Re: [asterisk-users] Polycom SP 601 Reboot Issue- Help!
We had a situation where the 601 base went missing and the electrical connection between the side cars and the 601 was broke. Might be worth a look to see if the phone got damaged. -Original Message- From: Jerry Jones [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tue, 24 Apr 2007 12:27:46 -0500 Subject: Re: [asterisk-users] Polycom SP 601 Reboot Issue- Help! The only reboot issue I have with 1 sidecar is the side car deciding to randonly rebbot, not the phone itself Perhaps upgrading to 2.1 will help? On Apr 24, 2007, at 10:51 AM, J French wrote: I have a Polycom 601 with 3 expansion modules running 2.0.3. We have Buddywatch set up on around 42 users on the expansion modules. We are experiencing reboots on the 601. Today it happened twice after users paged through the phones. The page groups have about 23 phones each. There is a third page group comprising all 46 phones. I'm thinking it may be an issue with changing buddywatch state on so many buddies so quickly. Also, the cpu usage is pegged at 100% for around 3 minutes after it reboots, FWIW. Anyone else experiencing rebbots on the 601? Advice is really needed! 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 [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 [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] wget from within asterisk?
I believe asterisk requires the source/devel package. I'm afraid I'm not going to be much help on that. We use Gentoo and it was a simple 'emerge curl' and I was done. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 16:18:07 -0700 Subject: RE: [asterisk-users] wget from within asterisk? On version 1.2.12.1 running on FC4 with curl.i386 installed the asterisk CURL function is not registered, perhaps in need something else (curl-devel.i386 ?) From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? Make sure the curl library/package is installed, then re-compile asterisk. We're using it on 1.2. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:40:40 -0700 Subject: RE: [asterisk-users] wget from within asterisk? I saw CURL, but it does not register appear in show functions or show applications, deprecated or add-on? From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] wget from within asterisk? -Original Message- From: Damon Estep [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:32:53 -0700 Subject: [asterisk-users] wget from within asterisk? What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? Look at the CURL function. ___ --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] wget from within asterisk?
-Original Message- From: Damon Estep [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:32:53 -0700 Subject: [asterisk-users] wget from within asterisk? What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? Look at the CURL function. ___ --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] wget from within asterisk?
Make sure the curl library/package is installed, then re-compile asterisk. We're using it on 1.2. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:40:40 -0700 Subject: RE: [asterisk-users] wget from within asterisk? I saw CURL, but it does not register appear in show functions or show applications, deprecated or add-on? From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] wget from within asterisk? -Original Message- From: Damon Estep [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:32:53 -0700 Subject: [asterisk-users] wget from within asterisk? What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? Look at the CURL function. ___ --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] wget from within asterisk?
They both seem to work, but the Curl spits out warnings about being deprecated. Ours are all configured using CURL. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:52:35 -0700 Subject: RE: [asterisk-users] wget from within asterisk? Thx! I saw a note about Curl vs. CURL, is there a difference? From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? Make sure the curl library/package is installed, then re-compile asterisk. We're using it on 1.2. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:40:40 -0700 Subject: RE: [asterisk-users] wget from within asterisk? I saw CURL, but it does not register appear in show functions or show applications, deprecated or add-on? From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] wget from within asterisk? -Original Message- From: Damon Estep [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:32:53 -0700 Subject: [asterisk-users] wget from within asterisk? What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? Look at the CURL function. ___ --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] wget from within asterisk?
