Re: [Asterisk-Users] Call waiting
I'm having the same problem. I'm using a SIP phone (not sure if that matters or not). I tried doing a hook-flash, then dialing *0, and I just get a reorder message. I have threewaycalling=yes in my zapata.conf. I actually gave up a few weeks ago looking for the answer to this, but it appears that other people are having the same problem. Can this be done with a SIP phone? Swannie On Tuesday 23 March 2004 8:14 am, Ed Rubright wrote: > You need to hook-flash first...then dial *0. > > Ed > > _ > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Simon Brown > Sent: Tuesday, March 23, 2004 5:20 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Call waiting > > > I have "threewaycalling=yes" in my zapata.conf file. > When I hear the call waiting indicator I dial *0# on the phone, but it is > ignored. > > How can I get it to work? > > Simon > > _ > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Ed Rubright > Sent: Tuesday, 23 March 2004 14:46 > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Call waiting > > > You need to send a hook flash to the PSTN line attached to the X100P. This > is done by: > > hook-flash then *0 > > You'll need to have 3way calling enabled for it to work. > > NOTE: This info can be found in the wiki pages: > http://www.voip-ino.org/wiki-Asterisk > > Ed > > _ > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Simon Brown > Sent: Monday, March 22, 2004 7:28 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Call waiting > > > I have a problem with call waiting which I cannot find an answer for in any > of the information that I have searched through online. I cannot find > anything in the list archives either. > > My * server is connected to a PSTN line with an X100P. I have a call > waiting service on this PSTN line which requires a hook flash signal to > swap calls. > > I have Cisco 7940 phones working with my * server. When on an outside call > and another outside call comes in, I can hear the call waiting indicator, > but cannot find any way of swapping calls (sending a hook flash signal > through the X100P). > > Any help would be greatly appreciated. > > Simon Brown > - > This mail was content checked for malicious code and viruses > by GFI MailSecurity. ___ 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
Re: [Asterisk-Users] XML Phone book software.
Cool, thanks! I must have missed this in my searching. Thanks, Brian On Friday 12 March 2004 7:12 am, stan wrote: > On Thu, Mar 11, 2004 at 04:06:41PM -0600, Brian R. Swan wrote: > > I'm looking into writing a some phone book XML/PHP software for my Cisco > > phones. Specifically, I'd like to be able to use a web interface (on the > > computer) to maintain a contact list, and then dial from it on the phone. > > Maybe using MySql on the back end or something (to be determined). > > Before I start, and duplicate something else that exists, I wanted to see > > if anyone has heard of software like that? Searches of Sourceforge, > > Freshmeat, and Google didn't turn up much or anything. > > see the cmxml software section of > http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx > > ___ > 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 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] XML Phone book software.
Hi gang, I'm looking into writing a some phone book XML/PHP software for my Cisco phones. Specifically, I'd like to be able to use a web interface (on the computer) to maintain a contact list, and then dial from it on the phone. Maybe using MySql on the back end or something (to be determined). Before I start, and duplicate something else that exists, I wanted to see if anyone has heard of software like that? Searches of Sourceforge, Freshmeat, and Google didn't turn up much or anything. Thanks! Brian ___ 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] ZAP/Call Waiting...
Hi all, ...on to my next issue with Asterisk that I'm trying to get resolved. I have a Cisco SIP phone (7960), and I'm using Vonage through their Motorola box and an X100P. All the basics work well. The problem I'm having is that I'm not sure how to answer call waiting on the Vonage line from the Cisco phone. I see numerous ways to send a flash to the X100P in the docs, but I'm not sure how to "glue" them all together and make something that's easy to use from the cisco phone. Also, it would be nice to have the call waiting caller ID work as well (if possible). Thanks in advance for any help. Swannie ___ 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
Re: [Asterisk-Users] Call roll-over question...
Hi all,
Thanks everyone who replied for their help. I wanted to post and describe my
"final" config in case anyone in the future wants to accomplish something
similar.
Here's my "Standard extension" macro:
exten => s,1,Dial(${ARG2},15)
exten => s,2,Voicemail(u${ARG1})
exten => s,3,Hangup
exten => s,102,Dial(${ARG3},15)
exten => s,103,Voicemail(b${ARG1})
exten => s,104,Hangup
exten => s,203,Voicemail(b${ARG1})
exten => s,204,Hangup
And the line that calls the macro.
exten => 2001,1,Macro(stdexten,2001,SIP/2001,SIP/3001)
So, with call waiting turned off on the Cisco phone, I can call 2001, and if
it's busy it will roll over to 3001. If they're both busy is will go to
Voicemail. As a side note, I opted to not make the 3001 extension direct
dialable (is that a word?). So, any calls to 3001 will just get a re-order,
however, calls can still be placed from 3001.
Again, thanks everyone for their help with my config!
Swannie
On Saturday 06 March 2004 8:54 pm, Chris A. Icide wrote:
> At 03:52 PM 3/6/2004, you wrote:
> >try this in extentions.conf, it should do what you want...
