[asterisk-users] Polycom phone help needed
Hi All, I believe that a lot of the Sound Point Success story here. Can some one kindly let me know how to set up the Polycom Phones Sound Point or any reference to refer from... Your kindly help are appreciated. Thank you. ___ --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
[asterisk-users] Polycom phone references needed
Hi all, Any polycom phone v1.6 IP301 references? I had purchase three new phone and I cant connect them into Asterisk 1.2.11. I do appreciate if some one can point me how and where ? Thank you ___ --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] Asterisk and Siemens Legacy PBX
Hi James, James wrote; When I hit 9 on the siemens it does not get a dial tone from asterisk, I assume this is because I have not told asterisk to give it one. I might be wrong; My question is, are you sure your ISDN ( Asterisk span to Siemens ) is up logically? ISDN is no tone given, dial tone is actually produced by Legacy Side, when L1, L2 and L3 signals is up (eg coding, framing and timing), Legacy PBX will automatically make it self ready, and simulating the dial tone when user hit 9 to call out. I did try with Alcatel and Ericsson MD machine; both are simulating dial tone once L2 and L3 are working properly, so I assume that this is the Europe PBX standard. As fall as ISDN Legacy PBX is concern, it will throw out the digits if nothing wrong with the link. If possible, share with your Zapata.conf setting, may be group of us can help. Tq James wrote; Hi, I just realised I think I have missed a step Asterisk is not matching the extension from the siemens because the siemens has not even sent one yet, it is still waiting for a dial tone. When I hit 9 on the siemens it does not get a dial tone from asterisk, I assume this is because I have not told asterisk to give it one(dur!) How should I tell asterisk how to handle this, I have defined it a context and I know its making it this far, but I don9t know how to get the next bit coded. Any help appreciated ! Thanks James ___ --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] MWI from Asterisk to Meridian
If its control by serial port, you need TAPI license, and need some investigation period to integrate with Nortel system. ( you have to pay for the license, if your system can do so. ) Cheers. Andrew: The key here is try to create a way to integrate Asterisk Voicemail with existing Meridian PBX and send the MWI to M2616. I'm investigating the propietary protocol used by Nortel in the integration between Octel-Dialogic and Meridian for MWI. I believe is good idea to create an appl (and execute that appl in voicemail.conf like external application) for connect to the Meridian console port (Serial interface) and send the command to the end M2616. So, if you have some other idea is very welcome. Cheers. ___ --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] Ringing all extensions
Hi J. Oquendo [EMAIL PROTECTED], What LAN switch that you are using, and what type of IP phones that you are using? I've set up a ring all context on my gateway on extensions.conf: [EMAIL PROTECTED] ~]# grep *7 ast/extensions.conf exten = *7,1,Dial(SIP/1201SIP/1202SIP/1203,15,tr) Asterisk shows that it rings the lines but in reality nothing happens... Any thoughts on this? 2006-08-02 17:07:20 VERBOSE[7027] logger.c: -- Executing Dial(Zap/1-1, SIP/1201SIP/1202SIP/1203|15|tr) in new stack 2006-08-02 17:07:20 NOTICE[7027] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) 2006-08-02 17:07:20 VERBOSE[7027] logger.c: -- Called 1201 2006-08-02 17:07:20 VERBOSE[7027] logger.c: -- Called 1202 206-08-02 17:07:20 VERBOSE[7027] logger.c: -- Called 1203 2006-08-02 17:07:24 VERBOSE[7027] logger.c: == Spawn extension (main-aa, *7, 1) exited non-zero on 'Zap/1-1' 2006-08-02 17:07:24 VERBOSE[7027] logger.c: -- Hungup 'Zap/1-1' I truncated to phones to ring to 3 lines but in reality there are 42 lines that are supposed to ring at once when *7 is pressed. ___ --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
Subject: [asterisk-users] Slow dialing from PBX via E1
