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

Thu Jun 18 13:49:54 CDT 2009

It errors the same whether I use g or G. 

James Shigley

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at 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
<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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