Summary:	ASTERISK-13092: no dialling tone with AMI command originate & 1.6.0.1
Reporter:	Stephen Coles (psycho coles)	Labels:
Date Opened:	2008-11-19 11:07:29.000-0600	Date Closed:	2011-06-07 14:08:13
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Core/ManagerInterface
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) asterisk_configs.tar.gz ( 1) sip_1.4.18.1_originate_debug.txt ( 2) sip_1.6.0.1_originate_debug.txt
Description:	when i dial a number using AMI command originate in asterisk 1.6.0.1 my phone rings when i pick it up it connects to the number i want it to but i get no dialling tone\noise. in asterisk 1.4.18.1 when i use originate it works and i get a dialling tone\noise. attached is the sip debug from both systems. to test i set up: 192.168.16.116 - ext:220 my phone that i originate from 192.168.16.162 - ext: 222 the phone that i dial using originate command 192.168.16.4 - asterisk 1.4.18.1 box 192.168.16.8 - asterisk 1.6.0.1 box i dialled using the originate command setting channel variable to 220 and exten to 222
Comments:	By: Terry Wilson (twilson) 2008-11-19 18:40:43.000-0600 I am unable to duplicate the lack of ringing tone in 1.6.0.1 with: extensions.conf ------------ [default] exten => _600X,1,Dial(SIP/${EXTEN},30) sip.conf ------ [test](!) type=friend host=dynamic context=default secret=test canreinvite=no nat=yes [6001](test) [6002](test) [6003](test) orignate command --------------- Action: Originate Channel: SIP/6001 Context: default Exten: 6002 Priority: 1 After the originate, 6001 rings.  I answer.  I hear ringing on the line until I answer 6002.  I tested on 1.4, 1.6.0 SVN, 1.6.0.1 tarball and trunk.  Can you reduce this to the simplest extensions.conf and sip.conf settings that will reproduce the issue and post back with that info? By: Stephen Coles (psycho coles) 2008-11-20 16:38:47.000-0600 have set the system up like you said above - all i had was extensions.conf, sip.conf & manager.conf (so i could log onto the manager) in my config dir still the same, even tried your exact originate command with and without events. tried it on the aastra 55i firmware: 2.4.0.96 and on the polycom ip soundpoint 320 firmware: 3.1.1.0137 i ported my set up to the asterisk now 1.02  (asterisk version 1.4.18.1) and i got a ringing sound on my phone you gave me your sip.conf and extension.conf file what other conf file were in the asterisk dir can you show me them so i can use thos as well. how does the phone get the dialling sound, i guess asterisk must tell it to give out a dialling sound using SIP signalling. if so whats the different between 1.4.18.1 and 1.6.0.1 way it tells a phone to give out a dialling sound. could the phone not be understanding the SIP packet to give out a dialling sound? also i found i needed to mute the other phone ringing noise as not to mistake it for my phones dialling noise are you trying it on softphones as i'm doing this on a hardphone?  will try a softphone tomorrow, retry it on the polycom and setup you test system on my old 1.4.18.1 box By: Terry Wilson (twilson) 2008-11-20 16:55:08.000-0600 I see no difference in the SIP messaging logs you posted.  Could I get you to capture the traffic from the asterisk box with wireshark and post here? By: Stephen Coles (psycho coles) 2008-11-21 06:47:48.000-0600 will do when i get to work. do you want a wireshark of 1.6 box or the 1.4 box as well.  can i use my config or do you want me to use yours for both. also could you zip a copy of all the conf files you want me to use so i have a exact same conf as you what causes the phone to make a dialling noise, if for me it works in 1.4.18.1 but not in 1.6.0.1 asterisk must tell it to have a dialling noise when you originate when i dial on the phone directly i get the dialling noise but when i originate i do not, the first must be controlled by the phone because the phone knows i'm dialling but the second i thought would be controlled my the phone must be controlled by the asterisk box thanks for your help on this By: Terry Wilson (twilson) 2008-11-21 10:20:23.000-0600 attaching my entire /etc/asterisk directory--although all the pertinent information in the config files should be above.  The ringing indication on the phone in this situation, according to my captures is through the rtp--not through any SIP signaling.  So, when you do your capture, capture udp traffic--not just port 5060. By: Stephen Coles (psycho coles) 2008-11-21 12:35:14.000-0600 ignore i have fixed i got rid of the indications.conf file i checked lots of time but got no error messages on the CLI however when i put indications.conf in and then took back out i got errors on the CLI also i put indication.conf in the /etc/asterisk dir still no ringing indication when i issued 'restart now', i put your configuration in and got ringing indication, put my conf back in and it work (why i don't know) but it's solved - thank you for your help By: Terry Wilson (twilson) 2008-11-21 13:18:23.000-0600 Closing at request of reporter.