Re: [sipx-users] Help with Patton gateway

http://track.sipfoundry.org/browse/XX-8652 Once again Dale solves the mystery. Should this be reported to FreeSWITCH? From what I've seen they have a pretty good track record for fixing bugs. On 07/16/2010 07:49 PM, Josh Patten wrote: > The reason why this is an issue is because for end users mi

Re: [sipx-users] terminate registrations

From: sipx-users-boun...@list.sipfoundry.org [sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler [dhub...@ezuce.com] 2010/7/15 Miguel Jesús Díaz Macedo : > How can I terminate SIP registrations manually?, I have restarted de > registar se

Re: [sipx-users] Random Dropped Registrations

From: sipx-users-boun...@list.sipfoundry.org [sipx-users-boun...@list.sipfoundry.org] On Behalf Of m...@grounded.net [m...@grounded.net] The only ntpd mention is the running process which never changes pid. grep "ntpd" really-big-file 6374 ?SLs

Re: [sipx-users] Help with Patton gateway

Also... What if you were transferring hoping the person picked up, it went to their VM, so then you wanted to go ahead and go through with the transfer once their VM picked up. Maybe not the best way to do it, but I'm sure someone will try it. -Original Message- From: Josh Patten Send

Re: [sipx-users] Help with Patton gateway

The reason why this is an issue is because for end users migrating from another system, as in my case, They are used to the "Transfer->dial number->Transfer" method of transferring calls. The "Transfer->Blind->Dial Number" concept is foreign to them so out of force of habit they use the former

Re: [sipx-users] Random Dropped Registrations

> (One thing to check: is ntpd.conf's modification date reasonable? Looks right. That's about the time I last changed anything on it. # ls -la /etc/ntp.conf -rw-r--r-- 1 root root 784 Jul 13 17:36 /etc/ntp.conf # more /etc/ntp.conf tinker panic 0 restrict default kod nomodify notrap nopeer noque

Re: [sipx-users] Help with Patton gateway

I concur with Matt's last two posts. I just tested attended transfer as described and had the same results -- fine to VM, failed to AA. Blind transfers are fine. Running 4.0.4 w/Polycom 650 3.1.3c split, BootROM 4.2.2 Of course, one might wonder why we need an attended transfer to AA (other

Re: [sipx-users] Help with Patton gateway

But what would you expect? One would expect an attended transfer to AA or VM to work, that is the issue he is reporting. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems He

Re: [sipx-users] Help with Patton gateway

On 7/16/10 6:29 PM, Josh Patten wrote: >From any phone, call any other phone. Then do an attended transfer from either phone to a FreeSWITCH based service (auto attendant, voicemail, or conference). Report back results and vote on http://track.sipfoundry.org/browse/XX-8652

Re: [sipx-users] Help with Patton gateway

Attended for everything in what I described below. Blind is fine. On 7/16/2010 5:41 PM, Josh Patten wrote: > Attended transfer to VM or blind? > > I haven't seen any issues with blind transfer. > > On 07/16/2010 05:32 PM, Matthew Kitchin (public/usenet) wrote: > >> A little more info, 4.0.4 se

Re: [sipx-users] Help with Patton gateway

Attended transfer to VM or blind? I haven't seen any issues with blind transfer. On 07/16/2010 05:32 PM, Matthew Kitchin (public/usenet) wrote: > A little more info, 4.0.4 seems fine when transferring to someone's VM, > but fails when transferring to Auto attendant. IT totally fails going to > th

Re: [sipx-users] Help with Patton gateway

A little more info, 4.0.4 seems fine when transferring to someone's VM, but fails when transferring to Auto attendant. IT totally fails going to the AA. 4.2.1 seems to exhibit the same behavior (described below) when transferring to either VM or AA. On 7/16/2010 5:15 PM, Matthew Kitchin (public

