info@beprojects.com
2005-May-19 10:08 UTC
[Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail
Has anybody successfully (or I guess unsuccessfully for that matter) implemented Cisco Call Manager and used an * box for voicemail? I checked the wiki and google and I see some references to Call Manager Express and *, but CME is completely different than CM. If anybody has done this or has any insight, it would be appeciated. We are trying to migrate ~ 300 users off of Cisco Unity and onto * for voicemail so that we have more flexibility and a lot lower maintenance costs. Thanks. Peder
John Riek
2005-May-19 19:18 UTC
[Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail
What version of Call Manager are you using? __________________________________ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail
info@beprojects.com
2005-May-20 05:55 UTC
[Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail
3.3.6, so I would either have to use MGCP or H.323. John Riek wrote:> What version of Call Manager are you using? > > > > > __________________________________ > Do you Yahoo!? > Yahoo! Mail - You care about security. So do we. > http://promotions.yahoo.com/new_mail > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > . >
Scott Herrick
2005-May-25 19:39 UTC
[Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail
BUMP It's CM 3.3.6 MAN that would be sweet if * could take the place of Unity! Anybody? :-) info@beprojects.com wrote:> Has anybody successfully (or I guess unsuccessfully for that matter) > implemented Cisco Call Manager and used an * box for voicemail? I > checked the wiki and google and I see some references to Call Manager > Express and *, but CME is completely different than CM. If anybody has > done this or has any insight, it would be appeciated. We are trying to > migrate ~ 300 users off of Cisco Unity and onto * for voicemail so that > we have more flexibility and a lot lower maintenance costs. Thanks. > > Peder > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > . >
Paul Davidson
2005-May-26 05:40 UTC
[Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail
> Date: Thu, 26 May 2005 16:50:02 +1000 > From: Shaun Ewing <sewing@gmail.com> > Subject: Re: [Asterisk-Users] Cisco Call Manager & Asterisk for > Voicemail > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <897da30e0505252350617e4ff3@mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On 5/26/05, Scott Herrick <scott@angvall.com> wrote: > > BUMP > > It's CM 3.3.6 > > > > MAN that would be sweet if * could take the place of Unity! > > > > Anybody? > > > > :-) > > I've got it working with Callmanager 3.3(5) and Asterisk (connected > with chan_oh323). > > Not totally integrated - one still needs to set call forwarding > (busy/no answer) on each extension that needs voicemail, but MWI works > and so does the messages button (eg: on 7960G) to retrieve VM. > > If somebody can tell me how to send a call in Callmanager to (for > example) extension 27000 when 7000 is unavailable by checking the > voicemail box (rather than entering an individual number for each > extension), it'll be perfect. > > I can share my progress so far if it will be beneficial. > > -ShaunYou've done the hard bits. The bad news is that, under CCM, there's really not much in the way of VM configuration. You should set up the VM Pilot stuff to your extension for the Asterisk voicemail- this allows you to click the 'voicemail' box on each extension rather than keying it in- but you still have to touch each extension. You can use their automated tools to make systemwide changes to all extensions- but I don't trust them at all, and I don't think that would help you in this case. I'd love to see how you configured the MWI and how you've set your dialplan- from the way it looks, you're using a different extension for each mailbox. Theoretically, there should be fields on the PDUs from h.323 that show the forwarding number- that's the way Unity does it- and you go into VM for the forwarding number, not for the extension dialed. I'm not sure without playing if any of the h323 channel drivers make the forwarding number available as a channel variable- if they don't, it should be a relatively trivial patch, assuming CCM sends it across (which I'm pretty sure it does- again, time to set some debugs and watch the PDUs). -pbd
trixter http://www.0xdecafbad.com
2005-May-26 20:16 UTC
[Asterisk-Users] International Caller ID?
On Thu, 2005-05-26 at 22:05 -0500, Nathan E. Pralle wrote:> Greetings. > > My wife and I make a LOT of International calls -- mostly to Australia. I > just got Caller ID on my landline, but no numbers/names get passed on > international calls. Is it even possible to get this? Is it a special > request to the phone company, or is it just not possible at all?Most dont offer it on landlines. VoIP is a special case becuase of how its done. Name is also gonna be a problem because of how that is looked up, odds are the telco wont be able to provide that even if they can provide the number. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050526/4bbda343/attachment.pgp
Dan Austin
2005-May-26 20:50 UTC
[Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail
While not widely used in the Cisco product line, SCCP can be used for more than handsets. Newer VoIP gateways support SCCP trunking. SIP in the 4.X series of CCM is a nice addition, but it is rather limited at the moment. G7.11 only, requires a MTP for DTMF, hold and transfer, etc. Dan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul Davidson Sent: Thursday, May 26, 2005 7:59 PM To: Scott Herrick Cc: Asterisk Users Mailing List - Non-Commercial Discussion; Shaun Ewing Subject: Re: [Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail On 5/26/05, Scott Herrick <scott@angvall.com> wrote:> My environment is a standard Cisco Call Mangler VoIP solution that has > reached the capacity of the Unity VM system. The cost of anotherUnity> box is enough to prompt the decision makers to look for othersolutions.> > All the calls come into a Cisco 6509 and then to the CM. > > The more I look at the H.323 the less I like it. Rather, I didn'tknow> how much I didn't know. And I thought SIP was a thick protocol. > > I have not thought this through but would SCCP on the * box help?Might> there be a way to have * "talk" SCCP to CM for VM and MWI? > > Back to the books > ScottMy environment could be a practical clone of yours. Truth be told, under CCM, h.323 is the best option out there, at least until you get to CCM 4.0. I share your trepidation with SIP- and it's not really an option under CCM 3- but SIP and H.323 share many of the same problems regarding NAT traversal, etc. H.323 is more of a logical descendant from ISDN, however, and as such is a little more robust- but it's still pretty cryptic. SCCP will not help at all. CCM uses SCCP only for communication with handsets, and to my knowledge, the protocol was not designed to handle server to server communication. MGCP would be an option, except it's possibly worse than h.323, and Asterisk doesn't yet support it in the way CCM would need to see Asterisk as an MGCP gateway. As an interesting aside, we just learned from Cisco that later generation FXO modules for the router VIC modules support Caller-ID back to Callmanager- but *ONLY* if you define the VIC as an h.323 gateway- MGCP doesn't support passing the fields back to CCM. Store that one away, those of you who are using CCM. MWI, on the other hand, is one area that I think Cisco got right, at least from an implementation standard under CCM. If you want to turn on the MWI, simply place a call to a extension of your choosing, while setting caller ID info for the call to be the extension you want to turn on- CCM does the rest. So, a simple call file will take care of the whole mess- off, on.. that's what Shaun spent the time putting together. It requires at least a signalling handshake between CCM and Asterisk- it actually doesn't need a voice channel setup from Asterisk to CCM, but of course, if you can do signalling, having voice is more than a bonus. The good news? You don't really have to know all that much about h.323 or sip to configure the trunks and get Asterisk and CCM humming to the same tune. You've already got a strong networking environment (6509s are nice iron), so connecting CCM and Asterisk really isn't much of a process- follow Shaun's example to the letter, possibly with my modification if it helps you, and you're golden- once you've got it configured, many other things will start to click. Good luck. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
info@beprojects.com
2005-May-29 14:11 UTC
[Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail
Do you have a sample h323.conf file? I finally got my * and CCM talking to each other, but I had to use a gatekeeper and have them both go through that. Without a GK, they both just sat there like they didn't know what to do. My guess was always that my h323.conf file was wrong, but there is so little useful info on how to set that up that I had no idea what to even try. Thanks. Peder