# AGI "Got SIP response 486 "Busy Here" back from xx.xx.xx.xx" **URL:** \ **Category:** Asterisk Support\ **Created:** [January 18, 2010, 9:10am UTC](https://community.asterisk.org/t/agi-got-sip-response-486-busy-here-back-from-xx-xx-xx-xx/30268 "2010-01-18T09:10:19Z")\ **Posts on this page:** 4\ **Page:** 1
**Author:** ![salimafsar](https://avatars.discourse-cdn.com/v4/letter/s/ad7895/32.png) [@salimafsar](https://community.asterisk.org/u/salimafsar)\ **Post date:** [January 18, 2010, 9:10am UTC](https://community.asterisk.org/t/agi-got-sip-response-486-busy-here-back-from-xx-xx-xx-xx/30268/1 "2010-01-18T09:10:19Z")
Hello, I can make calls with x-lite but when I make calls with AGI, I got this error. It may be related with Codec setting. I setup G729 but it is still same. I couldn’t figure it out. If I open x-lite for example with 2000 user. And then if I set Extension=2000 with AGI it is calling x-lite and when I Answer x-lite, x-ite calls my cellphone. This is very strange… The onlt thing I guess it is codec problem. Thanks for any help. Salim ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [January 18, 2010, 11:54am UTC](https://community.asterisk.org/t/agi-got-sip-response-486-busy-here-back-from-xx-xx-xx-xx/30268/2 "2010-01-18T11:54:59Z")
Busy here is not an appropriate response for a codec problem. You will need to suppy the contents of the invite and the responses to get a better idea of what is happening (sip set debug on, with the debug category logged, or something like wireshark). ---
**Author:** ![salimafsar](https://avatars.discourse-cdn.com/v4/letter/s/ad7895/32.png) [@salimafsar](https://community.asterisk.org/u/salimafsar)\ **Post date:** [January 18, 2010, 12:07pm UTC](https://community.asterisk.org/t/agi-got-sip-response-486-busy-here-back-from-xx-xx-xx-xx/30268/3 "2010-01-18T12:07:46Z")
Hi, I started sip debug. And you can see the output: INVITE sip:05305235244@XX.XX.XX.XX SIP/2.0 Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK1507ea1f;rport From: “02122134114” sip:02122134114@XX.XX.XX.XX;tag=as5d4ca7b8 To: sip:05305235244@XX.XX.XX.XX Contact: sip:02122134114@YYY.YYY.YYY.YYY Call-ID: [297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX](mailto:297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX) CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 18 Jan 2010 11:57:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 315 v=0 o=root 3419 3419 IN IP4 YYY.YYY.YYY.YYY s=session c=IN IP4 YYY.YYY.YYY.YYY t=0 0 m=audio 19636 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv * * * [Jan 18 13:57:51] VERBOSE[4444] logger.c: \<— SIP read from XX.XX.XX.XX:5060 —\> SIP/2.0 100 Trying Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;rport=5060;received=YYY.YYY.YYY.YYY;branch=z9hG4bK1507ea1f From: “02122134114” sip:02122134114@XX.XX.XX.XX;tag=as5d4ca7b8 To: sip:05305235244@XX.XX.XX.XX Call-ID: [297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX](mailto:297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX) CSeq: 102 INVITE Content-Length: 0 \<-------------\> [Jan 18 13:57:51] VERBOSE[4444] logger.c: — (7 headers 0 lines) — [Jan 18 13:57:53] VERBOSE[4444] logger.c: \<— SIP read from XX.XX.XX.XX:5060 —\> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;rport=5060;received=YYY.YYY.YYY.YYY;branch=z9hG4bK1507ea1f To: sip:05305235244@XX.XX.XX.XX;tag=3472804677-628605 From: “02122134114” sip:02122134114@XX.XX.XX.XX;tag=as5d4ca7b8 Call-ID: [297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX](mailto:297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX) CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH Contact: sip:05305235244@XX.XX.XX.XX:5060 Call-Info: sip:XX.XX.XX.XX;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 208 v=0 o=btnxtmsc1 14207 14207 IN IP4 XX.XX.XX.XX s=sip call c=IN IP4 62.244.254.150 t=0 0 m=audio 39944 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=ptime:20 \<-------------\> [Jan 18 13:57:53] VERBOSE[4444] logger.c: — (11 headers 10 