Re: [asterisk-users] SIP not answering on one trunk.

2012-11-28 Thread Joshua Colp

Ron Wheeler wrote:

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.


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,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com   www.asterisk.org

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] SIP not answering on one trunk.

2012-11-28 Thread Ron Wheeler

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

On 28/11/2012 1:15 PM, Joshua Colp wrote:

Ron Wheeler wrote:

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.


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,




--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] SIP not answering on one trunk.

2012-11-27 Thread Ron Wheeler

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_866nnn]  IAX that works
include = DID_866nnn_default
[DID_866nnn_default]
exten = 866n,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: -- DAHDI/2-1 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_ARG1=102) 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/102-0002 
is ringing
[2012-11-27 14:44:01] VERBOSE[3589] app_dial.c: -- SIP/102-0002 
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)

--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users