Hi to all, I'v got this AS5300 Switch with 4 E1, I'm using it for voice over IP traffic, now I need to edit existing E1 configuration, and need to add new Voip Carriers, I saw some tutorial on www, the guy was using some kind of interface to do that, something called Cisco Voice Manager, can sombody tell me where to get that software, or at least give me some hints on how to add new voip sip cariers on that cisco device. or how to edit existing ones. Please. Any help is aprisciated. thanks.
Well I have 1 WTL telecom Swithc with 48E1, and 1 cisco AS5300 4E1, what I want to do is use the cisco as extra voip handling source, that means, I will move few voip cariers from WTL switch to cisco, the cisco will only accept incoming calls filter the prefix if the prefix is right cisco will send the call to WTL switch with same prefix. In this case I have a sip carrier wich is sending traffic from 2 IP's with prefix let's say : 44444, that is coming in via internet, all 4 E1's on cisco are alredy configured and connected to WTL switch. so the traffic comes in from internet cisco filter is and if prefix and IP is right than cisco sends it to WTL swithc, thats wat I want to do, add new carriers, edit existing ones.
Thanks for you help.
1st make your PRIs go up.
isdn switch-type "primary-ni2" <-- could be something else
controller t1 0/0/1:23 <--- repeat for all ctrls, put E1 is using that
pri-group timeslots 1-23 <--or 1-30 if E1
voice-port 0/0/1:23 <-- repeat for all voice-ports
dial-peer voice 4444 voip
incoming called-number 4444. <-matches your prefix
codec g711u <-- codec used by sip carrier,
dial-peer voice 4445 pots
destination-pattern 4444. <-prefix again
no digit-strip <-- you send the full number as received
This is the basics. Do "show isdn status" and check all controllers to be "multiframe_established". Do "show controllers t1" and check no errors, no slip seconds.
If something wrong with the above, some additional settings may be needed.
Then, start receiving calls. If something doesn't work, do "term mon", "debug ccsip message" and "debug isdn q931", and we will what is happening. Good luck!
I gues I'm lost, first I need to get an view of the hierarchie of this system, for example how can I see wich voip carriers are alredy configured? cause now there are 2-3 cariers working on this AS5300, there is traffic on it.
Steps to see the list of alredy existing working carriers. their configurations.
Second how to add a new carrier. This carriers connect via internet. Letssay there is a new carrier :
IP : 126.96.36.199
Protocol : SIP
Prefix : 44444
What are the steps to add this carier.
Cisco IOS doesn't work like that. Normally there is no explicit definition of a carrier. When a call via VoIP comes in, you don't actually need to do anything special in preparation. The basic configuration construct is instead "dial-peer", if you do "show run" you will see that. If you have more advanced requirements like authentication, OSP, settling, billing, blocking, special routing etc, we can discuss how to do that, but once again, none of that is necessary to basic call routing from VoIP to PRI.
So the commands to make carX work are the ones I gave you before.
Well the cisco has 4 E1 and all 4 are connected to another telecom switch wich handles all : billing, outgoing carriers and stuff. So this cisco is used only to accept voip filtr the prefix if its right, and IP is right than pass it thrue to the other telecom SWithc. so the parts of 4E1 to otherSwithc are configured, and few voip carriers are laredy sending trafic via internet, that works, thats configured by sombody alse, so now I need to learn how to add new voice carriers, or manage existing ones.
I suggest the following two documents to begin learning IOS voice features:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml (with config examples)
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ok, after reading this, and other stuff, and lil help from some frend, I'm more further now. Now I can add new carriers, delete them, edit, and so on. alredy made some trunk groups, tomorow will go on, but!!! I need to do 1 important thing: I need to make sure that call coming in on a carrier gets a specified cid (caller id) and goes on to another switch. In my situation I have 4 E1 ports, all 4 of them are configured and connected to WTL switch IPNX. I alredy have 4 inbound and few outbound carriers, well those 4 inbound Carriers are defined per port, what means, carier 1 goes to e1-1 carier 2 to e1-2 and so on. so maximaly I can add 4 carriers, cause on the billing side on wtl switch it bill per E1, so I need to connect more inbound carriers and be able to bill them, so need to send a fake or specified callerID to bill them based on caller id. Can sombody tell me how to do that.
sip call comes in on AS5300> gets filtered and sent to specific E1,> gets to WTL switch, > wtl switch cheks on wich E1 it comes, and bills it.
I need :
sip call comes in on AS5300> gets filtered on prefix or IP, AS5300 puts specified caller id to that call and sends it further to > WTL , WTL switch checks caller id, bills on caller ID.
