# Asterisk playback a file but no audio **URL:** \ **Category:** Asterisk Support\ **Created:** [February 15, 2011, 3:08pm UTC](https://community.asterisk.org/t/asterisk-playback-a-file-but-no-audio/34163 "2011-02-15T15:08:57Z")\ **Posts on this page:** 1\ **Page:** 1
**Author:** ![afaylona](https://avatars.discourse-cdn.com/v4/letter/a/dec6dc/32.png) [@afaylona](https://community.asterisk.org/u/afaylona)\ **Post date:** [February 15, 2011, 3:08pm UTC](https://community.asterisk.org/t/asterisk-playback-a-file-but-no-audio/34163/1 "2011-02-15T15:08:57Z")
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](mailto: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](mailto: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](mailto: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](mailto: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](mailto: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](mailto: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 pluto_CLI\> \<— 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](mailto: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 \<------------\> – \ 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](mailto: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](mailto: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](mailto: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](mailto: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](mailto: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](mailto: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](mailto: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)