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All, I would like to play an announcement to the user on what external line a call came in, right before this call get bridged to this user. How would I go about implementing this in * ? Regards, Hans -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may

...mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. Compuware Limited (company number 1522537) is a company registered in England and Wales whose registered office is at 163 Bath Road, Slough SL1 4AA, Berkshire, United Kingdom. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/...

Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my "features.conf" . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let

...- Brad _____ From: asterisk-users-bounces@lists.digium.com on behalf of David Thomas Sent: Fri 3/17/2006 5:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: DUNDi .... Halfway and CLUSTERING On 3/17/06, Watkins, Bradley <Bradley.Watkins@compuware.com> wrote: > I understand what you're saying now. While I have absolutely no proof of > this, I have to believe that it's something they've solved. I've got > several production systems (since early December of last year) using the > type of cluster that I'...

...- Brad _____ From: asterisk-users-bounces@lists.digium.com on behalf of David Thomas Sent: Fri 3/17/2006 3:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: DUNDi .... Halfway and CLUSTERING On 3/17/06, Watkins, Bradley <Bradley.Watkins@compuware.com> wrote: > Do you mean the peristence of connecting a specific phone to a specific > server? If so, then it's relatively easy. The ldirectord has a persistence > setting that does that. If I'm misunderstanding you, then could you explain > further what you mean? &...

When I downloaded the sip and bootrom from Polycom website, I noticed a file called SoundPointIPWelcome.wav. However, I have no idea where and when it was used. I played the wav file but I have never heard the phone using this wav file before. Does anyone know what it is used for?

...fway and CLUSTERING Brad, How are you able to overcome the Call-ID stickiness problem when loadbalancing with Ultramonky? As I understand it LVS does not properly support SIP in that it doesn't always use the same source path. regards, David On 3/17/06, Watkins, Bradley <Bradley.Watkins@compuware.com> wrote: > At the moment I'm out of the office, but when I return I'll be certain > to do that. Note that my solution is different from what you are > working on with regexten, though I suspect some of the challenges that > I've faced and overcome are not. I'...

...-gate.so.1 => (0xffffe000) libpthread.so.0 => /lib/libpthread.so.0 (0xb7f60000) libc.so.6 => /lib/libc.so.6 (0xb7e2d000) /lib/ld-linux.so.2 (0x80000000) ------------------------------ Message: 3 Date: Tue, 1 Sep 2009 14:55:35 +1000 From: "Lee, John (Sydney)" <John.Lee at compuware.com> Subject: Re: [asterisk-users] Inquiry:Problem with Call Parking To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <1AD5F1954C21244283D099883A2B3AB9FC05D9 at apac-syd-ex001.apac.cpwr.corp> Content-Type: te...

...connection between an Ericsson MD110 and * with analog trunks. Problem with this is, that all CallerID info is send over a serial link (ICU). Is there anyone who knows if there is support for this on * or to find the specification of ICU somewhere? Regards, Roelof Dijkstra Network Engineer EMEA Compuware Europe BV -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please noti...

Why can't people learn to turn off this crap for lists? I don't care if you are on vacation and I'm sure that everyone else on this list doesn't care either. -Matthew ----- Original Message ----- From: "Dijkstra, Roelof" <Roelof.Dijkstra@nl.compuware.com> To: "Matthew Boehm" <mboehm@cytelcom.com> Sent: Thursday, January 06, 2005 9:24 AM Subject: Out of Office AutoReply: [Asterisk-Users] asterisk addson > Currently i'm on holiday. I'm back in the office on the 24th of Jan. > If you have any urgent questions ,...

I did not really spend too much time on looking at call forwarding and wonder if someone could help me. It seems that for setting call forwarding on the Polycom phone itself, only "forward all calls" will work. The other call forward function like "forward if no-answer for n rings" or "forward if busy" does not work at all on the phone. If that is the case, it

We are setting up an office in Malaysia. We contacted Telekom Malaysia and are surprised to be told that ISDN-30 is no longer available. They are yet to give us information of the replacement technology. Does anyone have any experience and information with this? Thanks in advance.

Hi all, My console is no longer responding to commands, I''m using the web-based console running off of a server. I have tried to interrupt R, I have deleted the data and profile files in the user directory, and restarted the server, relogged in, flushed the cache on the browser, but the console is not responding. I even move the .rstudio directory When I try to run the code, it looks

...lite is not able to register to the asterisk server. Is there anything which needs to be tweaked on Asterisk side to get this working? Please help. Thanks, Jagan ------------------------------ Message: 13 Date: Fri, 11 Mar 2005 11:31:29 +0100 From: "Vledder, Hans" <Hans.Vledder@nl.compuware.com> Subject: RE: [Asterisk-Users] Asterisk and USB ISDN controllers ... To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <D913221A882FD31198D90008C75D69090F140249@cwnl-ams-pri01.nl.compuware.co m> Content-Type: text/plai...

I am using Polycom IP600 phone. If I call a phone which has DND (do not disturb) enabled, the message to the caller will be "The person on extension ... is on the phone, please leave a message ...". Is there a way to pick the "person ... not available" message instead?

I setup a number of remote phones on public IPs using the web interface. Now my question is how do I change the default Polycom:456 password via the web interface. Is there a hidden way or does it have to be done via FTP TFTP? Thanks, Steve Totaro

I remember seeing something somewhere that described how I could get SIP header information with Asterisk. It was a command or a variable. Anyone know what it is? Thanks. Doug.

I am testing analog phone and fax machine plugged into Rhino Channel Bank which is connected to TE412P card. This site is in China. I am running RHEL 5, Asterisk 1.4.21.2, Zaptel 1.4.11 and libpri 1.4.4 I ran into a problem which is analog phone can hear dial tone and can make outgoing calls. Another phone (ether internal or external) can call the analog phone ***but the phone does not

I am testing Asterisk 1.4.22, zaptel 1.4.10.1 and libpri 1.4.4 on RHEL5 on DELL PE2950 and using ISDN-10. I thought about cutting over to production tonight when I noticed a serious problem. SIP calls are fine but if I dialed to outside (Dial(Zap/g1)) a few times or someone called in a few times, Asterisk just froze (cannot enter anything on the CLI console) and then even the machine had to be

Hi i have problem with AddQueueMember logic. I need login Agent(Member) in asterisk. use this option: for example: AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13) and now i want to call to this Agent: exten => _1XX,1,Dial(Agent/${EXTEN:1}) call to 113 and asterisk should call to Agent => 13 on interface SIP/ekiga. This doesn't work, How can i do this on Asterisk 1.4(not