Hi,
Just wanted to ask some help on my problem regarding no audio on asterisk. For the past 5 months, my asterisk is working properly until yesterday. There were no audio that’s being heard at the other end of the line(PSTN). the asterisk is configured to play a certain audio file. I haven’t change a thing in my asterisk server, but it was not confirmed if changes happened on my partner. My asterisk is setup in externip ip. This was the original config since the first day the myasterisk work.Yesterday morning, it was all working fine, until 3:00 PM(manila time). When dialing a access_code the file is being played by asterisk but no audio is heard on other end. Hope you could help me out with my simple problem. thank you very much…
My call setup is something like this
GSM PHONE-> TELCO SwiTCH -> PARTNERS ZTE SOFTSwiTCH -> MYasterisk as Audio Playback server
My asterisk’s sip.conf
externip = 203.187.77.187
localnet=10.0.0.0/255.0.0.0
[203.175.223.25] – PARTNER’S ZTE SOFTSWITCH
type=friend
host=203.175.223.25
context=gle_inc
insecure=very
nat=yes
fromdomain=203.175.223.25
qualify=yes
canreinvite=yes
disallow=all
allow=ulaw
allow=alaw
call-limit=120
trustpid=yes
sendrpid=yes
rtptimeout =5
ASTERISK CLI LOGS
asterisk -vvvr
Asterisk 1.4.29, Copyright © 1999 - 2009 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 ‘/var/lib/sekavoice/prbt_conf/asterisk/asterisk.conf’: Found
== Parsing ‘/var/lib/sekavoice/prbt_conf/asterisk/extconfig.conf’: Found
Connected to Asterisk 1.4.29 currently running on pluto (pid = 9226)
Verbosity is at least 3
Reliably Transmitting (NAT) to 203.175.223.25:5060:
OPTIONS sip:203.175.223.25 SIP/2.0
Via: SIP/2.0/UDP 203.187.77.187:5060;branch=z9hG4bK14d38739;rport
From: “asterisk” sip:asterisk@203.187.77.187;tag=as4e9759ca
To: sip:203.175.223.25
Contact: sip:asterisk@203.187.77.187
Call-ID: 1aecba557b1f42a06cbd81bf71cb811e@203.187.77.187
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 15 Feb 2011 14:43:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
pluto*CLI>
<— SIP read from 203.175.223.25:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.187.77.187:5060;received=203.187.77.187;rport=5060;branch=z9hG4bK14d38739
To: sip:203.175.223.25;tag=ac9579e4sa
From: "asterisk"sip:asterisk@203.187.77.187;tag=as4e9759ca
Call-ID: 1aecba557b1f42a06cbd81bf71cb811e@203.187.77.187
CSeq: 102 OPTIONS
Supported: 100rel
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,NOTIFY,REFER,PRACK,UPDATE
Accept: application/sdp,application/ISUP,multipart/mixed
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0
<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘1aecba557b1f42a06cbd81bf71cb811e@203.187.77.187’ Method: OPTIONS
pluto*CLI>
<— SIP read from 203.175.223.25:5060 —>
INVITE sip:*0341111@203.187.77.187:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 203.175.223.25:5060;branch=z9hG4bK47243dd651e773e5
Via: SIP/2.0/UDP 10.109.0.2:5060;received=10.109.0.2;branch=z9hG4bK0c063fc7.0
To: "*0341111"sip:*0341111@203.187.77.187
From: "09178866042"sip:09178866042@203.175.223.25;tag=0a6d0002-0000056000000079
