Hi..... I have just successfully setup Asterisk with 2 Cisco 7940 phones (converted for SIP) and a SIP softphone on a W2K box.....and it all seems to work very well.....to those who wrote this software, it is really cool. Anyway, I am new to this software, and I have a lot of questions which I am hoping someone on the mailing list might be able to answer for me.....I am basically trying to get an idea of how/what I can do with Asterisk that I am already doing with our existing phone system.... Sorry about the length of the mail....the docs don't seem to cover some of the topics below..... Thanx in advance for any help. Chris. * We currently have a Cisco IP telephony system (using their CallManager).....am I right in saying that Cisco phones using Skinny will not work with asterisk? Is it ever likely too? * When we connect and power on a Cisco 79X0 phone for the first time, it automatically registers with the CallManager and is assigned a temporary number. We then do into the CallManager admin interface and assign it to its owner, give it its permanent number etc. Among the things which happen are the TFTP files for the phone (eg: SEP<mac_address>.cnf) get created as part of the automatic registration. When I converted the phones to SIP, I had to manually create config files for each phone (SIP<mac_address>.cnf etc.).....is there any way I can have this happen automatically? * We have an 8 port E1 card in a Cisco 6509 which takes our main phone trunk to the public network. Can we connect an Asterisk PBX server with an E1 card to this? If so, could we then connect the Asterisk PBX to the callmanager? (Perhaps with another extension range).....and if so, how? * .....or could we connect Asterisk to the 6509 over IP and so make it part of the main phone system? * We have a Nortel Meridian PBX on our other campus which is connected to our IP telephony system via an E1 link to a Cisco Vg200 voice H.323 gateway.......would there be any way to point asterisk at this gateway and make it part of our main phone system that way? ....again if so how? _________________________________________________________________ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail
>Hi..... > >I have just successfully setup Asterisk with 2 Cisco 7940 phones >(converted for SIP) and a SIP softphone on a W2K box.....and it all >seems to work very well.....to those who wrote this software, it is >really cool. > >Anyway, I am new to this software, and I have a lot of questions >which I am hoping someone on the mailing list might be able to >answer for me.....I am basically trying to get an idea of how/what >I can do with Asterisk that I am already doing with our existing >phone system.... > >Sorry about the length of the mail....the docs don't seem to cover >some of the topics below..... > >Thanx in advance for any help. > >Chris. > > >* We currently have a Cisco IP telephony system (using their >CallManager).....am I right in saying that Cisco phones using Skinny >will not work with asterisk? Is it ever likely too?Skinny is not included as a channel in Asterisk at this time. There are reports of a Skinny channel well into development - see http://www.sf.net/projects/sccp and we await further testing.>* When we connect and power on a Cisco 79X0 phone for the first >time, it automatically registers with the CallManager and is >assigned a temporary number. We then do into the CallManager admin >interface and assign it to its owner, give it its permanent number >etc. Among the things which happen are the TFTP files for the phone >(eg: SEP<mac_address>.cnf) get created as part of the automatic >registration. When I converted the phones to SIP, I had to manually >create config files for each phone (SIP<mac_address>.cnf >etc.).....is there any way I can have this happen automatically?Yes and no. You still will have to create a file called SIP<mac address>.cnf which contains the "extensions" that you expect the phone to use. However, if you have an RFC compliant DHCP server, you should be able to make everything happen automatically except for the generation of that extension. There are almost no hooks between any of the very sophisticated Cisco configuration files and Asterisk; they are _separate_ systems. Asterisk simply deals with SIP devices and their SIP transactions - Asterisk does _not_ configure SIP devices, and Asterisk is not Cisco-specific in any treatment of SIP transactions. I seem to recall that there is a Cisco 79xx administration tool in the http://www.vovida.org/ pages somewhere.>* We have an 8 port E1 card in a Cisco 6509 which takes our main >phone trunk to the public network. Can we connect an Asterisk PBX >server with an E1 card to this? If so, could we then connect the >Asterisk PBX to the callmanager? (Perhaps with another extension >range).....and if so, how?Yes. The manual should explain further details.>* .....or could we connect Asterisk to the 6509 over IP and so make >it part of the main phone system?I don't know. Does the 6509 talk SIP?>* We have a Nortel Meridian PBX on our other campus which is >connected to our IP telephony system via an E1 link to a Cisco Vg200 >voice H.323 gateway.......would there be any way to point asterisk >at this gateway and make it part of our main phone system that way? >....again if so how?Yes. That's too complex to explain adequately here, but you should try setting it up to answer the question yourself. JT
>>* .....or could we connect Asterisk to the 6509 over IP and so make >>it part of the main phone system?>I don't know. Does the 6509 talk SIP?It doesn't appear to. I would love to be wrong. It does support MGCP, though.