The Curl/CURL is an asterisk dialplan distinction. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 15:06:49 -0700 Subject: RE: [asterisk-users] wget from within asterisk? Options I am aware of for installing curl are yum install in FC4 or download fromcurl.haxx.se, neither option distinguishes between curl and CURL, can someone offer me the slap in the head I need? Damon From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfDamon Estep Sent: Friday, November 17, 2006 2:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? Thanks a bunch, this seems to be a simple solution, I just did not have CURL installed before I built asterisk. From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 2:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? They both seem to work, but the Curl spits out warnings about being deprecated. Ours are all configured using CURL. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:52:35 -0700 Subject: RE: [asterisk-users] wget from within asterisk? Thx! I saw a note about Curl vs. CURL, is there a difference? From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? Make sure the curl library/package is installed, then re-compile asterisk. We're using it on 1.2. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:40:40 -0700 Subject: RE: [asterisk-users] wget from within asterisk? I saw CURL, but it does not register appear in show functions or show applications, deprecated or add-on? From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] wget from within asterisk? -Original Message- From: Damon Estep [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:32:53 -0700 Subject: [asterisk-users] wget from within asterisk? What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? Look at the CURL function. ___ --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] Polycom autoprovision behind a NAT
I can confirm that the linksys routers cause ftp problems. Is your FTP server set to use pasive mode? -rb -Original Message- From: "Curt Shaffer" [EMAIL PROTECTED] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.com Date: Mon, 6 Nov 2006 19:19:48 -0600 Subject: [asterisk-users] Polycom autoprovision behind a NAT I am having an issue with doing FTP auto provisioning of Polycom 501’s when they are behind a NAT. If I put the phone on the same subnet as the provision server it loads the configs and changes fine but as soon as I put in behind a NAT it comes up with cannot contact boot server. I have tried behind and replicated this behind a PIX 501, a Linksys SOHO router and a Motorolla SOHO router. Any ideas? Curt ___ --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] Polycom autoprovision behind a NAT
I if you like, I can take a config file(s) and put up over here as a test. Our ftp is working. It might be informative. -Original Message- From: "Curt Shaffer" [EMAIL PROTECTED] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.com Date: Mon, 6 Nov 2006 20:17:07 -0600 Subject: RE: [asterisk-users] Polycom autoprovision behind a NAT To be honest I don’t know for sure. I am using VSFTPD. I have never needed to set this with setups I have used it before and there is nothing in the config that says passive. So I’m guessing that its not. After you asked this I have googled passive FTP and it seems to be on the money as to what is going on so I will try passive and see if that helps. Thanks! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Monday, November 06, 2006 7:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom autoprovision behind a NAT I can confirm that the linksys routers cause ftp problems. Is your FTP server set to use pasive mode? -rb -Original Message- From: "Curt Shaffer" [EMAIL PROTECTED] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.com Date: Mon, 6 Nov 2006 19:19:48 -0600 Subject: [asterisk-users] Polycom autoprovision behind a NATI am having an issue with doing FTP auto provisioning of Polycom 501’s when they are behind a NAT. If I put the phone on the same subnet as the provision server it loads the configs and changes fine but as soon as I put in behind a NAT it comes up with cannot contact boot server. I have tried behind and replicated this behind a PIX 501, a Linksys SOHO router and a Motorolla SOHO router. Any ideas? Curt ___ --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] Polycom autoprovision behind a NAT
I'm not sure. We ended up putting in a d-link router to get around the ftp problem. In most of our sites we have netscreen 5gt routers and they work fine. -Original Message- From: "Curt Shaffer" [EMAIL PROTECTED] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.com Date: Mon, 6 Nov 2006 20:35:27 -0600 Subject: RE: [asterisk-users] Polycom autoprovision behind a NAT If you want that is fine. But as I mentioned when I put the phone on the same subnet as the ftp server with no NAT it works like a charm. Is there something in the config that deals with NAT traversal with regards to how it is provisioned? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Monday, November 06, 2006 8:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom autoprovision behind a NAT I if you like, I can take a config file(s) and put up over here as a test. Our ftp is working. It might be informative. -Original Message- From: "Curt Shaffer" [EMAIL PROTECTED] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.com Date: Mon, 6 Nov 2006 20:17:07 -0600 Subject: RE: [asterisk-users] Polycom autoprovision behind a NATTo be honest I don’t know for sure. I am using VSFTPD. I have never needed to set this with setups I have used it before and there is nothing in the config that says passive. So I’m guessing that its not. After you asked this I have googled passive FTP and it seems to be on the money as to what is going on so I will try passive and see if that helps. Thanks! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Monday, November 06, 2006 7:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom autoprovision behind a NAT I can confirm that the linksys routers cause ftp problems. Is your FTP server set to use pasive mode? -rb -Original Message- From: "Curt Shaffer" [EMAIL PROTECTED] To: "' Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.com Date: Mon, 6 Nov 2006 19:19:48 -0600 Subject: [asterisk-users] Polycom autoprovision behind a NAT I am having an issue with doing FTP auto provisioning of Polycom 501’s when they are behind a NAT. If I put the phone on the same subnet as the provision server it loads the configs and changes fine but as soon as I put in behind a NAT it comes up with cannot contact boot server. I have tried behind and replicated this behind a PIX 501, a Linksys SOHO router and a Motorolla SOHO router. Any ideas? Curt ___ --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] Hitachi WIP-5000/IP-5000 firmware
Good suggestion. It now seems to roam between access points nicely, even while a call is in progress. What access pooints are you using? -rb ___ 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] Polycom IP500 phones do not update time from time server
Kanuri, Seshu (Company IT) wrote: Does anyone know how Polycom 500s will be able to update their time. My setup for a time sync with Public domain Time servers is not successful. Seshu We had a user with a Sonic Wall Firewall who needed to set the snpt server to the IP address of his firewall in order to get the time to update. Not sure of the fw version. -rb ___ 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] Polycom 600 problem
Andrei (MPI) wrote: Hi Jared, Thank you for your reply. That server is for asterisk only, things like X-windows and samba were not even installed there. I limited what I could from system point of view. Digium support has qualified the box as clean for TDM400P operations. It is not clicks and pops, it's just brief moments of silence when I cannot hear the other side, while they can hear me okay. I will play more with configs and ideas you guys suggested. Also it is really strange, that the problem always appears for a specfic phone. 9 phones are ALWAYS working just fine, while 1 phone is doing this interruptions for all outgoing calls, 2 more phones are having this problem from time to time. All the phones are same model and are configured in the same way, difference is only in extensions, voicemail box and user/passwords settings. Echo cancellation? ulaw/alaw? Anything else? Thanks for your ideas. We had a similar problem. I moved the connection one switch port over the problem went away. I didn't really look into why the port should be a problem, nor did I do a cable test. -rb ___ 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] VoicemailMain can't read from phone keyboard!
