Only Difference is one side card is ECHO Cancellation supported and other is non-ECHO cancellation. Is there any issue ?
@Asterisk1 Sangoma A102 (non-ECHW) @Asterisk2 Sangoma A102D (ECHW) -Satish From: [email protected] To: [email protected] Date: Fri, 25 Mar 2011 20:41:09 +0000 Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue Okay! i have changed context at master side ; Span 1: WPT1/0 "wanpipe1 card 0" (MASTER) switchtype = national ; commonly referred to as NI2 context = from-internal group = 1,24 echocancel = yes signalling = pri_net channel => 1-23 Same error nothing change.. satish-desktop*CLI> core set verbose 10 Verbosity was 0 and is now 10 satish-desktop*CLI> core set debug 999 Core debug was 0 and is now 999 == Using SIP RTP CoS mark 5 -- Executing [7527@from-sip:1] Dial("SIP/7623-00000000", "DAHDI/g1/527") in new stack [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) == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/7623-00000000' status is 'CONGESTION' > From: [email protected] > To: [email protected] > Date: Fri, 25 Mar 2011 15:35:21 -0500 > Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue > > On Friday 25 March 2011 14:40:40 satish patel wrote: > > Following is my scenario to connect back to back PRI of two asterisk > > server. PRI cards are Sangoma A102D > > > > [Asterisk1]------------[PRI]-Cross Cable---------[Asterisk2] > > > > Asterisk1 > > > > ; Span 1 (MASTER) > > switchtype = national ; commonly referred to as NI2 > > context = from-pstn > > group = 0,24 > > echocancel = yes > > signalling = pri_net > > channel => 1-23 > > > > > > Asterisk2 > > > > ; Span 1 > > switchtype = national ; commonly referred to as NI2 > > context = from-pstn > > group = 0,24 > > echocancel = yes > > signalling = pri_cpe > > channel => 1-23 > > Here's one confusing part. You're saying that calls that come from the > master to the slave end up in context from-pstn (on the slave), but calls > from the slave to the master ALSO end up in from-pstn (on the master). > Seems like one of them should be "from-internal" or the like. I'm sure > some of your problem emanate from these settings. > > > satish-desktop*CLI> > > [Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable > > to create channel of type 'DAHDI' (cause 34 - Circuit/channel > > congestion) > > Check the other side for error messages. > > > [Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor: > > Cannot connect [Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115 > > __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned > > -1: Invalid argument [Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115 > > __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned > > -1: Invalid argument > > This problem is due to a misconfiguration. Asterisk cannot handle the local > network being addressed as the 0.0.0.0 network. You need to use the full > local address. > > -- > Tilghman > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
