thr3ads.net - asterisk users - [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 [Mar 2015]

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Salaheddine Elharit

2015-Mar-25 12:35 UTC

[asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

hello list,

i have asterisk 11.15.0 and i have some trunks sip from my provider

we have some ip phone astra 6731i

each Ip-phone is configured with trunk and we call

no ihave configured another trunk from the same provider in my asterisk

i can call all numbers just the numbers are configured in thses ip phones.

but when i configured the same trunk in x-lite i can call theses ip-phones
without issue
 the problem just when i configure the trunk in my server and i use
extension

all the ip-phone and x-lite and server asterisk in the same network
192.168.1.x

 == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/FD/0033149XXXXXX
    -- SIP/FD-000000b9 is making progress passing it to SIP/306-000000b8
       > 0x2afec424c430 -- Probation passed - setting RTP source address to
192.168.1.212:57592
       > 0xc5922b0 -- Probation passed - setting RTP source address to
217.195.xx.xxx:29674
    -- Got SIP response 556 "No address found" back from
217.195.XX.XXX:5060
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s at macro-dialout-trunk:23]
NoOp("SIP/306-000000b8", "Dial
failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34")
in new stack
    -- Executing [s at macro-dialout-trunk:24]
GotoIf("SIP/306-000000b8",
"0?continue,1:s-CONGESTION,1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing [s-CONGESTION at macro-dialout-trunk:1]
Set("SIP/306-000000b8", "RC=34") in new stack
    -- Executing [s-CONGESTION at macro-dialout-trunk:2]
Goto("SIP/306-000000b8", "34,1") in new stack
    -- Goto (macro-dialout-trunk,34,1)
    -- Executing [34 at macro-dialout-trunk:1]
Goto("SIP/306-000000b8",
"continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue at macro-dialout-trunk:1]
NoOp("SIP/306-000000b8",
"TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to
other trunks") in new stack
    -- Executing [continue at macro-dialout-trunk:2]
Set("SIP/306-000000b8",
"CALLERID(number)=306") in new stack
    -- Executing [0149XXXXXX at from-internal:7]
Macro("SIP/306-000000b8",
"outisbusy,") in new stack
    -- Executing [s at macro-outisbusy:1] Progress("SIP/306-000000b8",
"") in
new stack
    -- Executing [s at macro-outisbusy:2] GotoIf("SIP/306-000000b8",
"0?emergency,1") in new stack
    -- Executing [s at macro-outisbusy:3] GotoIf("SIP/306-000000b8",
"0?intracompany,1") in new stack
    -- Executing [s at macro-outisbusy:4] Playback("SIP/306-000000b8",
"all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
[2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:701
ast_openstream_full: File all-circuits-busy-now does not exist in any format
[2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:1017
ast_streamfile: Unable to open all-circuits-busy-now (format (ulaw)): No
such file or directory
[2015-03-25 12:18:31] WARNING[25161][C-0000006d]: app_playback.c:484
playback_exec: ast_streamfile failed on SIP/306-000000b8 for
all-circuits-busy-now&pls-try-call-later, noanswer
[2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:701
ast_openstream_full: File pls-try-call-later does not exist in any format
[2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:1017
ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No such
file or directory
[2015-03-25 12:18:31] WARNING[25161][C-0000006d]: app_playback.c:484
playback_exec: ast_streamfile failed on SIP/306-000000b8 for
all-circuits-busy-now&pls-try-call-later, noanswer
    -- Executing [s at macro-outisbusy:5]
Congestion("SIP/306-000000b8", "20")
in new stack
[2015-03-25 12:18:31] WARNING[25161][C-0000006d]: channel.c:4862 ast_prod:
Prodding channel 'SIP/306-000000b8' failed
  == Spawn extension (macro-outisbusy, s, 5) exited non-zero on
'SIP/306-000000b8' in macro 'outisbusy'
  == Spawn extension (from-internal, 0149XXXXXX, 7) exited non-zero on
'SIP/306-000000b8'
    -- Executing [h at from-internal:1] Hangup("SIP/306-000000b8",
"") in new
stack
  == Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/306-000000b8'
  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording SIP/306-000000b8
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Matthew Jordan

