[asterisk-users] Checking status of a cell phone
Hi, I am looking for a way to check the status of a cell phone. Found one way that worked for me and would like to have some feedback or suggestion of improvments. Below example is for a “Swedish” cell phone, dont know if it works in the same way for other countries. I could define “redirecting” numbers for 3 traffic cases when u dial my mobile (073-302 59 75): NOT_INUSE call forward to A INUSE call forward to B in my case 010-602 4975 UNAVAILABLE call forward to C in my case 010-602 4976 From manager: Action: Originate\r\nChannel: OOH323/00733025975@Avaya\r\nExten: 0106024000\r\nContext: inputinterior.se\r\nPriority: 1\r\nTimeout: 1000\r\nCallerID: 106024000\r\n\r\n DBPut\r\nFamily: DS\r\nKey: 0733025975\r\nVal: NOT_INUSE\r\n\r\n Wait a second... Action: DBGet\r\nFamily: DS\r\nKey: 0733025975\r\n\r\n In the dialplan: exten = 0106024975,1,Set(DB(DS/0733025975)=INUSE) exten = 0106024975,n,Hangup() exten = 0106024976,1,Set(DB(DS/0733025975)=UNAVAILABLE) exten = 0106024976,n,Hangup() Just a short call to my cell phone, to se if i get anything back, my cell phone doesn’t even ring. Wait a second if the call is redirected, then check to se if the status has changed from NOT_INUSE to something else. Dont know if it is a stupid idea, but it worked on my cell phone, and the switchboard girls was very happy to be able “to ask” my cell phone “what I am doing” Most of the day i am INUSE so they dont need to transfer calls to me ehen they know I am INUSE. Ofc there is some delay from asking to getting the answer, but as the girls said, we could live with the delay, 2seconds compared to be “blind” is nothing.wlEmoticon-smile[1].png-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pbx.c: We were unable to say the number
Hi Mohammad, which application do you use to say the number [n]? you can use *SayAlpha()*application to say the number you want. Please read link: *http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SayAlpha* for more infomation. Phuonghd On Sat, Mar 26, 2011 at 11:15 AM, Mohammad Khan beepl...@gmail.com wrote: Hello, Occasionally, I get the following warning in my asterisk log, pbx.c: We were unable to say the number [n], is it too large? n is two or one digit number, which doesn't look like large to me! Could anybody please tell more about this warning, like in what scenario I may have this warning. Thanks, Mohammad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Checking status of a cell phone
On Sat, 26 Mar 2011 10:50:19 +0100, magnu...@inputinterior.se wrote: I am looking for a way to check the status of a cell phone. Found one way that worked for me and would like to have some feedback or suggestion of improvments. I'd like to check I understood: Your Asterisk server is connected to a landline and can call your cellephone (073-302 59 75). When a call comes in from the landline, Asterisk checks whether your cellphone is available and redirects the call; If not available, it calls a landline number (010-602 4975). If this landline number is not available, it tries a third number (010-602 4976)? Is the AMI code below enough to check if the cellphone is available/in-use? Action: Originate Channel: OOH323/00733025975@Avaya\r\nExten: 0106024000 Context: inputinterior.se Priority: 1 Timeout: 1000 CallerID: 106024000 DBPut Family: DS Key: 0733025975 Val: NOT_INUSE -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the most stable version of asterisk?
