Re: [sipx-users] Cisco 7970G Firmware Upgrade

then either you didnt unzip the files in put them in tftp root or you have a bad sip firmware download. On Wed, Apr 7, 2010 at 2:36 AM, Rhon wrote: > Hi Tonny, > > I made every steps below but still unable to meet load the firmware > successfully. There is an attempt to load the firmware and you

Re: [sipx-users] Cisco 7970G Firmware Upgrade

Hi Tonny, I made every steps below but still unable to meet load the firmware successfully. There is an attempt to load the firmware and you can see it via the Cisco Phone screen. On Wed, Apr 7, 2010 at 1:42 PM, Tony Graziano wrote: > you cant use sccp firmware on a sip system. > > Yes, I know.

Re: [sipx-users] Cisco 7970G Firmware Upgrade

Hi Ben, I have all the files pointed properly in the term70.default.loads. I checked the file names very carefully and match it with the firmware files. So it shouldn't be the problem. All the files are in the tftp server. And it located properly in the tftp-root. I can see the progress screen d

Re: [sipx-users] Cisco 7970G Firmware Upgrade

you cant use sccp firmware on a sip system. 1) Have tftpd32 running as both tftp and dhcp server 2) boot the phone and hold down the # key until the buttons cycle orange 3) press 3 4 6 1 6 7 2 8 5 0 * # 4) light the blue touch paper and stand well back 5) the phone will download the image from

Re: [sipx-users] Cisco 7970G Firmware Upgrade

Rhon, ensure that the term70.default.loads file is directing the phone to the following files: jar70sip.8-5-4TH1-6.sbn cnu70.8-5-4TH1-6.sbn apps70.8-5-4TH1-6.sbn dsp70.8-5-4TH1-6.sbn cvm70sip.8-5-4TH1-6.sbn and also that those files are available within the root directory of the tftp server. the

Re: [sipx-users] Cisco 7970G Firmware Upgrade

Hi Everyone, I'm stuck with the firmware upgrade. :( I tried reverting to SCCP and load the firmware via tftp, everything went well! No hassle at all. But every time we load the SIP-8.5.4 firmware we only get the term70.default.loads file and reboot in circles. I even made a direct connection be

Re: [sipx-users] Dial Tone delayed after when phone is hanged

sounds like a dialplan ont he phone. On Wed, Apr 7, 2010 at 1:11 AM, Rhon wrote: > Hi, > > We noticed after calling or receiving a call there was at least more that > 10secs of delay before getting a new dial tone? > > We already put down the handset, however the phone is still busy whenever a >

Re: [sipx-users] polycom 650 HD voice

makes me sound like a rock star On Wed, Apr 7, 2010 at 12:34 AM, Josh Patten wrote: > Many people I talk to say it sounds like the person they are talking to is > in the room with them. A few people (I think they may have hearing loss > issues) think it sounds like the other caller is in a tunne

[sipx-users] Dial Tone delayed after when phone is hanged

Hi, We noticed after calling or receiving a call there was at least more that 10secs of delay before getting a new dial tone? We already put down the handset, however the phone is still busy whenever a call is established again on the same phone no. Is there a way to adjust this? We are using C

Re: [sipx-users] polycom 650 HD voice

Many people I talk to say it sounds like the person they are talking to is in the room with them. A few people (I think they may have hearing loss issues) think it sounds like the other caller is in a tunnel for a few calls, then they get used to it and are glad they can hear everything that's

Re: [sipx-users] Recommendations for FXS, ATA

The Mediant 1000 is rock solid for me. I've never had to restart it or take the PRI circuit down for any reason. I bought an extra power supply just in case one blew out (it can take 2 power supplies) but haven't had an issue yet. I also agree about the configuration. One day you just "get it"

Re: [sipx-users] polycom 650 HD voice

> Isn't the 330 a non-HD unit? I have 650's and 335's and calling between a > 335 and a 650 results in the little 'HD' logo on the line appearance and > great sounding audio (handset or speakerphone). Yes, it's non HD. My test was just to see if the two polycom's talking together would sound any

Re: [sipx-users] Recommendations for FXS, ATA

>- If you decide you want to manually configure because sipxecs doesn't > cover the way you want to use it you may want to kill yourself with a dull > butter knife (the configuration is not obvious, and the documentation is of What I found in learning about these things was that the initial le

