* You are viewing Posts Tagged ‘stack’

Detecting Fax Without Aswer()ing The Call First?

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
— Goto (process-callerid,s,4)
— Executing [s@process-callerid:4] GotoIf(“SIP/66.193.176.35-000000b8″,
“0?8″) in new stack
— Executing [s@process-callerid:5] GotoIf(“SIP/66.193.176.35-000000b8″,
“0?8″) in new stack
— Executing [s@process-callerid:6] GotoIf(“SIP/66.193.176.35-000000b8″,
“1?7:8″) in new stack
— Goto (process-callerid,s,7)
— Executing [s@process-callerid:7] Set(“SIP/66.193.176.35-000000b8″,
“CALLERID(num)009806858″) in new stack
— Executing [s@process-callerid:8] Return(“SIP/66.193.176.35-000000b8″, “”)
in new stack
— Executing [youngandson-test@incoming:3]
Macro(“SIP/66.193.176.35-000000b8″, “stdexten,210,sip/ra2501″) in new stack
— Executing [s@macro-stdexten:1] Dial(“SIP/66.193.176.35-000000b8″,
“sip/ra2501,360″) in new stack
== Using SIP RTP CoS mark 5
— Called sip/ra2501
— SIP/ra2501-000000b9 is ringing
[2013-02-24 17:05:12] WARNING[6554]: chan_sip.c:8979 process_sdp: ignoring
‘video’ media offer because port number is zero
— SIP/ra2501-000000b9 answered SIP/66.193.176.35-000000b8
— Locally bridging SIP/66.193.176.35-000000b8 and SIP/ra2501-000000b9
[2013-02-24 17:05:31] WARNING[6554]: chan_sip.c:8979 process_sdp: ignoring
‘video’ media offer because port number is zero
[2013-02-24 17:05:31] WARNING[6554]: chan_sip.c:8945 process_sdp: ignoring
‘audio’ media offer because port number is zero
— Started music on hold, class ‘default’, on channel
‘SIP/66.193.176.35-000000b8′
[2013-02-24 17:05:31] WARNING[6532]: res_musiconhold.c:659 monmp3thread: poll()
failed: Interrupted system call
[2013-02-24 17:05:31] WARNING[6532]: res_musiconhold.c:659 monmp3thread: poll()
failed: Interrupted system call
….
….
….
[2013-02-24 17:05:34] WARNING[6532]: res_musiconhold.c:659 monmp3thread: poll()
failed: Interrupted system call
[2013-02-24 17:05:34] WARNING[6532]: res_musiconhold.c:659 monmp3thread: poll()
failed: Interrupted system call
== Redirecting ‘SIP/66.193.176.35-000000b8′ to fax extension due to CNG
detection
— Stopped music on hold on SIP/66.193.176.35-000000b8
== Spawn extension (incoming, fax, 1) exited non-zero on
‘SIP/66.193.176.35-000000b8′ in macro ‘stdexten’
== Spawn extension (incoming, fax, 1) exited non-zero on
‘SIP/66.193.176.35-000000b8′
— Executing [fax@incoming:1] Gosub(“SIP/66.193.176.35-000000b8″,
“receive-fax,fax,1″) in new stack
— Executing [fax@receive-fax:1] Verbose(“SIP/66.193.176.35-000000b8″,
“3,Incoming fax from 18009806858″) in new stack
Incoming fax from 18009806858
— Executing [fax@receive-fax:2] Set(“SIP/66.193.176.35-000000b8″,
“FAXDEST=/var/spool/fax/incoming”) in new stack
— Executing [fax@receive-fax:3] Set(“SIP/66.193.176.35-000000b8″,
“FAX-FILENAME 130224-170534 Incoming Fax”) in new stack
— Executing [fax@receive-fax:4] ReceiveFAX(“SIP/66.193.176.35-000000b8″,
“/var/spool/fax/incoming/20130224-170534 Incoming Fax.tif”) in new stack
— Channel ‘SIP/66.193.176.35-000000b8′ receiving FAX
‘/var/spool/fax/incoming/20130224-170534 Incoming Fax.tif’
== Using UDPTL CoS mark 5
== Spawn extension (receive-fax, fax, 4) exited non-zero on
‘SIP/66.193.176.35-000000b8′

Fax is suddenly detected and received! (Im not sure why all these warnings came up, something misconfigured in music on hold…)

Is there any way to make Asterisk “listen” for CNG tone during the connected call, eliminating the need for Answer() and Wait()?
Is the fax detection completely impossible when compressed codec (g729, gsm…)
is in use? I’ve read its unreliable but does not work at all for me.
(Asterisk 1.8.13 installed from Debian repository)

Thanks …. Martin

Queue Joinempty, Even After AddQueueMember

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.

503 Unable To Load

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)
– Executing [34@macro-dialout-trunk:1] Goto(“SIP/1002-00000993″,
“continue,1″) in new stack
– Goto (macro-dialout-trunk,continue,1)
– Executing [continue@macro-dialout-trunk:1] GotoIf(“SIP/1002-00000993″,
“1?noreport”) in new stack
– Goto (macro-dialout-trunk,continue,3)
– Executing [continue@macro-dialout-trunk:3] NoOp(“SIP/1002-00000993″,
“TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 – failing through to other trunks”) in new stack
– Executing [continue@macro-dialout-trunk:4] Set(“SIP/1002-00000993″,
“CALLERID(number)02″) in new stack
– Executing [8834404@from-internal:6] Macro(“SIP/1002-00000993″,
“outisbusy,”) in new stack
– Executing [s@macro-outisbusy:1] Progress(“SIP/1002-00000993″, “”) in new stack
– Executing [s@macro-outisbusy:2] Playback(“SIP/1002-00000993″,
“all-circuits-busy-now,noanswer”) in new stack
– Playing ‘all-circuits-busy-now.ulaw’ (language ‘en’)
– Executing [s@macro-outisbusy:3] Playback(“SIP/1002-00000993″,
“pls-try-call-later,noanswer”) in new stack
– Playing ‘pls-try-call-later.ulaw’ (language ‘en’)
[2012-11-05 11:24:37] WARNING[29848]: file.c:766 ast_readaudio_callback:
Failed to write frame
== Spawn extension (macro-outisbusy, s, 3) exited non-zero on
‘SIP/1002-00000993′ in macro ‘outisbusy’
== Spawn extension (from-internal, 8834404, 6) exited non-zero on
‘SIP/1002-00000993′
– Executing [h@from-internal:1] Hangup(“SIP/1002-00000993″, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on
‘SIP/1002-00000993′
== Extension Changed auto_hint_1002[from-internal] new state Idle for Notify User 1002
== Extension Changed auto_hint_1002[from-internal] new state Idle for Notify User 1001

Darin

RemoveQueueMember And Realtime Queues

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.

Unable to create channel of type ‘IAX2′ (cause 20 – Unknown)

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)

Diagnosing call hangups

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 and the call simply terminates (FreePBX dialplan in
place):

Examples from our lab today:

[Dec 21 14:17:31] VERBOSE[12854] logger.c: — SIP/astnorth-00000f70
answered SIP/100-00000f6f

LOCAL SIDE HUNGUP

[Dec 21 14:17:37] VERBOSE[12854] logger.c: — Executing
[h@macro-dialout-trunk:1] Macro(“SIP/100-00000f6f”, “hangupcall|”) in
new stack