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


 
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salah.elharit200 at gm...
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PostPosted: Wed Mar 25, 2015 7:36 am    Post subject: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPC Reply with quote

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@macro-dialout-trunk:23] NoOp("SIP/306-000000b8", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack
    -- Executing [s@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@macro-dialout-trunk:1] Set("SIP/306-000000b8", "RC=34") in new stack
    -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/306-000000b8", "34,1") in new stack
    -- Goto (macro-dialout-trunk,34,1)
    -- Executing [34@macro-dialout-trunk:1] Goto("SIP/306-000000b8", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@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@macro-dialout-trunk:2] Set("SIP/306-000000b8", "CALLERID(number)=306") in new stack
    -- Executing [0149XXXXXX@from-internal:7] Macro("SIP/306-000000b8", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/306-000000b8", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/306-000000b8", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/306-000000b8", "0?intracompany,1") in new stack
    -- Executing [s@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@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@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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mjordan at digium.com
Guest





PostPosted: Wed Mar 25, 2015 7:59 am    Post subject: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPC Reply with quote

On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit
<salah.elharit200@gmail.com> wrote:
Quote:
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
Quote:
0x2afec424c430 -- Probation passed - setting RTP source address to
192.168.1.212:57592
Quote:
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@macro-dialout-trunk:23] NoOp("SIP/306-000000b8", "Dial
failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34")
in new stack
-- Executing [s@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@macro-dialout-trunk:1]
Set("SIP/306-000000b8", "RC=34") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2]
Goto("SIP/306-000000b8", "34,1") in new stack
-- Goto (macro-dialout-trunk,34,1)
-- Executing [34@macro-dialout-trunk:1] Goto("SIP/306-000000b8",
"continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@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@macro-dialout-trunk:2] Set("SIP/306-000000b8",
"CALLERID(number)=306") in new stack
-- Executing [0149XXXXXX@from-internal:7] Macro("SIP/306-000000b8",
"outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/306-000000b8", "") in
new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/306-000000b8",
"0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/306-000000b8",
"0?intracompany,1") in new stack
-- Executing [s@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@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@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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salah.elharit200 at gm...
Guest





PostPosted: Wed Mar 25, 2015 8:23 am    Post subject: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPC Reply with quote

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@digium.com (mjordan@digium.com)>:
Quote:
On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit
<salah.elharit200@gmail.com (salah.elharit200@gmail.com)> wrote:
Quote:
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@macro-dialout-trunk:23] NoOp("SIP/306-000000b8", "Dial
failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34")
in new stack
     -- Executing [s@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@macro-dialout-trunk:1]
Set("SIP/306-000000b8", "RC=34") in new stack
     -- Executing [s-CONGESTION@macro-dialout-trunk:2]
Goto("SIP/306-000000b8", "34,1") in new stack
     -- Goto (macro-dialout-trunk,34,1)
     -- Executing [34@macro-dialout-trunk:1] Goto("SIP/306-000000b8",
"continue,1") in new stack
     -- Goto (macro-dialout-trunk,continue,1)
     -- Executing [continue@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@macro-dialout-trunk:2] Set("SIP/306-000000b8",
"CALLERID(number)=306") in new stack
     -- Executing [0149XXXXXX@from-internal:7] Macro("SIP/306-000000b8",
"outisbusy,") in new stack
     -- Executing [s@macro-outisbusy:1] Progress("SIP/306-000000b8", "") in
new stack
     -- Executing [s@macro-outisbusy:2] GotoIf("SIP/306-000000b8",
"0?emergency,1") in new stack
     -- Executing [s@macro-outisbusy:3] GotoIf("SIP/306-000000b8",
"0?intracompany,1") in new stack
     -- Executing [s@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@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@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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asterisk_list at earth...
Guest





PostPosted: Wed Mar 25, 2015 8:48 am    Post subject: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPC Reply with quote

** THIS IS NOT WHERE YOUR REPLY BELONGS **

On Wednesday 25 Mar 2015, Salaheddine Elharit wrote:
Quote:
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
Back to top
salah.elharit200 at gm...
Guest





PostPosted: Wed Mar 25, 2015 11:24 am    Post subject: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPC Reply with quote

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@earthshod.co.uk (asterisk_list@earthshod.co.uk)>:
Quote:
** THIS IS NOT WHERE YOUR REPLY BELONGS **

On Wednesday 25 Mar 2015, Salaheddine Elharit wrote:
Quote:
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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