Re: [sipx-users] Help with Patton gateway

From any phone, call any other phone. Then do an attended transfer from either phone to a FreeSWITCH based service (auto attendant, voicemail, or conference). Report back results and vote on http://track.sipfoundry.org/browse/XX-8652 On 07/16/2010 05:21 PM, Michael Scheidell wrote: On 7/16/10

Re: [sipx-users] Help with Patton gateway

I can test anything for about the next 10 min. I have a personal interest in it too... I'm tentatively planning an upgrade from 4.0.4 to 4.2.1 this weekend. I"ll go through with it if I know it doesn't at least make things worse. On 7/16/2010 5:17 PM, Tony Graziano wrote: > I can test with 4.2.1

Re: [sipx-users] Help with Patton gateway

On 7/16/10 6:17 PM, Tony Graziano wrote: I can test with 4.2.1 and sipxbridge tomorrow. i can test with a cisco and polycom :-) so my brain isn't fried, and I don't need to re-read everything, exactly what do you want tested? -- Michael Scheidell, CTO Phone: 561-999-5000, x 1259 > *| *

Re: [sipx-users] Help with Patton gateway

I can test with 4.2.1 and sipxbridge tomorrow. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Custom

Re: [sipx-users] Help with Patton gateway

I just tested. I think I'm seeing the same thing. On 4.2.1, if I do attended transfer to the auto attendant, the call tranfers, but appears to stay on hold on the transferring phone. On 4.0.4, it appears to be even more broken. The call stays on hold and it doesn't actually transfer the call eith

Re: [sipx-users] Help with Patton gateway

Can I get other people on this list to test this scenario as well? It'll only take a couple of minutes. I know all of you have at least two phones on your desk :-P Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 7/16/2010 4:55 PM, Tony Graziano wrote: > I h

Re: [sipx-users] Help with Patton gateway

I hope its fixable with a config change and not a deep down inside issue. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.842

Re: [sipx-users] Help with Patton gateway

Something tells me that's not the case. It seems all the major open source VoIP platforms (Asterisk, FreeSWITCH, YaTE, etc.) all have some kind of problem dealing with REFER in some way or another. This is why I had to buy an Audiocodes gateway instead of continuing to use my Adtran because REF

Re: [sipx-users] Help with Patton gateway

I'm only talking about attended transfers to FreeSWITCH media services (AKA conference, voicemail, and auto attendant) not phone-to-phone transfers. Phone-to-phone transfers are working fine. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 7/16/2010 4:48 P

Re: [sipx-users] Help with Patton gateway

On 7/16/10 5:45 PM, Josh Patten wrote: I'm running 4.2.1 I have just confirmed this has issues with Aastra phones as well. I've been saying for a while that FreeSWITCH has issues with the way attended transfers are handled. http://wiki.sipfoundry.org/display/xecsuserV4r2/Custom+FreeSWITCH+progr

Re: [sipx-users] Help with Patton gateway

I'm running 4.2.1 I have just confirmed this has issues with Aastra phones as well. I've been saying for a while that FreeSWITCH has issues with the way attended transfers are handled. http://wiki.sipfoundry.org/display/xecsuserV4r2/Custom+FreeSWITCH+programming is a prime example. Fix the Fr

Re: [sipx-users] Help with Patton gateway

Sipxbridge Is not involved in the call scenario, which he details out. This surely needs to be addressed though. I'll try to compare a call tomorrow on a 4.0.4 system with the same gateway. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi.

Re: [sipx-users] Help with Patton gateway

There was an in issue with sipxbridge and attended transfers back in 4.0.2 I believe. There was a patch for it and it was fixed in 4.0.4. Not sure if this is your issue, I don't see what version you are running on this system. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-bo

Re: [sipx-users] Help with Patton gateway

My apologies, the download is posted and made available on June 30, but only to customers with maintenance, guess that means we can start a clock. I'll see what I can do to put my hands on it next week. On Fri, Jul 16, 2010 at 5:03 PM, Josh Patten wrote: > Again, no 3.3.0 > > Josh Patten > A

