Re: [asterisk-users] Best USB Handset and Softphone Combination
Hello Steve, I personally use an 'ultra-portable' headset from Logitech, I would recommend this for any end-users that have laptops: http://www.logitech.com/index.cfm/webcam_communications/internet_headsets_phones/devices/223cl=us,en There is also this model for desktops: http://www.logitech.com/index.cfm/webcam_communications/internet_headsets_phones/devices/3622cl=us,en I've been using Counterpath Eyebeam with the USB headset for a year, and I never had a problem. Eyebeam automatically switches it's profile to use the USB mic and headset when it detects thats it's plugged in. We haven't deployed Eyebeam to our clients because of the risk involved with running a phone on an unmanaged desktop. Most of our customers have little desktop management, if any. But I would recommend either Eyebeam or Bria, they have a provisioning system for easier management and the phones can be rebranded. Good luck ;-) Omar On 10/19/07, Steve Totaro [EMAIL PROTECTED] wrote: I have a client that want to try the softphone with USB handsets route to see if hardphones will even be needed. I always push for hardphones (Polycom) so I am not sure about softphones or USB handsets. This is going to be for a 300+ seat call center onsite and many offsite, I plan on using OpenVPN for the offsite machines. Any advice on softphones, handsets, or practical experience with this sort of deployment? It would be very nice if there was a central way of provisioning the phones. All machines are fairly new (newer than two years), they have very strict policies on downloads and streaming. Thanks in advance. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 lights not working on subscription
I used to deploy these phones, it was these types of issues that forced me to drop it. It took way too long to troubleshoot the problems and there was a general lack of documentation. This was 2 years ago, things might have changed. If I remember correctly, it was this issue you are having that was the final straw. Good luck, Omar On 10/22/07, Carlos Maimone [EMAIL PROTECTED] wrote: Dear friends, I am working around with a Snom 360 and Asterisk 1.4 + FreePBX In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. any idea of what's going on? IN SNOM 360 I am currently using firmware 6.5.12 I am pretty sick dealing with this issue. thanks and regards, Charlie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 lights not working on subscription
Hey Mike, We started deploying exclusively Polycom and Linksys. The Polycom's support presence, they call it 'Buddy List'. I am not sure about the Linksys phones, I don't think they do although I did see support for SLA (Shared Line Appearance). Omar On 10/23/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: I also have problems with these phones. I have deployed many of them and have had nothing but problems. Omar, what phones did you switch to? I needed some of the features of the snom phones, like the multiple buttons with prescence lights. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: Monday, October 22, 2007 9:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom 360 lights not working on subscription I used to deploy these phones, it was these types of issues that forced me to drop it. It took way too long to troubleshoot the problems and there was a general lack of documentation. This was 2 years ago, things might have changed. If I remember correctly, it was this issue you are having that was the final straw. Good luck, Omar On 10/22/07, Carlos Maimone [EMAIL PROTECTED] wrote: Dear friends, I am working around with a Snom 360 and Asterisk 1.4 + FreePBX In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. any idea of what's going on? IN SNOM 360 I am currently using firmware 6.5.12 I am pretty sick dealing with this issue. thanks and regards, Charlie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Refrigerator Alarms
Balu, Do you want events passed to Asterisk from the refrigerator? Or does a reminder type phone call need to be placed on an interval? Please be more specific, since this sounds like a special purpose refrigerator, does it have any way of passing events to an external device? Omar A. Sabek On 10/17/07, Balu Raman [EMAIL PROTECTED] wrote: Hi, I want asterisk to call a person on the phone for monitoring the refrigerator storing vaccines. I am clueless where to look. Can someone clue me in ? Thanks, balu raman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID to hunt group?