Re: [sipx-users] polycom 650 HD voice

Isn't the 330 a non-HD unit? I have 650's and 335's and calling between a 335 and a 650 results in the little 'HD' logo on the line appearance and great sounding audio (handset or speakerphone). -Eric On Apr 6, 2010, at 11:40 AM, m...@grounded.net wrote: > I have a number of 330's and a while

Re: [sipx-users] Recommendations for FXS, ATA

A couple of things to watch out for using Audiocodes: - They do not post docs and firmware on their open website, you have to buy support (even for a new unit) to get access; they do allow their distributors to 'give away' the firmware (if you bought it from an official distributor that migh

Re: [sipx-users] Remote Polycom registration - not working

Tony, Firstly, thanks for the quick response. I think I covered, or at least tried to, most of what you have suggested already. You must have also enabled sipx to support remote workers and the server is behind NAT. "I have all my internal and external SRV records configured, and have foll

Re: [sipx-users] Recommendations for FXS, ATA

Sorry about that… this(these) would be used for one, two or three cordless handset extensions that can be used when folks are away from their desks. They won’t be assigned to a particular user so no v-mail box will be configured and thus MWI support will not be necessary. Likely the extensions wil

Re: [sipx-users] Recommendations for FXS, ATA

It really helps if you describe what it will be used for. I would not want to use just ANY fxs device for a fax machine, for example. This also matters with what you are expecting from an end user perspective (immediate dial, ringown device, mwi support, etc.). On Tue, Apr 6, 2010 at 10:00 PM,

Re: [sipx-users] Recommendations for FXS, ATA

I always use the audiocodes MP-11x gateways because they're well documented and "supported" on sipX. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 4/6/2010 9:00 PM, Paul Herron wrote: Can folks share their opinions of the best FXS ATAs to use with SipX

[sipx-users] Recommendations for FXS, ATA

Can folks share their opinions of the best FXS ATAs to use with SipXecs? In particular, I’m interested in which products are both robust and can easily be configured and up and running in a short period of time. I would think that the ability to configure via the SipX administration web console wo

[sipx-users] cdr entries

I had some problems with my server today, all the phones lost registration and the server wouln't accept calls from the trunk. I restarted it with /etc/init.d/sipxecs restart and it all came back, just noticed some entries in the cdr log From To Start Duration Status sip 011442076380

Re: [sipx-users] ACD sound quality / clicking noise

On 2010-04-07, at 00:53, Remco van Vugt (Raffel Internet) wrote: > Hi, > > I've been testing Sipxecs (4.0.4) running on both Xen (plenty of resources) > and a dual-PIII - 1.4 GHz with 4GB of RAM. Sipxecs turns out to be a great > piece of software, and nearly ready for production use, except

Re: [sipx-users] ACD sound quality / clicking noise

The ACD does certainly have its faults. I have not seen this problem however when I have used it. Mike -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of mkitchin.pub...@gmail.com Sent: Tuesday, April 06, 2010 7:31

Re: [sipx-users] Cisco 7970G Firmware Upgrade

Hi, Here's the result of /var/log/messages: Apr 7 08:23:43 sipxgw xinetd[6785]: Started working: 1 available service Apr 7 08:24:31 sipxgw dhcpd: DHCPREQUEST for 10.9.20.147 from 00:19:56:d6:6f:c9(SEP001956D66FC9) via eth0 Apr 7 08:24:31 sipxgw dhcpd: DHCPACK on 10.9.20.147

Re: [sipx-users] Remote Polycom registration - not working

You must have also enabled sipx to support remote workers and the server is behind NAT. On Tue, Apr 6, 2010 at 7:56 PM, John Clark wrote: > Hoping maybe someone has some ideas about why I cannot get a Polycom 450 > remote phone to register. > > > > I have a test remote network setup, and have v

Re: [sipx-users] Remote Polycom registration - not working

Then don't do any of those things. 1. Confirm your firewall at the remote end does not have an ALG or SPI turned on. 2. Confirm your firewall where sipx is installed is handling NAT via symmetric port nat. Try and register a softphone FIRST. After your softphone registers properly, make sure you