lines) — [Jan 18 13:57:53] VERBOSE[4444] logger.c: Found RTP audio format 18 [Jan 18 13:57:53] VERBOSE[4444] logger.c: Found RTP audio format 101 [Jan 18 13:57:53] VERBOSE[4444] logger.c: Peer audio RTP is at port 62.244.254.150:39944 [Jan 18 13:57:53] VERBOSE[4444] logger.c: Found audio description format telephone-event for ID 101 [Jan 18 13:57:53] VERBOSE[4444] logger.c: Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) [Jan 18 13:57:53] VERBOSE[4444] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 18 13:57:53] VERBOSE[4444] logger.c: Peer audio RTP is at port 62.244.254.150:39944 [Jan 18 13:57:53] DEBUG[4444] chan\_sip.c: Oooh, we need to change our audio formats since our peer supports only 0x100 (g729) and not 0x4 (ulaw) [Jan 18 13:57:54] VERBOSE[4444] logger.c: \<— SIP read from XX.XX.XX.XX:5060 —\> SIP/2.0 180 Ringing Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;rport=5060;received=YYY.YYY.YYY.YYY;branch=z9hG4bK1507ea1f To: sip:05305235244@XX.XX.XX.XX;tag=3472804677-628605 From: “02122134114” sip:02122134114@XX.XX.XX.XX;tag=as5d4ca7b8 Call-ID: [297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX](mailto:297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX) CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH Contact: sip:05305235244@XX.XX.XX.XX:5060 Call-Info: sip:XX.XX.XX.XX;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 208 v=0 o=btnxtmsc1 14207 14207 IN IP4 XX.XX.XX.XX s=sip call c=IN IP4 62.244.254.150 t=0 0 m=audio 39944 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=ptime:20 \<-------------\> [Jan 18 13:57:54] VERBOSE[4444] logger.c: — (11 headers 10 lines) — [Jan 18 13:57:54] VERBOSE[4444] logger.c: Found RTP audio format 18 [Jan 18 13:57:54] VERBOSE[4444] logger.c: Found RTP audio format 101 [Jan 18 13:57:54] VERBOSE[4444] logger.c: Peer audio RTP is at port 62.244.254.150:39944 [Jan 18 13:57:54] VERBOSE[4444] logger.c: Found audio description format telephone-event for ID 101 [Jan 18 13:57:54] VERBOSE[4444] logger.c: Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) [Jan 18 13:57:54] VERBOSE[4444] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 18 13:57:54] VERBOSE[4444] logger.c: Peer audio RTP is at port 62.244.254.150:39944 [Jan 18 13:57:56] VERBOSE[4444] logger.c: \<— SIP read from XX.XX.XX.XX:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;rport=5060;received=YYY.YYY.YYY.YYY;branch=z9hG4bK1507ea1f To: sip:05305235244@XX.XX.XX.XX;tag=3472804677-628605 From: “02122134114” sip:02122134114@XX.XX.XX.XX;tag=as5d4ca7b8 Call-ID: [297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX](mailto:297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX) CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH Contact: sip:05305235244@XX.XX.XX.XX:5060 Call-Info: sip:XX.XX.XX.XX;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 208 v=0 o=btnxtmsc1 14207 14207 IN IP4 XX.XX.XX.XX s=sip call c=IN IP4 62.244.254.150 t=0 0 m=audio 39944 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=ptime:20 \<-------------\> [Jan 18 13:57:56] VERBOSE[4444] logger.c: — (11 headers 10 lines) — [Jan 18 13:57:56] VERBOSE[4444] logger.c: Found RTP audio format 18 [Jan 18 13:57:56] VERBOSE[4444] logger.c: Found RTP audio format 101 [Jan 18 13:57:56] VERBOSE[4444] logger.c: Peer audio RTP is at port 62.244.254.150:39944 [Jan 18 13:57:56] VERBOSE[4444] logger.c: Found audio description format telephone-event for ID 101 [Jan 18 13:57:56] VERBOSE[4444] logger.c: Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) [Jan 18 13:57:56] VERBOSE[4444] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 18 13:57:56] VERBOSE[4444] logger.c: Peer audio RTP is at port 62.244.254.150:39944 [Jan 18 13:57:56] VERBOSE[4444] logger.c: list\_route: hop: sip:05305235244@XX.XX.XX.XX:5060 [Jan 18 13:57:56] DEBUG[4444] chan\_sip.c: Strict routing enforced for session [297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX](mailto:297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX) [Jan 18 