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

2005-03-31 Thread 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 [EMAIL PROTECTED]

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 )  H.323  cisco
call-manager (extension 302)  E1 PSTN



Iif I dial now with the Sip-phone:  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 KristoferssonMario Spendier wrote: Hi all,Im running Asterisk since two days, and its 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? Im 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 ) in new stack  -- Executing Dial(SIP/12345-454d, OH323/  ) in new stack  -- H.323 call to  with codec alaw  -- Called   -- H.323 call 'ip$localhost/27230' cleared, reason 24 (Call ended  with Q.931 cause)  -- Hungup 'OH323/L27230'Thanks a lot!!!Mario     ___ Asterisk-Users mailing list Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

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

2005-03-27 Thread Maron Kristfersson
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,
 

Im running Asterisk since two days, and its 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? Im 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 ) in new stack
-- Executing Dial(SIP/12345-454d, OH323/  ) in new stack
-- H.323 call to  with codec alaw
-- Called 
-- 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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