[Asterisk-Users] Re:Remote Call Forwarding
Philipp, I already have that call-forwarding feature set into asterisk. What I am looking is how to set that feature remotely by calling into your voicemail or any given no. so that person can set call-forwarding remotely. Few of our sales people want this kind of feature, because if they are stuck in traffic and expecting important call, so that, they can call from there mobile into asterisk and set call-forward to there mobile. With the current call-forwarding feature, person has to be there physically to set this feature from there extension. If somebody has any example, it would be great help. Regards, KD Date: Thu, 20 May 2004 11:02:31 +0200 From: Philipp von Klitzing [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Remote Call Forwarding To: [EMAIL PROTECTED] Organization: AEGEE Reply-To: [EMAIL PROTECTED] Hi! I am trying to find remote call forwarding feature in asterisk. I don't know is it possible or any one had already done it. The Wiki is your friend: http://www.voip-info.org/wiki-Asterisk+call+forwarding Cheers, Philipp ___ 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] Remote Call Forwarding
Hi, I am trying to find remote call forwarding feature in asterisk. I don't know is it possible or any one had already done it. SBC (local Telco) provide such feature. I can call into my voicemail number, and set the remote-call-forward to my cell or another number. It is like person can remotely manage to set the call-forward or DND to his/her extension. Can this be doable in asterisk? KD ___ 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] Re: Random Echo
TC, Appreciated your help and will try out TelLabs card and see if we can get rid of echo. Yesterday I did some changes in TX and RX attenuation setting on Channel bank and it reduces the echo, but it is not yet vanished as we wanted. Any way Thanks. Regards, KD Date: Wed, 17 Mar 2004 19:47:16 -0800 From: TC [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Random Echo To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] I did some google search but didn't find any details, about how to configure between Adtran 750 and T100P. If you have already done, please give us some details. not sure what level of dtl you want its quite straight fwd It varies depending on the chasis but in general there are T1 in and T1 out DB-15's for each T1 circuit you want to echo cancel. A straight T1 rj-45 cable goes to the channel bank other end is db-15 in to Tellab then a T1 X over goes from the Tellab db-15 out to the T100p card Then there are external Mode and Chan switchs that allow you to configure the T1 circuit (the line bld, framing, and coding/signaling), and then other setting to allow channels FXO/FXS LS, GS and enable echo cancel channel by channel, I do it for all fxs/fxo ports and turn ALL * echo cancel in zapata OFF.. ___ 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] Re: Random Echo
Interesting, how do you configure T1 Tellabs 2572's? We also have same issue and I am tried resolving this echo issue. I did some google search but didn't find any details, about how to configure between Adtran 750 and T100P. If you have already done, please give us some details. Regards, KD Date: Wed, 17 Mar 2004 15:52:09 -0800 From: TC [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Random Echo To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] I have a user that complains of echo for the duration of a ZAP-SIP call. I have experienced it too, but it only happens randomly. Is there any way to debug the echo cancellation other than doing a zap show channel X? I am using a T100P -- Adtran 750. Has anyone done anything to successfully stop echo with the Adtran 750? Yup inline T1 hardware echo cans :) I am using some older 64ms T1 tellabs 2572's, stops echo in its tracks ___ 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] Re: Adtran 750 DID question.
Scott, You can't see DNIS on any channels/Line? If I understand correctly you can't see digits coming in from your Telco? And it falls into your S context? Try changing your Adtran Firmware, I tried on L35 and L36 it didn't worked properly, so I am using L34. Also, try changing the wink time in Zapata.h from 150MS to 250MS, once you change the timing you have to recompile Zapata. Kd Subject: RE: [Asterisk-Users] Re: Adtran 750 DID question. Date: Fri, 30 Jan 2004 19:16:26 -0500 From: Bisker, Scott (7805) [EMAIL PROTECTED] To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Yes. immediate=3Dno is in zapata.conf before the channel declaration. = This makes absolutely no sense at all. -sb ___ 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] Re: Adtran 750 DID question.