13:57:56] VERBOSE[4444] logger.c: set\_destination: Parsing sip:05305235244@XX.XX.XX.XX:5060 for address/port to send to [Jan 18 13:57:56] VERBOSE[4444] logger.c: set\_destination: set destination to XX.XX.XX.XX, port 5060 [Jan 18 13:57:56] VERBOSE[4444] logger.c: Transmitting (NAT) to XX.XX.XX.XX:5060: ACK sip:05305235244@XX.XX.XX.XX:5060 SIP/2.0 Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK0113a5cb;rport From: “02122134114” sip:02122134114@XX.XX.XX.XX;tag=as5d4ca7b8 To: sip:05305235244@XX.XX.XX.XX;tag=3472804677-628605 Contact: sip:02122134114@YYY.YYY.YYY.YYY Call-ID: [297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX](mailto:297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX) CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 [Jan 18 13:57:56] VERBOSE[4516] logger.c: \> Channel SIP/MySIP-09587a30 was answered. [Jan 18 13:57:56] VERBOSE[4518] logger.c: – Executing [05305235244@MySIP:1] Set(“SIP/MySIP-09587a30”, “CALLERID(all)=Telegami \<02124448546\>”) in new stack [Jan 18 13:57:56] WARNING[4518] pbx.c: Can’t find trailing parenthesis? [Jan 18 13:57:56] VERBOSE[4518] logger.c: – Executing [05305235244@MySIP:2] Set(“SIP/MySIP-09587a30”, “CALLFILENAME=5305235244-Mon Jan 18 13:57:56 2010||%Y%m%d-%H%M%S)}-”) in new stack [Jan 18 13:57:56] WARNING[4518] pbx.c: Setting multiple variables at once within Set is deprecated. Please separate each name/value pair into its own line. [Jan 18 13:57:56] WARNING[4518] pbx.c: Ignoring entry ‘’ with no = (and not last ‘options’ entry) [Jan 18 13:57:56] VERBOSE[4518] logger.c: – Executing [05305235244@MySIP:3] Monitor(“SIP/MySIP-09587a30”, “wav|5305235244-Mon Jan 18 13:57:56 2010|m”) in new stack [Jan 18 13:57:56] VERBOSE[4518] logger.c: – Executing [05305235244@MySIP:4] Dial(“SIP/MySIP-09587a30”, “SIP/05305235244@MySIP”) in new stack [Jan 18 13:57:56] VERBOSE[4518] logger.c: Audio is at YYY.YYY.YYY.YYY port 17596 [Jan 18 13:57:56] VERBOSE[4518] logger.c: Adding codec 0x100 (g729) to SDP [Jan 18 13:57:56] VERBOSE[4518] logger.c: Adding codec 0x4 (ulaw) to SDP [Jan 18 13:57:56] VERBOSE[4518] logger.c: Adding codec 0x8 (alaw) to SDP [Jan 18 13:57:56] VERBOSE[4518] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 18 13:57:56] VERBOSE[4518] logger.c: Reliably Transmitting (NAT) to XX.XX.XX.XX:5060: INVITE sip:05305235244@XX.XX.XX.XX SIP/2.0 Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK78955bf6;rport From: “Telegami” sip:02124448546@XX.XX.XX.XX;tag=as05dd652c To: sip:05305235244@XX.XX.XX.XX Contact: sip:02124448546@YYY.YYY.YYY.YYY Call-ID: [5004eaab24eebe5e7fc7177d1c4ee299@XX.XX.XX.XX](mailto:5004eaab24eebe5e7fc7177d1c4ee299@XX.XX.XX.XX) CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 18 Jan 2010 11:57:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 315 v=0 o=root 3419 3419 IN IP4 YYY.YYY.YYY.YYY s=session c=IN IP4 YYY.YYY.YYY.YYY t=0 0 m=audio 17596 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv * * * [Jan 18 13:57:56] VERBOSE[4518] logger.c: – Called 05305235244@MySIP [Jan 18 13:57:56] VERBOSE[4444] logger.c: \<— SIP read from XX.XX.XX.XX:5060 —\> SIP/2.0 100 Trying Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;rport=5060;received=YYY.YYY.YYY.YYY;branch=z9hG4bK78955bf6 From: “Telegami” sip:02124448546@XX.XX.XX.XX;tag=as05dd652c To: sip:05305235244@XX.XX.XX.XX Call-ID: [5004eaab24eebe5e7fc7177d1c4ee299@XX.XX.XX.XX](mailto:5004eaab24eebe5e7fc7177d1c4ee299@XX.XX.XX.XX) CSeq: 102 INVITE Content-Length: 0 \<-------------\> [Jan 18 13:57:56] VERBOSE[4444] logger.c: — (7 headers 0 lines) — [Jan 18 13:57:56] VERBOSE[4444] logger.c: \<— SIP read from XX.XX.XX.XX:5060 —\> SIP/2.0 486 Busy here Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;rport=5060;received=YYY.YYY.YYY.YYY;branch=z9hG4bK78955bf6 To: sip:05305235244@XX.XX.XX.XX;tag=3472804680-801306 From: “Telegami” sip:02124448546@XX.XX.XX.XX;tag=as05dd652c Call-ID: [5004eaab24eebe5e7fc7177d1c4ee299@XX.XX.XX.XX](mailto:5004eaab24eebe5e7fc7177d1c4ee299@XX.XX.XX.XX) CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH