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Hi,
I used C3640, but It was changed, because of few DSP in it. However, configuration is same. It also depends on used IOS version. Here are fragments from configurations: AS5300: ! clock timezone GMT 0 ; in some Docs = necessary ! isdn switch-type primary-net5 ; I`m in Europe :-) isdn voice-call-failure 0 ! ! voice call send-alert voice rtp send-recv ! voice service voip ! voice class codec 3 codec preference 1 g711alaw codec preference 2 g711ulaw ! controller E1 0 clock source line primary pri-group timeslots 1-31 description to-PSTN ! translation-rule 2 ; type of number (subs/national/international) depend on your telco provider Rule 0 .... 021111 ANY subscriber Rule 10 any 0211111111 ANY subscriber ! ! translation-rule 10 ; type of number (subs/national/international) depend on your telco provider Rule 0 ^421211110... 0 ANY subscriber Rule 1 ^421211111... 1 ANY subscriber Rule 2 ^421211112... 2 ANY subscriber Rule 3 ^421211113... 3 ANY subscriber Rule 4 ^421211114... 4 ANY subscriber Rule 5 ^421211115... 5 ANY subscriber Rule 6 ^421211116... 6 ANY subscriber Rule 7 ^421211117... 7 ANY subscriber Rule 8 ^421211118... 8 ANY subscriber Rule 9 ^421211119... 9 ANY subscriber Rule 10 any 1234 ANY subscriber ! interface Serial0:15 description PRI-D-CHANNEL-to-PSTN no ip address no logging event link-status isdn switch-type primary-net5 isdn guard-timer 3000 isdn map address 0.* plan isdn type subscriber isdn send-alerting isdn sending-complete no cdp enable ! voice-port 0:D input gain -6 output attenuation 14 echo-cancel coverage 32 echo-cancel suppressor cptone SK description E1 bearer-cap Speech ! dial-peer voice 8 pots tone ringback alert-no-PI destination-pattern 00T port 0:D prefix 00 ! dial-peer voice 10 pots tone ringback alert-no-PI destination-pattern 0[1-9]........ port 0:D prefix 00421 ! dial-peer voice 20 pots tone ringback alert-no-PI destination-pattern 00421[1-9]........ port 0:D prefix 00421 ! dial-peer voice 999 voip numbering-type international incoming called-number . voice-class codec 3 session protocol sipv2 dtmf-relay cisco-rtp h245-signal h245-alphanumeric fax rate 7200 ip qos dscp cs5 media no vad supplementary-service pass-through ! dial-peer voice 1 pots incoming called-number . direct-inward-dial port 0:D ! dial-peer voice 42121111 voip destination-pattern 42121111.... translate-outgoing called 10 voice-class codec 3 session protocol sipv2 session target ipv4:1.2.3.4:5060 ; IP address of Asterisk ip qos dscp cs5 media no vad ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:1.2.3.4:5060 ; IP address of Asterisk ! ntp server 1.2.3.5 ! I`m not sure, if all things are necessary and correct, but... it`s working :-). I can place calls from asterisk to PSTN via AS5300, and also receive calls from pstn. In this configuration, i have DDI prefix from my telco as 42121111xxxx. 421 = international prefix 2 (02) = national prefix, 1111xxxx is my DDI prefix in which i can use 10 000 numbers. I`m using 4 digit extensions in my numbering plan at Asterisk, so I could have DID in 1:1 mapping. Fragments of very simple asterisk configurations: Extensions.conf [globals] CISCOSIPGW=2.2.2.2 ;(IP address of AS5300) [outgoing-cisco-pstn] exten => _90NXXXXXXXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],180) ; local calls Sip.conf [2.2.2.2] type=friend host=2.2.2.2 nat=no canreinvite=yes dtmfmode=rfc2833 disallow=all allow=alaw allow=ulaw In this cas, only 10 digit numbers are allowed (only national calls) to dial via Cisco, through number 9 as an prefix for outbound calls. Hope, that this samples will be usefull for you. PS: sorry for english, i hope, you could understand it :-) -b ----- Original Message ----- From: "Anton Krall" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, May 11, 2005 7:08 PM Subject: RE: [Asterisk-Users] Asterisk and Cisco AS5300 or 3600 > Hey Barney > > What are the steps necessary to make that work on the cisco AS5300? Any > configs I need to check to make it work? And what do I need on asterisks > side? > > Ever used cisco 3600? > > |-----Original Message----- > |From: [EMAIL PROTECTED] > |[mailto:[EMAIL PROTECTED] On Behalf Of barney > |Sent: Mi�rcoles, 11 de Mayo de 2005 05:22 a.m. > |To: Asterisk Users Mailing List - Non-Commercial Discussion > |Subject: Re: [Asterisk-Users] Asterisk and Cisco AS5300 or 3600 > | > |> Just in case you don't know, AS5350 supports SIP *and* H323 > |after IOS > |> version > |> 12.3 (maybe a little earlier). > |> It allows you to use both at the same time, without needing > |to set it > |> up for one system specifically. > |> Haven't tried it with Asterisk yet though. > | > | > |I have tried it. I have SIP trunk between Asterisk and AS5300 > |(C3640 before), and it`s working good. > |It`s quite good solution, but its much more expensive as some > |PCI card direct in Asterisk (i`m using PRI interconnect to PSTN). > | > |-b > | > |PS: sorry for poor english > | > | > | > |> On Wednesday 11 May 2005 11:23, Anton Krall wrote: > |>> I need some advice on some h323 issues. I need to test connectivity > |>> from Asterisk to a Cisco AS5300 that has PSTN lines and to > |cisco 3600 > |>> voip routers. > |>> > |>> H323 needs to be used here but I was wondering if anybody > |has linked > |>> Asterisk to these Cisco routers before? > |> Just in case you don't know, AS5350 supports SIP *and* H323 > |after IOS > |> version > |> 12.3 (maybe a little earlier). > |> It allows you to use both at the same time, without needing > |to set it > |> up for one system specifically. > |> Haven't tried it with Asterisk yet though. > |> > |> Richard. > |> _______________________________________________ > |> Asterisk-Users mailing list > |> [email protected] > |> http://lists.digium.com/mailman/listinfo/asterisk-users > |> To UNSUBSCRIBE or update options visit: > |> http://lists.digium.com/mailman/listinfo/asterisk-users > |> > | > |_______________________________________________ > |Asterisk-Users mailing list > |[email protected] > |http://lists.digium.com/mailman/listinfo/asterisk-users > |To UNSUBSCRIBE or update options visit: > | http://lists.digium.com/mailman/listinfo/asterisk-users > | > | > > |
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