Is this possible?
What you want to do is carrier based routing. Kind of advanced configuration. See:
Hope this hels, please rate post if it does!
Well I just started reading this link wich you gived, and even now I'm much further in understanding how this thing works, actualy simple but still much to understand and leearn, here by I want to thank you very very much for pointing me to this article. I'm going to read it.
Note that if the carriers can send the called number with a prefix (different for each carrier), you can do a simple match using basic dial-peer and strip the prefix before connecting the call via PRI interface.
If the can't do that, the only method for mapping 1 to 1 is the one in the document above.
Thank you for the nice rating and good luck!
ok, now I have a lil problem again. The problem is that after reading that document, I cant realy understand the connection between some things like : profiles, trunk groups, and the importantnes of them. Is there somewhere a page where I can see the real call route to understand what are the real use of this thing, like where I can see the sceme of incoming or outgoing call and see what happens first, second, third, how is the call handled how is the right route choosed?? I need that to understand what I'm doing, cause now I know how to add, delete, edit all those things, but I'm lil confused about the use of all of that. and also I want to understand better where is de configuration differance in 2 configuration models:
1. I can have only 4 inbound and 4 outbound carriers. thats it.
2. I can have only 4 outbound and more inbound carriers.
My situation now:
I have 4 E1 cisco AS5300, the config now is 4 inbound 4 outbound.
I need to have more inbound carriers and I need to bill them on other side, thus I need to configure it a way that when inbound carier X (voip) calls in, he gets an specified Caller ID (cli: 0031-123445 )added in place of he's own cli and goes further to e1 wich leads the call to WTL switch in WTL switch the billing is set on CLI ( caller id ) so it will get billed. and outbound carriers are set on WTL on PORT base, so its a call sent to a specific E1 port wich is fisicaly connected to E1 on cisco and dependent on ( i gues ) prefix or some other config the call is being sent to right carier.
Please help me understand what is what , I'm realy sorry for this stupid quiestion. I'm sitting and reading this stuuff hier again to see if I didnt miss something. This is lil new for me, I studiet Cisco networking and stuff but newer had any studies on Device configurations. I have now lil problem alredy few days, one of those alredy configured carriers on AS530 is not using the 2cond port wich was reserved vor hem, so I need to chenge the settings and add another inbound carrier in he's place. but when I started I couldnot understand the relation and use of profiles and trunk groups.
I realize that an advanced call routing configuration can be confusing for anyonw. I think with a little more time you will be able to understand how things work.
Unfortunately beside digging and reading more on CCO documents I cannot think of a good way of learning.
Now for your problem, if you want to replace the calling number, that is easy, you can do that in the outgoing dial-peer with "clid network-number". Or, if you want to send always the same number on a give interface, configure "isdn calling-number ..." under the interface. There are more advanced ways using translation-profile and rules.
Regarding the fact that the 2nd carrier doesn't go to the 2nd PRI as you want, this can be resolved only looking at the configuration, and making sure that the access list is correct to match the IP address from where the voip call comes from.
It seems to me that the carriers are sending a prefix so it shouln't really be be difficult.
But the configuration seems a little incomplete, for example you defined trunkgroup ATT-OUT, but no DP outgoing uses it ?
Then is not clear to me what you are trying to do with translation-profiles, consider that by default, and outgoing DP pots will strip the matched digits, so there shouldn't be need for translation in this case.
translation-rules 7452, 19937, and 90380 are a litte wrong. Beside there is no need to match /^/, if you want to always prepend some digits (but why?), just match the empty string,that is //.
Or just state what is not working. remember, do "term mon" and "debug isdn q931" and you will what you are sending to the switch, then is is just a matter of configuring right to send the right numbers.
Well thats the problem, I dont understand the connection and use of it in this config, this cisco is configured by one frend, he configured what I needed, and said that this is temporary to use only 4X4 inbound/outbound, and later after I learn more, we ware plannning to reconfigure it to use 4outbound, many inbound and inbound should be on cli base( caller ID ) so if sombody calls in he gets specified caller ID added and sent to E1 to our other Switch, so I started learning learnin and still learning and tryng to understand... so that was my problem to, cause I wanted first to understand the use off all those things, thats why I was looking for smoe sceme, some route way picture, where I can see what happens with incoming or outgoing call visualy. so I can see the importance of things.
please go over the documents I've linked at the beginning of this thread. Perhaps you can browse the the CCO website for more tutorials. Or ask your friend to explain the DP basics again.