Hello Nuurul,
> Hai,
>
> Today, I have observed a network eng from a vendor
> install a E&M Type II card on a Cisco 3600 and
> connected to my E&M II Toshiba card. We did a loop
> test from port 1 and port 2 and it works. The Eng
> configured the first port dial peer as 300..., second
> port dial peer as 301.., 302.. and 303.. . So when we
> did a call from our Key phone unit to another remote
> site connecetd to a FSX int card we were able to get
> connected using both FXS numbers was 311 and 312.
>
> Funny thing is when the other side dial the first port
> i.e 300242--> 242 is the ext. We were able to see the
> call get into the router and pass through to the KST.
> From the ext 242 we can see the light showing a call
> but the call was never establish. The other side
> advice getting busy tone. But when we test the other
> dial peer number i.e 301242, 302242 it works. And when
> we need do a local loop from one key phone to another
> one we must add up another number i.e 3011242, insted
> off 301242.
>
> As I observed the sh run for both router, the config
> for the dial peer was the same and the port was config
> correctly (just observed since it by another vendor).
> But each time the net eng configured or change any
> setting he never do any shutdown and no shut.
>
> Thus my question is :
>
> a) Is there any barr from using 300 flow by the wild
> card ...?.
>
I don't think 300 have any special meaning.
But I guest there is some overlap dial peer cause the problem.
Let's use this command to verify :
show dial-plan number 300242
> b) When I asked the eng why he keep congif a new dial
> peers insted off duing any debug he reply that for
> VOIP there is no debug commands. I may not be VOIP
> experts but as a CCNA i did observed a few debug
> commands for VOIP. Any info on this.
You can use
debug voip ccapi inout
You may also simulate a call from the Cisco router :
csim start xxxxxxx
But, DO NOT pick up the call at your key phone. This will cause the router reboot on
3640 platform.
(There is no problem on other platform such as AS5300, I have ask Cisco TAC for this,
but their answer is : "csim is a hidden command, so we won't provide any support on
this
issue....")
> c) Would the command no shut and shutdown help in this
> situation.
No, the change is effect once you type the command and press enter.
> d) What would happend to a 64Kbps line with 32 CIR
> with 4 voice channels compress up to 11 KBPS?. We only
> test the voice quality with extended ping as dumy
> data. If the first, second and third channels is busy
> the fourth persons who is using the voice channel will
> have a good or bad quality off voice. Heck after this
> any problem the first people to be blame will be the
> PBX guy.
Since most of my previous VOIP config are work for IDD service provider,
I have face this problem before. I will try to find the answer.
Quncy Lau
System Engineer
CCNP
Hong Kong
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