I posted this below message few days ago, but some how it didn't show up in the mailing list. When some one calls into your DID Trunk line what symptom do you see on Adtran as well as on Asterisks console? Asterisk winks properly in FXO DPO mode, we had checked this things with our Telco instruments. If your PBX doesn't wink to Telco, your DID line's will become busy, that is one way to check whether Asterisk is winking properly or not. What series FXS card you are using? Is it L1 or L2? Because only L1 series cards work with our DID trunk line. I don't know why and still Adtran technical support is not able to figure out. I have same configuration running in my office and finally with all the help from Digium and adtran, problem seems to be less. Still fully it is not resolved yet because FXS L2 is not working. Hopefully Adtran will release new firmware. Regards, Kekin -Original Message- From: Kekin Dand Sent: Tuesday, January 27, 2004 5:24 PM To: [EMAIL PROTECTED] Subject: Re:Incoming DID call Voice Problems I had similar problem and it took all most 2 months to resolve it. Few things you have check in your Adtran 750 configuration. 1. For incoming DID trunk line it has to terminated on FXS card. (Which I think you already did) 2. FXS card needs to be set on FXS DPO mode in order to work properly. 3. Your DID trunk line should be configured in Asterisk as em in zaptel.conf and em_w in Zapata.conf. Reason you are facing this problem either your battery is not getting reversed and Telco can't see Answer Supervision on your line when the calls get connected. If you have ohms meter or multimeter check your DID line voltages. When inbound call comes in and both parties go off hook you should see positive voltage on that line, in idle situation(on hook) you will negative 48V. Yes, local calls are different then long distance. Answer Supervision is not required for local calls. Only for long distance it is required, so that Telco can start billing. Hope this should resolve your problem. Regards, Kekin Subject: RE: [Asterisk-Users] Incoming DID call Voice Problems Date: Mon, 26 Jan 2004 09:31:32 -0500 From: Bisker, Scott (7805) [EMAIL PROTECTED] To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] I have an updated question on this one. It seems that only inbound long = distance calls (calls from outside the local calling area) on our DID = trunk have one-way voice. I have my adtran 750 fxs lines configured as = FXS Loopstart with all the defaults. Again, the problem is that once = the call bridges, the outside caller can hear the person they called, = but the inside person can't hear the caller. This happens regardless of = the internal technology, SIP, Zap, H323. Could it be possible that inbound long distance calls are signalled = different than inbound local calls? Inbound calls on the PRI work = flawlessly. Any ideas -sb ___ 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] RE: DID Trunk Lines and Caller ID
Michael, We also have the same DID trunk lines and was told by Telco, on this line we can't get callerID facility. I don't know is there some way to do it. I would like to ask you a question, what hardware are you use to terminate all 14 DID trunk line, into asterisk. I am facing little problem with this line and my Telco is not able to resolve the problem. Weather the problem is my side or there end don't know yet. Regards, Kekin From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Mon, 5 Jan 2004 08:28:08 -0700 Subject: [Asterisk-Users] DID Trunk Lines and Caller ID Reply-To: [EMAIL PROTECTED] This is a multi-part message in MIME format. --=_NextPart_000_01C0_01C3D365.D94BFFA0 Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit I have an installation which is currenly using 14 DID Trunk Lines. I need to be able to use Caller ID information and currently it is not available on these lines. Is there another way to access this information? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming calls randomly hangup and blank calls
Hi, I have little problem and it is so embracing when u r talking to some one and line get hang-up. When some one calls from out of state or out of country my calls gets randomly hang-up with in few seconds and it happens with most of the calls. It's happening randomly I got few calls, which worked. I have also observer that with quite few call I can't hear the person and person can hear me. If they try to leave the voicemail and this problem occurs then, voicemail will hang-up on user, because it think there is no voice, so we get half voicemail of that person or blank voicemail. I don't know weather this two thinks are related to each other problems or I have two different problems. I have removed busydetect=yes and callprogress=yes then also same thing, tried to set busycount=6 but no luck. Has any one facing this kind of problem in * My setup is DID inbound line, incoming only, to Adtran TA750 (FXS-CARD)connecting to * on T1 with t100p card. Outgoing calls are working fine, since it going through normal CO lines and FXO card. Can somebody put there input, if they had faced these problems. Any help is appreciated. Regards, Kekin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID line with Adtran TA750 and T100p
Hello, I new to this, but with the help of mailing lists archives and IRC I am able to build my PBX. Thanks to all who had help me to reach till here. I am stuck at a point where I can't find the solution on mailing lists or even on IRC. I have individual 4 DID (Direct Inward Line) coming from Telco and terminating into TA 750 to FXS card. Many of them told that Phone instrument terminates to FXS card that is correct. When I check with Adtran Tech they told it should be terminated into FXS card, with DPO mode on each DID circuit. I checked this on there website also here it is: How do I extend a DID (Direct Inward Dial) line from the telephone company using voice FXO and FXS cards? This connection seems backwards when compared to the OPX line. Remember with a DID line, the telco acts like the switch (FXO) and the customer supplies the battery (FXS). The customer connects the telco DID line to our FXS card and the DID trunk of the PBX to the FXO card. These voice lines originate from telco and terminate into the PBX. They will never originate from the PBX. When a call comes into the telco's switch with your telephone number, the telco closes a switch connected to your cable pair. This causes loop current to flow from the FXS card. The FXS card sends signal bits across the T1 to the FXO card who then closes his switch causing loop current to flow from the DID interface card on the PBX. The PBX then signals the telco (with a wink) that it is ready for the call. The PBX does this by reversing the battery's polarity. When the Telco sees this wink, the Telco then passes the DNIS digits through the talk path into the PBX. The PBX uses the DNIS digits to route the call to the appropriate phone. The call can terminate from either end. If the person at the PBX hangs up the loop current (from the PBX to the telco) will stop flowing and Telco will return to an idle condition by opening their switch. The call can also be terminated from the telco side if the incoming caller hangs up. When this happens, the telco opens their switch and loop current stops flowing. The PBX then returns to an idle condition. My PBX communicates with TA750 on T1 with T100p card. I don't know what signaling needs to be set in Zaptel.conf and Zapata.conf files, so that once Telco send the signal it will see the PBX and rings that DID extensions. Has anyone had done this? Please let me know, I have to put this machine live by Monday and I stuck where I have no clue what to do. Thanks Regards, KD ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