2015-Mar-25 12:59 UTC

head link

[asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit
<salah.elharit200 at gmail.com> wrote:> hello list,
>
> i have asterisk 11.15.0 and i have some trunks sip from my provider
>
> we have some ip phone astra 6731i
>
> each Ip-phone is configured with trunk and we call
>
> no ihave configured another trunk from the same provider in my asterisk
>
> i can call all numbers just the numbers are configured in thses ip phones.
>
> but when i configured the same trunk in x-lite i can call theses ip-phones
> without issue
>  the problem just when i configure the trunk in my server and i use
> extension
>
> all the ip-phone and x-lite and server asterisk in the same network
> 192.168.1.x
>
>  == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
>     -- Called SIP/FD/0033149XXXXXX
>     -- SIP/FD-000000b9 is making progress passing it to SIP/306-000000b8
>        > 0x2afec424c430 -- Probation passed - setting RTP source address
to
> 192.168.1.212:57592
>        > 0xc5922b0 -- Probation passed - setting RTP source address to
> 217.195.xx.xxx:29674
>     -- Got SIP response 556 "No address found" back from
217.195.XX.XXX:5060
>   == Everyone is busy/congested at this time (1:0/1/0)
>     -- Executing [s at macro-dialout-trunk:23]
NoOp("SIP/306-000000b8", "Dial
> failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE =
34")
> in new stack
>     -- Executing [s at macro-dialout-trunk:24]
GotoIf("SIP/306-000000b8",
> "0?continue,1:s-CONGESTION,1") in new stack
>     -- Goto (macro-dialout-trunk,s-CONGESTION,1)
>     -- Executing [s-CONGESTION at macro-dialout-trunk:1]
> Set("SIP/306-000000b8", "RC=34") in new stack
>     -- Executing [s-CONGESTION at macro-dialout-trunk:2]
> Goto("SIP/306-000000b8", "34,1") in new stack
>     -- Goto (macro-dialout-trunk,34,1)
>     -- Executing [34 at macro-dialout-trunk:1]
Goto("SIP/306-000000b8",
> "continue,1") in new stack
>     -- Goto (macro-dialout-trunk,continue,1)
>     -- Executing [continue at macro-dialout-trunk:1]
NoOp("SIP/306-000000b8",
> "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through
to
> other trunks") in new stack
>     -- Executing [continue at macro-dialout-trunk:2]
Set("SIP/306-000000b8",
> "CALLERID(number)=306") in new stack
>     -- Executing [0149XXXXXX at from-internal:7]
Macro("SIP/306-000000b8",
> "outisbusy,") in new stack
>     -- Executing [s at macro-outisbusy:1]
Progress("SIP/306-000000b8", "") in
> new stack
>     -- Executing [s at macro-outisbusy:2]
GotoIf("SIP/306-000000b8",
> "0?emergency,1") in new stack
>     -- Executing [s at macro-outisbusy:3]
GotoIf("SIP/306-000000b8",
> "0?intracompany,1") in new stack
>     -- Executing [s at macro-outisbusy:4]
Playback("SIP/306-000000b8",
> "all-circuits-busy-now&pls-try-call-later, noanswer") in new
stack
> [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:701
> ast_openstream_full: File all-circuits-busy-now does not exist in any
format
> [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:1017
> ast_streamfile: Unable to open all-circuits-busy-now (format (ulaw)): No
> such file or directory
> [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: app_playback.c:484
> playback_exec: ast_streamfile failed on SIP/306-000000b8 for
> all-circuits-busy-now&pls-try-call-later, noanswer
> [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:701
> ast_openstream_full: File pls-try-call-later does not exist in any format
> [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:1017
> ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No such
> file or directory
> [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: app_playback.c:484
> playback_exec: ast_streamfile failed on SIP/306-000000b8 for
> all-circuits-busy-now&pls-try-call-later, noanswer
>     -- Executing [s at macro-outisbusy:5]
Congestion("SIP/306-000000b8", "20")
> in new stack
> [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: channel.c:4862 ast_prod:
> Prodding channel 'SIP/306-000000b8' failed
>   == Spawn extension (macro-outisbusy, s, 5) exited non-zero on
> 'SIP/306-000000b8' in macro 'outisbusy'
>   == Spawn extension (from-internal, 0149XXXXXX, 7) exited non-zero on
> 'SIP/306-000000b8'
>     -- Executing [h at from-internal:1]
Hangup("SIP/306-000000b8", "") in new
> stack
>   == Spawn extension (from-internal, h, 1) exited non-zero on
> 'SIP/306-000000b8'
>   == MixMonitor close filestream (mixed)
>   == End MixMonitor Recording SIP/306-000000b8
>
The verbose output states why your call is congested:

    -- Got SIP response 556 "No address found" back from
217.195.XX.XXX:5060

The far end came back with a 556 response to the outbound INVITE
request. It doesn't think that whatever you dialled exists.

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

Salaheddine Elharit

2015-Mar-25 13:23 UTC

head link

[asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

tnaks for your response but the number dialed exist and i can call this
number when i configure the trunk directly in x-lite and i call call also
this number from my cell phone .
any help
thanks and regards