Re: [sipx-users] Help with Patton gateway

Also I don't think it's a Polycom problem. This only happens when doing attended transfer to FreeSWITCH services. Attended transfer to everything else (including park) works fine. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 7/16/2010 4:00 PM, Josh Patt

Re: [sipx-users] Help with Patton gateway

Again, no 3.3.0 Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 7/16/2010 4:03 PM, Tony Graziano wrote: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html look here. On Fri, Jul 16, 2010 at 5:00 PM, Josh Patten

Re: [sipx-users] Help with Patton gateway

http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html look here. On Fri, Jul 16, 2010 at 5:00 PM, Josh Patten wrote: > http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip550.html > > I don't see it >

Re: [sipx-users] Help with Patton gateway

http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip550.html I don't see it Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 7/16/2010 3:49 PM, Tony Graziano wrote: FWIW - Firmware 3.3.0 is now posted... though you may still have the same proble

Re: [sipx-users] Help with Patton gateway

FWIW - Firmware 3.3.0 is now posted... though you may still have the same problem. On Fri, Jul 16, 2010 at 12:36 PM, Josh Patten wrote: > I have now confirmed this is not a problem with the gateways. I posted a > ticket here: > http://track.sipfoundry.org/browse/XX-8652 > Even with firmware 3.1.3

Re: [sipx-users] Inbound SIP URI Dialing

From: sipx-users-boun...@list.sipfoundry.org [sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tim Byng [...@missioninc.com] I just set up sip.mycompany.com as a domain alias in sipXecs. I don't have the DNS records register

Re: [sipx-users] Inbound SIP URI Dialing

I don't pretend to know how a 3cx does things, but sip uri dialing should do a dns srv lookup, which is not the same thing as a hostname. Why don't you dial something globally dialable and see if you get out? 1...@loligo.com (screaming monkeys followed by echo test) BTW - If the 3CX softphone u

[sipx-users] Inbound SIP URI Dialing

I am currently testing inbound SIP URI dialing, and it appears to be partially working. I just set up sip.mycompany.com as a domain alias in sipXecs. I don't have the DNS records registered yet, so I updated my windows hosts file on an external machine that has a 3xc softphone installed. If I dial

Re: [sipx-users] Random Dropped Registrations

I think it would be good to understand if the sipx startup script will start whether or not ntp is running. I think he is simply vaguely referencing the fact that ntp is checked. Is there a dependency on ntp or ntpd in sipx that would be causing it to update itself or update his system? I ask thi

Re: [sipx-users] Random Dropped Registrations

Any of a million things could be happening. (One thing to check: is ntpd.conf's modification date reasonable? If its date got mis-set *into the future*, some process might be trying to restart NTP over and over, because it always thinks ntp.conf was just changed.) But to find out "what is rest

[sipx-users] Upgrading from 4.0.4 to 4.2.1 - any last minute words of wisdom?

I need to upgrade from 4.0.4 to 4.2.1 in the next week or so. I have tested everything I know to test, and don't see any issues I can't deal with. I'm currently running -4.0.4 -CentOS 5.4 64 Bit. -Polycom Soundpoint 450s and 550s -Polycom Soundstation 6000s -All Polycoms running Bootrom 4.2.1 and

Re: [sipx-users] 5 digit caller ID going out on all calls, but need 10

Er... I'm not so sure. I would not transform the extension or anything, simply let sipx send 10 digits as the callerid for a single account. It might simply be how your CM is looking at the invite and how your trunk is configured. If it is setup as an unmanaged gateway it will probably send 10 di

Re: [sipx-users] Recommended Polycom SoundPoint SIP Application and BootROM versions for sipXecs 4.2.0

On Thu, Jul 15, 2010 at 11:46 PM, Josh Patten wrote: > 3.2.2 has the SLA bug > 3.2.3 has the latency bug > 3.1.3RevC has an issue with EHS adapters for wireless headsets where if a > monitored extension is ringing then it signals the headset to ring as well. > 3.1.3RevC is stable but is missing my

