Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue
[0]) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 27418/0x6B1A) (Terminator) Message type: ALERTING (1) [1e 02 81 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 30 (cs0, Progress Indicator) q931.c:3590 q931_receive: call 60186 on channel 2 enters state 4 (Call Delivered) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 27418/0x6B1A) (Terminator) Message type: CONNECT (7) [1e 02 81 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 30 (cs0, Progress Indicator) q931.c:3620 q931_receive: call 60186 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 27418/0x6B1A) (Originator) Message type: CONNECT ACKNOWLEDGE (15) !! Got reject for frame 69, but we have nothing -- resetting! NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3009 q931_disconnect: call 60186 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 27418/0x6B1A) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 27418/0x6B1A) (Terminator) Message type: RELEASE (77) q931.c:3795 q931_receive: call 60186 on channel 2 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 27418/0x6B1A) (Originator) Message type: RELEASE COMPLETE (90) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null !! Got reject for frame 71, but we have nothing -- resetting! James Shigley -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres Sent: Wednesday, June 17, 2009 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue James A. Shigley wrote: Never saw this appear on the list. So just resending it. You might get more help if you include a PRI Debug that shows the call being rejected. Andres http://www.neuroredes.com 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 ___ -- 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 ___ -- 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
Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue
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 Shigley4099819213) 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=X
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= X
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= _NXX,1,Dial(${belltd}/409${EXTEN})
exten= 9819213,1,Dial(${inf}/409${EXTEN}
[from_test] ; noted but not repaired.
exten= _NXX,1,Set(CALLERID(name)=James Shigley)
exten= _NXX,2,Set(CALLERID(number)=4099819213)
exten=
_NXX,3,Set(CALLERID(all)=${CALLERID(name)}${CALLERID(num)})
exten= _NXX,4,Dial(${bell}/${EXTEN})
[from_test] ; noted but not repaired.
exten= _NXX,1,Set(CALLERID(name)=James Shigley)
exten= _NXX,2,Set(CALLERID(number)=4099819213)
exten= _NXX,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
Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue
(len=30, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x16, 0x02, 0x01, 0xFC, 0x02, 0x01, 0x00, 0x80, 0x0E, 'CHARLOT,DANIEL' ] PROTOCOL 1F 8B 0001 00 (CONTEXT SPECIFIC [11]) A1 0016 (CONTEXT SPECIFIC [1]) 02 0001 FC (INTEGER: 252) 02 0001 00 (INTEGER: 0) 80 000E 43 48 41 52 4C 4F 54 2C 44 41 4E 49 45 4C (CONTEXT SPECIFIC [0]) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 27418/0x6B1A) (Terminator) Message type: ALERTING (1) [1e 02 81 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 30 (cs0, Progress Indicator) q931.c:3590 q931_receive: call 60186 on channel 2 enters state 4 (Call Delivered) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 27418/0x6B1A) (Terminator) Message type: CONNECT (7) [1e 02 81 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 30 (cs0, Progress Indicator) q931.c:3620 q931_receive: call 60186 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 27418/0x6B1A) (Originator) Message type: CONNECT ACKNOWLEDGE (15) !! Got reject for frame 69, but we have nothing -- resetting! NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3009 q931_disconnect: call 60186 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 27418/0x6B1A) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 27418/0x6B1A) (Terminator) Message type: RELEASE (77) q931.c:3795 q931_receive: call 60186 on channel 2 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 27418/0x6B1A) (Originator) Message type: RELEASE COMPLETE (90) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null !! Got reject for frame 71, but we have nothing -- resetting! James Shigley -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres Sent: Wednesday, June 17, 2009 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue James A. Shigley wrote: Never saw this appear on the list. So just resending it. You might get more help if you include a PRI Debug that shows the call being rejected. Andres http://www.neuroredes.com 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 ___ -- 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 ___ -- 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
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.comwrote:
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 Shigley4099819213) 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=X
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= X
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= _NXX,1,Dial(${belltd}/409${EXTEN})
exten= 9819213,1,Dial(${inf}/409${EXTEN}
[from_test] ; noted but not repaired.
exten= _NXX,1,Set(CALLERID(name)=James Shigley)
exten= _NXX,2,Set(CALLERID(number)=4099819213)
exten= _NXX,3,Set(CALLERID(all)=${CALLERID(name)}${CALLERID(num)})
exten= _NXX,4,Dial(${bell}/${EXTEN})
[from_test] ; noted but not repaired.
exten= _NXX,1,Set(CALLERID(name)=James Shigley)
exten= _NXX,2,Set(CALLERID(number)=4099819213)
exten= _NXX,3,Dial(${bell
Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue
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 Shigley4099819213) 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${EXTEN}). I've tried everything I can think of and
googled for a good while trying to find an explanation for got hangup,
cause 50. What is cause 50?
