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Salaheddine Elharit

2015-Mar-25 13:23 UTC

[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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A J Stiles

2015-Mar-25 13:47 UTC

head link

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

** THIS IS NOT WHERE YOUR REPLY BELONGS **

On Wednesday 25 Mar 2015, Salaheddine Elharit wrote:> 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
Make sure you are sending the number in the correct format, when you Dial() 
via your trunk.  Some providers want you to omit the leading zero from the STD 
code.  Others want you to include it.  Others still want you to include the 
IDD code  (and then definitely leave out the 0, just like you were phoning home 
from abroad).

My home phone number is (01332) XXXXXX.  To call it, you might have to Dial() 
any of the following  (assuming OUTSIDE is defined elsewhere):

Dial(${OUTSIDE}/01332XXXXXX, 60)		; with leading 0
Dial(${OUTSIDE}/1332XXXXXX, 60)		; without leading 0
Dial(${OUTSIDE}/441332XXXXXX, 60)	; with IDD code

If you don't know what format your telco are expecting and have to determine
by experiment, it probably would be easiest to set up an extension which just 
makes a call to one fixed number -- your own mobile is as good as anything 
else.

To remove the leading 0 from ${EXTEN} , you can use ${EXTEN:1} which omits one 
digit from the beginning.  

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

Salaheddine Elharit

2015-Mar-25 16:24 UTC

head link

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

thank you for your response but i think that the issue is related to the
RTP because i can call all numbers with the same format

when i call any number except 0033149xxxxxx i get the same adress from
provider  only with this number cnfigurerd in ip-phone in our network i get
this error

best regards

number works without issue

 Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/FD/0033661223291
    -- SIP/FD-0000011f is making progress passing it to SIP/306-0000011e
       > 0x2afee8182fa0 -- Probation passed - setting RTP source address to
192.168.1.212:12728         ip adress of my x-lite
       > 0x2afee822e480 -- Probation passed - setting RTP source address to
217.195.31.148:43486        ip adress of provider
        SIP/FD-0000011f answered SIP/306-0000011e
       > 0x2afee822e480 -- Probation passed - setting RTP source address to
217.195.31.148:43486         the same ip adress and the same port




number with error

 Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5


- Called SIP/FD/0033149xxxxxx
   SIP/FD-0000011d is making progress passing it to SIP/306-0000011c
     > 0x2afee8182fa0 -- Probation passed - setting RTP source address to
192.168.1.212:47452    ip adress of my x-lite
     > 0xc7452e0 -- Probation passed - setting RTP source address to
217.195.31.146:23392        ip adress of provider
     Got SIP response 556 "No address found" back from
217.195.31.129:5060
                      not the same ip and port

2015-03-25 13:47 GMT+00:00 A J Stiles <asterisk_list at earthshod.co.uk>:
> ** THIS IS NOT WHERE YOUR REPLY BELONGS **
>
> On Wednesday 25 Mar 2015, Salaheddine Elharit wrote:
> > 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
>
> Make sure you are sending the number in the correct format, when you Dial()
> via your trunk.  Some providers want you to omit the leading zero from the
> STD
> code.  Others want you to include it.  Others still want you to include the
> IDD code  (and then definitely leave out the 0, just like you were phoning
> home
> from abroad).
>
> My home phone number is (01332) XXXXXX.  To call it, you might have to
> Dial()
> any of the following  (assuming OUTSIDE is defined elsewhere):
>
> Dial(${OUTSIDE}/01332XXXXXX, 60)                ; with leading 0
> Dial(${OUTSIDE}/1332XXXXXX, 60)         ; without leading 0
> Dial(${OUTSIDE}/441332XXXXXX, 60)       ; with IDD code
>
> If you don't know what format your telco are expecting and have to
> determine
> by experiment, it probably would be easiest to set up an extension which
> just
> makes a call to one fixed number -- your own mobile is as good as anything
> else.
>
> To remove the leading 0 from ${EXTEN} , you can use ${EXTEN:1} which omits
> one
> digit from the beginning.
>
> --
> AJS
>
> Note:  Originating address only accepts e-mail from list!  If replying off-
> list, change address to asterisk1list at earthshod dot co dot uk .
>
> --
> _____________________________________________________________________
> -- 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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