Call-ID: 00002b4200006451-0010-0854@10.109.0.2
CSeq: 4047 INVITE
Max-Forwards: 19
Contact: sip:09178866042@203.175.223.25:5060
Record-Route: sip:203.175.223.25:5060;lr
Record-Route: sip:10.109.0.2:5060;lr
Supported: 100rel
P-Asserted-Identity: sip:09178866042@203.175.223.25
User-Agent: ZTE Softswitch/1.0.0
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,PRACK,UPDATE
P-Charging-Vector: icid-value=Netun-20110215153250-002af988
Content-Type: application/sdp
Content-Length: 135
v=0
o=ZTE 2149 26839 IN IP4 203.175.223.25
s=phone-call
c=IN IP4 203.175.223.25
t=0 0
m=audio 41148 RTP/AVP 0 8 18 4
a=ptime:20
<------------->
— (18 headers 7 lines) —
Sending to 203.175.223.25 : 5060 (NAT)
Using INVITE request as basis request - 00002b4200006451-0010-0854@10.109.0.2
Found peer '203.175.223.25’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 203.175.223.25:41148
Looking for *0341111 in gle_inc (domain 203.187.77.187)
list_route: hop: sip:203.175.223.25:5060;lr
list_route: hop: sip:10.109.0.2:5060;lr
<— Transmitting (NAT) to 203.175.223.25:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.175.223.25:5060;branch=z9hG4bK47243dd651e773e5;received=203.175.223.25
Via: SIP/2.0/UDP 10.109.0.2:5060;received=10.109.0.2;branch=z9hG4bK0c063fc7.0
Record-Route: sip:203.175.223.25:5060;lr
Record-Route: sip:10.109.0.2:5060;lr
From: "09178866042"sip:09178866042@203.175.223.25;tag=0a6d0002-0000056000000079
To: "*0341111"sip:*0341111@203.187.77.187
Call-ID: 00002b4200006451-0010-0854@10.109.0.2
CSeq: 4047 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:*0341111@203.187.77.187
Content-Length: 0
<------------>
– Executing [*0341111@gle_inc:1] Set(“SIP/203.175.223.25-00000014”, “CALLERID(all)=639178866042”) in new stack
– Executing [*0341111@gle_inc:2] Goto(“SIP/203.175.223.25-00000014”, “comedy|*0341111|1”) in new stack
– Goto (comedy,*0341111,1)
– Executing [*0341111@comedy:1] NoOp(“SIP/203.175.223.25-00000014”, “”) in new stack
– Executing [*0341111@comedy:2] Goto(“SIP/203.175.223.25-00000014”, “com-filter|s|1”) in new stack
– Goto (com-filter,s,1)
– Executing [s@com-filter:1] Set(“SIP/203.175.223.25-00000014”, “CALLERID(dnid)=*0341111”) in new stack
– Executing [s@com-filter:2] GotoIf(“SIP/203.175.223.25-00000014”, “0?30”) in new stack
– Executing [s@com-filter:3] GotoIf(“SIP/203.175.223.25-00000014”, “1?20:10”) in new stack
– Goto (com-filter,s,20)
– Executing [s@com-filter:20] Goto(“SIP/203.175.223.25-00000014”, “mobe-comedy|s-com1|1”) in new stack
– Goto (mobe-comedy,s-com1,1)
– Executing [s-com1@mobe-comedy:1] Set(“SIP/203.175.223.25-00000014”, “EPCODE=COMEDY1”) in new stack
– Executing [s-com1@mobe-comedy:2] Set(“SIP/203.175.223.25-00000014”, “CDR(userfield)=*0341111”) in new stack
– Executing [s-com1@mobe-comedy:3] Set(“SIP/203.175.223.25-00000014”, “CDR(accountcode)=COMEDY1”) in new stack
– Executing [s-com1@mobe-comedy:4] Macro(“SIP/203.175.223.25-00000014”, “querydb”) in new stack
– Executing [s@macro-querydb:1] Set(“SIP/203.175.223.25-00000014”, "strtdate=“2011-02-15"”) in new stack
– Executing [s@macro-querydb:2] MYSQL(“SIP/203.175.223.25-00000014”, “Connect connid localhost seka_voice seka_voice 2929_mobfiles”) in new stack