1.2 is not active either. Both are solid. I am loving SNOM phones and OpenVPN software. Only port(s) open is what I assign to OpenVPN. CallWeaver was way ahead of asterisk at the time but you are right it died. Generally, the newer, the worse. 1.2 was very solid except for a few strange things that could be worked around. Newer versions have are like Fedora Core (or FC X) you are just testing beta software for Digium's commercial products. Thanks, Steve Totaro On Fri, Mar 25, 2011 at 10:25 AM, Douglas Mortensen d...@impalanetworks.com wrote: Based on the following URL, it seems that CallWeaver may not still be an active project?? http://www.callweaver.org/blog/20 From a security standpoint, I would usually expect it is safer to be with an active project, than a dead one. Otherwise who is going to patch vulnerabilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stot...@totarotechnologies.com] Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What is the most stable version of asterisk? On Wed, Mar 23, 2011 at 12:58 PM, Douglas Mortensen d...@impalanetworks.com wrote: 1.2? 1.4? 1.6? 1.8? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 Callweaver? http://www.voip-info.org/wiki/view/CallWeaver. I believe they forked somewhere in the 1.2 release. Many features ahead of Asterisk. Although I didn't see anything on FreeSwitch stating anything anything about deadlocking, I know that was one of the main reasons for BKW, as seasoned asterisk developer and folks to start from scratch. That and the hybrid dual license in Asterisk. http://www.sofaswitch.org/docs/How%20does%20FreeSWITCH%20compare%20to%20Asterisk.pdf Read the whole piece. I know it isn't Asterisk but BKW who contributed and I believe is still helping Asterisk Besides, I feel that FreeSwitch is the most stable. I like 1.2 so I went with Callweaver for many installations. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Streaming Hold Music
On Thu, Mar 24, 2011 at 03:08:11PM -0700, Chris Davis wrote: Does anyone have a good solution to stream hold music to the Asterisk/FreePBX server? I currently have setup WinAMP using ShoutCast but it appears to a very touch and go stream, being that it seems to periodically drop the connection. Stream from where? Why would you need WinAMP in the mix? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Checking status of a cell phone
Setup as below: Celluar Network - E1 - Avaya - OOH323 - Asterisk It works like this, some1 (we can call her Åsa) wants to know if i am avaiable (my cell phone 073-302 59 75 is NOT_INUSE) She have a web-app (just a simple form), where she enter my extension and hits enter. The web-app originates the call as i wrote and waits for the status then ofc presents it to Åsa. I was writing the app (probably the worst written code i have done so dont ask me to post it) late thursday and let Åsa use it on Friday. And yes, the AMI code was enough, everytime she should transfer a call to me or just call me , she used the web-app first, and she was very happy. When she saw that I was INUSE she sent me a mail that mr X has been looking for me, i got the mail while I was talking in the phone so I know that she used the web-app to determine my status, not just transfering the call. -Ursprungligt meddelande- From: Gilles Sent: Saturday, March 26, 2011 11:37 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Checking status of a cell phone On Sat, 26 Mar 2011 10:50:19 +0100, magnu...@inputinterior.se wrote: I am looking for a way to check the status of a cell phone. Found one way that worked for me and would like to have some feedback or suggestion of improvments. I'd like to check I understood: Your Asterisk server is connected to a landline and can call your cellephone (073-302 59 75). When a call comes in from the landline, Asterisk checks whether your cellphone is available and redirects the call; If not available, it calls a landline number (010-602 4975). If this landline number is not available, it tries a third number (010-602 4976)? Is the AMI code below enough to check if the cellphone is available/in-use? Action: Originate Channel: OOH323/00733025975@Avaya\r\nExten: 0106024000 Context: inputinterior.se Priority: 1 Timeout: 1000 CallerID: 106024000 DBPut Family: DS Key: 0733025975 Val: NOT_INUSE -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pbx.c: We were unable to say the number
On Fri, Mar 25, 2011 at 11:15 PM, Mohammad Khan beepl...@gmail.com wrote: Hello, Occasionally, I get the following warning in my asterisk log, pbx.c: We were unable to say the number [n], is it too large? n is two or one digit number, which doesn't look like large to me! Could anybody please tell more about this warning, like in what scenario I may have this warning. Thanks, Mohammad Please post the relevant context that is being executed, that'll give us not only the actual application, but more info as to how it's being passed. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisks with ss7 problem