[sipx-users] Remote Polycom registration - not working

Hoping maybe someone has some ideas about why I cannot get a Polycom 450 remote phone to register. I have a test remote network setup, and have verified that if I send any traffic from it to my sipx proxy I can see the traffic at my local firewall. I am allowing all traffic from the remote netw

Re: [sipx-users] ACD sound quality / clicking noise

I can only say I don't have this problem with our ACD queue. It is for our helpdesk with 5 agents and they field several hundred calls a day. I'm using polycom phones and Cisco switches. Do wireshark captures show anything? Maybe try a different switch and handset vendor. Others may know more, b

[sipx-users] ACD sound quality / clicking noise

Hi, I've been testing Sipxecs (4.0.4) running on both Xen (plenty of resources) and a dual-PIII - 1.4 GHz with 4GB of RAM. Sipxecs turns out to be a great piece of software, and nearly ready for production use, except for one annoying problem: a clicking sound is audible on calls being bridged

Re: [sipx-users] FCC loses key ruling on Internet `neutrality' - wtop.com

At the same time SIP is predictable torrent traffic is able to make thousands of connections and saturate circuits unchecked. The FCC never had the authority and needed to be reined in. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@m

Re: [sipx-users] Disable ring on presence speed dial for parked call

I thought I checked everywhere, but I could have easily missed it. I'm at the airport now, but I will check again tomorrow. --Original Message-- From: Picher, Michael To: Matthew Kitchin (public/usenet) To: sipx-users@list.sipfoundry.org Subject: RE: [sipx-users] Disable ring on presence

Re: [sipx-users] FCC loses key ruling on Internet `neutrality' - wtop.com

Not likely to affect commercial subscribers at all, just residential offerings where this is commonly practiced. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: T

Re: [sipx-users] Retain polycom ring volume and speaker phone volume at reboot

I don't think ring volume will stick... none will stick through a reboot... only from call to call... From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew Kitchin (public/usenet) Sent: Tuesday, April 06, 2010 1:03 PM To: Tony Grazia

[sipx-users] FCC loses key ruling on Internet `neutrality' - wtop.com

This COULD mean that your ISP or broadband provider, who would like to sell you their VOIP services might be able to throttle or block SIP, and other VOIP services. (its usually best to get your internet connection from the same provider that sells you SIP trunks) http://www.wtop.com/?sid=1927

Re: [sipx-users] Disable ring on presence speed dial for parked call

Did you check the sipXecs settings in the Polycom around ringing? You might be able to make it ring on primary extension only. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew Kitchin (public/usenet) Sent:

Re: [sipx-users] Audiocodes M1000/SipX stand alone survivability

did you set the registrar or the outbound proxy? Only change the outbound proxy, make sure to leave the registrar alone. Also make sure the outbound proxy port is set to 5080 on the phone. Could you also post a screenshot of your SAS settings from the mediant web interface? Josh Patten Assistan

Re: [sipx-users] Audiocodes M1000/SipX stand alone survivability

Shameless bump :) Any ideas? Looks to me like I'm missing one piece of the jigsaw to get SAS working TIA Huw - Original Message - From: "Huw Wyn Jones" To: "Josh Patten" Cc: sipx-users@list.sipfoundry.org Sent: Tuesday, 6 April, 2010 16:15:28 GMT +00:00 GMT Britain, Ireland, Portugal S

[sipx-users] Disable ring on presence speed dial for parked call

I'm running Sipx 4.0.4, sixbridge, Verizon VOIP, No firewall (not needed, private connection), Polycom 450s and 550s - bootrom 4.2.1, firmware 3.1.3C split. This facility has one person answer almost every call and put it on a parked extension. I set the main admins 550 to monitor the three busi

Re: [sipx-users] Retain polycom ring volume and speaker phone volume at reboot

That only shows handset, headset, and handsfree. Handsfree is checked, and that is sticking between calls. Handsfree is not sticking through a reboot and I don't see any option for ring volume sticking. Am I missing something? On 4/6/2010 11:32 AM, Tony Graziano wrote: 1. Go to Devices -> Ph

Re: [sipx-users] polycom 650 HD voice

HD only works when calling another HD handset. When you call another HD handset and there is nothing blocking the codec from being passed or negotiated, it will work. For Example: Calling an internal subscriber locally will pass HD. Calling via the PSTN will likely not work as most ITSP's won't a

