[asterisk-users] Back-to-back asterisk PRI issue

[satish patel]

sometime i am getting following error also. what is this means?

[Mar 25 17:11:52] WARNING[5961]: sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available!  Using Primary channel as D-channel anyway!



From: satish_lx at hotmail.com
To: asterisk-users at lists.digium.com
Date: Fri, 25 Mar 2011 21:04:45 +0000
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue








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: satish_lx at hotmail.com
To: asterisk-users at lists.digium.com
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 at 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: tilghman at meg.abyt.es
> To: asterisk-users at lists.digium.com
> 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
> 
> --
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