Hey Rich, This functionality is included in freepbx. It's called ringall-prim and can be found as an option for ringing phones in a Ring Group. You may want to it out for a solution to your issue or install it to manage your system ;-) Omar On 10/16/07, Rich [EMAIL PROTECTED] wrote: Asterisk 1.4.2 I have spent much of today trying to make a DID (from SIP GW) ring to 4 extensions in a hunt (roll-over) group. Results from searching docs and forums seem to indicate it is doable and so trivial no one includes an actual example. I can make all 4 exts ring at once with the like of exten = _1655,1,Ringing() exten = _1655,2,Dial(SIP/1655SIP/1656SIP/1657SIP/1658) exten = _1655,3,Hangup() exten = _1655,103,Congestion() but I want to ring the 1st ext, if it is busy then ring the 2nd, etc... I am trying to emulate a 10-line 20 phone key system. The end user does not like all the phones ringing at once. I have tried lots of combinations in extensions.conf including the following that I would think should work... the 2nd line never rings, just dead air, or I hear a click and dead air... where am I going wrong !! This did not work, not sure why exten = _1655,1,Ringing() exten = _1655,2,Dial(SIP/1655) exten = _1655,103,Dial(SIP/1656) exten = _1655,104,Hangup() exten = _1655,204,Dial(SIP/1657) exten = _1655,205,Hangup() exten = _1655,305,Dial(SIP/1658) exten = _1655,306,Hangup() exten = _1655,406,Congestion() !! this didn't work , I hear no ringing, 2nd call gave some wierd error about codec and slin exten = _1655,1,Answer() exten = _1655,2,Dial(SIP/1655,30) exten = _1655,3,Hangup() exten = _1655,103,Dial(SIP/1656,20) exten = _1655,104,Hangup() exten = _1655,204,Dial(SIP/1657,10) exten = _1655,205,Hangup() exten = _1655,305,Dial(SIP/1658) exten = _1655,306,Hangup() exten = _1655,406,Congestion() Any clues will be greatly appreciated! Thanks, Rich ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7970 SIP - few questions
Restarting the 7970 is like unlocking it twice, *-*-# to unlock, *-*-# to reboot. I don't believe hint functionality works on the SIP firmware for the 7970. Omar A. Sabek On 4/18/06, Tomislav Parčina [EMAIL PROTECTED] wrote: - How to restart the phone? (On 7960 it is *+6+Settings) - How to setup working dtmf? - How to setup hinting? For line is line button=4 featureID9/featureID ... For speeddial is line button=5 featureID2/featureID featureLabel341/featureLabel speedDialNumber341/speedDialNumber /line How to define hinting? - How to login true ssh? I have setup username and password, and when I try to log in it sends me challenge!?! login as: root [EMAIL PROTECTED]'s password: login: root challenge: YDXWGXMTpassword: Invalid Username/Password Entry. login: That is all, for now :)) -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?
It seems the proxy address is added to all incoming calls to the Cisco phone. On 3/16/06, Tim Connolly [EMAIL PROTECTED] wrote: I'm not sure this is the issue. Every call seem to get the proxy address added whether it's the main proxy or the backup. What has to match to make the phone NOT append the proxy address? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Connolly Sent: Wednesday, March 15, 2006 1:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? That's probably what is happening on my end. Any suggestions on how to fix this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Tuesday, March 14, 2006 7:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? We only had the problem when the call was redirected from one server to another. So if a phone was called from another phone on the server, the called worked perfectly, but if it was redirected from another server, we got the proxy added to the end. Doesn't help when you're trying to make the existence of multiple servers transparent. Aaron Chris Stenton wrote: Maybe I have something strange in my dial plan but I have no problem just hitting dial from missed calls under 8.2. Chris - Original Message - From: Aaron Daniel [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 8:44 PM Subject: Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? We rolled back to 7.4 cause of that too. 7.5 has a strange bug where if the server loses connection, the phone's just don't try re-registering. Aaron Tim Connolly wrote: Just curious, why not 7.5 ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Monday, March 13, 2006 2:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? I'm using P0S3-08-2-00.. I noticed the callerID started showing up with the number, then @proxy-addr... So the callerID on the phone looks like: [EMAIL PROTECTED] which of course is logged in the missed calls exactly like that, and completely foobars the dialing string if you try to dial a missed call by simply hitting the dial button. Can anyone else verify this problem? Yeah, that bothered me so I rolled back to SIP 7.4. Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time
