.
check_prop_file_exists() {
local exists=0
if ! test -f "$1"
then
echo "Property file not found: '$1'" >&2
exists=1
fi
return ${exists}
}
get_prop_value() {
# ensure property file exists and then pull out the
# requested property value
check_prop_file_exists "$1" \
&& perl -n \
-e 'use English;' \
-e 's/#.*$//;' \
-e "/^\\s*$2\\s*=\\s*/ && print join( ' ', split( /[\\s,]+/,
\$POSTMATCH ));" \
$1
}
MAILSTORE_DIR=/var/sipxdata/mediaserver/data/mailstore
DAYS=90
# Override the DAYS variable with optional command line argument
if [ "$1" == "--days" ]; then
if [[ "$2" == [1-9] ]]; then
if [[ "$2" < "$DAYS" ]]; then
DAYS=$2
fi
fi
fi
CleanList=`mktemp -t voicemail_inbox_clean.XX`
trap "rm ${CleanList} 2>/dev/null" 0
if [ -d ${MAILSTORE_DIR} ]
then
for deleted_dir in `find ${MAILSTORE_DIR} -maxdepth 2 -type d -name
inbox `
do
if cd "${deleted_dir}" > /dev/null 2>&1
then
# Find all voice messages that are more than $DAYS old.
Base the test
# on the last modified date for the voice message
"envelope" file.
# echo ${deleted_dir}
cat /dev/null > ${CleanList}
for name_prefix in `find . -mtime +${DAYS} -name "*-*.xml"
| cut -d - -f 1`
do
# Remove all files with a .sta, .wav or .xml extension
that have the
# same filename prefix as the old voice message envelope.
for expired in ${name_prefix}-*.{sta,wav,xml}
do
test -f $expired && echo $expired >> ${CleanList}
done
done
if [ -s ${CleanList} ]
then
# Now that we've deleted messages, the summary.xml file
is no longer
# accurate. Delete it so that it gets recreated next
time it is accessed.
test -f summary.xml && echo summary.xml >> ${CleanList}
# rm -f `cat ${CleanList}`
# cat $CleanList
touch -cm `cat ${CleanList}`
mv -t ../deleted `cat ${CleanList}`
fi
else
echo "failed to cd to '${deleted_dir}'" 1>&2
fi
done
else
echo "Mailstore '${MAILSTORE_DIR}' not found" 1>&2
exit 1
fi
--
Regards
--
Gerald Drouillard
Technology Architect
Drouillard& Associates, Inc.
http://www.Drouillard.biz
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] How to troubleshoot phone registrations
On 12/31/2011 9:49 AM, Scott Howell wrote: > > > > I have a system setup for testing and after a while my > phones simply loose their registration. I have two Polycom > 331's setup and they will be working fine. Last night they > were working and when I woke up they just say URL Call > Disabled when you try to go off hook. In the registrations > screen they show expired. > > I would assume it has to do with a reg timer of some kind, > but just don't know where to begin with this issue. > > Did your server go to sleep also? Look at the log files on the server to see if it went to sleep. What is the boot/firmware on the polycom phones? Were the phones provisioned via sipx or did you do it manually? -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Sip Hacking?
On 12/30/2011 7:16 PM, Ken Ridley wrote: Gerald, I have not used fail2ban before, what I got from the quick search I did, is that it blocks connections from IPs The calls are originating from sip:200@(WAN IP of our Router) If fail2ban has other options, can you please direct me to the information I copied the necessary files to http://www.drouillard.biz/fail2ban.tar.gz -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Sip Hacking?
On 12/30/2011 5:17 PM, Ken Ridley wrote: I found about 30 failed calls in our CDR history from user 200 to 5 different numbers There is no User 200 on our system We have remote users, so there are ports opened on the router to allow them to connect, no VPNs Is this something to be expected? All of the calls failed with a 483 error, is this the sipx way of blocking invalid users, or did I just get luck this time If I'm just lucky what can I do to prevent this? ** ** You could probably create a fail2ban rule for this. -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Anyone suddenly unable to register with voip.ms?
On 12/20/2011 2:41 AM, Todd Hodgen wrote: My own Voip.ms account seem to be functioning fine. You may not notice if you don't have the alarms being emailed to you, or you don't have constant voice traffic. -- Regards ------ Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Anyone suddenly unable to register with voip.ms?
On 12/19/2011 9:51 PM, Philippe Laurent wrote: I have two locations, both running sipX and registering with different voip.ms <http://voip.ms> accounts, that have both stopped registering with voip.ms <http://voip.ms>, with no changes to sipX or Voip.ms account settings. Anyone else experiencing the same? If so, have you resolve it, and what did you do to do so? We can register, but in the last month or so we have been loosing our registrations voip.ms from time to time, effectively stopping the in/outbound calling for 5-10 mins. It almost seems like we loose the registration at each site about once a day now. We have tried different POP's and settings in the gateway. Voip.ms doesn't seem to take it seriously. I even had a capture when we lost registration that was passed off as a "normal challenge". I believe there is something going on with their authentication. It is getting bad enough now that we are probably going to switch away from them. It is effecting 4.2.1 and 4.4 systems. -- Regards ---------- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Cisco Linksys SPA8000 - any issues?
On 12/17/2011 11:47 AM, Krzysztof Ślazyk wrote: > > > > Hello, > > Does anybody use SPA8000 with sipx? > Do you have any issues with it? > > We have a few 8800 in use. We had 1 out of 6 die after 1 year. I don't like the fact that each outbound FXO line has to be configure as a gateway, but with the 8000 there are no FXO ports. -- Regards ---------- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] SIPX - RAID Management
On 12/13/2011 8:43 AM, Black, Dave (CallPoint Canada) wrote: I was wondering if anyone had experience with SIPX installed alongside the LSI MegaRaid MSM utilities. The configuration under consideration is a re-purposed ASUS RS500 server with a ASUS Pike 1064E raid controller based on LSI MegaRaid. What we are attempting to achieve is the installation of the MegaRaid server software alongside SIPX for ongoing management and alerting functions. Is anyone doing this, any pit falls, or concerns regarding ongoing configuration management relating to SIPX upgrades... I would suggest using software raid instead of the hardware raid. That way you can take the drives and plug them into a new machine without too much concern about adapter compatibility. -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
[sipx-users] Linux NAT / Firewalls / Shorewall
Just a quick note on Linux firewalls and in this case t.38 aka fax. We had a issue with t.38 working properly on a new install. Everything else was fine. We were seeing a 200 ok message from the ITSP on the phone server just before sending t.38 that was telling the phone server the Owner/Creator was a unknown IP address, meaning the IP address was unknown to both parties. The phone server would then send the t.38 to the unknown address as directed. The ITSP showed the 200 message leaving their system did not have the unknown IP. It turns out the client's linux firewall/server kernel has sip helpers loaded and kicked in only for the t.38. The following clears up the problem: http://www.shorewall.net/FAQ.htm#faq77 or basically: rmmod nf_nat_sip rmmod nf_conntrack_sip Maybe a nice note for the wiki? -- Regards -- Gerald Drouillard Technology Architect Drouillard& Associates, Inc. http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
