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 4 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 2.4 years ago 0 Answer

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 2.6 years ago 8 Answer

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 2.7 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 2.9 years ago 1 Answer