Re: [sipx-users] Can not transfer from Auto Attendant

You need to look at the firewall then. On Mar 26, 2012 4:02 PM, "Stiles Watson" wrote: > Well, I looked at the voip.ms registration status and it lists my WAN IP > and port 5080 so it does not seem like a port issue. > > On 03/26/2012 12:30 PM, Gerald Drouillard wrote: > > On 3/26/2012 11:00 AM,

Re: [sipx-users] Can not transfer from Auto Attendant

Well, I looked at the voip.ms registration status and it lists my WAN IP and port 5080 so it does not seem like a port issue. On 03/26/2012 12:30 PM, Gerald Drouillard wrote: 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

Re: [sipx-users] Can not transfer from Auto Attendant

There has always been a voip.ms template. You do need to modify the lines to point it (in two places) to the pop you register to with them. Look at your registration status at their portal. What is it telling you and what firewall are you using? On Mar 26, 2012 2:40 PM, "Stiles Watson" wrote: >

Re: [sipx-users] Can not transfer from Auto Attendant

Read this please... http://blog.myitdepartment.net/?p=191 If you register from sipx and your firewall is configured correctly, it will register TO port 5080 with voip.ms. You can verify this with voip.ms in the portal and see the IP:port you are registred to them. It should show your :5080. If i

Re: [sipx-users] Can not transfer from Auto Attendant

Hmm, That's likely. Here's some info from voip.ms: SIP: 5060 UDPRTP Range: 10001-2 UDP Is there a way to make voip.ms match sipx or do I have to change sipx to match voip.ms (and change my finewall)? When I set up voip.ms, I used this how-to from Tony. However, some things have chang

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 Drouillar

Re: [sipx-users] Can not transfer from Auto Attendant

Anyone else have an idea why I lose audio when retrieving a call from hold or canceling a transfer? Stiles On 03/21/2012 05:37 PM, Stiles Watson wrote: Yes, under Account Settings-->Inbound Settings, I set Device Type to "IP PBX Server, Asterisk or Softswitch". Is this the only place to set i

Re: [sipx-users] Can not transfer from Auto Attendant

Follow the howto. Make sure all the Nat and moh stuff is set according to it. On Mar 21, 2012 5:46 PM, "Stiles Watson" wrote: > Yes, under Account Settings-->Inbound Settings, I set Device Type to "IP > PBX Server, Asterisk or Softswitch". Is this the only place to set it? I > was assuming this

Re: [sipx-users] Can not transfer from Auto Attendant

Yes, under Account Settings-->Inbound Settings, I set Device Type to "IP PBX Server, Asterisk or Softswitch". Is this the only place to set it? I was assuming this was an account level setting and not a DID setting. Stiles On 03/21/2012 05:19 PM, Michael Picher wrote: in the voip.ms

Re: [sipx-users] Can not transfer from Auto Attendant

http://blog.myitdepartment.net/?p=191 On Mar 21, 2012 5:29 PM, "Michael Picher" wrote: > in the voip.ms did you change the account to be 'Asterisk or SIP PBX' > (i'm paraphrasing). > > mike > > On Wed, Mar 21, 2012 at 5:04 PM, Stiles Watson wrote: > >> We'll I signed up for voip.ms and set up th

Re: [sipx-users] Can not transfer from Auto Attendant

What is the firewall? What is the bootrom/firmware if the phones? On Mar 21, 2012 5:14 PM, "Stiles Watson" wrote: > We'll I signed up for voip.ms and set up the gateway for it and now all > calls work in and out of sipxecs... but now resume from hold and canceling > a transfer result in losing

Re: [sipx-users] Can not transfer from Auto Attendant

in the voip.ms did you change the account to be 'Asterisk or SIP PBX' (i'm paraphrasing). mike On Wed, Mar 21, 2012 at 5:04 PM, Stiles Watson wrote: > We'll I signed up for voip.ms and set up the gateway for it and now all > calls work in and out of sipxecs... but now resume from hold and cance