Chris, you may have a different and simpler setup. Internal calls work fine here, since the proxy server on the CallerID is the same proxy server used for all internal users. I was referring to calls that originate outside of the enterprise. I should have been more clear. Omar On 3/14/06, Chris Stenton [EMAIL PROTECTED] wrote: Yes it does display caller id as callingnumber@ip of calling party but that does not interfere with me hitting dial from missed calls. Seems the Cisco phone sends the sip INVITE as callingnumber@ip of calling party rather than callingnumber@ip address of defaultproxyserver but asterisk ignores the info after the @? Chris - Original Message - From: Omar A. Sabek [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 11:45 PM Subject: Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time The 8.2 firmware displays Caller ID as callingnumber@proxyaddr... this becomes problematic for users that want to dial from their 'Missed Calls' log. Omar On 3/13/06, Nathan Bowyer [EMAIL PROTECTED] wrote: On 3/13/06, Chris Stenton [EMAIL PROTECTED] wrote: I have had no issues with 8.2 so far! Chris Except the Caller ID issue reported in another thread? This issue has been fixed in SIP firmware 7.5 Omar A. Sabek Yes, and I read that SIP 7.5 firmware have some other issues. They recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware. Tomislav ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time
The 8.2 firmware displays Caller ID as callingnumber@proxyaddr... this becomes problematic for users that want to dial from their 'Missed Calls' log. Omar On 3/13/06, Nathan Bowyer [EMAIL PROTECTED] wrote: On 3/13/06, Chris Stenton [EMAIL PROTECTED] wrote: I have had no issues with 8.2 so far! Chris Except the Caller ID issue reported in another thread? This issue has been fixed in SIP firmware 7.5 Omar A. Sabek Yes, and I read that SIP 7.5 firmware have some other issues. They recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware. Tomislav ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time
To be totally honest, I have 7.5 running on many phones and I have yet to receive a report on a firmware related issue. Omar On 3/13/06, Tomislav Parcina [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: 9. ozujak 2006 18:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time This issue has been fixed in SIP firmware 7.5 Omar A. Sabek Yes, and I read that SIP 7.5 firmware have some other issues. They recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware. Tomislav ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time
This issue has been fixed in SIP firmware 7.5 Omar A. Sabek On 3/9/06, Nathan Bowyer [EMAIL PROTECTED] wrote: On 3/9/06, Greg Oliver [EMAIL PROTECTED] wrote: On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote: Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Add the following to SIPDefault.cnf or SIPMAC.cnf: sntp_server: time.nrc.ca sntp_mode: unicast time_zone: EST On my 7960 with 7.4 firmware, the time automagically disappears for some unknown reason. The phone still functions, but the time goes away until I reboot it. Not a big deal to me, so I have not investigated it further. -Greg I use anycast. Seems like I read something about directbroadcast not working in recent SIP versions. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
I just started looking at the differences between 8.2 and 7.5. In addition to the new Security Configuration, there is also compatibility added for provisioning the phone to CCM. The firmware appears to be working fine. Also, the copywright date range has been changed to 2000-2006. Omar A. Sabek On 3/9/06, C F [EMAIL PROTECTED] wrote: Does that mean that since CCM supports SIP, Cisco will just make sure that their SIP images work with CCM? On 3/9/06, Aaron Daniel [EMAIL PROTECTED] wrote: The image is located in the non-sip section, go figure. They're harping that this is for their new sip ccm... Aaron C F wrote: Why can't I find an 8.2 on Ciscos site? the latest I can find is 7.5. Are you guys talking about SIP? On 3/9/06, Mailing List [EMAIL PROTECTED] wrote: - Original Message - From: Nabeel Jafferali [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 10:42 AM Subject: RE: [Asterisk-Users] 7940/60 SIP 8.2 So has anybody tried installing the new SIP version? It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM v5.0. Just downloaded it after your email and got it working on the first try. Give me a few minutes to write up the procedure. OK, I had a 7960 running SIP 7.4 and managed to upgrade to SIP 8.2. I haven't tried any of the new features, but can make and receive calls fine. Sweet, guess I'll give it a go. _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Program Buttons on Cisco 79xx Phones