– Executing [s@macro-querydb:3] MYSQL(“SIP/203.175.223.25-00000014”, "Query resultid 1 select content_path from pull_keywords where call_code = '0341111’ and start_date<=‘2011-02-15’ and end_date >= ‘2011-02-15’") in new stack
– Executing [s@macro-querydb:4] MYSQL(“SIP/203.175.223.25-00000014”, “Fetch fetchid 2 playbacks”) in new stack
– Executing [s@macro-querydb:5] MYSQL(“SIP/203.175.223.25-00000014”, “Disconnect 1”) in new stack
– Executing [s@macro-querydb:6] Wait(“SIP/203.175.223.25-00000014”, “1”) in new stack
– Executing [s@macro-querydb:7] Playback(“SIP/203.175.223.25-00000014”, “/var/lib/sekavoice/mobfiles_files/comedy/020811/bobita1”) in new stack
Audio is at 203.187.77.187 port 10586
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
plutoCLI>
<— Reliably Transmitting (NAT) to 203.175.223.25:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.175.223.25:5060;branch=z9hG4bK47243dd651e773e5;received=203.175.223.25
Via: SIP/2.0/UDP 10.109.0.2:5060;received=10.109.0.2;branch=z9hG4bK0c063fc7.0
Record-Route: sip:203.175.223.25:5060;lr
Record-Route: sip:10.109.0.2:5060;lr
From: "09178866042"sip:09178866042@203.175.223.25;tag=0a6d0002-0000056000000079
To: "*0341111"sip:*0341111@203.187.77.187;tag=as006e8e1c
Call-ID: 00002b4200006451-0010-0854@10.109.0.2
CSeq: 4047 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:*0341111@203.187.77.187
Content-Type: application/sdp
Content-Length: 210
v=0
o=root 9226 9226 IN IP4 203.187.77.187
s=session
c=IN IP4 203.187.77.187
t=0 0
m=audio 10586 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
– <SIP/203.175.223.25-00000014> Playing ‘/var/lib/sekavoice/mobfiles_files/comedy/020811/bobita1’ (language ‘en’)
pluto*CLI>
<— SIP read from 203.175.223.25:5060 —>
ACK sip:*0341111@203.187.77.187:5060 SIP/2.0
Via: SIP/2.0/UDP 203.175.223.25:5060;branch=z9hG4bKa7fceb0476ed34e0
Via: SIP/2.0/UDP 10.109.0.2:5060;received=10.109.0.2;branch=z9hG4bK661c542e.0
To: "*0341111"sip:*0341111@203.187.77.187;tag=as006e8e1c
From: "09178866042"sip:09178866042@203.175.223.25;tag=0a6d0002-0000056000000079
Call-ID: 00002b4200006451-0010-0854@10.109.0.2
CSeq: 4047 ACK
Max-Forwards: 19
User-Agent: ZTE Softswitch/1.0.0
Content-Length: 0
<------------->
— (10 headers 0 lines) —
Reliably Transmitting (NAT) to 203.175.223.25:5060:
OPTIONS sip:203.175.223.25 SIP/2.0
Via: SIP/2.0/UDP 203.187.77.187:5060;branch=z9hG4bK74fd6062;rport
From: “asterisk” sip:asterisk@203.187.77.187;tag=as1286712c
To: sip:203.175.223.25
Contact: sip:asterisk@203.187.77.187
Call-ID: 7b1a76ff4acaa73c138ab71436fe5a3a@203.187.77.187
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 15 Feb 2011 14:44:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
pluto*CLI>
<— SIP read from 203.175.223.25:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.187.77.187:5060;received=203.187.77.187;rport=5060;branch=z9hG4bK74fd6062
To: sip:203.175.223.25;tag=c2176821sa
From: "asterisk"sip:asterisk@203.187.77.187;tag=as1286712c
Call-ID: 7b1a76ff4acaa73c138ab71436fe5a3a@203.187.77.187
CSeq: 102 OPTIONS
