TRUNK Dial Failed Due To CONGESTION HANGUPCAUSE: 34

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Asterisk Users 4 Comments

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”, “RC4”) 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)06”) 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

4 thoughts on - TRUNK Dial Failed Due To CONGESTION HANGUPCAUSE: 34

  • 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.

  • 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 :

  • ** THIS IS NOT WHERE YOUR REPLY BELONGS **

    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.

  • 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 :