Steven Wang wrote:
Hello
I try to set up voicemails for extension. When VoicemailMain gets called, it
prompts for mailbox and password. It seems not able to read from the phone.
So the authentication always fails.
I desparately need help to understand what is wrong. Here is a part of my
extensions.conf:
exten = _8500, 1, Wait(2)
exten = _8500, 2, VoicemailMain(${CALLERIDNUM})
exten = _8500, 3, Hangup
You don't mention the type of phone you're using, but on our setup with
SIP phones, we add a sipdtmfmode(inband) to what you have above. You
might try fiddling with that.
-russ
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Re: [Asterisk-Users] Confirm MWI doesnt work with SIP RealTime?
Matthew Boehm wrote: Can someone else confirm that your phone does not recieve MWIs when using SIP and RealTime? I can confirm that. Polycoms and CVS-Head from a month or so ago. I haven't tried it since. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ODBC Message Waiting Indicator
Hi, everyone: I was playing with the ODBC configuration to pull sip and voicemail config info from a MSSQL2000 server. Everything works great except for the message waiting indication on the Polycom phones (all three models). If I move the sip registration info to sip.conf, the MWI starts working again. I'm just asking if this is a bug or should I poke around some more to get it to work. Any pointers in the right direction are appreciated. Thanks... -rb ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Polycom IP 500 Voicemail
Wiley E. Siler wrote: I have a solution that allows me to assign a soft key with no problems. However, it seems like a waste the the hard button labeled Voice Mail is not dialing right into voice mail. Is there a known way yo do this? I have tried everything in the manual but it doesn't seem to work. I have IP 500s and I want to be able to use all three display lines for just lines on the phone. I think that feature is inly available on the 1.2.0 sip firmware. It works on ours but when you press it, you still have to pick a line, then connect. The line button goes right to the voicemail. Also, do you know if it is possible to program the buttons along the bottom of the screen like normal soft buttons? Probably, but I haven't looked into it enough And finally... Is there a way to make the system dial without having to hit the Send key after dialing a number? look at the digitmap in sip.cfg -rb ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Polycom IP 500 Voicemail
Wiley E. Siler wrote: Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using up.bypassInstantMessage=1 up.oneTouchVoicemail=1 in the XML to no avail. Has anyone been able to get a Polycom 500 to use the hardbutton to retrieve voice mail and drop directly into voice mail without going through all the menus? We programmed line 3 (line 6 on the IP 600s) on each phone with its own context/registration and set the IP 500 to auto dial into voicemail. extensions.conf: [voicemail] exten = 5501,1,voicemailmain2,[EMAIL PROTECTED] The phone.cfg file has a setting for autodial. I assume you can get a phone registered, but make sure dtmfmode is set to inband and set a mailbox= line to get MWI working. -rb ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Polycom IP600 stops to send/receive calls
Jorge Mendoza wrote: Hi, I'm testing a Polycom IP600. With firmware version 1.1 the phone reboots at any time. With firmware version 1.2, the first reboot was an endless reboot. Then I moved the phone to another lan port, then it worked fine. Then I installed again in the initial lan port and the phone works well. However after some time of inactivity (1 hour?), the IP600 stops to send and receive calls. After a reboot is works fine again. We have a * box with many BT101 and softphones working for months without any problem. I'm missing something? it is a bad config file? or it is a phone bug? We had one do the same thing. Changing the registration timeout in the phone.cfg file down to 20 seems to have fixed the problem. -rb ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 phones
Tomica Crnek wrote: Hi everyone, I have to test few models of SIP IP phones with Asterisk. I have seen on voip-info.org that there are lot of phones that work ok with Asterisk. But, I want to ask for suggestion - which models are the best for Asterisk? We have a dozen or so Polycom IP 500's and IP 600's working great. I highly recommend them. -russ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Good source for Polycom IP Phones
Matthew John Darnell wrote: Aloha, Does anyone have a good source for Polycom SoundPoint® IP 600/500/300 phones? http://www.reviewvideo.com/ Usually has them in stock. I've had good luck ordering from them. -rb ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk configuration inside a DMZ w/SIP