Sip Login information
[test]
username=test
type=friend
secret=X
callerid=
host=dynamic
nat=no
canreinvite=no
context=from_test
;codecs
disallow=all
allow=ulaw
Also had it as
[test]
username=test
type=friend
secret= X
callerid= James Shigley 4099819213
host=dynamic
nat=no
canreinvite=no
context=from_test
;codecs
disallow=all
allow=ulaw
My From Context has changed several times here is some of the iterations
I've tried.
inf=DAHDI/g2
bell=DAHDI/G3
[from_test] ; noted but not repaired.
exten= _NXX,1,Dial(${belltd}/409${EXTEN})
exten= 9819213,1,Dial(${inf}/409${EXTEN}
[from_test] ; noted but not repaired.
exten= _NXX,1,Set(CALLERID(name)=James
Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue
It errors the same whether I use g or G.
James Shigley
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 Shigley4099819213) 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${EXTEN}). I've tried everything I can think of
and googled for a good while trying to find an explanation for got
hangup, cause 50. What is cause 50?
Sip Login information
[test]
username=test
type=friend
secret=X
callerid=
host=dynamic
nat=no
canreinvite=no
context=from_test
;codecs
disallow=all
allow=ulaw
Also had it as
[test]
username=test
type=friend
secret= X
callerid= James Shigley 4099819213
host=dynamic
nat=no
canreinvite=no
context=from_test
;codecs
disallow=all
allow=ulaw
My From Context has changed several times here is some of the iterations
I've tried.
inf=DAHDI/g2
bell=DAHDI/G3
[from_test] ; noted but not repaired.
exten= _NXX,1,Dial(${belltd}/409${EXTEN})
exten= 9819213,1,Dial(${inf}/409${EXTEN}
[from_test] ; noted but not repaired.
exten= _NXX,1,Set(CALLERID(name)=James Shigley)
exten= _NXX,2,Set(CALLERID(number
Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue
Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 27418/0x6B1A) (Originator) Message type: CONNECT ACKNOWLEDGE (15) !! Got reject for frame 69, but we have nothing -- resetting! NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request The remote switch does not like your CONNECT ACKNOWLEDGE message. I have no idea why but my first guess would be to play around with the 'switchtype' in your chan_dahdi.conf. Another thing to try is to enable/disable 'facilityenable' as well to see if it changes anything. Andres http://www.neuroredes.com ___ -- 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
[asterisk-users] Asterisks, Sip to Local PRI/PTSN issue
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 Shigley4099819213) 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${EXTEN}). I've tried everything I can think of
and googled for a good while trying to find an explanation for got
hangup, cause 50. What is cause 50?
Sip Login information
[test]
username=test
type=friend
secret=X
callerid=
host=dynamic
nat=no
canreinvite=no
context=from_test
;codecs
disallow=all
allow=ulaw
Also had it as
[test]
username=test
type=friend
secret= X
callerid= James Shigley 4099819213
host=dynamic
nat=no
canreinvite=no
context=from_test
;codecs
disallow=all
allow=ulaw
My From Context has changed several times here is some of the iterations
I've tried.
inf=DAHDI/g2
bell=DAHDI/G3
[from_test] ; noted but not repaired.
exten= _NXX,1,Dial(${belltd}/409${EXTEN})
exten= 9819213,1,Dial(${inf}/409${EXTEN}
[from_test] ; noted but not repaired.
exten= _NXX,1,Set(CALLERID(name)=James Shigley)
exten= _NXX,2,Set(CALLERID(number)=4099819213)
exten=
_NXX,3,Set(CALLERID(all)=${CALLERID(name)}${CALLERID(num)})
exten= _NXX,4,Dial(${bell}/${EXTEN})
[from_test] ; noted but not repaired.
exten= _NXX,1,Set(CALLERID(name)=James Shigley)
exten= _NXX,2,Set(CALLERID(number)=4099819213)
exten= _NXX,3,Dial(${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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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Shigley4099819213) 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${EXTEN}). I've tried everything I can think of
and googled for a good while trying to find an explanation for got
hangup, cause 50. What is cause 50?
