Re: [Asterisk-Users] Avaya firmware
I had no problems updating the firmware. Is anyone able to use the MWI light on their Avaya 4602? Thats the only thing on mine that I cant do. On Wed, 18 Aug 2004 08:38:18 -0700, Aaron Johnson [EMAIL PROTECTED] wrote: Tenorio, Leandro wrote: Just guessing, but 've you tried the to rename Sip_4602ap1_0.ebin to appsip.ebin I did. The problem turned out to be with my HTTP server. I switched HTTP servers and everything is now running fine. ___ 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 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: Re: [Asterisk-Users] First Post: Any existing AVAYA Switch - Asterisk Voicemail configs?
Avaya refers to this as a 'green feature' and it has to be enabled by Definity in order to send CID to an analog port. In addition, you must also be using the TN2793B analog circuit pack. -CaNaBiS On Sat, 07 Aug 2004 09:57:17 +0800, Leo Ann Boon [EMAIL PROTECTED] wrote: I believe the Definity sends the voicemail integration signal (CID etc) via DTMF and not the regular FSK CID supported by *. Mike Cathey wrote: On Fri, 2004-08-06 at 06:02, Jason Williams wrote: It won't work on the FXO We verified that the Definity sends CID on analog ports (that's how we're connecting it to our existing VM system). * doesn't seem to see the CID for some reason though. :\ However it should work fine on the PRI check the definity trunk form and set send number to y We have it configured to send CID, however, it only seems to actually send CID when a call is routed from the PSTN through the Definity to *. Is there something else we're missing here? Someone mentioned that we'd need Q.sig. Cheers, Mike ___ 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 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 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] Avaya IP Phones and Asterisk
I am using an Avaya 4602SW SIP phone with Asterisk. Works nicely. On Wed, 11 Aug 2004 16:25:02 -0300, Tenorio, Leandro [EMAIL PROTECTED] wrote: Be carefull, what's you phone type, most of the Avaya IP Phones use custom h.323 protocol. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Katerina Sadri Sent: Wednesday, August 11, 2004 4:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Avaya IP Phones and Asterisk I have added H.323 support for ASTERISK PBX using the OpenH323 library. I thought that since the Avaya IP phones are H.323 phones I will be able to make them register to Asterisk server. Does anyone know if this is possible ? Katerina ___ 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 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 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] Avaya and Asterisk
You can load the newly released Avaya firmware on the Avaya 4602's to register with Asterisk. I am doing this and it works. On Wed, 11 Aug 2004 11:53:07 -0700, Katerina Sadri [EMAIL PROTECTED] wrote: So far I have not found a way that I can register the Avaya phone with Asterisk. From what I have found so far is that Avaya phone needs the Avaya Media Server and Avaya Gateway. Looking at the h.323.conf (in Asterisk) and the file 46xxsettings.txt (avaya file located in tftpboot) there are no settings to make the phone initialize. I have sent an email to the Asterisk Users Mailing List to see if anyone has done it before. I will investigate some more and I hope that someone from the mailing list will answer. Katerina ___ 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 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] First Post: Any existing AVAYA Switch - Asterisk Voicemail configs?
