Issue With Inbound Route

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hello liste

i have creat i trunk sip and inboun route for inbound calls the issue whe i use the DID in inboud route i have a error No DID or CID Match.

but when i leave this DID field blank i can route the call without any issue

how can ido in order to use DID in route inboud "i use elastix"

Executing [s@from-trunk:1] NoOp("SIP/358-106-000000c0", "No DID or CID Match") in new stack -- Executing [s@from-trunk:2] Answer("SIP/358-106-000000c0", "") in new stack -- Executing [s@from-trunk:3] Wait("SIP/358-106-000000c0", "2") in new stack > 0x2add5020a390 -- Probation passed - setting RTP source address to 217.xxx.xx.xxx:207xx --…

Asterisk Users 6 months ago 1 Answer

Detecting Fax Without Aswer()ing The Call First?

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Trying to make the fax detection work. My current setup (with no fax) is done without Answer(), so the call is answered only when someone actually picks-up the phone. But when the incoming call is fax, I can her the tone and call is never forwarded to "Fax" extension. But... Strange thing happens when I (mistakenly) put a call on hold:

-- Executing [youngandson-test@incoming:2] Gosub("SIP/66.193.176.35-000000b8", "process-callerid,s,1") in new stack -- Executing [s@process-callerid:1] Verbose("SIP/66.193.176.35-000000b8", "3,- Original CallerID: "FREE CALL TOLL" <18009806858> ") in new stack -- - Original CallerID: "FREE CALL TOLL" <18009806858> -- Executing [s@process-callerid:2] GotoIf("SIP/66.193.176.35-000000b8", "1?4") in new stack…

Asterisk Users 2.6 years ago 0 Answers

Queue Joinempty, Even After AddQueueMember

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Hello,

I add a member to a Queue with AddQueueMember, but the Queue still indicates "joinempty" :

Add member to queue :

/-- Executing [queueadd@sub-GetParams:2] AddQueueMember("SIP/sip17-00005c1e", "myqueue11,member3") in new stack -- Executing [queueadd@sub-GetParams:3] NoOp("SIP/sip17-00005c1e", "AQMSTATUS = ADDED") in new stack/

... but JOINEMPTY when entering the Call Queue :

/-- Executing [queue@pbx-routing:4] Queue("SIP/SipIncoming-00005da9", "myqueue11,,,,60") in new stack -- Executing [queue@pbx-routing:5] NoOp("SIP/SipIncoming-00005da9", "queuestatus == JOINEMPTY") in new stack/

How is this possible ?



Kind regards, Jonas.

Asterisk Users 2.8 years ago 8 Answers

503 Unable To Load

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Can any one suggest me what I have to do for this issue. There is no nat as i have directly connected to internert without firewall.

Got SIP response 503 "Unable to load gateways" back from xxx.xxx.xxx.xxx:5060 -- SIP/outbound-00000994 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-dialout-trunk:23] NoOp("SIP/1002-00000993", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack -- Executing [s@macro-dialout-trunk:24] Goto("SIP/1002-00000993", "s-CONGESTION,1") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/1002-00000993", "RC4") in new stack -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/1002-00000993", "34,1") in new stack -- Goto (macro-dialout-trunk,34,1) --…

Asterisk Users 2.9 years ago 1 Answer

RemoveQueueMember And Realtime Queues

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Hello,

using asterisk 1.8.11.1 using realtime Queues

When trying to remove a queue member, I get the following :

-- Executing [122@from-TESTCORP:2] RemoveQueueMember("SIP/testcorp5-0000000c", "testcorpq1,SIP/testcorp7") in new stack WARNING[18788]: app_queue.c:5653 rqm_exec: Unable to remove interface from queue 'testcorpq1': 'SIP/testcorp7' is not a dynamic member

How can one remove a queue member when using realtime queues ?

Extra question : adding a queue member to a queue, will AddQueueMember work ?

