G looks up 1,2,3,4,5, g looks up 5,4,3,2,1 so yes, at least in theory. If you only have one open line, no harm no foul.
_____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Thursday, June 18, 2009 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue I don't feel like looking it up but does a capital G and lowercase g in your DAHDI/group make a difference? Just a thought. On Thu, Jun 18, 2009 at 2:28 PM, James A. Shigley <j...@answeringserv.com> wrote: I didn't have a limit set, but I put one on of 5 for testing sake that didn't change a thing. James Shigley Monroe Telephone Answering Service From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, June 17, 2009 2:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue Is your SIP call-limit set to 1? That might explain the busy/congest message. _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A. Shigley Sent: Wednesday, June 17, 2009 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue Never saw this appear on the list. So just resending it. Alright I've been having an issue when trying to dial out locally when coming from SIP. This used to work no problem, now it doesn't. Now the local PRI to Bell Is working fine I have calls coming in and out of it constantly right now. BUT if I try and make a local call from SIP (from X-Lite or one of our Linksys SPA2102s) It fails every time with errors like these == Using SIP RTP CoS mark 5 -- Executing [9819...@from_test:1] Dial("SIP/test-b636a620", "DAHDI/G3/9819213") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G3/9819213 -- Channel 0/23, span 3 got hangup, cause 50 -- Hungup 'DAHDI/71-1' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/test-b636a620' status is 'CHANUNAVAIL' == Using SIP RTP CoS mark 5 -- Executing [9819...@from_test:1] Dial("SIP/test-b6369010", "DAHDI/G3/4099819213") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G3/4099819213 -- Channel 0/23, span 3 got hangup, cause 50 -- Hungup 'DAHDI/71-1' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/test-b6369010' status is 'CHANUNAVAIL' == Using SIP RTP CoS mark 5 -- Executing [9819...@from_test:1] Set("SIP/test-09f23d18", "CALLERID(name)=James Shigley") in new stack -- Executing [9819...@from_test:2] Set("SIP/test-09f23d18", "CALLERID(number)=4099819213") in new stack -- Executing [9819...@from_test:3] Set("SIP/test-09f23d18", "CALLERID(all)=James Shigley<4099819213>") in new stack -- Executing [9819...@from_test:4] Dial("SIP/test-09f23d18", "DAHDI/G3/9819213") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G3/9819213 -- Channel 0/22, span 3 got hangup, cause 50 -- Hungup 'DAHDI/70-1' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/test-09f23d18' status is 'CHANUNAVAIL' Oh and sometimes it will also have this in the errors though no always [Jun 17 10:52:53] WARNING[27549]: app_dial.c:842 wait_for_answer: Unable to forward voice or dtmf On the second error above has the 409 added by the dialplan to see if Bell wanted full 10 digits. For the third I've tried a variety of ways of setting the CID thinking maybe that was the issue this was just my most recent. The odd thing is that I can send the call down one of my other PRI ports to our Amtelco Infinity system. (via exten=> 9819213,1,Dial(${inf}/409${EXTEN}). I've tried everything I can think of and googled for a good while trying to find an explanation for "got hangup, cause 50". What is cause 50? Sip Login information [test] username=test type=friend secret=XXXXXXXXX callerid= host=dynamic nat=no canreinvite=no context=from_test ;codecs disallow=all allow=ulaw Also had it as [test] username=test type=friend secret= XXXXXXXXX callerid= "James Shigley" <4099819213> host=dynamic nat=no canreinvite=no context=from_test ;codecs disallow=all allow=ulaw My From Context has changed several times here is some of the iterations I've tried. inf=DAHDI/g2 bell=DAHDI/G3 [from_test] ; noted but not repaired. exten=> _NXXXXXX,1,Dial(${belltd}/409${EXTEN}) exten=> 9819213,1,Dial(${inf}/409${EXTEN} [from_test] ; noted but not repaired. exten=> _NXXXXXX,1,Set(CALLERID(name)=James Shigley) exten=> _NXXXXXX,2,Set(CALLERID(number)=4099819213) exten=> _NXXXXXX,3,Set(CALLERID(all)=${CALLERID(name)}<${CALLERID(num)}>) exten=> _NXXXXXX,4,Dial(${bell}/${EXTEN}) [from_test] ; noted but not repaired. exten=> _NXXXXXX,1,Set(CALLERID(name)=James Shigley) exten=> _NXXXXXX,2,Set(CALLERID(number)=4099819213) exten=> _NXXXXXX,3,Dial(${bell}/${EXTEN}) Note I didn't include the full context only the lines relevant to local dialing. LD dialing which is sent out sip works just fine. Also I tried using g3 instead of G3 thinking maybe there was an issue with the high channels. Though when I do a core show channels there isn't near close to all the channels used. One final note. I did try calling other numbers beyond just 9819213 the errors and issue was the same regardless of the local number dialed. I think that's all the information you might need, If I forgot something just let me know. Oh and this is on * 1.6.0.6 James Shigley Monroe Telephone Answering Service 409-981-9213 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype)
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users