Hey Brian, my name is Brian too. I too administer a Definity. I use Asterisk for personal use, but to use Comedian Mail with Asterisk should be about the same as using anything else like Intuity Audix. Create your hunt group in the Definity, and then assign how ever many analog ports you want, like for the Intuity we use 8 ports. I assume you would have to use zap analog cards in the Asterisk server. Then, from the Definity just plug your analog extensions into your zap ports and then configure Asterisk to send all zap calls to the extension you define for voicemail. Give your users the extenion of the hunt group, and then the Definity will dial first available zap extension. I think you would need FXO cards in your asterisk box, but someone else should confirm that. - Original Message - From: Brian Hudson [EMAIL PROTECTED] Date: Mon, 2 Aug 2004 17:01:42 -0400 Subject: [Asterisk-Users] First Post: Any existing AVAYA Switch - Asterisk Voicemail configs? To: [EMAIL PROTECTED] This is my first post, so please feel free to direct me to another list if needed. I am in the early stages of researching Asterisk. I administer a small Avaya Definity G3 switch (~400 users). Can anyone point my to resources/documents/actual implementation notes of using Asterisk's voicemail with an Avaya Definity switch? Many thanks, Brian Hudson ___ 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] Successfully Using $135 Avaya sip phone
I have: [EMAIL PROTECTED] I emailed [EMAIL PROTECTED] about it, they said that I need to open a trouble ticket and go through the standard escalation process which means they want the sold-to number of my Avaya switch...well, what I have tried to explain to them a dozen times is that I am not using it with an Avaya switch, and if their SIP phone was intended to meet SIP standards they should try to help me fugure out why its not working properly. Let me know how it goes when you get yours. I am pleased with the overall quality of mine aside from this one issue...again, it may not be Avaya, it may not be Asterisk, it may just be something in my configuration. On Thu, 29 Jul 2004 00:23:00 +0200, Andy Powell [EMAIL PROTECTED] wrote: Brian Elton wrote: The phone stops working after about 20-30mins if I have mailbox=context in Asterisk; when I do have mailbox=contect in asterisk the sip debug returns 481 extension does not exist. Anyone willing to help me figure out why? what do you mean : mailbox=context (or contect) this should be [EMAIL PROTECTED] or mailbox=number to use the default context. eg: [EMAIL PROTECTED] where sales is a context in voicemail.conf Andy ___ 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 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] Successfully Using $135 Avaya sip phone
I think I am the first to use the $135 Avaya 4602 SIP phone, but I need some support from the community to fix one problem I have with it. The phone stops working after about 20-30mins if I have mailbox=context in Asterisk; when I do have mailbox=contect in asterisk the sip debug returns 481 extension does not exist. Anyone willing to help me figure out why? I.E. Is it an Asterisk issue? Is it a configuration issue on my side? Is it a bug with the phone? Any help would be greatly appreciated and it would open the way for an excellent, economically priced phone to the Asterisk community. --Asterisk CVS-HEAD-05/25/04-15:08:06 ___ 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] Successfully Using $135 Avaya sip phone
What I have is: [EMAIL PROTECTED] Where [avaya] is the context in voicemail.conf Is that right? When I had it as just mailbox=2002 that had the same problems. On Tue, 27 Jul 2004 23:11:51 -0400, James W. Brinkerhoff [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 27 July 2004 10:50 pm, Brian Elton wrote: I think I am the first to use the $135 Avaya 4602 SIP phone, but I need some support from the community to fix one problem I have with it. The phone stops working after about 20-30mins if I have mailbox=context in Asterisk; when I do have mailbox=contect in asterisk the sip debug returns 481 extension does not exist. Anyone willing to help me figure out why? I.E. Is it an Asterisk issue? Is it a configuration issue on my side? Is it a bug with the phone? Any help would be greatly appreciated and it would open the way for an excellent, economically priced phone to the Asterisk community. I believe it's supposed to be [EMAIL PROTECTED] so if you had [mycontext] 1001=.. in your voicemail.conf that would be [EMAIL PROTECTED] - -jwb -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBBxl3HyXYB+SEybkRAkjKAJ9v8cga7YfVm0JomU7daojU8afmRACffKQZ RVF7C2Eu27ivPmGCPDpAmUQ= =x9Kl -END PGP SIGNATURE- ___ 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 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] Help w/ SIP response 481
OK, I think I have my problem narrowed down on my Avaya 4602SW SIP hardphone. When I reset the phone the phone works perfect up to the point until I get the following error in the CLI: -- Got SIP response 481 Call Does Not Exist back from my.home.external.ip This is how I have the SIP extension setup: [2002] type=friend username=2002 secret=mypassword host=dynamic context=from-sip mailbox=2002 nat=yes qualify=yes dtmfmode=info reinvite=no canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm callwaiting=1 The SIP phone is behind a wireless router with no ports forwarded to it, and the Asterisk server is straight on the Internet. I also have a Sipura behind the wireless router and it works just fine. If anyone could help me that would be great, this is driving nuts. Thanks ___ 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] Cheap PoE switches/injectors?