Brian D'Arcy wrote: Hello all, Im having a nightmare of a time trying to get stable results with SIP clients on Asterisk. I cant seem to find a configuration that works! In our office, we run a Sonicwall Pro 200, which is a sip aware, stateful firewall. We've discovered that certain versions of the sonic wall products do strange things with SIP. For example the TC170 with standard firmware works fine (Public Asterisk, Polycom IP600 behind the Sonic wall). Upgrade that box to the enhanced version and suddenly transfer and hold stop working. It's not just SIP, either. SNTP on the IP600 through the Sonic Wall gear changes the time by 10 hours. These things have been reported to Sonic Wall, but no word on a patch. -rb ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SoundPointR IP 300
It was my understanding that the SIP version is not available until May or June. My IP600's work great , though... -- Original Message -- From: Shad Mortazavi [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Fri, 16 Apr 2004 07:07:45 -0400 Dear Group, Does any one have experience using SoundPoint(r) IP 300? I have one call center on Snom 200's I'm adding a second and was looking at the SoundPoint, but needed some input. Thanks Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Sydney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any Polycom Experts Out There?
Hi, all We're using Asterisk CVS 3-19-04 with four polycom IP600s. The work very well and we're quite happy with them. They register fine and all four are able to place and receive calls, BUT two of them are behind NAT routers and when they place a call on hold, the call is dropped within 5 seconds. I couldn't find any relevant items in the archive search using terms like SIP, NAT and HOLD. The server has a public static IP, two phones have public static IPs and the two with the hold problem have dynamic NAT'ed IPs. Not sure what other info might be helpful. Any pointers in the right direction would be appreciated. -russ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] FW: Voice/Data mixed routing over Digium E1/T1 Card
FW: Voice/Data mixed routing over Digium E1/T1 CardWe are using it in three sites where the T-1 is pure IP and calles are routed in/out over SIP IAX2 and then to a channel bank. As a router the T400/T100 works great; I would highly recommend it. As voice server it works great. As a combined router/voice server (using only IP voice trunks) we've been having some issues (which are being worked on...Thanks Mark) Russ Beaupre, P.E. BoTech Communications Corp. - Original Message - From: Ray Burkholder To: [EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 9:01 AM Subject: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card The documentation mentions that the Digium channels can be split into some voice channels and the remainder of the channels used for routing IP traffic. Does any one have this in use in conjunction with Asterisk? Does it work well? Would you recommend it for a production server? Obviously, if this works, this makes for a cost effective platform where you obtain one E1/T1 to a provider, and they can provide TDM and data over the one circuit. No separate router required. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses dangerous content at One Unified and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940 Music on Hold Does not work
I had a similar problem when using an AS5300 with SIP. I'm not familiar with the 7940, but on the AS5300 there is a Voice-Activity-Detection setting. Disabling the VAD on the AS5300 corrected the SIP music on hold problem. Russ Beaupre, P.E. - Original Message - From: Babak Pasdar [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 05, 2003 1:16 AM Subject: [Asterisk-Users] 7940 Music on Hold Does not work I have a pair of 7940 ip phones and a standard analog phone on my test system. I have a few issues: 1. When dialing out (using m option) via SIP to another another phone Zap or SIP, I do not hear anything moh or ringing. 2. When two SIP phones are talking and either phone is placed on hold by the other, no moh is heard. The analog phone works just fine to hear music on hold and ringing or music when calling or on hold. It seems that * starts moh properly, as indicated below, however the ciscos dont respond. appropriately. -- Executing Answer(SIP/Desk2.1-2ec8, ) in new stack -- Executing Dial(SIP/Desk2.1-2ec8, SIP/Desk1.1|20|Ttm) in new stack -- Called Desk1.1 -- Started music on hold, class 'default', on SIP/Desk2.1-2ec8 -- SIP/Desk1.1-cbd4 is ringing -- SIP/Desk1.1-cbd4 answered SIP/Desk2.1-2ec8 -- Stopped music on hold on SIP/Desk2.1-2ec8 -- Attempting native bridge of SIP/Desk2.1-2ec8 and SIP/Desk1.1-cbd4 -- Started music on hold, class 'default', on SIP/Desk2.1-2ec8 -- Stopped music on hold on SIP/Desk2.1-2ec8 Any suggestion is appreciated. Babak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
