RE: [Asterisk-Users] Sipura-841 Problems
Wow... sort of a horror story there. Not sure I have a lot of input other then we have two 841's that we've been using for about a month now without out problems. (I've actually been rather impressed.) One thing that comes to mind though, have you tried updating the firmware to the latest? Or is that impossible giving their behavior.. -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of LES.NET (1996) INC. Sent: Thursday, March 10, 2005 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sipura-841 Problems Hello Everyone. Please let me know if this is the incorrect forum for this question. I recently acquired 3 Sipura-841 phones for use with my asterisk system. I provisioned and deployed one. It worked for 3 days, then uddenly, it stopped working. Upon reboot, the MWI light blinks 20 times and that's the end of it. It has never registered again since then. So I replaced the phone with another that I had. I provisioned it, it went online for a split second, then it too died! MWI light blinks 20 times, and the phone is dead. So Sipura sent me out 3 replacement phones the next week. Now, I've provisioned a replacement phone. I was talking on it for 2 hours when it died. (guess what) Same problem mostly. MWI light blinks 20 times, and the phone just doesn't work. (Except this one will reboot itself every 5 minutes. About every 10th reboot, it will go online for a few minutes, then die again) Curious if anyone has experienced the same problem? I have tried all these phones in four totally unique environments. (ie: differetn dhcp servers, differnet hubs/switches, different ISPs, etc) Problem is consistent across them all. I've tried cvs-head, and asterisk-1.05, same problem. One other thing.. If I use the exact same SIP credentials/config on a Linksys PAP2, or Linksys RT31P2, it works fine, or even any softphone. Any direction would be appreciated. Les ___ 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 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] Sipura 841 Headset microphone volume?
I think you're in luck :) Sipura has a new firmware out for the 841 now, version 3.1.1a. (Quite the jump from 0.9.5) It includes -6db, 0db, +6db options for input gain on the handset, headset and speakerphone now. Release notes and downloads are available on their site. -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Bussinger Sent: Thursday, March 10, 2005 9:09 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Sipura 841 Headset microphone volume? We're setting up some Sipura 841 phones and they're working pretty well, but the microphone volume on the headset (not the handset) is too loud with our Plantronics headsets. Is there some way to turn down the amplification on the headset mic? The microphones are picking up the sound of someone walking on the floor across the room and every little movement or shuffle of the user. I found places to control the output volume of handset, headset, speaker, and ring but I haven't found anything for input control. Can anyone suggest anything? Thanks! ___ 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 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] Single FXS port with multiple phones ?
Hello all.. I have what I think is a rather simple question here. We're using * for home and business lines with great success so far. (inbound, outbound, several hard phones, voicemail, IVR, etc) My next project is to get some of my traditional POTS phones at home on the system. I understand this requires an FXS interface(s), which really is my question. If I don't care about having individual extensions for every line (as most homes don't) then I'm thinking it should be possible to just have a single FXS interface that would replace the telco's normal line connection to my home wiring after the demarc (assuming it could provide the juice required by my phones). Stop me here if I'm wrong. :) In trying to research this further, I'm looking at getting an SPA-3000 to do the FXS duties (as I could also use the FXO interface for something else), and it gives a variety of FXS specs. After a bunch of Googling it would seem that REN (ringer equivalence number) is one I need to pay attention to. The SPA-3000 says it can provide 3 REN. So as long as I stay under that total for my phones am I pretty much safe or are there other specs that I'm overlooking? Like possibly if more then one phone is off hook to talk on the line at the same time? In a semi-related question, I have a bunch of 2.4 Ghz cordless sets here and the base stations say Ring Equivalence: 0.0B. Could this truly be the case? Like the phones just monitor the line for a ring signal but don't draw current from it when it's ringing. I can see how it's possible as they have their own AC, but I just wanted to confirm. Sorry for the long post, I thought it would be shorter. -James ___ 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] Single FXS port with multiple phones ?