Hi, I am trying to set up asterisk with ss7. Whenever I try to load module chan_dahdi.so, I get the error [Mar 26 17:33:27] ERROR[10437]: chan_dahdi.c:10458 mkintf: Unable to find linkset -1 I have compiled dahdi, libss7, asterisks (am using asterisk 1.6) in that order. Have already set signalling to ss7 in dahdi_channels.conf How do I sort this out? Thanks for your help in advance. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisks with ss7 problem
Hi Peter, Check the configuration in chan_dahdi.conf file, its properly matching the cic and channels with system.conf Thanks Vinod Dharashive Sent from BlackBerry® on Airtel -Original Message- From: Otandeka Simon Peter sotand...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Sat, 26 Mar 2011 17:36:29 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisks with ss7 problem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pbx.c: We were unable to say the number
I am using asterisk 1.4.38 I am getting this warning occasionally when executing SayNumber in a macro with argument which is less than 100. On Sat, Mar 26, 2011 at 11:03 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Fri, Mar 25, 2011 at 11:15 PM, Mohammad Khan beepl...@gmail.comwrote: Hello, Occasionally, I get the following warning in my asterisk log, pbx.c: We were unable to say the number [n], is it too large? n is two or one digit number, which doesn't look like large to me! Could anybody please tell more about this warning, like in what scenario I may have this warning. Thanks, Mohammad Please post the relevant context that is being executed, that'll give us not only the actual application, but more info as to how it's being passed. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pbx.c: We were unable to say the number
Again, the relevant dialplan code is important. It is quite possible that there's an issue with the dialplan code that you (as the person who's dealing with the issue) may have missed. It happens all the time. On Sat, Mar 26, 2011 at 1:25 PM, Mohammad Khan beepl...@gmail.com wrote: I am using asterisk 1.4.38 I am getting this warning occasionally when executing SayNumber in a macro with argument which is less than 100. On Sat, Mar 26, 2011 at 11:03 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Fri, Mar 25, 2011 at 11:15 PM, Mohammad Khan beepl...@gmail.comwrote: Hello, Occasionally, I get the following warning in my asterisk log, pbx.c: We were unable to say the number [n], is it too large? n is two or one digit number, which doesn't look like large to me! Could anybody please tell more about this warning, like in what scenario I may have this warning. Thanks, Mohammad Please post the relevant context that is being executed, that'll give us not only the actual application, but more info as to how it's being passed. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spa8000 t38 faxing
Perhaps this will help. I have a SPA8800 which has 4 x FXS 4 x FXO ports. It took me some time to produce a working configuration. In Asterisk I have the following where 904 is the extension of the fax-modem and itsp is you VoIP Service Provider. sip.conf [general] . . faxdetect=no t38pt_udptl=yes,redundancy,maxdatagram=400 . . [904] ; Cisco SPA8800 FXS Port 4 ; Analogue FAX Modem attached type=friend defaultuser=904 secret=secret call-limit=2 qualify=yes canreinvite=no directmedia=no directrtpsetup=no ignoresdpversion=yes transport=udp,tcp host=dynamic context=your_context faxdetect=no . . [itsp] . . faxdetect=yes ignoresdpversion=yes . . I am including information from my SPA8800 for one of the FXS ports I have a Fax Modem attached to, the key to getting it to work I believe is the FAX Tone Detect Mode. Audio Configuration Preferred Codec: G711a Second Preferred Codec: Unspecified Third Preferred Codec: UnspecifiedUse Pref Codec Only: no Silence Supp Enable: yes Silence Threshold: medium G729a Enable: no Echo Canc Enable: yes G723 Enable: no Echo Canc Adapt Enable: yes G726-16 Enable: no Echo Supp Enable: yes G726-24 Enable: no FAX CED Detect Enable: yes G726-32 Enable: no FAX CNG Detect Enable: yes G726-40 Enable: no FAX Passthru Codec: G711a DTMF Process INFO: yes FAX Codec Symmetric: yes DTMF Process AVT: yes FAX Passthru Method: ReINVITE DTMF Tx Method: AVT DTMF Tx Mode: Strict DTMF Tx Strict Hold Off Time: 40FAX Process NSE: no Hook Flash Tx Method: None FAX Disable ECAN: no Release Unused Codec: yes FAX Enable T38: yes FAX T38 Redundancy: 1 FAX Tone Detect Mode: callee only Symmetric RTP: yes Supplementary Service Settings CW Setting: noBlock CID Setting: no Block ANC Setting: noDND Setting: no CID Setting: yes CWCID Setting: yes Dist Ring Setting: yesSecure Call Setting: no Message Waiting: noAccept Media Loopback Request: automatic Media Loopback Mode: sourceMedia Loopback Type: media Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
