Hello All,
I just installed Asterisk 1.8.7.1 on fresh FreeBSD 8.2
To install trunk communication between my Avaya and * I set all needed settings on Avaya. (see below)
is this normal * crashes with all default configs? I only installed and started it with no change any config file.
Any call routed through Avaya trunk crashes Asterisk. See asterisk log before it die below.
the problem fixed on FreeBSD 7.1 with asterisk18-1.8.7.1 also
any ideas?
Thank you!
Muu
avaya settings:
change signaling-group 13 Page 1 of 5
SIGNALING GROUP
Group Number: 13 Group Type: h.323
Remote Office? n Max number of NCA TSC: 4
SBS? n Max number of CA TSC: 4
Trunk Group for NCA TSC: 13
Trunk Group for Channel Selection: 13
Supplementary Service Protocol: a
T303 Timer(sec): 10
Near-end Node Name: clanv Far-end Node Name: ipvoice
Near-end Listen Port: 1720 Far-end Listen Port: 1720
Far-end Network Region: 2
LRQ Required? n Calls Share IP Signaling Connection? n
RRQ Required? n
Media Encryption? n Bypass If IP Threshold Exceeded? n
DTMF over IP: in-band Direct IP-IP Audio Connections? y
IP Audio Hairpinning? n
Interworking Message: PROGress
change trunk-group 13 Page 1 of 23
TRUNK GROUP
Group Number: 13 Group Type: isdn CDR Reports: y
Group Name: ipvoice COR: 1 TN: 1 TAC: 813
Direction: two-way Outgoing Display? n Carrier Medium: IP
Dial Access? n Busy Threshold: 255 Night Service:
Queue Length: 0
Service Type: tie Auth Code? n TestCall ITC: rest
Far End Test Line No:
TestCall BCC: 4
TRUNK PARAMETERS
Codeset to Send Display: 0 Codeset to Send National IEs: 6
Max Message Size to Send: 260 Charge Advice: none
Supplementary Service Protocol: c Digit Handling (in/out): enbloc/enbloc
Trunk Hunt: cyclical
Digital Loss Group: 18
Incoming Calling Number - Delete: Insert: Format:
Bit Rate: 1200 Synchronization: async Duplex: full
Disconnect Supervision - In? y Out? y
Answer Supervision Timeout: 0
change trunk-group 13 Page 2 of 23
TRUNK FEATURES
ACA Assignment? n Measured: none Wideband Support? n
Internal Alert? n Maintenance Tests? y
Data Restriction? n NCA-TSC Trunk Member:
Send Name: n Send Calling Number: y
Used for DCS? n
Suppress # Outpulsing? n Format: public
Outgoing Channel ID Encoding: preferred UUI IE Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Send Connected Number: n
Hold/Unhold Notifications? n
Send UUI IE? y Modify Tandem Calling Number? n
Send UCID? n
Send Codeset 6/7 LAI IE? y
Network (Japan) Needs Connect Before Disconnect? n
change trunk-group 13 Page 7 of 23
TRUNK GROUP
Administered Members (min/max): 1/4
GROUP MEMBER ASSIGNMENTS Total Administered Members: 4
Port Code Sfx Name Night Sig Grp
1: T00136 13
2: T00137 13
3: T00138 13
4: T00139 13
5:
list node-names
NODE NAMES
Type Name IP Address
IP clanv 172.25 .23 .210
IP ipvoice 172.25 .4 .227
IP medpro 172.25 .23 .211
change ip-network-region 2 Page 1 of 19
IP NETWORK REGION
Region: 2
Location: Home Domain: Telefon vneshnij 1
Name:
Intra-region IP-IP Direct Audio: yes
AUDIO PARAMETERS Inter-region IP-IP Direct Audio: yes
Codec Set: 2 IP Audio Hairpinning? n
UDP Port Min: 2048
UDP Port Max: 32767 RTCP Reporting Enabled? y
RTCP MONITOR SERVER PARAMETERS
DIFFSERV/TOS PARAMETERS Use Default Server Parameters? y
Call Control PHB Value: 46
Audio PHB Value: 46
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 6
Audio 802.1p Priority: 6 AUDIO RESOURCE RESERVATION PARAMETERS
H.323 IP ENDPOINTS RSVP Enabled? n
H.323 Link Bounce Recovery? y
Idle Traffic Interval (sec): 20
Keep-Alive Interval (sec): 5
Keep-Alive Count: 5
change ip-network-region 2 Page 3 of 19
Inter Network Region Connection Management
src dst codec direct Dynamic CAC
rgn rgn set WAN WAN-BW-limits Intervening-regions Gateway
2 1 2 y :NoLimit
2 2 2
2 3
2 4
2 5
2 6
2 7
change ip-codec-set 2 Page 1 of 2
IP Codec Set
Codec Set: 2
Audio Silence Frames Packet
Codec Suppression Per Pkt Size(ms)
1: G.711A n 2 20
2: G.729B n 2 20
3: G.711MU n 2 20
4: G.729A n 2 20
5:
6:
7:
Media Encryption
1: none
2:
3:
=========================
asterisk log:
aster# /usr/local/sbin/asterisk -rvvv
Asterisk 1.8.7.1, Copyright © 1999 - 2011 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.
