[asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

Thu Jun 18 13:27:41 CDT 2009

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 <jas at 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-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at 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-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at 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 [9819213 at 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 [9819213 at 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 [9819213 at from_test:1] Set("SIP/test-09f23d18",
> "CALLERID(name)=James Shigley") in new stack
>
>     -- Executing [9819213 at from_test:2] Set("SIP/test-09f23d18",
> "CALLERID(number)=4099819213") in new stack
>
>     -- Executing [9819213 at from_test:3] Set("SIP/test-09f23d18",
> "CALLERID(all)=James Shigley<4099819213>") in new stack
>
>     -- Executing [9819213 at 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**
>
>
>
>
>
>
>
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-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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