Dylan VanHerpen
2003-Jun-20 14:10 UTC
[Asterisk-Users] Can Zaptel cards be used with other Linux apps?
Can Zaptel T1/E1 cards also be used with other Linux apps (for instance as the WAN interface on a Linux based router, or with Bayonne, Vocal)? Thks, Dylan.
Steven Critchfield
2003-Jun-20 14:21 UTC
[Asterisk-Users] Can Zaptel cards be used with other Linux apps?
On Fri, 2003-06-20 at 16:10, Dylan VanHerpen wrote:> Can Zaptel T1/E1 cards also be used with other Linux apps (for instance > as the WAN interface on a Linux based router, or with Bayonne, Vocal)?As a router, absolutely. As a interface for other voice products, well you will probably have to do some programming there, but there is now technical reason stopping it from happening. Just beware the licenses. -- Steven Critchfield <critch@basesys.com>
Thanx for the info....unfortunately, I think we would need an Communications Media Module....which we don't have.... Chris.>From: <tim.mcqueen@qualisys.biz> >Reply-To: asterisk-users@lists.digium.com >To: <asterisk-users@lists.digium.com> >Subject: RE: [Asterisk-Users] Newbie questions..... >Date: Fri, 20 Jun 2003 15:21:53 -0500 > > >>* .....or could we connect Asterisk to the 6509 over IP and so make > >>it part of the main phone system? > > >I don't know. Does the 6509 talk SIP? > >It doesn't appear to. I would love to be wrong. It does support MGCP, >though. > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users_________________________________________________________________ The new MSN 8: smart spam protection and 2 months FREE* http://join.msn.com/?page=features/junkmail
Hi.... Thanx for the info.....sorry to hassle you, but I have follow on questions below.....>I seem to recall that there is a Cisco 79xx administration tool in the >http://www.vovida.org/ pages somewhere.Had a look at this.....this tool will certainly make managing Cisco SIP phones easier....thanx....>Yes. The manual should explain further details.I am not a voice comms expert and our Cisco IP Telephony system was installed by an outside company.... Are you referring to the Asterisk manual here? Just looking at the manual, it seems to me that I could use a Zaptel E1 card and configure the zapata.conf file appropriately......am I correct? If so, I would just need to learn how to configure an E1 port on the 6509.....and configure the CallManager to know where Asterisk is in the number range....>>* We have a Nortel Meridian PBX on our other campus which is connected to >>our IP telephony system via an E1 link to a Cisco Vg200 voice H.323 >>gateway.......would there be any way to point asterisk at this gateway and >>make it part of our main phone system that way? ....again if so how? > >Yes. That's too complex to explain adequately here, but you should try >setting it up to answer the question yourself.Again, I am at a bit of a loss since I am not a voice comms expert....where would I begin in Asterisk...is there a H.323 channel (as there is for SIP) in Asterisk?...or is this a silly question.....? I don't see any mention of H.323 in the conf files.... ....and lastly, one further question.....I got voicemail working, but the red light on the Cisco 79X0 phone doesn't light when a voice mail is waiting.....is there a way to enable this? Thanx very much for the info, and thanx in advance for any further info.... Chris. _________________________________________________________________ MSN 8 helps eliminate e-mail viruses. Get 2 months FREE*. http://join.msn.com/?page=features/virus
