The wiki states: http://wiki.sipfoundry.org/display/sipXecs/Managing+Device+Firmware
Activating Device Files You can activate uploads from either the upload edit page or the manage uploads page. You can only activate one type of upload at a time and once activated, you cannot upload or delete files until you inactivate them. Inactivating an upload will make the files unavailable to the device. On Thu, Mar 10, 2011 at 8:34 PM, Tim Ingalls <[email protected]> wrote: > I figured out how to get the application and BootROM changed on the Polycom > phone. First you have to deactivate the device files. Then you have to make > the changes. Then you have to activate the device files. Then send the > profiles to the phones. It even rolled back the BootROM to 4.2.2. Hooray! > > > The skipping is still audible, but barely. It's mostly noticeable now when I > use the speakerphone. > > Thanks, > > Tim Ingalls > Shared Communications, Inc. > 801-618-2102 Office > > > > Tim Ingalls wrote: > > Tony, the system is a testing system in service for my home office with only > a few phones. > > I've tried rolling the firmware back, but it won't download and change the > version. When I delete the newer device file and then load the 3.1.3C file > and send the profile to the phone or reboot it, the phone just loads the > same application file that's already on the phone. Can you help me solve > that? How do I force it to download the new app file? I'm running BootROM > spip_ssip_BootROM_4_3_0_release_sig.zip and can't manage to downgrade that > either. Are the 3.1.3C app and the 4.3.0 BootROM incompatible or something? > > I'll give VLAN prioritization a try (if I can figure out how to do it on the > old Cisco 2900 I bought for $10), but there really isn't a lot of traffic on > my LAN, so I don't think that's the issue. It only happens at the beginning > of a sound file, but it would seem that if VLAN prioritization is the > problem then the stuttering would happen throughout the playback of the > sound file. > > Why does sipX need so much RAM? My Trixbox/Asterisk system only needs 512KB > and runs fine, including the audio. I thought I read that FS does a lot more > with less RAM than Asterisk. > > Would it help to switch the kernel to a realtime version for CentOS? > > Thanks, > > Tim Ingalls > Shared Communications, Inc. > 801-618-2102 Office > > > > Tony Graziano wrote: > > Your ram setting is a bit small. I run 2GB in a lab environment with > 2-3 phones for testing, I wouldn't in production run less than 4GB (64 > bit OS preferred). Your vlan for voice (assuming you have one) needs > to be prioritized over your other vlans, etc. > > There is nothing wrong with the wav files. > > /usr/share/www/doc/stdprompts_en > > At the same time, FS (what josh said, use firmware 3.1.3 RevC) has no > fix date. It's not fixed in sipxecs 4.4 because the FS guys have not > prioritized it yet. > > On Wed, Mar 9, 2011 at 11:17 PM, Josh M. Patten <[email protected]> > wrote: > > > Also if you tell FreeSWITCH not to use G.722 this problem goes away. > ________________________________ > From: [email protected] > [[email protected]] on behalf of Tim Ingalls > [[email protected]] > Sent: Wednesday, March 09, 2011 9:05 PM > To: Discussion list for users of sipXecs software > Subject: Re: [sipx-users] Voicemail IVR Voice Stutters During Playback > > Sorry this is such an old issue, but I'm revisiting it since I'm still > having the problem even on other phones. > > I'm wondering if the problem comes from an OS or hardware problem. I wonder > if my CPU has enough power to process the files. It only happens at the > beginning of a phrase. I don't know if everything is just a .wav file and > that maybe the delay is caused from a delay at reading the file in question > or processing it. I have a single processor Pentium4 3.2 GHz w/ 2GB RAM and > a SATA hard drive. > > I think the issue is that the Polycom phone is reproducing the sound so well > that it shows all of the skips. If I listen close on other phones (like an > analog phone connected to a Grandstream HT386 ATA) I can hear the skipping > but it's not as pronounced. > > Where do I find all of the files that make up the voicemail and AA prompts? > I'll download and open them to make sure they aren't bad somehow. > > Thanks, > > Tim Ingalls > Shared Communications, Inc. > www.sharedcom.com > 801-618-2102 Office > > > > Michael Picher wrote: > > is this just at the very beginning of the message? > > could you download the message to your PC to see if the actual message like > that? This will determine if it is a recording issue or a playback issue. > > Mike > > On Tue, Jan 18, 2011 at 6:13 AM, Tony Graziano > <[email protected]> wrote: > > > There is a one right way, and multiple wrong ways to upgrade > bootrom/firmware in sipxecs. > When you upload polycom firmware or bootrom, it should still be zipped, > and uploaded using the "polycom files" dropdown box" > Are you using lldp for phone discovery or SLA? If not, firmware 3.1.3RevC > s safest to use (IMO). > > On Mon, Jan 17, 2011 at 11:24 PM, Tim Ingalls <[email protected]> wrote: > > > I'm running BootROM 4.3.0.0246 (which is Rev. B) and SIP 3.2.1.0054. Do I > need to roll back the BootROM to 4.2.2? > > I tried loading 4.2.2 into sipX-config (Devices >> Device files) and > rebooting the phones, but the same BootROM (4.3.0) is still coming up when I > go to the phone's status page. What's the correct way to roll back the > firmware? > > By the way, when you say "firmware" are you referring to the sip.ld > application or the BootROM? > > Thanks, > > Tim Ingalls > Shared Communications, Inc. > 801-618-2102 Office > > > > Tony Graziano wrote: > > Most importantly, there are issues of certain versions of polycom > firmware. What are the phones running? > > On Mon, Jan 17, 2011 at 7:20 PM, Tim Ingalls <[email protected]> wrote: > > > Hi. I'm not sure why, when I play back voicemail messages, the IVR voice > stutters. It sounds like it is stuttering at the beginning of each > phrase. It sounds like really bad jitter/latency. The actual voice > message sounds just fine. It's just the IVR voice that is the problem. > > Playback is on a Polycom IP 670 and a Polycom IP 450 using the default > g.722 codec. I'm running sipXecs 4.2.1 on a Pentium 4 running a 3.0 GHz > single-core processor and 2GB of DRAM. My install is based on the latest > 4.2.1 ISO. I have plenty of free RAM and am not using the swap memory. > I'm running Asterisk on a box that has less horsepower and I have no > such problems. > > Here's the performance output from the top command: > > top - 17:09:43 up 22 days, 20:02, 1 user, load average: 0.07, 0.07, > 0.02 > Tasks: 130 total, 1 running, 128 sleeping, 0 stopped, 1 zombie > Cpu(s): 0.2%us, 0.2%sy, 0.0%ni, 99.7%id, 0.0%wa, 0.0%hi, 0.0%si, > 0.0%st > Mem: 2067248k total, 1896720k used, 170528k free, 302592k buffers > Swap: 6289436k total, 0k used, 6289436k free, 692344k cached > > While checking VM or leaving a VM message, the CPU never appears to go > over 1% to 5% usage. > > Does anyone have any ideas on how to fix this? > > -- > Thanks, > > Tim Ingalls > Shared Communications, Inc. > 801-618-2102 Office > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.326.5325 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > > Helpdesk Contract Customers: > http://support.myitdepartment.net > Blog: > http://blog.myitdepartment.net > > Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.326.5325 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > > Helpdesk Contract Customers: > http://support.myitdepartment.net > Blog: > http://blog.myitdepartment.net > > Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > -- > There are 10 kinds of people in this world, those who understand binary and > those who don't. > > [email protected] > blog: http://www.sipxecs.info > call: sip:[email protected] > > ________________________________ > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > > > ________________________________ > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
