[Asterisk-Users] Re: Asterisk as Cisco Call-Manager - dial out to PSTN

[Mario Spendier]

Hi Maron, 

 

Thank you for your answer! I use a simple cisco router 2621XM as call
manager with the following configuration:

 

interface Loopback79

 description ALT-VoIP-Gateway

 ip address 10.xxx 255.255.255.255

 h323-gateway voip interface

 h323-gateway voip id Ldnxxx ipaddr 10.xxx 1719 priority 120

 h323-gateway voip h323-id Altxxx at xxx.com

 h323-gateway voip tech-prefix 301

 h323-gateway voip bind srcaddr 10.xxx

 

The structure is ...

 

Sip-phone --> SIP --> Asterisk as call-manager  (extension 399) --> H.323
--> cisco gatekeeper (extension 6666) --> H.323 --> cisco call-manager
(extension 302) --> E1 PSTN

 

Iif I dial now with the "Sip-phone": 6666 302 [PSTN number (handy number,
....)] I should be able to telephone the the PSTN of the call manager with
the extension 302. It works within cisco devices perfectly but not with
asterisk. Can you tell me your experiences and practices??

 

Thanks a lot!!

 

Mario

 

 

 

 

 

Hi Mario.
 
What kind of Cisco gateway are you using, I swapped an Cisco Call 
Manager 4.0 for Asterisk, and am using 12 gateways worldwide for PSTN 
access.  However using SIP, which the gateways (Call Manager Express on 
1760 routers) support very well for trunking.
 
I've found that H323 is even buggy between the CME gateways from Cisco.
 
Regards,
 
Maron Kristofersson
 
Mario Spendier wrote:
> Hi all,
> 
>  
> 
> I'm running Asterisk since two days, and it's really one of the phatest 
> software available on the net!!! Respect!!! I have connected Asterisk as 
> a call manager for a cisco gatekeeper. Everything works fine internal, 
> but if I want to ring to a PSTN over another call manager, which is 
> connected over ISDN, I get the following output. Has anyone experience 
> in this or can help me? I'm running against closed doors in this 
> problem!!! If I phone over a Cisco call manager it works, so the failure 
> is Asterisk based.
> 
>  
> 
> -- Executing NoOp("SIP/12345-454d", ""call for "XXXX") in new stack
> 
>     -- Executing Dial("SIP/12345-454d", "OH323/ XXXX ") in new stack
> 
>     -- H.323 call to XXXX with codec alaw
> 
>     -- Called XXXX
> 
>     -- H.323 call 'ip$localhost/27230' cleared, reason 24 (Call ended 
> with Q.931 cause)
> 
>     -- Hungup 'OH323/L27230'
> 
>  
> 
> Thanks a lot!!!
> 
>  
> 
> Mario
> 
> 
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