The only two programmable buttons are the 'Messages' and 'Services' and 'Directory buttons'. They are all configured in sip_default: messages_uri: number to dial services_url: xml file to load directory_url: xml file to load Cheers, Omar On 3/4/06, Kevin Steil [EMAIL PROTECTED] wrote: Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware…thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys WIP300 WiFi Phone
I purchased my WIP300 from voipsupply.com, as far as I know they are the only ones carrying it. Cheers, Omar On 2/27/06, Me [EMAIL PROTECTED] wrote: Where are you folks getting the best deal on this phone right now? After my experience with the last Wifi phone, I am a little gunshy at the moment. I am not sure if I should wait to see how this product plays out or not. On 2/27/06, Philip Edelbrock [EMAIL PROTECTED] wrote: Omar A. Sabek wrote: Like BJ, I'm sorry you had bad luck Phil. I have been playing with this phone all weekend, and I have had minor problems. The voice quality is as good as my cisco and polycom sip phones. I asked a friend to guess what kind of phone I was talking on and he said it sounded like a regular home or office phone. I have been very happy with the voice quality. My first day was a huge disappointment. Three crashes, calls wouldn't work over my work's wifi (eventhough it registered ok), short battery time, lost settings after a crash, etc. However, after I went in and cleared my settings back to default, the troubles went away! I'm been using it for over three days without a glitch. So, I would recommend to anybody else who is getting one of these phones, to immediately set all settings back to 'default' (under the Tools menu) before spending too much time configuring it. I reported on the voip-info page dismal talk times but it must have been an anomoly. Today I spoke for over an hour on the phone and still had plenty of juice left. My battery life seems to have improved as well. I don't know if that's was a glitch fixed by setting things back to the defaults, or if cycling the battery is helping. I also have less of a tendency to play with the menus, and the backlight could be a power drainer (it is quite bright). All-in-all this phone is a winner. It works with Asterisk flawlessly. As long as my troubles don't come back, I would agree. I think my phone was shipped to me in a funny state causing it not to work right. It's a winner now. There are some little things I would wish for, but I'm quite happy with it. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys WIP300 WiFi Phone
Like BJ, I'm sorry you had bad luck Phil. I have been playing with this phone all weekend, and I have had minor problems. The voice quality is as good as my cisco and polycom sip phones. I asked a friend to guess what kind of phone I was talking on and he said it sounded like a regular home or office phone. I have been very happy with the voice quality. The only problem I had was my phone would not connect to my wireless network this morning. I was a bit puzzled because it worked last night and all my settings were correct. In a move of frustration I deleted all the 'preset' profiles that came with the phone and recreated the current wi-fi profile. It picked up my network and connected to * no problem. The GUI is much like a cell phone's. It has a center joystick that takes a bit to get used to handling, it's pretty sensitive. One great thing about the configuration of the phone is the seperation between 'network profile' and 'sip account'. In other words, after configuring the network settings, you select the SIP account you want to use when on that network. This is especially useful for users that have unique wi-fi settings per location but a common SIP proxy like *. I reported on the voip-info page dismal talk times but it must have been an anomoly. Today I spoke for over an hour on the phone and still had plenty of juice left. All-in-all this phone is a winner. It works with Asterisk flawlessly. Don't look for much support from Linksys though as I called them when I received the phone and they told me they dont carry any item with that model number. That should be a temporary thing though, I have a feeling these wi-fi linksys phones are going to be pretty popular. Thanks, Omar A. Sabek On 2/26/06, Me [EMAIL PROTECTED] wrote: How is the voice quality? I've just plugged mine back into the charger after having used it nearly all day. I didn't have any of the problems you've described. Sorry you're having such bad luck with it. I'm not certain what the phones are rated to do, but I probably got better than 3 hours talk time on it today which is definitely the best I've gotten with any of the WiFi phones up to this point. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] simulating a few thousand SIP clients?
