similar to: Distinctive ringing on Cisco 79xx

Hi All, I am running into a problem setting the ringtones via _ALERT_INFO on the Cisco 7960 phone. I am using * 1.2.1 and have tried setting the variable to several values. I have also tried setting the phone's software to both 7.5 and 8.2 thinking that it might be a version issue, but with no success. I have examined the packets and do see the ALERT_INFO header being sent, but the

Hi list! I need help with distinctive ring on Cisco 7940 phone. I'm using Asterisk 1.2.5 (I know, I should upgrade) and in dial plan I have: exten => _64X,n,Set(_ALERT_INFO=Chirp2) exten => _64X,n,Dial(SIP/${EXTEN},30,wWtT) On Cisco in Settings => Ring type I have "Chirp1" and "Chirp2". By default phone is ringing sound "Chirp1". For internal calls

Hi Can anyone with distinctive ring on their 7960's possibly post how they've got it to work? I understand that the ALERT_INFO variable is involved but using the examples for the variable value from the WiKi I'm just getting an error message from the Asterisk concole. Thanks in advance. P

Hi, I'm after two very specific ringtones for the 79xx's... A dog barking, and a horse either galloping or neighing. I've tried making the sounds, but for some bizarre reason they're not working. I used to make quite a few ringtones for the 79xx's, but I seem to have forgotten how to do it! And to top things off, I can't even find the documentation on Cisco's site

Hi all, Has anyone gotten custom ring tones to work using ALERT_INFO with the Cisco 7940 SIP phone? I've read the wiki, but just can't get this to work. I'm currently using the 7.2 SIP image. Thanks, Chris

I've lurked for a while, but I think this is one of my first "pleas" for help. I'm having issues where a parked call using the macro below is getting "stuck". Users park the call via a blfxfer key on an Aastra phone. If the call is a blind transfer, it tries to park the call. If it isn't a blind transfer, it tries to unpark the call. Only 2 extensions (2759 and

CVS HEAD/Asterisk 1.2: Is there a way to have the entire extensions.conffile coming from the realtime? It appears that RealTime for the extensions.conf file is on a context by context basis, but you have to create each new context in the extensions.conf file then add a "switch => Realtime" line (then reload). I want to be able to add phones without having to edit any files.

Dear all, Has anyone seen this before and can suggest a solution? I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. Could

Hello, I'm still looking for any ideas on this problem: I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. I have

I apologize for the double post. I am curious as to what the usefullness is of the multiple line appearance feature on Polycom phones. I setup our phones to register one line per extension but I hear the IP501's can do three line appearances. Why and how could this feature be applied? Thanks again all. Kenny ______________________________________________________ Click here to

How do I set each extension to play it's own voicemail prompts? I have vm working in that it plays the standard "person at extension 1234 is not available....." and takes the message. I've recorded seperate .gsm files for each user but can not figure out how to use them. - Gary Edison Information Technologies www.EdisonInfo.com P.O. Box 554

Are there any standards for setting up pbx dialplans for businesses/offices? What I mean is that, which numbers are reserved for a specific use ex. 0 for operator ? Putting Zero for operator in the dialplan seems to be the common practice of businesses. If there is such a standard, * and # are used for what ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a particular reason for doing this unconditionally on both 32- and 64-bits in xenolinux? It would seem to me that this feature should be disabled at most in unprivileged kernels... Thanks, Jan _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com http://lists.xensource.com/xen-devel

Address lookup canonical name asterisk.org. aliases addresses 216.27.40.102 Service scan FTP - 21 Error: TimedOut SMTP - 25 Error: ConnectionRefused HTTP - 80 Error: ConnectionRefused POP3 - 110 Error: TimedOut NNTP - 119 Error: TimedOut digium.com is ok though Address lookup canonical name digium.com. aliases addresses 216.207.245.1 Service scan FTP - 21 Error: TimedOut SMTP - 25

Hi all! I would like to solve some problems: I have a sip provider that lets me make pstn calls after listening some stuff and entering a pin number: 1) How can I make Asterisk enter the pin number? Then wait 1 second and enter the phone number? I have in extensions.conf: exten => 6*,1,Dial,SIP/2002@myprovider,60,tr I have tried with w (like with ZAP channels) but it does not work, nor

Hi, I'm working with this issue for a while, Now I already solve the dialplan issues, but I still have a question about the Callgroups, I read at www.voip-info.org that , there is a 63 limit of callgroups. And I'm wondering why?? and if the 1.2.0beta version supported more than 63 Groups?? (I did'nt find any Changelog for 1.2) or If not There is any unoficial patch for that ? Thanks

In case people on the list don't know it, A Vorbis tester (who's email I've lost! If you're on the list, speak up!) is maintaining a vorbis audio quality test page: http://hem.passagen.se/ingets1/vorbis.htm He's doing us the great service of hunting down samples that break Vorbis (so we don't have to :-) In any case, it turns out that the trumpets in the dr4 sample

I have SAMBA & Winbind working quite well. Users are able to login with DOMAIN+username. With SAMBA sharing using the following config snipit, each user can see their home directory from their Windows machine, but when they try to access the directory, they are prompted for username/password. No combination of username or password seems to work. comment = Home Directories browseable

I am trying to get distinctive ringing going again with these phones, depending on the outside line the call comes in on. I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) But

Anyone have an example XML file for the PAP2T? Cheers, j