== Parsing ‘/usr/local/etc/asterisk/asterisk.conf’: == Found
== Parsing ‘/usr/local/etc/asterisk/extconfig.conf’: == Found
Connected to Asterisk 1.8.7.1 currently running on aster (pid = 35268)
Verbosity was 0 and is now 3
asterCLI>
asterCLI>
asterCLI>
[Dec 9 14:28:07] WARNING[35268]: res_rtp_asterisk.c:433 create_new_socket: Unable to allocate RTP socket: Protocol not supported
[Dec 9 14:28:07] WARNING[35268]: chan_h323.c:980 __oh323_rtp_create: Unable to create RTP session: Protocol not supported
[Dec 9 14:28:07] ERROR[35268]: chan_h323.c:1946 external_rtp_create: No RTP stream is available for call ip$172.25.23.210:18176/17664 (17664) ERROR: on_external_rtp_create failure
[Dec 9 14:28:07] WARNING[35268]: res_rtp_asterisk.c:433 create_new_socket: Unable to allocate RTP socket: Protocol not supported
[Dec 9 14:28:07] WARNING[35268]: chan_h323.c:980 __oh323_rtp_create: Unable to create RTP session: Protocol not supported
[Dec 9 14:28:07] ERROR[35268]: chan_h323.c:1946 external_rtp_create: No RTP stream is available for call ip$172.25.23.210:18176/17664 (17664) ERROR: on_external_rtp_create failure
[Dec 9 14:28:07] WARNING[35268]: res_rtp_asterisk.c:433 create_new_socket: Unable to allocate RTP socket: Protocol not supported
[Dec 9 14:28:07] WARNING[35268]: chan_h323.c:980 __oh323_rtp_create: Unable to create RTP session: Protocol not supported
[Dec 9 14:28:07] ERROR[35268]: chan_h323.c:1946 external_rtp_create: No RTP stream is available for call ip$172.25.23.210:18176/17664 (17664) ERROR: on_external_rtp_create failure
[Dec 9 14:28:07] WARNING[35268]: res_rtp_asterisk.c:433 create_new_socket: Unable to allocate RTP socket: Protocol not supported
[Dec 9 14:28:07] WARNING[35268]: chan_h323.c:980 __oh323_rtp_create: Unable to create RTP session: Protocol not supported
[Dec 9 14:28:07] ERROR[35268]: chan_h323.c:1946 external_rtp_create: No RTP stream is available for call ip$172.25.23.210:18176/17664 (17664) ERROR: on_external_rtp_create failure
[Dec 9 14:28:07] WARNING[35268]: res_rtp_asterisk.c:433 create_new_socket: Unable to allocate RTP socket: Protocol not supported
[Dec 9 14:28:07] WARNING[35268]: chan_h323.c:980 __oh323_rtp_create: Unable to create RTP session: Protocol not supported
[Dec 9 14:28:07] ERROR[35268]: chan_h323.c:1946 external_rtp_create: No RTP stream is available for call ip$172.25.23.210:18176/17664 (17664) ERROR: on_external_rtp_create failure
[Dec 9 14:28:07] WARNING[35268]: res_rtp_asterisk.c:433 create_new_socket: Unable to allocate RTP socket: Protocol not supported
[Dec 9 14:28:07] WARNING[35268]: chan_h323.c:980 __oh323_rtp_create: Unable to create RTP session: Protocol not supported
[Dec 9 14:28:07] ERROR[35268]: chan_h323.c:1946 external_rtp_create: No RTP stream is available for call ip$172.25.23.210:18176/17664 (17664) ERROR: on_external_rtp_create failure
[Dec 9 14:28:07] WARNING[35268]: res_rtp_asterisk.c:433 create_new_socket: Unable to allocate RTP socket: Protocol not supported
[Dec 9 14:28:07] WARNING[35268]: chan_h323.c:980 __oh323_rtp_create: Unable to create RTP session: Protocol not supported
– Executing [s@default:1] Wait(“H323/ip$172.25.23.210:18176/17664”, “1”) in new stack
[Dec 9 14:28:07] WARNING[35268]: res_rtp_asterisk.c:433 create_new_socket: Unable to allocate RTP socket: Protocol not supported
asterCLI>
Disconnected from Asterisk server
Executing last minute cleanups