Re: [sipx-users] polycom 650 HD voice

Both handsets need to support HD audio - otherwise they will negotiate to a non-HD codec Nathaniel Watkins IT Director Garrett County Government 316 East Alder Street, Suite 2 Oakland, MD 21550 Telephone: 301-334-5001 Fax: 301-334-5021 E-mail: nwatk...@garrettcounty.org -Original Message--

[sipx-users] polycom 650 HD voice

I have a number of 330's and a while back, I bought a 650 so that I could test HD voice. I've not bought another 650 so guessing I'm not getting the full HD experience but figured I should get some darn good quality even 650 to 330. Both of the phone sound incredible on hands free but pretty muc

Re: [sipx-users] Retain polycom ring volume and speaker phone volume at reboot

1. Go to Devices -> Phone Groups (or Phones) 2. Pick the group (or phone) 3. Find your Phone Model 4. Select, if you are using Polycom, firmware Version to 2.0 5. Click on Voice/Codecs on the left 6. Scroll down to section labeled Volume Persistence 7. Enable the options you'd like to

Re: [sipx-users] Newbie: AudioCodes configuration help

When I was working on the MP-118, all I had to do is request access from Audicodes to their site. Took a little time, but I had access to their site for about 3 days for documentation and firmware. Andrew _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list

Re: [sipx-users] Newbie: AudioCodes configuration help

I totally agree, and didn't mean to imply a request for direct posts or other actitivity against license agreements. I have meanwhile found what we need. What beats me (us), is that we have been trying to work directly with Audiocodes, and they refuse to get us the firmware without a support c

[sipx-users] Retain polycom ring volume and speaker phone volume at reboot

I'm running Sipx 4.0.4, sixbridge, Verizon VOIP, No firewall (not needed, private connection), Polycom 450s and 550s - bootrom 4.2.1, firmware 3.1.3C split. This is a medical clinic and I have had to turn the ringer volume all the way up and the speaker phone volume all the way up (for paging p

Re: [sipx-users] Newbie: AudioCodes configuration help

If it's a M1000 you are looking for, perhaps googling: http://www.google.com/search?sourceid=chrome&ie=UTF-8&q=M1000_SIP_F5.40A Will yield something. On Tue, Apr 6, 2010 at 11:43 AM, Tony Graziano wrote: > I think gettin

Re: [sipx-users] Newbie: AudioCodes configuration help

I think getting AC firmware from somewhere other than an aUthorized source for your unit is a big no-no with AC licensing agreements. It makes sense for people to suggest how you might find it, but directly providing it might put them in some sort of liability with AC, which is not in anyone's inte

Re: [sipx-users] Newbie: AudioCodes configuration help

I asked for mediant firmware many times and I believe that people are very hesitant to help with firmware. The most common reply was: "Search the Internet and you will find"... And I did in fact :). On Tue, 06 Apr 2010 08:29:19 -0700, Johan Reinalda wrote: > That is the conclusion we have come

Re: [sipx-users] Cisco 7970G Firmware Upgrade

Hi Dylan, I appreciate the tip, thanks! I'll do that tom and will post any progress. Rhon On Tue, Apr 6, 2010 at 11:01 PM, Dylan Ebner wrote: > We use cisco phones, (not 7970s) and figuring out if they are actually > getting the tftp files is very very important. Here is how we confirm that

Re: [sipx-users] Dial Tone delayed after when phone is hanged

FYI, We are using Cisco as Gateway + FXO, SipXecs 4.1.7. Calls were made from/to outside number Thanks. On Tue, Apr 6, 2010 at 11:29 PM, Rhon wrote: > Hi, > > We noticed after calling or receiving a call there was at least more that > 10secs of delay before getting a new dial tone? > We alre

Re: [sipx-users] Newbie: AudioCodes configuration help

That is the conclusion we have come to as well. The units have 5.8 right now. Can any help me in getting Audiocodes firmware v5.4 for the M1K and MP112 units ? :-) Johan On 4/6/2010 6:08 AM, Huw Wyn Jones wrote: > I *think* you need to downgrade the Audiocodes Mediant 1000 firmware to 5.4 >