Hello Roy, Have you heard of Sipp? http://sipp.sourceforge.net/. I am pretty sure it can do what you desire. Also a commercial tool from Empirix, Hammer NXT. (http://www.empirix.com/default.asp?action=articleID=64) Cheers, Omar On 1/29/06, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: sure, but I need to simulate the SIP REGISTER and OPTION traffic sent by ATAs as well. Best regards Roy Sigurd Karlsbakk [EMAIL PROTECTED] --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way. On Jan 29, 2006, at 7:26 PM, Wai Wu wrote: Set up another * and use the manager api to make lots of calls to the other one. You can even make hundresd calls at a time. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Roy Sigurd Karlsbakk Sent: Sunday, January 29, 2006 1:19 PM To: Asterisk Non-Commercial Discu Subject: [Asterisk-Users] simulating a few thousand SIP clients? hi i'm setting up a rig to handle quite a few SIP clients, so i need a way to simulate, say, 20k SIP ATAs. Does anyone know how? This should of course be as close as possible to 'reality', meaning n% calls per client and the usual REGISTER/OPTION traffic. thanks Best regards Roy Sigurd Karlsbakk [EMAIL PROTECTED] --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Odd Polycom Dialing Problem
Modify or delete the digitmap. The digitmap allows the phone to intelligently detect numbers dialed (and presumably save the user from hitting send after a specified number of seconds). The digitmap is available on the web interface under SIP and Local Settings. The default is: [2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT We pass all digits to Asterisk, which handles it intelligently. Omar A. Sabek On 1/18/06, calvis [EMAIL PROTECTED] wrote: I am starting to play around with my first Polycom phone. I have it registered to my asterisk box, but the problem is dialing. When I try to dial with the handset I get my provider telling me that they can not complete my call afterwards Allison pops in and tells me all circuits are busy and can not complete my call. However, if I punch in the numbers and then hit the dial button the call completes with no problem. Does anyone have a clue what is going on? This is a SoundPoint IP 601 phone and I am using Asterisk at Home 2.2. Charles Alvis Internet Technology Group, Inc. Redmond, WA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voipsupply - my experience
I also have nothing but wonderful things to say about voipsupply.com, they are first on my list for equipment needs. They have great methods of communication and they free up my time to better serve my clients. Omar A. Sabek On 12/15/05, Dennis Gilmore [EMAIL PROTECTED] wrote: On Thursday 15 December 2005 11:55, Terry H. Gilsenan wrote: Hi, I would just like to let everyone know about the support I have rec'd from voipsupply. I travelled from .au to the US to do some urgent server upgrades for $EMPLOYER. During this trip I has impressed upon me (with absolutely zero notice) the requirement for a IP Phone link to the .au office. With only a couple of day left in the US I was stumped. I called Voipsupply and left a message (it was 9:30PM CST - Houston) at 10:30PM Mark emailed me and sent me his Cell phone number. The upshot is: I placed an order for some Digium kit at Midnight, on the 14th and I have just rec'd the goods in Houston now. Mark: Fantastic service! I appreciate the help. Regards, Having moved from AU to US last year I know how important a good reliable and cheap link back home is. for this i use digium equipment and asterisk with an AU iax provider. i have a local number in Brisbane. I have also purchased some VoIP equipment from Voipsupply for my employer and have received fantastic service and support from them for that also. Dennis ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Slightly OT - Anyone know of an external ringer compatible with Cisco phones
Hey Cory, I havent come across a voip ring amplifier or visual indicator. Here are some amplifiers and visual inidicators for an environment using ATAs: http://www.soundbytes.com/Merchant2/merchant.mvc?Screen=CTGYStore_Code=SBCategory_Code=PhoneRingAmplifier Omar A. SabekOn 11/22/05, Cory Andrews [EMAIL PROTECTED] wrote: Have an application where Cisco phones are being used in a noisyenvironmentlooking for some type of external ringer or amplifier sousers can hear the phones ringing over the background noise.Anyonefamiliar with such a device? Thanks,--Cory J AndrewsPartner / Purchasing+++VOIPSupply.com - Everything you need for VOIP454 Sonwil DriveBuffalo, NY 14225+++tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22f - 716.630.1548e - [EMAIL PROTECTED]AIM - b2Cory___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 Multiple Line Appearance