Re: [sipx-users] Can not transfer from Auto Attendant

We'll I signed up for voip.ms and set up the gateway for it and now all calls work in and out of sipxecs... but now resume from hold and canceling a transfer result in losing the call. Again, I'm using Polycom SoundPoint IP 335 phones. While on an active call, I can receive the call and put it

Re: [sipx-users] Can not transfer from Auto Attendant

I would not put much emphasis on the item you looked up. If the call was not acked it means there was a sip trace or pcap associated in order to make that determination. If the call is coming in FROM the ITSP using sipxbridge as a trunk manager, whether you are REGISTERED or NOT, it should send to

Re: [sipx-users] Can not transfer from Auto Attendant

While searching the old messages on this list to see if I can get any insight on this issue, I found this 2010 note from Tony Graziano in a response to a different question: "It means they are not acking the call. I suspect this is because sipxbridge may not be involved in the call, and only

Re: [sipx-users] Can not transfer from Auto Attendant

Ignore the tls messages. They are cosmetic and not in use (and fixed post 4.2, hint). You need to ensure the call comes into sipx on port 5080 (invite) or have flow route configured for your ip/port. A pcap or sip trace would be needed beyond that. I register to flow route and can receive calls f

Re: [sipx-users] Can not transfer from Auto Attendant

OK, thanks. DNS is running on the sipXecs server. I re-installed and used the fully qualified domain during the setup. The auto generated DNS zone file is correct and all the tests pass. However, I still have the initial problem of not being able to transfer to an incoming call to an ext from

Re: [sipx-users] Can not transfer from Auto Attendant

Nettica.com should work. If you only have one server, you don't need srv records, but dns should be running on sipx server and it should be configured automatically On Mar 19, 2012 3:53 PM, "Stiles Watson" wrote: > Are there instructions for setting up SRV records with Godaddy? I do not > plan

Re: [sipx-users] Can not transfer from Auto Attendant

you dont absolutely need the rr record. http://blog.myitdepartment.net/?p=104 On Mon, Mar 19, 2012 at 3:53 PM, Stiles Watson wrote: > Are there instructions for setting up SRV records with Godaddy? I do not > plan to ever ad servers, but if this is the preferred way. If no one like > Godaddy fo

Re: [sipx-users] Can not transfer from Auto Attendant

Are there instructions for setting up SRV records with Godaddy? I do not plan to ever ad servers, but if this is the preferred way. If no one like Godaddy for this, are there DNS providers which work well for this setup? I tried a couple of months ago to get SRV recs set up, but I had a probl

Re: [sipx-users] Can not transfer from Auto Attendant

you only need port 5060 open if you are supporting remote users. the fact that the call transfer fails makes me wonder if flowroute is sending you the inbound call to port 5080 or not. Godaddy supports SRV and SRV at subdomains too. After you fix the internal stuff you might consider chasing dow

Re: [sipx-users] Can not transfer from Auto Attendant

OK, I can do that. But maybe http://wiki.sipfoundry.org/display/sipXecs/DNS+Concepts+for+sipXecs should be edited to say that instead of instructing you to change it: "under *system administration*, go into the *System* menu and select the *Domain* menu item. Change the *domain name* to the fu

Re: [sipx-users] Can not transfer from Auto Attendant

Changing the SIP domain is a bad thing to do... If you want to use A-Records, you should start that way. I'd suggest another reinstall and do it with A-Records Be aware if you use A-Records you can not just add servers to the cluster. This assumes SRV records. Mike On Mon, Mar 19, 2012 at

[sipx-users] Can not transfer from Auto Attendant

This is a follow up question to a problem I submitted about 2 months ago. Long story short, I reinstalled sipxecs 4.2 and made the sipxecs server the DHCP and DNS servers as well. This solved some of my issues, but not all. I still am unable to transfer incoming calls to any extension. I can di