|Summary:||ASTERISK-06133: The NewChannelEvent: State: Ringing is sent to the call manager ONLY if the phone is directly connected to the Asterisk PBX.|
|Reporter:||Robert Krzyminski (bercik)||Labels:|
|Date Opened:||2006-01-19 04:04:39.000-0600||Date Closed:||2006-05-09 10:07:12|
|Environment:||Attachments:||( 0) sip_debug_log|
|Description:||When I'm using the call manager I am able to receive the NewChannelEvent: with the State: Ringing only if the phone is directly connected to the Asterisk PBX.|
When I'm calling using SIP to the phones directly connected to the Asterisk PBX everything is fine and as the called phone starts to ring, the event is sent to the Asterisk Manager console.
But when I'm calling to PSTN using Aculab card, or a SIP proxy provider, the event is not sent to the console.
I can see that the Aculab card is receiving the ringing state from the network, and I'm sure that the driver passes the event to the Asterisk using the ast_queue_control function.
|Comments:||By: Olle Johansson (oej) 2006-02-02 01:17:50.000-0600|
The aculab driver needs to call ast_set_state, so please report it to them.
By: Olle Johansson (oej) 2006-02-02 01:19:20.000-0600
For the SIP part, I need a bit more information. Reading the source, we do not differ between locally attached peers and other incoming calls for setting state (which generates the manager event). Can you please try again and capture logs so we can try to find the difference between calling through a SIP proxy and calling a local device?
Moving this issue to the SIP category.
By: Olle Johansson (oej) 2006-02-09 08:12:33.000-0600
I need an answer from you or I have to close this bug report due to lack of feedback.
By: Sebastian Nocetti (snocetti) 2006-02-14 13:55:11.000-0600
this is what my console shows when I place a call to a cisco 5300, as you see making progress is there, I dont know why sip channel does not change state to RINGING... or maybe the problem is in other place...
Every time I send a call to a Cisco 5300, SIP channels does not change state to RINGING when I hear Ring...
calls comes on this form:
SIPURA ATA (1502) send a call... ASTERISK uses DIAL app to call to PSTN through a CISCO 5300.
-- Executing Dial("SIP/1502-e517", "SIP/01144445555@gwCisco1|100|C") in new stack
-- Called 01144445555@gwCisco1
after a while (3-4 seconds) I start to hear ringing tone:
-- SIP/gwCisco1-93dd is making progress passing it to SIP/1502-e517
I type "show channels" command to see line status and it is what it shows:
Channel Location State Application(Data)
SIP/gwCisco1-93dd (None) Down AppDial((Outgoing Line))
SIP/1502-e517 0111558908604@callce Ring Dial(SIP/01144445555@gw
I dont see RINGING state, and I cant see any event RINGING on manager...
so, I can't know when a phone is ringing!...
it only happens sending calls to cisco 5300, If I send a call to another sipura or another phone attached to asterisk (registered), all is OK.
By: Serge Vecher (serge-v) 2006-02-14 14:12:59.000-0600
snocetti: you will have to capture a _SIP debug_ of the session as OEJ requested and attach the log to the bug.
By: Robert Krzyminski (bercik) 2006-02-15 02:00:39.000-0600
see attached log. Sorry that it took me so long.
By: Sebastian Nocetti (snocetti) 2006-02-22 10:20:44.000-0600
somebody can fix this bug please?, I really want it solved!
By: Olle Johansson (oej) 2006-03-02 15:50:57.000-0600
If we are getting the ring signal as early media, we never get a ringing indication. The cisco sends the audio with ringing tone instead of actually sending signalling that tells us that the other end is ringing. I guess this can be configured in the cisco.
Without a ringing indication, we can't do much.
By: Sebastian Nocetti (snocetti) 2006-03-03 09:06:30.000-0600
I noticed that CISCO sends SIP 183 - SESSION PROGRESS...
What's de difference between 180 and 183???
By: Sebastian Nocetti (snocetti) 2006-03-07 09:42:03.000-0600
any news on this??
By: Sebastian Nocetti (snocetti) 2006-03-07 09:46:00.000-0600
maybe you can add a new event like:
state: SessionProgress or similar instead RINGING... on cases 183 is received...
By: Serge Vecher (serge-v) 2006-05-01 15:40:38
Let's try to revisit this: Is this still an issue with either v.188.8.131.52 or the latest trunk? If so, please provide a sip trace with set debug 4, set verbose 4. Thanks.
By: Serge Vecher (serge-v) 2006-05-09 10:07:12
Closing this issue for now. If you are still able to reproduce this issue with most recent 1.2 code, please indicate what revision you are using and upload a sip debug log showing the problem. Thanks.