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 <[email protected]>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:* [email protected] [mailto: > [email protected]] *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:* [email protected] [mailto: > [email protected]] *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)
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