Hi Chris I've done a lot of things with Cisco AVVID solutions in the past.> CallManager).....am I right in saying that Cisco phones using > Skinny will > not work with asterisk? Is it ever likely too?Cisco own Skinny Protocol is not supported directly. But Cisco SE always told me that Skinny is a subset of H.323. But I would'nt count on it to be functioning correctly.> * We have an 8 port E1 card in a Cisco 6509 which takes our > main phone trunk to the public network. Can we connect an > Asterisk PBX server with an E1 card to this? If so, could > we then connect the Asterisk PBX to the callmanager? > (Perhaps with another extension range).....and if so, how?Yes, it's possible. I've done it to trombone calls through a Operator system (Trio) and to interconnect with both Ericsson, Nortel PBXs and iPBXs. On CM you set it up as a trunk line, create a route map which forward all calls to that specific E1 / T1 port on the 6608, which are connected to the Asterisk Pri port. On the Asterisk you do the same. Beware that top-down, bottom-up is opposite on the 2 system. Then you should be able to get it working. There could be a few minor adjustments which needs to be done on the CCM - but trying to recall the configuration Page from memory is'nt that easy :-) But it was actually very few steps in getting it to interconnect with Nortel Meridians and Ericsson MD-110 through PRI trunks. But you actually dont need it. You can specify H.323 trunks/endpoints/zones in CCM where you can specify The IP address and numbering plan of the Asterisk system. This way would be better because you have to apply codecs (involve DSP) four times using the 6608 board. This would probably give very long delays more than 300 ms depending on the payload.> * .....or could we connect Asterisk to the 6509 over IP and > so make it part of the main phone system?Already answered - But asterisk can not directly communicate with 6608 on the 6509. All out going calls using the 6608 has to go through the Callmanager.> * We have a Nortel Meridian PBX on our other campus which is > connected to > our IP telephony system via an E1 link to a Cisco Vg200 voice H.323 > gateway.......would there be any way to point asterisk at > this gateway and > make it part of our main phone system that way? ....again if so how?Hm.... This one is a bit difficult to answer. As I recall vg200 is a limited voice gateway which only can be used by Cisco Callmanager. Which makes your scenario to be : Meridian -> PRI trunk -> VG200 -> Router > WAN/LAN -> Router -> Cisco Callmanger -> 6608 ->PSTN Am I correct ? The you can replace the Meridian and VG200 with Asterisk or any of these with Asterisk. Or you can add it separately on the LAN on same location as the VG200, and specify asterisk as a H.323 Trunk/endpoint on the Cisco callmanager. In general all solutions depends on the equipments ability to apply to the standards. If some of it does'nt you would loose functionality. But I would really think twice before connecting too many different iPBX Components in one VoIP solution. I can understand if you're going to test the asterisk project as an alternative to either Cisco AVVID solution or the PBX's in your campus network. But a working enviroment I would really be very careful. It would be almost impossible to troubleshoot. It is very important to know your current setup in details, and what you hope to gain by using Asterisk. If my English is to bad then I apologize ... :-) Cheers, Johnny Witt Netv?rksspecialist CSIS - Combined Services & Integrated Solutions Majsmarken 9 ? DK2680 Solr?d Strand ? Denmark Tlf : (+45) 56 13 11 83 ? Mobil (+45) 28 66 28 48 ? jwi@csis.dk -------------- next part -------------- A non-text attachment was scrubbed... Name: Johnny Witt (jwi@csis.dk).vcf Type: text/x-vcard Size: 416 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030628/ef836a54/JohnnyWittjwicsis.dk.vcf