Contact: sip:05305235244@XX.XX.XX.XX:5060 Call-Info: sip:XX.XX.XX.XX;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Length: 0 \<-------------\> [Jan 18 13:57:56] VERBOSE[4444] logger.c: — (10 headers 0 lines) — [Jan 18 13:57:56] VERBOSE[4444] logger.c: – Got SIP response 486 “Busy here” back from XX.XX.XX.XX [Jan 18 13:57:56] VERBOSE[4444] logger.c: Transmitting (NAT) to XX.XX.XX.XX:5060: ACK sip:05305235244@XX.XX.XX.XX SIP/2.0 Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK78955bf6;rport From: “Telegami” sip:02124448546@XX.XX.XX.XX;tag=as05dd652c To: sip:05305235244@XX.XX.XX.XX;tag=3472804680-801306 Contact: sip:02124448546@YYY.YYY.YYY.YYY Call-ID: [5004eaab24eebe5e7fc7177d1c4ee299@XX.XX.XX.XX](mailto:5004eaab24eebe5e7fc7177d1c4ee299@XX.XX.XX.XX) CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 * * * [Jan 18 13:57:56] VERBOSE[4518] logger.c: – SIP/MySIP-095a81d0 is busy [Jan 18 13:57:56] VERBOSE[4518] logger.c: == Everyone is busy/congested at this time (1:1/0/0) [Jan 18 13:57:56] VERBOSE[4518] logger.c: == Auto fallthrough, channel ‘SIP/MySIP-09587a30’ status is ‘BUSY’ [Jan 18 13:57:56] VERBOSE[4444] logger.c: Really destroying SIP dialog '5004eaab24eebe5e7fc7177d1c4ee299@XX.XX.XX.XX’ Method: INVITE [Jan 18 13:58:07] VERBOSE[4518] logger.c: Scheduling destruction of SIP dialog '297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX’ in 10752 ms (Method: INVITE) [Jan 18 13:58:07] DEBUG[4518] chan\_sip.c: Strict routing enforced for session [297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX](mailto:297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX) [Jan 18 13:58:07] VERBOSE[4518] logger.c: set\_destination: Parsing sip:05305235244@XX.XX.XX.XX:5060 for address/port to send to [Jan 18 13:58:07] VERBOSE[4518] logger.c: set\_destination: set destination to XX.XX.XX.XX, port 5060 [Jan 18 13:58:07] VERBOSE[4518] logger.c: Reliably Transmitting (NAT) to XX.XX.XX.XX:5060: BYE sip:05305235244@XX.XX.XX.XX:5060 SIP/2.0 Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK00ee8f73;rport From: “02122134114” sip:02122134114@XX.XX.XX.XX;tag=as5d4ca7b8 To: sip:05305235244@XX.XX.XX.XX;tag=3472804677-628605 Call-ID: [297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX](mailto:297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX) CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: User busy X-Asterisk-HangupCauseCode: 17 Content-Length: 0 * * * [Jan 18 13:58:07] DEBUG[4518] res\_monitor.c: monitor executing ( nice -n 19 sox -m “/var/spool/asterisk/monitor/5305235244-Mon Jan 18 13:57:56 2010-in.wav” “/var/spool/asterisk/monitor/5305235244-Mon Jan 18 13:57:56 2010-out.wav” “/var/spool/asterisk/monitor/5305235244-Mon Jan 18 13:57:56 2010.wav” && rm -f “/var/spool/asterisk/monitor/5305235244-Mon Jan 18 13:57:56 2010-”\* ) & [Jan 18 13:58:07] VERBOSE[4444] logger.c: \<— SIP read from XX.XX.XX.XX:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;rport=5060;received=YYY.YYY.YYY.YYY;branch=z9hG4bK00ee8f73 To: sip:05305235244@XX.XX.XX.XX;tag=3472804677-628605 From: “02122134114” sip:02122134114@XX.XX.XX.XX;tag=as5d4ca7b8 Call-ID: [297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX](mailto:297044a51f2b84ad4930f74330184e61@XX.XX.XX.XX) CSeq: 103 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH Contact: sip:05305235244@XX.XX.XX.XX:5060 Content-Length: 0 Thanks. ---
**Author:** ![bira\_more](https://avatars.discourse-cdn.com/v4/letter/b/bbce88/32.png) [@bira\_more](https://community.asterisk.org/u/bira_more)\ **Post date:** [January 18, 2010, 5:11pm UTC](https://community.asterisk.org/t/agi-got-sip-response-486-busy-here-back-from-xx-xx-xx-xx/30268/4 "2010-01-18T17:11:43Z")
You can try to do couple of things: 1. Start using g711 in all places 2. Remove monitoring. You could have problems from g729, because if you want monitoring, then you will need transcoding. And monitoring itself looks also not very good - you can try to use fixed name - at least while debugging.