Excellent, thanks for the info. That hiss is interesting; maybe all the extra wire makes it tougher for the SPA to drive. In any event it sounds like it will do what I want. Thanks.. :) -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luki Sent: Wednesday, March 09, 2005 6:52 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Single FXS port with multiple phones ? James, the SPA's have a FXS Port Power Limit which is 3 by default. The range is 1-8. I have a SPA 2000 with 6 phones on one port (entire house wired to one port) which works OK as long as your phones do not rely on power from the line (cordless phones, etc. are fine). On the second port I have just one phone, no problem using both ports simultaneously. The only problem I have is the hissing noise on the line -- it seems to be higher than what I'm used to from regular telco lines. I'm confident it's due to the wiring in the house, not the number of phones. I had all six plugged in directly, works fine, no hiss; if I leave just one plugged in and still use the house wiring, more hiss. It's probably to do with the impedance setting, but neither one seems to make a noticeable difference. --Luki ___ 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 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] Paging and Intercom using Sipura SPA-841
Well, I'm using intercom with a couple SPA-841. Currently using * CVS from
a couple weeks ago and a macro in extensions.conf. It should give you some
idea:
[macro-intercom]
; ${ARG1} - Extension
exten = s,1,SetCDRUserField(intercom)
exten = s,2,SIPaddheader(Call-Info: \;answer-after=0)
exten = s,3,Dial(${ARG1})
exten = s,4,Congestion
exten = s,104,Busy
-James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kavit Munshi
Sent: Wednesday, March 09, 2005 10:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Paging and Intercom using Sipura SPA-841
I want to implement a one way announcement and paging facility using
Asterisk and Sipura phones. The wiki says Sipura phones only support
Auto Answer using the Call-Info header which is no lone shipped with
asterisk stable since 1.0.4.
I would like to ask if anyone has implemented a similiar facility
using Sipura SPA-841 or any other SIP phones. If I could take a look
at how they have done this with a snippet of the extensions.conf I will
be very grateful.
regards,
Kavit
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RE: [Asterisk-Users] SJphone on PDA registering with Asterisk???
Well, given your setup and the fact that you aren't seeing anything on the console with verbose debugging on, I'm going to guess there is a network/routing issue here. I'd try getting the PDA on line and just doing some simple ping tests to the 192.168.250.x network from it. (including to the * server). If you can reach it then is surely should let you register or at least give you info on the console. -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Sunday, March 06, 2005 8:59 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SJphone on PDA registering with Asterisk??? James Pooton wrote: I'm all so using SJphone on my x50v, works surprisingly well :). Is voip.elmit.com also in the 192.168.1.X NAT space that your PDA is in? There might be the problem: I have the server at two ethernet cards reachable: Extern with a public IP Intern with 192.168.250.20 on this internal LAN is a wireless accesspoint, which in return changes the IP address to a network 192.168.1.x There is a NAT between the internal server IP and the PDA, and there is a nat between internal IP and Internet. Do you have host=dynamic in your * sip.conf entry for 701 ? Actually might help to toss your sip.conf entry out here for 701 without the secret. [701] ; Test phone 701 type=friend username=701 secret=very_secret nat=yes host=dynamic context=test_phone canreinvite=yes disallow=all allow=ulaw allow=alaw dtmfmode=rfc2833 qualify=1000 [EMAIL PROTECTED] pickupgroup=1 qualify=yes Do you see any connection attempts on the console? (ie starting * with -gcvv) No, not at all!! bye Ronald Your not far off.. -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Sunday, March 06, 2005 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SJphone on PDA registering with Asterisk??? C. Tomlinson wrote: Ronald, You will need to give *more* information than that I have SJphone on my PDA, and have setup a SIP account on *, and it works fine :-) I take it you have setup sjphone to register to *. I take it your PDA has a network connection? I have setup a sip account at asterisk (701:password) I have an asterisk (voip.elmit.com with an IP address) I have setup a new profile on the PDA sip-elmit: Initialization: as suggested Sip proxy: Proxy domain: my IP address Port 5060 Userdamain: voip.elmit.com Advanced options (nothing set) Sip: Expose software version Enable STUN unsage Redirection: nothing selected STUN: as suggested Use elimit-sip elmit-sip in use (save changes) Display shows: elmit-sip SIP: registering as sip:[EMAIL PROTECTED] ... Host address: 192.168.1.101 NAT/Firewall: Full Cone NAT -- Ronald (office) (Ro) sip:[EMAIL PROTECTED] click on dial Nothing happens, .. not registered in *, ... What have I done wrong? bye Ronald -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ 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 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] Recommended Phone for beginner
I've been running a pair of Sipura 841's with my business partner. So far I've been very impressed for the money. Having multiple lines (with multiple registrations) has also been really nice as we both have our work line and home line off the phones. (And you can update them for 2 more lines for a total of 4 also). Custom ringers you can upload, 100 number phone directory and intercom capability really make it fun. About the only thing I'd like to see on it is a backlit display. You can find out a lot more online at Sipura's site, but it's definitely a model to consider. Good luck.. -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Burke Sent: Monday, March 07, 2005 7:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Recommended Phone for beginner Hello everyone, I've been watching this list for a while, but it is the first time I've posted. I'ved decided to setup a * server for my house and will need 3 phones (one main, one for my wife, and one for my office). I was wondering if there was a particular brand that people reommended? I'd like ot get an actual SIP phone, instead of an adapter like the Sipura SPA-2000. I've been looking at the Grandstream BudgetTone 100 series but after looking at the Wiki for setting up * with that phone it looks like it might be more trouble than its worth. Of course I would love a Cisco 79* but I'd like to keep the cost at a minimum but get a good amount of flexibility in tersm of features. Hopefully once I get over the learning hump I can start contributing to this list. Any input would be appreciated. Thanks, Ryan ___ 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 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] SJphone on PDA registering with Asterisk???