2015-03-25 12:59 GMT+00:00 Matthew Jordan <mjordan at digium.com>:
> On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit
> <salah.elharit200 at gmail.com> wrote:
> > hello list,
> >
> > i have asterisk 11.15.0 and i have some trunks sip from my provider
> >
> > we have some ip phone astra 6731i
> >
> > each Ip-phone is configured with trunk and we call
> >
> > no ihave configured another trunk from the same provider in my
asterisk
> >
> > i can call all numbers just the numbers are configured in thses ip
> phones.
> >
> > but when i configured the same trunk in x-lite i can call theses
> ip-phones
> > without issue
> >  the problem just when i configure the trunk in my server and i use
> > extension
> >
> > all the ip-phone and x-lite and server asterisk in the same network
> > 192.168.1.x
> >
> >  == Using SIP RTP TOS bits 184
> >   == Using SIP RTP CoS mark 5
> >     -- Called SIP/FD/0033149XXXXXX
> >     -- SIP/FD-000000b9 is making progress passing it to
SIP/306-000000b8
> >        > 0x2afec424c430 -- Probation passed - setting RTP source
address
> to
> > 192.168.1.212:57592
> >        > 0xc5922b0 -- Probation passed - setting RTP source address
to
> > 217.195.xx.xxx:29674
> >     -- Got SIP response 556 "No address found" back from
> 217.195.XX.XXX:5060
> >   == Everyone is busy/congested at this time (1:0/1/0)
> >     -- Executing [s at macro-dialout-trunk:23]
NoOp("SIP/306-000000b8",
> "Dial
> > failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE
> 34")
> > in new stack
> >     -- Executing [s at macro-dialout-trunk:24]
GotoIf("SIP/306-000000b8",
> > "0?continue,1:s-CONGESTION,1") in new stack
> >     -- Goto (macro-dialout-trunk,s-CONGESTION,1)
> >     -- Executing [s-CONGESTION at macro-dialout-trunk:1]
> > Set("SIP/306-000000b8", "RC=34") in new stack
> >     -- Executing [s-CONGESTION at macro-dialout-trunk:2]
> > Goto("SIP/306-000000b8", "34,1") in new stack
> >     -- Goto (macro-dialout-trunk,34,1)
> >     -- Executing [34 at macro-dialout-trunk:1]
Goto("SIP/306-000000b8",
> > "continue,1") in new stack
> >     -- Goto (macro-dialout-trunk,continue,1)
> >     -- Executing [continue at macro-dialout-trunk:1]
> NoOp("SIP/306-000000b8",
> > "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing
through to
> > other trunks") in new stack
> >     -- Executing [continue at macro-dialout-trunk:2]
> Set("SIP/306-000000b8",
> > "CALLERID(number)=306") in new stack
> >     -- Executing [0149XXXXXX at from-internal:7]
Macro("SIP/306-000000b8",
> > "outisbusy,") in new stack
> >     -- Executing [s at macro-outisbusy:1]
Progress("SIP/306-000000b8", "")
> in
> > new stack
> >     -- Executing [s at macro-outisbusy:2]
GotoIf("SIP/306-000000b8",
> > "0?emergency,1") in new stack
> >     -- Executing [s at macro-outisbusy:3]
GotoIf("SIP/306-000000b8",
> > "0?intracompany,1") in new stack
> >     -- Executing [s at macro-outisbusy:4]
Playback("SIP/306-000000b8",
> > "all-circuits-busy-now&pls-try-call-later, noanswer") in
new stack
> > [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:701
> > ast_openstream_full: File all-circuits-busy-now does not exist in any
> format
> > [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:1017
> > ast_streamfile: Unable to open all-circuits-busy-now (format (ulaw)):
No
> > such file or directory
> > [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: app_playback.c:484
> > playback_exec: ast_streamfile failed on SIP/306-000000b8 for
> > all-circuits-busy-now&pls-try-call-later, noanswer
> > [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:701
> > ast_openstream_full: File pls-try-call-later does not exist in any
format
> > [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: file.c:1017
> > ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No
> such
> > file or directory
> > [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: app_playback.c:484
> > playback_exec: ast_streamfile failed on SIP/306-000000b8 for
> > all-circuits-busy-now&pls-try-call-later, noanswer
> >     -- Executing [s at macro-outisbusy:5]
Congestion("SIP/306-000000b8",
> "20")
> > in new stack
> > [2015-03-25 12:18:31] WARNING[25161][C-0000006d]: channel.c:4862
> ast_prod:
> > Prodding channel 'SIP/306-000000b8' failed
> >   == Spawn extension (macro-outisbusy, s, 5) exited non-zero on
> > 'SIP/306-000000b8' in macro 'outisbusy'
> >   == Spawn extension (from-internal, 0149XXXXXX, 7) exited non-zero on
> > 'SIP/306-000000b8'
> >     -- Executing [h at from-internal:1]
Hangup("SIP/306-000000b8", "") in
> new
> > stack
> >   == Spawn extension (from-internal, h, 1) exited non-zero on
> > 'SIP/306-000000b8'
> >   == MixMonitor close filestream (mixed)
> >   == End MixMonitor Recording SIP/306-000000b8
> >
>
> The verbose output states why your call is congested:
>
>     -- Got SIP response 556 "No address found" back from
> 217.195.XX.XXX:5060
>
> The far end came back with a 556 response to the outbound INVITE
> request. It doesn't think that whatever you dialled exists.
>
> --
> Matthew Jordan
> Digium, Inc. | Director of Technology
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
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