-- Executing [122@from-TESTCORP:5] AddQueueMember("SIP/testcorp5-0000000c", "testcorpq1,SIP/testcorp7") in new stack WARNING[18788]: app_queue.c:5708 aqm_exec: Unable to add interface 'SIP/testcorp7' to queue 'testcorpq1': Already there -- Executing [122@from-TESTCORP:6] NoOp("SIP/testcorp5-0000000c", "AQMSTATUS = MEMBERALREADY") in new stack



Kind regards, Jonas.

Asterisk Users 3.1 years ago 1 Answer

Unable to create channel of type 'IAX2' (cause 20 - Unknown)

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I have two asterisk servers connected via iax.
home_server < => IAX2 < => clinic_server I'm just testing, calling from "home_server" via "clinic_server" but I'm getting an error message:
Call gets through to "clinic_server" but will not call back. Dial("IAX2/home_server-957", "IAX2/home_server/218,30,rw") in new stack
[Apr 14 16:59:45] WARNING[27870]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)

Asterisk Users 3.4 years ago 9 Answers

Diagnosing call hangups

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Hi, I have customers complaining of random call hangups, and I am trying to
determine in each case what happened to the call. I normally run a
customer's PBX with verbose set to 3 and debug off, and looking at each
case in the "full" log (default full in logger.conf) I don't see
anything special between the call being answered and the call
terminating - the log lines look exactly the same whether the local side
hung up on purpose, the remote side hung up on purpose, or something
unknown happened…

Asterisk Users 3.7 years ago 1 Answer

AGI Problem

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On Sep 23, 2011, at 8:01 PM, Mehmet Avcioglu wrote:
> I have an AGI script that occasionally disappears without completing its action and asterisk logs the following.
>
> AGI Script script.php completed, returning 4
> Spawn extension (context, 0123456, 2) exited non-zero on 'Local/0123456@context-f46e;1'
I also changed the dialplan and added a line to print AGISTATUS, but when this "returning 4" happens, asterisk stops there and doesn't execute any further dialplan actions, so I don't even see AGISTATUS value. exten => h,1,AGI(script.php,${ANSWEREDTIME},,,)
exten => h,n,NoOp(${AGISTATUS}) Executing [h@context:1] AGI("Local/0123456@context-4b79;1",…

Asterisk Users 4 years ago 6 Answers

Error deleting key from database

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Hi Everybody, Don't know why this DBdeltree error appears on Asterisk CLI.Good part is, it
does remove family entry from AstDB. Sample Dialplan.... exten => 1212,1,Noop()
same => n,Set(TEST=1234)
same => n,Set(DB(${TEST}/TESTSTART)=${STRFTIME(${EPOCH},,%Y-%m-%d
%H:%M:%S)})
same => n,DBdeltree(${TEST})
same => n,Hangup() Asterisk CLI output.... [Jun 1 14:30:39] == Using SIP RTP CoS mark 5
[Jun 1 14:30:39] -- Executing [1212@incoming-did:1]
NoOp("SIP/29-0000002f", "") in new stack
[Jun 1 14:30:39] -- Executing [1212@incoming-did:2]
Set("SIP/29-0000002f", "TEST=1234") in new stack
[Jun 1 14:30:39] -- Executing [1212@incoming-did:3]
Set("SIP/29-0000002f", "DB(1234/TESTSTART)=2011-06-01 14:30:39") in new stack
[Jun…

Asterisk Users 4.3 years ago 0 Answers

BRI confiugration error

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Hi sir, I was installed Goautodial server and I have b410p BRI card. BRI card
showing OK with dahdi_tool, this NT mode.
whenever I am dialing from server i am not able to connect the call . in Cli
below mention warning is comming .
please what is the mistake with me . help me
Executing [0559566768@default:1] AGI("Console/dsp", "agi://
127.0.0.1:4577/call_log") in new
stack

Asterisk Users 4.3 years ago 4 Answers