If I understand this thread correctly, they are discussing monitoring the status of other agents. This cannot be done with the SIP firmware on the CP-79x0 phones. We are currently working on testing the SCCP firmware with the revisions made to the chan-sccp module, but I am stuck on getting the phone to accept the CTLSEPmac.tlv file. Does anyone have a working version of the xml contents of this file? Thanks, Omar A. SabekOn 11/16/05, Chris Stenton [EMAIL PROTECTED] wrote: - Original Message -From: Sergio Chersovani [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Tuesday, November 15, 2005 2:25 PMSubject: Re: [Asterisk-Users] Cisco 7960 Multiple Line Appearance Matt Hoskins ha scritto: I'm currently using asterisk 1.0.7 with cisco 7960 SIP 7.5 phones.I'd like to use this phone for a receptionist so that she can take calls for 4 other people.Is this possible? The SIP firmware does not support it. You have to use SCCP to do that Is there any way to do this with SIP and the 7960?I've seen the 7914 but then I'd have to use SCCP and I'm not sure if it is stable enough for production use. Well give it a chance :-) http://chan-sccp.berlios.deI've got all sixSIP lines registered as extention 201 and I've had all six in use at one time.Chris___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Opinions
Curren, Can you tell us a little more about the environment you are deploying these phones in, how many phones and what kind of setup? Omar A. SabekOn 11/12/05, Curren C. Calhoun [EMAIL PROTECTED] wrote: Thanks that's the type of info I'm looking for...I've heard some early grumblings but wanted to see if anything else has comeup... From: Remco Barende [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sat, 12 Nov 2005 09:50:01 +0100 (CET) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Snom 360 Opinions I'm not too pleased with the phones, I have about 40 of them, some of the displays tend to die and the dial pad feels to 'mushy' IMHO, just like the keys on a good old ZX80 computer Also I'm having some issues with sound quality on some phones, but I still need to switch some phones to see if that is really an issue of the phone. Also if you want to use * call files, with the 360 you will run into a big where the call is being redialled as if it failed while in fact the call is ongoing. Annoying and haven't found out if that is an * bug or Snom bug. The Snom 190's do not have this problem. Just my $0.02 (which is really not a lot these days!) :) On Sat, 12 Nov 2005, Curren C. Calhoun wrote: I¹m looking to add in some Snom 360 phones, could anyone give thoughts or opinions about the speakerphone, general quality... Also the phone would need to be powered over Ethernet... I like some of the listed features and the expandability of the phone but am open to any other suggestions as well... Thanks Curren___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please recommend a phone
The Cisco CP-7940/60 flashes it's MWI during incoming calls. If you are using an ATA, there are several devices that can display flashing/blinking lights during incoming calls by simply putting it between the ATA and phone. OmarOn 10/19/05, Christian Stredicke [EMAIL PROTECTED] wrote: Take a look at snom.com...CS -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Jesse Keating Sent: Wednesday, October 19, 2005 5:31 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Please recommend a phone On Wed, 2005-10-19 at 16:39 -0400, Jesus Mogollon wrote: I'm in need of a phone that would blink a led to let the callee know that there is an incoming call. The GXP-2000 does this but I want an alternative to Grandstream. Any help is appreciated. Polycom IP301s and 501s have a red LED that blinks when calls are coming in. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] merchant account