Sip Login information
[test]
username=test
type=friend
secret=X
callerid=
host=dynamic
nat=no
canreinvite=no
context=from_test
;codecs
disallow=all
allow=ulaw
Also had it as
[test]
username=test
type=friend
secret= X
callerid= James Shigley 4099819213
host=dynamic
nat=no
canreinvite=no
context=from_test
;codecs
disallow=all
allow=ulaw
My From Context has changed several times here is some of the iterations
I've tried.
inf=DAHDI/g2
bell=DAHDI/G3
[from_test] ; noted but not repaired.
exten= _NXX,1,Dial(${belltd}/409${EXTEN})
exten= 9819213,1,Dial(${inf}/409${EXTEN}
[from_test] ; noted but not repaired.
exten= _NXX,1,Set(CALLERID(name)=James Shigley)
exten= _NXX,2,Set(CALLERID(number)=4099819213)
exten=
_NXX,3,Set(CALLERID(all)=${CALLERID(name)}${CALLERID(num)})
exten= _NXX,4,Dial(${bell}/${EXTEN})
[from_test] ; noted but not repaired.
exten= _NXX,1,Set(CALLERID(name)=James Shigley)
exten= _NXX,2,Set(CALLERID(number)=4099819213)
exten= _NXX,3,Dial(${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
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 Shigley4099819213) 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${EXTEN}). I've tried everything I can think of and
googled for a good while trying to find an explanation for got hangup,
cause 50. What is cause 50?
Sip Login information
[test]
username=test
type=friend
secret=X
callerid=
host=dynamic
nat=no
canreinvite=no
context=from_test
;codecs
disallow=all
allow=ulaw
Also had it as
[test]
username=test
type=friend
secret= X
callerid= James Shigley 4099819213
host=dynamic
nat=no
canreinvite=no
context=from_test
;codecs
disallow=all
allow=ulaw
My From Context has changed several times here is some of the iterations
I've tried.
inf=DAHDI/g2
bell=DAHDI/G3
[from_test] ; noted but not repaired.
exten= _NXX,1,Dial(${belltd}/409${EXTEN})
exten= 9819213,1,Dial(${inf}/409${EXTEN}
[from_test] ; noted but not repaired.
exten= _NXX,1,Set(CALLERID(name)=James Shigley)
exten= _NXX,2,Set(CALLERID(number)=4099819213)
exten= _NXX,3,Set(CALLERID(all)=${CALLERID(name)}${CALLERID(num)})
exten= _NXX,4,Dial(${bell}/${EXTEN})
[from_test] ; noted but not repaired.
exten= _NXX,1,Set(CALLERID(name)=James Shigley)
exten= _NXX,2,Set(CALLERID(number)=4099819213)
exten= _NXX,3,Dial(${bell}/${EXTEN})
Note I didn't include the full context only the lines relevant to local
dialing. LD dialing which is sent out sip works just fine. Also I tried
using g3 instead of G3 thinking maybe there was an issue with the high
channels. Though when I do a core show channels there isn't near close to
all the channels used.
One final note. I did try calling other numbers beyond just 9819213 the
errors and issue was the same regardless of the local number dialed.
I think that's all the information you might need, If I forgot something
just let me know. Oh and this is on * 1.6.0.6
James Shigley
Monroe Telephone Answering Service
409-981-9213
___
-- Bandwidth
Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue
James A. Shigley wrote: Never saw this appear on the list. So just resending it. You might get more help if you include a PRI Debug that shows the call being rejected. Andres http://www.neuroredes.com Alright I’ve been having an issue when trying to dial out locally when coming from SIP. This used to work no problem, now it doesn’t. Now the local PRI to Bell Is working fine I have calls coming in and out of it constantly right now. BUT if I try and make a local call from SIP (from X-Lite or one of our Linksys SPA2102s) It fails every time with errors like these ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue
James A. Shigley wrote:
snip
The odd thing is that I can send the call down one of my other PRI ports
to our Amtelco Infinity system. (via exten=
9819213,1,Dial(${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?
snip
According to these sites:
http://www.quintum.com/support/xplatform/ivr_acct/webhelp/Disconnect_Cause_Codes.htm
http://www.cisco.com/en/US/docs/ios/11_0/debug/command/reference/disdn.html
Cause code 50 is:
Requested facility not subscribed - The remote equipment supports the
requested supplementary service by subscription only.
I don't know what that really means, sorry. It could be a setting with
your switchtype or pridialplan in chan_dahdi.conf. Or, it could be
something's not set up right on the telco side.
-Dave
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