Hey man, I have a bunch of used power injectors that I would actually like to sell. If you are interested I will gather them all up and count them, but I know I have at least 24. I'd be glad to send you one to test. They are all Avaya/Lucent brand, they should work for any type of phone. Thanks, -Brian On Mon, 19 Jul 2004 09:03:49 -0700 (PDT), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Look out for 3c17205 switches from 3com and read the QOS thread posting here at the moment. P -Original Message- From: Scott Laird [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004, 7:58 AM To: '[EMAIL PROTECTED]' [EMAIL PROTECTED] Subject: [Asterisk-Users] Cheap PoE switches/injectors? I'm trying to spec out hardware for a new office, and I'd like to include power over Ethernet as an option. I've seen a handful of PoE injectors around $1000 for 24 ports and a couple switches up around $2500 for 24 ports. Are there any cheaper options, short of buying a boatload of 1-port injectors off of ebay? I don't really need more then 24 ports of PoE out of 48 total ports, so one of CIsco's big PoE switches is complete overkill. This is for a startup, where cheap is important. Thanks. Scott ___ 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 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 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] Flash Zap trunk from a Sipura
I think this has been an ongoing issue. When you figure out a solution let me know. The only solution I could come up with isnt feasible for everyone. I have another pbx that provides the dialtone for my asterisk box. I put two zap cards in my asterisk. On my other switch I set it so that if line 1 is busy it rolls to line 2, this way when my second call comes in I can switch over to it from my Sipura. It works for me, but isnt possible for everyone. On Mon, 19 Jul 2004 10:09:30 -0700, Trevor Peirce [EMAIL PROTECTED] wrote: Hello, In my quest to create several proof of concepts for what can be done with Asterisk, I've run into a bit of a problem. I have a pair of SPA-2000's acting as off premise extensions for an analog line. When a call waiting call comes in, the caller id information makes it though the ULAW codec and displays on the caller id box, however asterisk doesn't seem to want to pick up the hook-flash sent by the Sipura to answer that second call. I have configured the Sipura to send hook-flash messages to asterisk, and it does, but asterisk doesn't seem to know what to do with them. I've searched the wiki and google with no success. Thanks, Trevor ___ 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 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] IP Phone recommendation
I really wish a few users would help me with my Avaya 4602SW SIP phone as I believe I am the first to try it with Asterisk. Its a very modern looking phone, I've seen it on froogle for about $135, and it has a decent set of features, and Avaya is really good quality and has a good name in telephony. The Avaya 4602 has been out for a while, but they just released the SIP firmware for it about 1 week ago. The only issue so far that I have with it is that it stops working after I get a -- Got SIP response 481 Call Does Not Exist back from my.home.ip.address which occurs about 15 or 20 minutes after I plug the phone in, if I reset the phone it, again, will work for about 15-20 mins. On Mon, 19 Jul 2004 09:25:48 -0700, Harry McGregor [EMAIL PROTECTED] wrote: On Mon, 2004-07-19 at 09:04, Yiannis Costopoulos wrote: Hi, I am looking for some affordable IP Phones. Any experiences with the SipToneII by ipDialog? So far our experience with the IP Dialog SipToneII is not good. It locks up after hang up on us, and just does not play nice. If anyone has any suggestions on how to get it working, we are all ears. The IP Dialog phone is running $200, while the Zip 4x4 is running $280-300 (depending on qty). We are deploying ~60 phones. Originally we were going to try and do 20 Uniden UIP200 and 40 Zip 4x4. We were unable to get our hands on a Uniden, and found that it would not even be available for an august deployment, so we decided to try the IP Dialog phone. The Uniden would have been a very worth while cost savings, as it's $150 and the Zip is $280 for our qty, but the $80 savings of the IP Dialog is not worth it to us. Harry What about soft phones? Any recommendations there (for Windoze and Linux)? Have not tried it but what about PhoneGaim? Thanks, Yiannis ___ 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 -- Harry McGregor, Computing Manager Tucson Support Group - U.S. Geological Survey University of Arizona - Environment and Natural Resource Building 520-670-5574 (office) - [EMAIL PROTECTED] 520-661-7875 (Cell) - [EMAIL PROTECTED] The opinions/statements expressed herein are my own and should not be taken as a position, opinion, or endorsement of the University of Arizona or the U.S. Geological Survey. ___ 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 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