The CC merchant machines I've encountered require entry of the account number, exp date, total charge, etc. before dialing and transmitting the data. Even though we are able to pass DTMF successfully through the gateway, we still make the recommendation that any application that requires a negotiation phase (ie, fax machines, CC merchant machines, dial-up modem) remain on a traditional POTS line. And just like you mention, alternative methods are available including web access. Hopefully, the technological disconnect between voip and dial-up data transmission will come to pass sooner than later. Omar A. SabekOn 10/20/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: I am interested in hearing some user experiences of anyone using amerchant account.The constraints are that everything entered must beDTMF-able.Card number, CCV, exp, numeric portion of the streetaddress, zipcode are all easy. name however is not so easy. How have others solved this problem?Or have they only set up systemswhere web access is required?--Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402US +1 360 207 0479 or +1 516 687 5200FreeWorldDialup: 635378-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.1 (GNU/Linux)iD8DBQBDWE5s+1olxlzQw5cRAuJvAJ43RS5gsbSD5YYyzVdij/HlbpSp+wCdHRIK 8OQU0Acd7yf+rxuZF7frzeQ==9vGx-END PGP SIGNATURE-___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Cisco phone
Tom, Do you have VLANs or QoS configured on your switch? I had the same problem with my CP-7960 connected to a Catalyst 3524XL switch. I solved it by tweaking my QoS settings on the switch, there was an option to enable dot1p priority tagging on the voice VLAN. I am not sure if the Dell switch you are using has a similar feature, if it does check there. Also, how have you troubleshot this problem so far, does it happen on every switch you've tried? Is it happening on every CP-7960G you've tried? You're right about it being a simple problem, that aux switch port has always worked fine IMLE. Cheers, Omar A. SabekOn 10/20/05, Tom Tune [EMAIL PROTECTED] wrote: My Cisco 7960g SIP phones share an annoying feature: Anything I plug into their 2nd Ethernet or PC port loses connection every thirty seconds or so. I did a ping -t and can see regular drops. I do not have access to Cisco's tech support archives. I am pretty sure that this is a simple configuration problem. Can somebody point me in the right direction? Thanks in advance. ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] merchant account
Hey Bret, Most companies will get around this quandry by assigning their customers account numbers. This will then be x-referenced to a DB that contains that information (Name, Billing Addy, maybe last 4 SS). If this is not an option, there is always the Please say and spell your first and last name after the tone. But, not too clean from an automation point of view. Would switching to a CC processor that doesn't require a name be an option? I remember making international calls years ago by entering only my CC # and expiration. I am not sure if this is an option any longer. Omar A. SabekOn 10/20/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Thu, 2005-10-20 at 22:29 -0400, Omar A. Sabek wrote: The CC merchant machines I've encountered require entry of the account number, exp date, total charge, etc. before dialing and transmitting the data. Even though we are able to pass DTMF successfully through the gateway, we still make the recommendation that any application that requires a negotiation phase (ie, fax machines, CC merchant machines, dial-up modem) remain on a traditional POTS line. And just like you mention, alternative methods are available including web access. Hopefully, the technological disconnect between voip and dial-up data transmission will come to pass sooner than later. I should have added one more requirement.Internet authentication likeso many do.Although all that I have worked with require the name to beentered as text, something that isnt trivial to do (and asking people to tap out their name via dtmf would be too bothersome).I would like a merchant solution that would work 100% off someonecalling into the system and entering their data, authentication willthen occur via the internet. --Trixter http://www.0xdecafbad.com Bret McDanelUK +44 870 340 4605 Germany +49 801 777 555 3402US +1 360 207 0479 or +1 516 687 5200FreeWorldDialup: 635378 -BEGIN PGP SIGNATURE-Version: GnuPG v1.4.1 (GNU/Linux)iD8DBQBDWFOU+1olxlzQw5cRAljKAJsE111SudF9Q4+5+sWDo8nl4nWXkACdEUWLIa6KTRfeGDztsTr9Fdybarw==M8TZ-END PGP SIGNATURE- ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79XX and VLANS