I'm all so using SJphone on my x50v, works surprisingly well :). Is voip.elmit.com also in the 192.168.1.X NAT space that your PDA is in? Do you have host=dynamic in your * sip.conf entry for 701 ? Actually might help to toss your sip.conf entry out here for 701 without the secret. Do you see any connection attempts on the console? (ie starting * with -gcvv) Your not far off.. -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Sunday, March 06, 2005 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SJphone on PDA registering with Asterisk??? C. Tomlinson wrote: Ronald, You will need to give *more* information than that I have SJphone on my PDA, and have setup a SIP account on *, and it works fine :-) I take it you have setup sjphone to register to *. I take it your PDA has a network connection? I have setup a sip account at asterisk (701:password) I have an asterisk (voip.elmit.com with an IP address) I have setup a new profile on the PDA sip-elmit: Initialization: as suggested Sip proxy: Proxy domain: my IP address Port 5060 Userdamain: voip.elmit.com Advanced options (nothing set) Sip: Expose software version Enable STUN unsage Redirection: nothing selected STUN: as suggested Use elimit-sip elmit-sip in use (save changes) Display shows: elmit-sip SIP: registering as sip:[EMAIL PROTECTED] ... Host address: 192.168.1.101 NAT/Firewall: Full Cone NAT -- Ronald (office) (Ro) sip:[EMAIL PROTECTED] click on dial Nothing happens, .. not registered in *, ... What have I done wrong? bye Ronald C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: 06 March 2005 14:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SJphone on PDA registering with Asterisk??? I try to setup SJphone on my PDA, but it does not register with Asterisk. I have setup a sip account on asterisk, ... Can anybody give me a hint? bye Ronald ___ 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 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 -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ 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 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] IP Providers pass CallerID?
That's surprising; I thought they were one of the larger outfits. I have tried quite a few for outbound lately and the only one that has reliably passed Caller ID (using our 406 area code) is simpletelecom.com. They are also the only ones I've tried that respond to a support ticket in a reasonable amount of time. What is really odd is that I can pass caller ID with iax.cc/Sixtel in other area codes, but nothing in 406 which I need to do. (Our local numbers) Anyone know why that would be ? -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of TELUX Sent: Sunday, March 06, 2005 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IP Providers pass CallerID? Are there any IP Providers that will pass Caller ID? Broadvoice used to but no they dont. THX ___ 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 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] X100P Clone, Which one?
Well, considering I'm on topic, I shouldn't get flamed to badly for this. I have a bunch of these working well in my home experiments: http://www.laptops4me.com/product_info.php/products_id/1444 And yes that price is correct and they do arrive. :) Not everyone can justify buying the supported hardware to kick the tires and try it out * at home. On the commercial side support is worth every penny I'm sure. However I think it helps the community to have low cost entry options for people to learn. I know it helped me. -James ___ 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] Music on hold on timing sources
I think MOH works without a timing source, but can get end up sounding poor without one. (stuttering, etc) At least that has been my experience. Without a card, ztdummy becomes an option for timing if the box has the correct USB controller to make use of: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy Or you can add a cheap card. We have a box colo'd that I just threw an 8$ card in and works great for MOH, meetme, etc. As for no MOH on hold button pressing, I'd look to your * config for MOH.. -James From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Wednesday, March 02, 2005 2:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Music on hold on timing sources Hello: I have read that music on hold requires a timing source (which I never had to worry about previously since the server had zaptel hardware in it)...now I'm configuring a server in a colo which has no zaptel hardware. If I use the dialplan to run MusicOnHold(), I do get the music upon dialling that extension, but if I try to use the hold button on either a 7960 or X-Lite I get nothing. Is this the expected behavior? I figured that if a timing source was needed that MusicOnHold() should not work, but it does Thanks, Marty ___ 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
