11-06-2015 10:22 PM - edited 03-17-2019 04:50 AM
Hi all,
am new to As5300 Voice Gateway as i got used to work in Cisco CME and CUCM infrastructure.
am having As 5350 voice GW with internet connection, 2 cisco ip phones and POE switch.
I need to connect (voip calls) to another branch via internet connection, no E1/T1 or isdn wan connection are there.
Can i configure the AS5350 with dial peer, tftp-server (for ip phones), telephony service, ephone, ephone-dn
and vpn confiugration like the cme or the AS5300 doesn`t do that.
What is the actual scenario the AS5300 used for ???? can you provide an example resource please?!
can my above scenrio be done with only AS5350 GW and ip phones?
please help.
11-07-2015 10:25 AM
You should start by reading the data-sheet to understand what a product does
and NO, the AS5XXX are ONLY voice gateways, you cannot have CME on them.
The CME compatibility guides contain info of what platforms are supported for CME
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/requirements/guide/33matrix.html
11-07-2015 10:49 PM
Hi Jaime,
thanks for the data sheet, so it means the As5300 doesnt accomplish my above scenario. ??
so, there is nothing but cme to connect to another site using ip phones and vpn ???
is there another way ??
11-08-2015 08:13 AM
No, if you want to register phones, you need either CME or CUCM
If you want VPN, then you need an ISR or VPN dedicated solution, or an ISR with VPN module.
What you got, is just a gateway.
11-08-2015 11:48 PM
thanks Jaime
12-09-2015 11:25 PM
Hi again Jaime,
and sorry for being late. the below is the scenario which i want to ask about including my issue.
#############################
I`m having the blow scenario
Voice GW_1-->Internet-->Voice GW_2-->Sim Box (TELES.iGate)
the sim Box contains sim card in which it first receive the voip call on port 20 from the Voice GW_2 and then transfer the call via its sim card through port.40.
My issue is the call came to Voice GW_2 in which it doesn't move out the call to the sim box.There is ping between them and what only i did is a voip dial-beer as below.
!
dial-peer voice 44 voip
destination-pattern 0096773.......
session target ipv4:192.168.1.104
codec g711ulaw
dtmf-relay h245-alphanumeric h245-signal
no vad
!
the test call is for example calling a gsm number from the Voice GW_1 called no.(00967733606078), i want the call to move to the Voice GW_2 which then move it to the sim box, then the sim box will call the called no. using its sim number.
Sim box is configured with its ports and we use sip and H323 between it and the VoiceGW_2.
Is there anything i have to add to the VoiceGW_2 to to make it transfer the call to the sim box.
Thanks all.
12-09-2015 11:35 PM
Hi Mohammed,
You may probably try "debug voip ccapi inout" to see the cause why the call is failing to get forwarded.
Regards...
Ashok.
12-11-2015 10:23 PM
12-15-2015 09:28 AM
hi again Ashok,
i collect a debug file with changing dial-peer also i use sip then i remove it. i did a test call to the mobile and to the sims of the sim box.
i cant get if the call is received to my voiceGw_2 and tries to go to the sim box or not.
Please go through debug and assure me if the call is received to VoiceGW_2 or not.
thanks in advance.
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