> >
> >exten => 2001,1,ChanIsAvail(SIP/2001&SIP/3001)
> >exten => 2001,2,SubString,ToDial=${AVAILCHAN}|0|8
> >exten => 2001,3,Dial(${ToDial},20)
> >exten => 2001,4,Voicemail(u2001)
> >exten => 2001,5,Hangup
> >exten => 2001,104,Voicemail(b2001)
> >exten => 2001,105,Hangup
> >
> >exten => 3001,1,ChanIsAvail(SIP/3001&SIP/2001)
> >exten => 3001,2,SubString,ToDial=${AVAILCHAN}|0|8
> >exten => 3001,3,Dial(${ToDial},20)
> >exten => 3001,4,Voicemail(u3001)
> >exten => 3001,5,Hangup
> >exten => 3001,104,Voicemail(b3001)
> >exten => 3001,105,Hangup
>
> Another example that functions as I think you want is below. I like the
> example above, but if you want to stay away from variables for some reason,
> the function below does the same as above for calling 2001. You can copy
> and reverse the numbers for 3001 as well.
>
> exten => 2001,1,Dial(SIP/2001,20) ; Ring first line for 20 seconds if it's
> not in use
> exten => 2001,2,Voicemail(u2001) ; Line 1 rang for 20 seconds, no one
> answered, send to VM as unavailable
> exten => 2001,3,Hangup ; I always terminate a logical set of steps with
> hangup, just in case..
> exten => 2001,102,Dial(SIP/3001,20) ; Line 2001 was busy, try dialing line
> 3001.
> exten => 2001,103,Voicemail(b2001) ; No one answered 3001, but 2001 is
> busy, lets tell the caller we are on the phone
> exten => 2001,104,Hangup ; Just in case...
> exten => 2001,203,Voicemail(b2001) ; Line 2001 AND 3001 were busy, maybe
> I'm calling myself? Let the caller know we are on the phone
> exten => 2001,204,Hangup ; just in case...
>
> This should work just fine. I'm not sure how high priority can go before
> something goes poof. But two +101 jumps definitely work. I'll have to see
> how many times you can jump and how high a priority you can have.
>
> -Chris
>
>
> ___
> 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 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
Re: [Asterisk-Users] Call roll-over question...
Hmm, that would work, but then wouldn't both lines ring at the same time on my phone? I'd like to see if I can get it to roll over like described, but this is certainly an alternative. Thanks for the reply! Brian On Friday 05 March 2004 8:50 pm, Chris A. Icide wrote: > At 05:04 PM 3/5/2004, you wrote: > >I have another question for the group. I'm trying to make the following > >happen on my Cisco phone: > > > >I have two lines configured, 2001 and 3001. If I'm talking on 2001 and > > > > Try this > > exten => 2001,1,Dial(SIP/2001&SIP/3001,20) > > This will ring them both at the same time for 20 seconds > > >...any ideas? > > ___ > 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 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] Call roll-over question...
I have another question for the group. I'm trying to make the following happen on my Cisco phone: I have two lines configured, 2001 and 3001. If I'm talking on 2001 and someone tries to call me on 2001 I'd like the call to roll over to 3001 and then if I don't answer, it goes to Voice mail. I was able to accomplish this using the following sequence in extensions.conf (I'm doing this from memory, so I hope I got it right). exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Dial(SIP/3001,20) exten => 2001,3,Voicemail(u2001) Now, while this works if I'm talking on 2001, the obvious problem is that if none of the extensions are busy it will ring 2001 for 4 ring, then head over to 3001 for 4 rings until going to voice mail. So, I then tried the following: exten => 2001,1,ChanIsAvail(SIP/2001) exten => 2001,2,Dial(SIP/2001,20) exten => 2001,3,Voicemail(u2001) exten => 2001,102,Transfer(3001) exten => 2001,203,Voicemail(2001) exten => 3001,1,Dial(SIP/3001,20) exten => 3001,2,Voicemail(u2001) Which seems (to my newbie eyes) that it should work, but... it doesn't. If I pickup 2001 and call from another extension it goes straight to voice mail. Both extensions (2001,3001) work on their own, so I'm certain that they are configured correctly. Also, I have "call waiting" shut off on the cisco phone (so it should reject the SIP call to 2001 as busy). ...any ideas? Thanks! Swannie ___ 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] Newbie questions, call waiting/700 calling/etc...
Hi gang, I've just set up my Asterisk server with a X100P (talking to a Vonage Motorola do-dad), and a Cisco IP7960 SIP phone. All is working quite will with outbound and inbound calling. However, I have a few questions. First, regarding call waiting on Vonage/X100P, how do I "click over" to the call from my Cisco phone? I searched the archives and docs but can't seem to find any information on this. Along those same lines, is there any way to get call waiting caller ID working with this as well? I have what I believe are the appropriate lines in my Zapata.conf file (usecallerid=yes, callerid=asreceived, callwaiting=yes, calleridcallwaiting=yes), but it doesn't seem to work. Second, I got myself set up with an IAXtel number, I was wondering if there are any 700 test numbers out there that will read back the calling number, do an echo test, request a call back, etc. Mostly I want this for testing, and don't want to disturb any 700 number users. :) Third, are there any VoIP providers that I can have Asterisk talk to natively (i.e. via IAX or SIP, not the way I have Vonage set up now)? I'd be looking for a Chicago land number (630 specifically). I looked at VoicePulse, but they don't have any local numbers. Thanks for your help! Brian ___ 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
