similar to: Redirect and free the channel

Hi everyone! I'm wondering if anyone on the list had the opportunity to work with an NTT INS1500 ISDN PRI service before. You see, in Japan, if you receive a call that was just forwarded by another number, the call presentation not only includes the caller (ANI) and your number (DNIS), it will also usually include the forwarding number (REDIRECT). Does anybody know how to extract this field

The Asterisk Development Team has announced the release of Asterisk 1.4.36. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.36 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.4.36. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.36 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Hi, I have following problem. Asterisk is connected to ISDN router on BRI interface. ISDN PBX is connected to another channel of BRI interface. Now I'd like to route all incoming calls first to Asterisk and then if caller wants to talk to extension on ISDN PBX then I'd like to route call to another capi channel but free the current one. Is this possible at all or do I need to take 2 capi

Previously incorrectly sent to asterisk-dev list, sorry. I tried today while connected to a Jtec QSIG E1 card, with DAHDISendCallreroutingFacility with the following test dialplan: Extension 4888 is on the Fujitsu [incoming] exten => 8688,1,Answer() exten => 8688,n,Playback(connecting) exten => 8688,n,DAHDISendCallreroutingFacility(4888,8688) exten => 8688,n,Playback(goodbye)

When I use the Asterisk manager interface to redirect a call (Action: Redirect) I get an error with the message "Redirect failed, channel not up." This is especially troubling as it looks like this message was added to the code for the rather recent 1.2.x release. A quick google search implies that I'm not the only one experiencing this problem with 1.2.17, but me and

I am monitoring a channel... I then redirect that channel to a conf with lq as options. When playing back the gsm file I have all recording upto the point of redirect to the conference. How do I CONTINUE to record and not loose anything after redirecting to the conference? After redirecting I have dead air. asterisk 1.4.23. Thanks, Jerry

Hi guys, I've been looking on wiki, but i could find it only for chan_capi: http://www.voip-info.org/wiki/view/Asterisk+PBX+functions In the CAPI channel See Asterisk CAPI channels * Call Deflection (CD) (redirect without answering): Implemented by chan_capi How can i do it with my Softphone Xlite? Any one can help me? I want to redirect a call without answering it. Best regards,

Hello I am going to have a setup like this: One Asterisk-box with two TE121-cards, and an ISDN-30 line. On the other hand, I also have another box with VoiceGuide and Dialogic. As a temporary migration-solution i would like to redirect some of the ISDN30 channels from the Asterisk to the Dialogic-box. How would I do this?

Originally posted on asterisk-dev with no response for 5 days, so posting it to the wider audience now. Asterisk Release 1.6.1.1 Scenario:- 1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and 902 2. Using AMI, 901 is Originated 3. When 901 answers, it is Redirected to an extension "exten => dial,1,Dial(SIP/902)" 4. 902 rings, then answers 5.

Is there a work around for Caller ID information not being persisted when using the CLI or AMI Channel Redirect. A calls B (caller id is displayed), B transfers call to C (no caller id is displayed on phone c). Jacob Miles Software Engineer jacob.e.miles at l-3com.com 903.457.4422 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm looking forward to install an asterisk-server to perform ISDN-to-SIP- bridging. But for the times when I'm not available via SIP I also need call- forwarding-features. Afaik it is possible to directly forward a call on ISDN instead of opening a second ISDN-channel additional to the incoming-ISDN- channel and forward the call "by hand", right? What if I need features

Hi I am working on deploying voip for my company and would like to seek some advice on the number of E1 lines we need to rent. Our telco told us that there can be at most 30 concurrent channels on an E1 line. Typically, what is the maximum number of DIDs that we can allocate to that E1 line before users get frequent "all lines are busy"? We are running a support center with mostly

I haven't seen anything about this is the archives, so here we go. Sorry, its a long one. My setup: Dual Xeon 2.4 GHz. One TE410P card. Span 1 populated with a PRI. Span's 2-4 empty. zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us I'm currently having a problem where one of our PRI channels seems to keep a call even after it is hung up. We have

Does anyone know how I can send only calls that have been forwarded or redirected out on a specific channel? My PRI provider (BellSouth/AT&T) does not let me set the outgoing caller ID to anything other than our DID numbers, so when a user FWD's his Polycom phone to his mobile, it looks like his desk is calling his mobile phone rather than the Caller ID of the actual caller. I am

Hi, I have a Linksys SPA 3102 using as ATA. The call routing is : Phone -> PSTN -> SPA 3102 -> SIP Proxy -> Asterisk The problem I am having is that when the phone hangs up, SPA 3102 can't detect it and relay the CANCEL message. Is this problem with my SPA 3102 config or it just works like that by default? Thanks in advance for your help. Regards, Mark -------------- next

Hi, Is it possible to connect Asterisk with a mobile base station to handle call switching? What kind of protocol will I need to use to convert to sip? Any pointer or info will be greatly appreciated. Best Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081119/e74ef6b1/attachment.htm

Hi, Since a week ago I am trying to get chan_misdn working with asterisk 1.4.19, using HFC based ISDN card on Linux 2.6.22. My setup is done as detailed on wiki and FAQ. * mISDN and miSDNusers are 1.1.7.2, unpacked, build and installed. After installation and misdn-init, I have this: aragorn:root/pts/1: # misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) ->

Hello everyone. My land-line provider (Romtelecom) has a very nice offer for ISDN. All in all they offer me a digital land-line with 1 base number + 2 MSN's and that would make a grate addition to my full-time home office. Romtelecom say they're providing EURO-ISDN and the line is compatible with any euro-isdn compliant equipment. They say they'll install a NT at my office and this

Hi, I'm trying to get capiECT working. I'd like to transfer call to another ISDN router connected extension and free channel from router to Asterisk. I get incoming call on CAPI and would liek to transfer it to dialed local extension - 400 in this case: [outbound-capi-local] exten => _4XX,1,NoOp(Transferring to local PBX ISDN number ${EXTEN} on msn CAPI/${CALLERIDNUM}) exten =>