SIP Not Answering On One Trunk.

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Asterisk Users 2 Comments

I have 2 analog trunks.

They answer the incoming call, do the welcome message, ask for the extension, when a valid extension is entered it rings the right SIP
phone BUT
when the SIP phone is answered, the SIP phone keeps ringing and the call is not connected. If the phone is not answered it goes to voicemail correctly.

[DID_866nnnnnnn] IAX that works include = DID_866nnnnnnn_default
[DID_866nnnnnnn_default]
exten => 866nnnnn,1,Goto(voicemenu-artifact-en,s,1)

[DID_trunk_1]
include = DID_trunk_1_default
[DID_trunk_1_default]
exten => s,1,Goto(voicemenu-artifact-fr,s,1)

[DID_trunk_2]
include = DID_trunk_2_default
[DID_trunk_2_default]
exten => s,1,Goto(voicemenu-home,s,1)

[voicemenu-artifact-en]
;ArtifactEnglishFirst include = default include = conferences exten = s,1,Answer exten = s,n,Set(CALLERID(name)=Art-${CALLERID(name)})
exten = s,n,Wait(0.5)
exten = s,n,Background(record/HelloArtifactEnglish)
exten = s,n(menu),Background(record/DialExtensionEnglish)
exten = s,n,WaitExten(3)
exten = 0,1,Goto(inbound-reception,s,1)
exten = 9,1,Goto(changeLanguageFrArtifact,s,1)
exten = #,1,Directory(default,default,f)
exten = t,1,Goto(inbound-reception,s,1)
exten = i,1,Goto(voicemenu-artifact-en,s,menu)

My IAX trunks work.

Log of dialing in on Trunk2 – answering SIP 102 and waiting while it continued to ring.

[2012-11-27 14:43:52] verbose[3589] sig_analog.c: — Starting simple switch on ‘DAHDI/2-1’
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: — Executing
[s@DID_trunk_2:1] Goto(“DAHDI/2-1”, “voicemenu-home,s,1”) in new stack
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: — Goto (voicemenu-home,s,1)
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: — Executing
[s@voicemenu-home:1] Answer(“DAHDI/2-1”, “”) in new stack
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: — Executing
[s@voicemenu-home:2] Set(“DAHDI/2-1”, “CALLERID(name)=Home-ARTIFACT
LOGICI”) in new stack
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: — Executing
[s@voicemenu-home:3] Wait(“DAHDI/2-1”, “0.5”) in new stack
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: — Executing
[s@voicemenu-home:4] BackGround(“DAHDI/2-1”,
“record/HelloAnnetteAndRon”) in new stack
[2012-11-27 14:43:53] VERBOSE[3589] file.c: — Playing
‘record/HelloAnnetteAndRon.ulaw’ (language ‘en’)
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: == CDR updated on DAHDI/2-1
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: — Executing
[102@voicemenu-home:1] Macro(“DAHDI/2-1”, “stdexten,102,SIP/102”) in new stack
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: — Executing
[s@macro-stdexten:1] Set(“DAHDI/2-1”, “__DYNAMIC_FEATURES=”) in new stack
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: — Executing
[s@macro-stdexten:2] Set(“DAHDI/2-1”, “ORIG_ARG12”) in new stack
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: — Executing
[s@macro-stdexten:3] GotoIf(“DAHDI/2-1”, “0?6:4”) in new stack
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: — Goto (macro-stdexten,s,4)
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: — Executing
[s@macro-stdexten:4] Dial(“DAHDI/2-1”, “SIP/102,20,”) in new stack
[2012-11-27 14:43:56] VERBOSE[3589] netsock2.c: == Using SIP RTP CoS
mark 5
[2012-11-27 14:43:56] VERBOSE[3589] app_dial.c: — Called SIP/102
[2012-11-27 14:43:56] VERBOSE[3589] app_dial.c: — SIP/102-00000002
is ringing
[2012-11-27 14:44:01] VERBOSE[3589] app_dial.c: — SIP/102-00000002
answered DAHDI/2-1
Rang for a whole minute and a half until I hung up the DAHDI/2-1
[2012-11-27 14:45:42] VERBOSE[3589] pbx.c: — Executing
[h@voicemenu-home:1] Hangup(“DAHDI/2-1”, “”) in new stack
[2012-11-27 14:45:42] VERBOSE[3589] features.c: == Spawn extension
(voicemenu-home, h, 1) exited non-zero on ‘DAHDI/2-1’
[2012-11-27 14:45:42] VERBOSE[3589] app_macro.c: == Spawn extension
(macro-stdexten, s, 4) exited non-zero on ‘DAHDI/2-1’ in macro ‘stdexten’
[2012-11-27 14:45:42] VERBOSE[3589] pbx.c: == Spawn extension
(voicemenu-home, 102, 1) exited non-zero on ‘DAHDI/2-1’
[2012-11-27 14:45:42] VERBOSE[3589] sig_analog.c: — Hanging up on
‘DAHDI/2-1’
[2012-11-27 14:45:42] VERBOSE[3589] chan_dahdi.c: — Hungup ‘DAHDI/2-1’
(END)

2 thoughts on - SIP Not Answering On One Trunk.

  • Ron Wheeler wrote:

    I would suggest you grab a “sip set debug on” trace for this issue to confirm that the signaling is fine. When the phone answers it should send a 200 OK to Asterisk and then we respond with an ACK. If that all looks correct, as it seems to from the log you have provided thus far, I
    think you may have a phone specific issue.

    Cheers,

  • I did some more testing. The SIP phone does answer and it stops ringing. The incoming call
    (DAHDI) keeps broadcasting a ring tone to the caller’s handset even though the SIP phone stops ringing when the call is picked up and shows that the call is connected. Sorry for the confusion.

    If the SIP phone does not answer, the call goes through the voicemail process and takes a message.

    Does that help narrow it down?
    It looks like I have done something to the Asterisk configuration that is preventing the bridging of the incoming call to the local extension. It rings the local, the local picks up but the incoming caller is not connected and still hears the pbx ringing the SIP phone even though the SIP phone is no longer actually ringing.

    Ron