Supported: 100rel
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,NOTIFY,REFER,PRACK,UPDATE
Accept: application/sdp,application/ISUP,multipart/mixed
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0
<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘7b1a76ff4acaa73c138ab71436fe5a3a@203.187.77.187’ Method: OPTIONS
– Executing [s@macro-querydb:8] Hangup(“SIP/203.175.223.25-00000014”, “”) in new stack
== Spawn extension (macro-querydb, s, 8) exited non-zero on ‘SIP/203.175.223.25-00000014’ in macro ‘querydb’
== Spawn extension (mobe-comedy, s-com1, 4) exited non-zero on ‘SIP/203.175.223.25-00000014’
– Executing [h@mobe-comedy:1] Hangup(“SIP/203.175.223.25-00000014”, “”) in new stack
== Spawn extension (mobe-comedy, h, 1) exited non-zero on 'SIP/203.175.223.25-00000014’
Scheduling destruction of SIP dialog ‘00002b4200006451-0010-0854@10.109.0.2’ in 6400 ms (Method: ACK)
set_destination: Parsing sip:203.175.223.25:5060;lr for address/port to send to
set_destination: set destination to 203.175.223.25, port 5060
Reliably Transmitting (NAT) to 203.175.223.25:5060:
BYE sip:09178866042@203.175.223.25:5060 SIP/2.0
Via: SIP/2.0/UDP 203.187.77.187:5060;branch=z9hG4bK7167b509;rport
Route: sip:203.175.223.25:5060;lr,sip:10.109.0.2:5060;lr
From: "*0341111"sip:*0341111@203.187.77.187;tag=as006e8e1c
To: "09178866042"sip:09178866042@203.175.223.25;tag=0a6d0002-0000056000000079
Call-ID: 00002b4200006451-0010-0854@10.109.0.2
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
pluto*CLI>
<— SIP read from 203.175.223.25:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.187.77.187:5060;received=203.187.77.187;rport=5060;branch=z9hG4bK7167b509
To: "09178866042"sip:09178866042@203.175.223.25;tag=0a6d0002-0000056000000079
From: "*0341111"sip:*0341111@203.187.77.187;tag=as006e8e1c
Call-ID: 00002b4200006451-0010-0854@10.109.0.2
CSeq: 102 BYE
User-Agent: ZTE Softswitch/1.0.0
Content-Length: 0
<------------->
— (8 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘00002b4200006451-0010-0854@10.109.0.2’ Method: ACK
pluto*CLI>
<— SIP read from 203.175.223.25:5060 —>
OPTIONS sip:203.187.77.187:5060 SIP/2.0
Via: SIP/2.0/UDP 203.175.223.25:5060;branch=z9hG4bKffc5eadazxsbc
Via: SIP/2.0/UDP online-test:5060;maddr=10.109.0.2;received=10.109.0.2;branch=z9hG4bK1dea620e.0
To: sip:203.187.77.187
From: sip:203.175.223.25;tag=a6d0002-7720
Call-ID: 44614150-0010-0854@10.109.0.2
CSeq: 273444 OPTIONS
Max-Forwards: 20
Accept: application/sdp
Content-Length: 0
<------------->
— (10 headers 0 lines) —
Looking for s in gle_inc (domain 203.187.77.187)
pluto*CLI>
<— Transmitting (NAT) to 203.175.223.25:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 203.175.223.25:5060;branch=z9hG4bKffc5eadazxsbc;received=203.175.223.25
Via: SIP/2.0/UDP online-test:5060;maddr=10.109.0.2;received=10.109.0.2;branch=z9hG4bK1dea620e.0
From: sip:203.175.223.25;tag=a6d0002-7720
To: sip:203.187.77.187;tag=as155f4b57
Call-ID: 44614150-0010-0854@10.109.0.2
CSeq: 273444 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘44614150-0010-0854@10.109.0.2’ in 32000 ms (Method: OPTIONS)