Mario Spendier
2005-Mar-24 05:30 UTC
[Asterisk-Users] Asterisk as Cisco Call-Manager - dial out to PSTN
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$localhost/27230' cleared, reason 24 (Call ended with Q.931 cause) -- Hungup 'OH323/L27230' Thanks a lot!!! Mario -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050324/71bcd31e/attachment.htm
Maron Kristófersson
2005-Mar-28 00:30 UTC
[Asterisk-Users] Re: Asterisk as Cisco Call-Manager - dial out to PSTN
Hi Mario. What kind of Cisco gateway are you using, I swapped an Cisco Call Manager 4.0 for Asterisk, and am using 12 gateways worldwide for PSTN access. However using SIP, which the gateways (Call Manager Express on 1760 routers) support very well for trunking. I've found that H323 is even buggy between the CME gateways from Cisco. Regards, Maron Kristofersson Mario Spendier wrote:> Hi all, > > > > I?m running Asterisk since two days, and it?s really one of the phatest > software available on the net!!! Respect!!! I have connected Asterisk as > a call manager for a cisco gatekeeper. Everything works fine internal, > but if I want to ring to a PSTN over another call manager, which is > connected over ISDN, I get the following output. Has anyone experience > in this or can help me? I?m running against closed doors in this > problem!!! If I phone over a Cisco call manager it works, so the failure > is Asterisk based. > > > > -- Executing NoOp("SIP/12345-454d", ""call for "XXXX") in new stack > > -- Executing Dial("SIP/12345-454d", "OH323/ XXXX ") in new stack > > -- H.323 call to XXXX with codec alaw > > -- Called XXXX > > -- H.323 call 'ip$localhost/27230' cleared, reason 24 (Call ended > with Q.931 cause) > > -- Hungup 'OH323/L27230' > > > > Thanks a lot!!! > > > > Mario > > > ------------------------------------------------------------------------ > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Mario Spendier
2005-Mar-31 06:52 UTC
[Asterisk-Users] Re: Asterisk as Cisco Call-Manager - dial out to PSTN
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@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 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com<http://lists.digium.com/mailman/listinfo/asterisk-users>> http://lists.digium.com/mailman/listinfo/asterisk-users<http://lists.digium.com/mailman/listinfo/asterisk-users>> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users<http://lists.digium.com/mailman/listinfo/asterisk-users> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050331/5139bcfd/attachment.htm
Mario Spendier
2005-Mar-31 07:05 UTC
[Asterisk-Users] Asterisk as Cisco Call-Manager - dial out to PSTN
Hi Maron, Thank you for your answer! I use a simple cisco router 2621XM as call gateway 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@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 gateway (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 gateway 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 _____ From: Mario Spendier [mailto:Mario.Spendier@at.flextronics.com] Sent: Donnerstag, 24. M?rz 2005 13:30 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk as Cisco Call-Manager - dial out to PSTN 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050331/2c188a94/attachment.htm