Re: [sipx-users] Clarification needed for 4.0.1 to 4.2.0 update warnings and errors

On Thu, Jul 15, 2010 at 4:16 PM, Tony Graziano wrote: > I would have bumped to 4.0.4 before going to 4.2. > I just finished upgrading and testing my production box. Everything appears to be working as expected. Here's the upgrade path I took: 4.0.1 >> 4.0.2 >> 4.0.4 >> 4.2.0. Thanks for your he

Re: [sipx-users] 5 digit caller ID going out on all calls, but need 10

Good point. We'll do a trace at each point to confirm my assumption. Since sipX won't let me change the Extension pool to 10 digits, and 7 digits or 5 digits can be sent out with caller ID from the User Extension, I'm guessing sipX is the culprit. Still, it's best to be sure. Thanks. From: T

Re: [sipx-users] Help with Patton gateway

I have now confirmed this is not a problem with the gateways. I posted a ticket here: http://track.sipfoundry.org/browse/XX-8652 Even with firmware 3.1.3revC this is still happening. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 7/16/2010 10:29 AM, Tony Gr

Re: [sipx-users] 5 digit caller ID going out on all calls, but need 10

Rather than guessing as to what is sending what between these different devices, it might be prudent to get a PCAP on the server port and look at the individual packets to see what is being sent out of the server for caller ID. It might be sending out what you are requesting in the different flav

Re: [sipx-users] "Internal Error" trying to activate in Device files 3.1.3revC for Polycom phones

I have not been able to duplicate this. At any time did you touch or manually put any files in tftproot? Before you deleted the files did you verify they had the proper ownership and permissions? I've seen this type of behavior when people either uploaded or unzipped files in there manually, so I

Re: [sipx-users] "Internal Error" trying to activate in Device files 3.1.3revC for Polycom phones

Douglas, Tony G identified http://track.sipfoundry.org/browse/XX-6856 While the same error, the discussion on the conditions causing it (in the above) are a bit different. Regards, Don McIlvin Telecom Department Realogy/NRT/Coldwell Banker 52 Second Ave. (3rd Floor) Waltham, MA 02451 Office 78

Re: [sipx-users] Help with Patton gateway

So the question still remains if it happens with firmware 3.13RevC. Its the polycom complaining... 3.2 aint all that. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpd

Re: [sipx-users] Help with Patton gateway

EDIT - This also seems to be occuring with my Audiocodes gateways as well so apparently it's not isolated just to the Patton gateways. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 7/16/2010 10:23 AM, Josh Patten wrote: I'm forwarding the support request

Re: [sipx-users] Random Dropped Registrations

On Fri, 16 Jul 2010 10:00:52 -0500, Tran, Ly V. wrote: > What's with the Sat Dec 19, 2009 date and time? Not sure, something to do with the problem of time skipping back and forth because every time I look at the system time, it is correct. Might be related to the ntpd restarting? > -Origin

Re: [sipx-users] Random Dropped Registrations

Early in this thread, I was reminded that I should not run ntpdate when ntpd is running. I had a cron job running every hour, which would stop ntpd, run ntpdate, then restart ntpd. That of course was removed. Now, as it should, ntpd is started when the system boots and should maintain the clock,

Re: [sipx-users] 5 digit caller ID going out on all calls, but need 10

We are using a SIP trunk group to Avaya CM via a Session Manager server. No analog. I had also tried setting the User's caller ID to the 10 digit number again for giggles, and it sends only 5 digits still. I tried setting the number. The User extension seems to be the only thing that has any

Re: [sipx-users] Recommended Polycom SoundPoint SIP Application and BootROM versions for sipXecs 4.2.0

On Thu, Jul 15, 2010 at 11:46 PM, Josh Patten wrote: > 3.2.2 has the SLA bug > 3.2.3 has the latency bug > 3.1.3RevC has an issue with EHS adapters for wireless headsets where if a > monitored extension is ringing then it signals the headset to ring as well. > 3.1.3RevC is stable but is missing my