[sipx-users] Dial Tone delayed after when phone is hanged

Hi, We noticed after calling or receiving a call there was at least more that 10secs of delay before getting a new dial tone? We already put down the handset, however the phone is still busy whenever a call is established again on the same phone no. Is there a way to adjust this? Thanks in advan

Re: [sipx-users] Audiocodes M1000/SipX stand alone survivability

M. Partial success - partial failure! I set the outbound proxy for my line to the appropriate Audiocodes and after restarting the phone I can see it registered on the Mediant 1000 (;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"). Trouble

Re: [sipx-users] Audiocodes M1000/SipX stand alone survivability

Phone group I may have been doing something wrong though. Perhaps I'll play with it later to see if setting it at the group level will work. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 4/6/2010 9:59 AM, Tony Graziano wrote: I would assume this should

Re: [sipx-users] Cisco 7970G Firmware Upgrade

We use cisco phones, (not 7970s) and figuring out if they are actually getting the tftp files is very very important. Here is how we confirm that they are getting the files: 1. Edit your /etc/xinetd.d/tftp config file. In server_args add a "-v". This enables the logging on the tftp server 2. re

Re: [sipx-users] Audiocodes M1000/SipX stand alone survivability

I would assume this should be a phone group and not a user group. Which did you try? On Tue, Apr 6, 2010 at 10:55 AM, Josh Patten wrote: > I think it doesn't work. I specifically remember having to do it per user. > > Josh Patten > Assistant Network Administrator > Brazos County IT Dept. > (979)

Re: [sipx-users] Audiocodes M1000/SipX stand alone survivability

I think it doesn't work. I specifically remember having to do it per user. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 4/6/2010 9:55 AM, Tony Graziano wrote: Couldn't you add a group and make the changes by phone group and push them all out the same?

Re: [sipx-users] Audiocodes M1000/SipX stand alone survivability

Couldn't you add a group and make the changes by phone group and push them all out the same? On Tue, Apr 6, 2010 at 10:53 AM, Josh Patten wrote: > Neither. > > You have to configure the outbound proxy and port in the settings for > the line. Unfortunately with 4.0.x this must be done for every li

Re: [sipx-users] Audiocodes M1000/SipX stand alone survivability

Neither. You have to configure the outbound proxy and port in the settings for the line. Unfortunately with 4.0.x this must be done for every line on every phone. Devices >> Phones >> >> Lines >> >> Registration Once you get there change the Outbound Proxy and Outbound Proxy Port settings

Re: [sipx-users] Audiocodes M1000/SipX stand alone survivability

We're using Polycom SoundPoint phones (330's and 650's) so the following line caught my attention. >> Be aware that in sipXecs 4.0 the Polycom SoundPoint IP profiles by default >> include per-Line Outbound Proxy values, found under "Lines" - X - >> "Registration" (and "Show Advanced Settings".)

Re: [sipx-users] Audiocodes M1000/SipX stand alone survivability

There's really not much to it. You turn SAS on and set the phone's outbound proxy to the Audiocodes gateway port 5080 and that's it. Works every time. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 4/6/2010 8:43 AM, Huw Wyn Jones wrote: > Hi folks, > > Has

[sipx-users] Audiocodes M1000/SipX stand alone survivability

Hi folks, Has anyone seen or got a decent guide on setting up SAS for a SipX / Audiocodes Mediant 1000 combination? I've read http://sipx-wiki.calivia.com/index.php/HowTo_configure_AudioCodes_Stand-Alone_Survivability_Feature but it's a bit short on specifics :) Cheers Huw -- Huw Wyn Jones

Re: [sipx-users] Newbie: AudioCodes configuration help

I *think* you need to downgrade the Audiocodes Mediant 1000 firmware to 5.4 to get it to play nicely with SipX. We have a similar set-up in our college - SipX server on CentOS with three Mediant 1000 units (with FXS modules) distributed on three sites. Works brilliantly so far :) HTH Huw --

Re: [sipx-users] Cisco 7970G Firmware Upgrade

Well, the phones get told the TFTP server IP vial DHCP. It is the phone's responsibility to go look for its own firmware and macaddress.cfg files. What files you put there depends on the phone. I can't answer that for you because I don't use Cisco phones. Maybe somebody else here will pipe u