Hey Matthew, Simply turn on dot1p priority tagging on the switchports your phones are connected to. This won't interefere with the auxiliary phone switchport and it will enhance (to a certain degree) your voice QoS. My CP-7960s boot up in less than 5 seconds flat. Omar A. SabekOn 8/11/05, Matthew Boehm [EMAIL PROTECTED] wrote: Hey gang,We have about 30 Cisco 79XX phones all running latest 7.5 SIP. We arealso using all Cisco Switches and Routers. Everything works great exceptthat when you reboot a phone it takes like 3-5 minutes for it to come up. The phones spend tons of time 'Configuring VLAN..' We don't run anyVLANs. Is there some way to skip this?In the 'Network Settings' I have both 'Operational VLAN Id' and 'AdminVLAN Id' set to blank values. Any ideas?Thanks,Matthew___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Cisco phone
Unfortunately, I think you're right. It's a good thing you only have to reboot the phones for config and firmware changes. Omar A. SabekOn 10/20/05, Tom Tune [EMAIL PROTECTED] wrote: It happens on everything I have tried. I do not have any vlans or qos configured. Even on a generic 5 port with only the phone and asterisk server connected. I am beginning to think that it is a Microsoft style mutual upgrade scenario.. can't use the phones without the switch.On 10/20/05, Omar A. Sabek [EMAIL PROTECTED] wrote: Tom, Do you have VLANs or QoS configured on your switch? I had the same problem with my CP-7960 connected to a Catalyst 3524XL switch. I solved it by tweaking my QoS settings on the switch, there was an option to enable dot1p priority tagging on the voice VLAN. I am not sure if the Dell switch you are using has a similar feature, if it does check there. Also, how have you troubleshot this problem so far, does it happen on every switch you've tried? Is it happening on every CP-7960G you've tried? You're right about it being a simple problem, that aux switch port has always worked fine IMLE. Cheers, Omar A. SabekOn 10/20/05, Tom Tune [EMAIL PROTECTED] wrote: My Cisco 7960g SIP phones share an annoying feature: Anything I plug into their 2nd Ethernet or PC port loses connection every thirty seconds or so. I did a ping -t and can see regular drops. I do not have access to Cisco's tech support archives. I am pretty sure that this is a simple configuration problem. Can somebody point me in the right direction? Thanks in advance. ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] user name
I think you might be referring to the asterisk-users listserv. Please visit the following page and obtain a password reminder: http://lists.digium.com/mailman/listinfo/asterisk-users Omar A. SabekOn 10/21/05, Jerry Richmond [EMAIL PROTECTED] wrote: I am not spaning I just want to stop all this e-mail with out giving up my corprate e-mail address. HelpAndres [EMAIL PROTECTED] wrote: I don't get it either. He is either clueless or is someone spamming this list with nonsense we don't need to know about. Jonathan k. Creasy wrote: I don't get it. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] *On Behalf Of *Jerry Richmond *Sent:* Thursday, October 20, 2005 9:46 AM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] user name I am geting e-mail but asterisk doesn't know my user name or password. My user name has always Been Jerry Richmond, my e-mail address [EMAIL PROTECTED] I need a password of some kind. thanks___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 360 Phones - Administrator/User Feedback
I'm looking for some feedback on the Snom 360 phone. After deploying six of them to an office, I'm not as enthusiastic about them as I was when I was testing one before the deployment. The firmware seems to be consistently buggy, some of its problems are intermittent which makes it frustrating to troubleshoot and the support from Snom is lackluster to say the least. I find myself favoring Cisco and Polycom phones, not only from a user POV but also the automation of deployment... Does anyone share similar sentiments? Omar Sabek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Dialler
Hello Benni, Check out gnudialer (www.gnudialer.org), I was looking into this for a client but had to abandon the project. I remember reading that the dialer allows recording but I am not sure what the method is. It may be local, check it out. Omar SabekOn 10/6/05, Benni A. Aswin [EMAIL PROTECTED] wrote: Hi,Any of you have any experience with SIP softphone dialler that capableof local recording? (recording to files in harddrive)So far I only know eyeBeam and Express talk. eyebeam fine but thereare known error with recording. Express talk recording looks ok, but sometime it doesn't have incoming voice with *.CheersBenni-___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