Subject: [asterisk-users] Slow dialing from PBX via E1 Hi :) I have a 'slow dialing' problem. When I dial 200# for the 'echo test' application from my PBX extension 1010, I see this in the console the instant I press the # key: -- Starting simple switch on 'Zap/65-1' -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3 so Asterisk has accepted the call setup from the PBX. Then exactly 3 seconds elapses, and finally: -- Executing Playback(Zap/65-1, demo-echotest) in new stack -- Playing 'demo-echotest' (language 'en') at which point Allison announces 'You are about to enter an echo test..' How can I remove this 3 second pause? It's really annoying, and it doesn't happen when I dial out from the legacy PBX via an ISDN30 bearer not connected to Asterisk (nor does it happen with SIP phones on Asterisk). Even with debug + verbose both at 99, I see no extra information The extensions.conf is trivial: [general] static=yes writeprotect=yes [fromaxxess] exten = 200,1,Playback(demo-echotest) ; Let them know what's going on exten = 200,2,Echo ; Do the echo test exten = 200,3,Playback(demo-echodone) ; Let them know it's over This is with Asterisk 1.2.4 and Zaptel 1.2.3, on a Sangoma A104u (Sangoma support say their driver does no buffering and can't understand why this is happening) As ever, any advice warmly welcomed :) Cheers, Gavin. Hi Gavin, This is the default of setting of the Asterisk. If you wish to adjust the timing, please edit the source file of the Asterisk name call chan_zap.c, And look for the static int matchdigittimeout line to change the setting. The timing is in millisecond. Remember, compile your Asterisk after the changes. Good luck. ___ --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
[asterisk-users] RE: asterisk-users Digest, Vol 25, Issue 2
(Andrew Kohlsmith) wrote: Re: MWI from Asterisk to Meridian So, as I said, you are stuck using a Nortel ATA and an FXS port on Asterisk and using a hookflash *1 sequence to toggle it. Unfortunately the VM callback # will be the ATA's DN, so only one person at a time can access voicemail. Johann Steinwendtner wrote: ; If you need to have an external program, i.e. /usr/bin/myapp ; called when a voicemail is left, delivered, or your voicemailbox ; is checked, uncomment this: ;externnotify=/usr/bin/myapp Can your client accept that, Messages alert from 1 extension, and dial different number to access voicemail? Means omit the VM Call back. Will, like some brand alert from some extensions and vm call back will be different extensions. Nortel side, configured those vm alert port not accept the call from any extension. ( to avoid voicemail call back ) Create speed dial number, let user to access vm from Asterisk. You may look at this link it might help :) http://www.voip-info.org/wiki/index.php?page=Asterisk-Panasonic1232vm ___ --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
[asterisk-users] RE: alcatel ip touch 4068 ... sip?
Hi Cesc, Alcatel does not offer any SIP Terminal, it phone's are H.323 it's handle for their own features usages. ( like ADSI type ) From: Cesc [EMAIL PROTECTED] Subject: [asterisk-users] alcatel ip touch 4068 ... sip? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, Quickie ... is the alcatel ip touch 4068 (or any other in that series) sip enabled? If not, does alcatel have a sip-enabled phone? Cesc ___ --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
[asterisk-users] help for SPA-2100
Hi all, I got the SPA-2100 and I can dial call from other extension to this SIP ATA, however I got problem make out the call the Asterisk, I believe somethings to do with the Dial Plan in side the SPA-2100 configuration file. My setup simple topologies, ATA connected to my LAN, which same as other IP phones. Other IP phone can call SPA-2100, (no problem) but SPA-2100 can not call others IP phone. ---even though SIP.CONF file I place type=friend When I am make call from SPA-2100 ATA, CLI show nothing. Some one can help? ___ --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
[asterisk-users] meetme application doubt
Hi all, I had some doubt about meetme application, hope that some one can tell me what to do? (No GUI and CLI in this case) Say 5 user in the conference room 1000. Ofcoz, each user is holding userID in the conference room. Say user 1 SIP/200 Say user 2 SIP/201 3 SIP/202 4 SIP/203 5 SIP/204 Now, SIP/204 is the admin, how can SIP/204 know the phone extension is holding what user ID while there are conducting a conference call? Or mean. How can the SIP/204 kick out SIP/200 from conference room 1000 ___ --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
[asterisk-users] Polycom compatible phone for Asterisk
Hi all, Can some one provide me the infor about polycom phones model that compatible and stable to work with Asterisk? I intend to purchase IP 300, and IP 501 models. Tq ___ --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
