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Okay! i have changed context at master side<br><br><br><br>; Span 1: WPT1/0 "wanpipe1 card 0" (MASTER)<br>switchtype = national ; commonly referred to as NI2<br>context = from-internal<br>group = 1,24<br>echocancel = yes<br>signalling = pri_net<br>channel => 1-23<br><br><br><br>Same error nothing change..<br><br>satish-desktop*CLI> core set verbose 10<br>Verbosity was 0 and is now 10<br>satish-desktop*CLI> core set debug 999<br>Core debug was 0 and is now 999<br> == Using SIP RTP CoS mark 5<br> -- Executing [7527@from-sip:1] Dial("SIP/7623-00000000", "DAHDI/g1/527") in new stack<br>[Mar 25 16:39:47] WARNING[5336]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)<br> == Everyone is busy/congested at this time (1:0/1/0)<br> -- Auto fallthrough, channel 'SIP/7623-00000000' status is 'CONGESTION'<br><br><br><br>> From: tilghman@meg.abyt.es<br>> To: asterisk-users@lists.digium.com<br>> Date: Fri, 25 Mar 2011 15:35:21 -0500<br>> Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue<br>> <br>> On Friday 25 March 2011 14:40:40 satish patel wrote:<br>> > Following is my scenario to connect back to back PRI of two asterisk<br>> > server. PRI cards are Sangoma A102D<br>> > <br>> > [Asterisk1]------------[PRI]-Cross Cable---------[Asterisk2]<br>> > <br>> > Asterisk1<br>> > <br>> > ; Span 1 (MASTER)<br>> > switchtype = national ; commonly referred to as NI2<br>> > context = from-pstn<br>> > group = 0,24<br>> > echocancel = yes<br>> > signalling = pri_net<br>> > channel => 1-23<br>> > <br>> > <br>> > Asterisk2<br>> > <br>> > ; Span 1<br>> > switchtype = national ; commonly referred to as NI2<br>> > context = from-pstn<br>> > group = 0,24<br>> > echocancel = yes<br>> > signalling = pri_cpe<br>> > channel => 1-23<br>> <br>> Here's one confusing part. You're saying that calls that come from the<br>> master to the slave end up in context from-pstn (on the slave), but calls<br>> from the slave to the master ALSO end up in from-pstn (on the master).<br>> Seems like one of them should be "from-internal" or the like. I'm sure<br>> some of your problem emanate from these settings.<br>> <br>> > satish-desktop*CLI><br>> > [Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable<br>> > to create channel of type 'DAHDI' (cause 34 - Circuit/channel<br>> > congestion)<br>> <br>> Check the other side for error messages.<br>> <br>> > [Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor:<br>> > Cannot connect [Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115<br>> > __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned<br>> > -1: Invalid argument [Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115<br>> > __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned<br>> > -1: Invalid argument<br>> <br>> This problem is due to a misconfiguration. Asterisk cannot handle the local<br>> network being addressed as the 0.0.0.0 network. You need to use the full<br>> local address.<br>> <br>> -- <br>> Tilghman<br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br> </body>
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