Проблема с музыкой и односторонний звук после удержания звонка

Всем день добрый,

Перво на перво я использую астериск 1.8.21 на CentOS 6.4, сборка PIAF Есть две проблемы следующего характера: К серверу подключены 3 телефона: snom 300, snom 320 и grandstream 2010 Между ними разговоры идут используя SRTP и TLS, все нормально дозваниваются и говорят Теперь если на сноме использовать кнопку ХОЛД то должна быть музыка на удержании, но вместо нее тишина, В CLI я вижу следующее:

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: <--- SIP read from TLS:172.17.0.186:2059 ---> INVITE sip:999997@172.17.3.241:5061;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 172.17.0.186:2059;branch=z9hG4bK-7nze9as0gj61;rport From: "999999" <sip:999999@172.17.3.241>;tag=yk1iamm3ef To: <sip:999997@172.17.3.241;user=phone>;tag=as1ff254e3 Call-ID: 3c2a472f4b22-ixhjamkj85m0 CSeq: 5 INVITE Max-Forwards: 70 Contact: <sip:999999@172.17.0.186:2059;transport=tls;line=1n0coz8i>;reg-id=1 P-Key-Flags: keys="3" User-Agent: snom320/7.3.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 367 v=0 o=root 10291801 10291804 IN IP4 172.17.0.186 s=call c=IN IP4 172.17.0.186 t=0 0 m=audio 65418 RTP/SAVP 0 8 9 99 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendonly <-------------> [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: --- (18 headers 17 lines) --- [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Sending to 172.17.0.186:2059 (no NAT) [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 0 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 8 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 9 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 0 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 8 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 9 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 99 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 3 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 18 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 4 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 101 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 101 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format pcmu for ID 0 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format pcma for ID 8 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format pcma for ID 8 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format g722 for ID 9 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format g726-32 for ID 99 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format gsm for ID 3 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format gsm for ID 3 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format g729 for ID 18 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format g723 for ID 4 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format telephone-event for ID 101 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format telephone-event for ID 101 [2013-04-25 02:20:49] **WARNING[2155]** chan_sip.c: Rejecting secure audio stream without encryption details: audio 65418 RTP/SAVP 0 8 9 99 3 18 4 101 [2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: <--- Reliably Transmitting (no NAT) to 172.17.0.186:2059 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/TLS 172.17.0.186:2059;branch=z9hG4bK-7nze9as0gj61;received=172.17.0.186; rport=2059 From: "999999" <sip:999999@172.17.3.241>;tag=yk1iamm3ef To: <sip:999997@172.17.3.241;user=phone>;tag=as1ff254e3 Call-ID: 3c2a472f4b22-ixhjamkj85m0 CSeq: 5 INVITE Server: FPBX-2.9.0(1.8.21.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 Тишина из за реджекта SRTP, но я не могу понять как это исправить, может кто сталкивался, Кстати если HOLD запустить на гранстриме то все работает и на другом конце идет музыка. Тут есть еще одна проблема, если разговор длится более 8 минут, то если запустить HOLD, а потом отменить, то слышен либо шум, вместо разговора, либо тишина и разговор приходиться прерывать, в CLI видно следующее: [2013-04-25 02:35:45] VERBOSE[1721] chan_sip.c: <--- SIP read from UDP:172.17.3.143:5061 ---> ACK sip:999999@172.17.3.241:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.3.143:5061;branch=z9hG4bKa3c44852dabd7bad From: <sip:999997@172.17.3.143:5061;transport=udp>;tag=66a86b21e19fda81 To: "test-exten" <sip:999999@172.17.3.241>;tag=as43749bcc Contact: <sip:999997@172.17.3.143:5061;transport=udp> Supported: path X-Grandstream-PBX: true Call-ID: 3816661d337f21613f1c822625df9e9c@172.17.3.241:5060 CSeq: 32723 ACK User-Agent: Grandstream GXP2010 1.2.5.3 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [2013-04-25 02:35:45] VERBOSE[1721] chan_sip.c: --- (13 headers 0 lines) --- [2013-04-25 02:35:45] WARNING[7912] res_srtp.c: SRTP unprotect failed with: authentication failure 10 [2013-04-25 02:35:45] VERBOSE[1721] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.3.184:2054: OPTIONS sip:999998@172.17.3.184:2054;transport=tls;line=oi2k6svb SIP/2.0 Via: SIP/2.0/TLS 172.17.3.241:5061;branch=z9hG4bK01ca8748 Max-Forwards: 70 From: "Unknown" <sip:Unknown@172.17.3.241>;tag=as2e3228c2 To: <sip:999998@172.17.3.184:2054;transport=tls;line=oi2k6svb> Contact: <sip:Unknown@172.17.3.241:5061;transport=TLS> Call-ID: 438959a738d6c7aa349174e94bc5ee14@172.17.3.241:5061 CSeq: 102 OPTIONS User-Agent: FPBX-2.9.0(1.8.21.0) Date: Thu, 25 Apr 2013 06:35:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-25 02:35:45] VERBOSE[3272] chan_sip.c: <--- SIP read from TLS:172.17.3.184:2054 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.17.3.241:5061;branch=z9hG4bK01ca8748 From: "Unknown" <sip:Unknown@172.17.3.241>;tag=as2e3228c2 To: <sip:999998@172.17.3.184:2054;transport=tls;line=oi2k6svb> Call-ID: 438959a738d6c7aa349174e94bc5ee14@172.17.3.241:5061 CSeq: 102 OPTIONS Contact: <sip:999998@172.17.3.184:2054;transport=tls;line=oi2k6svb>;reg-id=1 User-Agent: snom300/7.3.30 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [2013-04-25 02:35:45] VERBOSE[3272] chan_sip.c: --- (14 headers 0 lines) --- [2013-04-25 02:35:46] VERBOSE[1721] chan_sip.c: Really destroying SIP dialog '438959a738d6c7aa349174e94bc5ee14@172.17.3.241:5061' Method: OPTIONS [2013-04-25 02:35:47] WARNING[7912] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [2013-04-25 02:35:47] VERBOSE[1721] chan_sip.c: <--- SIP read from UDP:172.17.3.250:5060 ---> OPTIONS sip:172.17.3.241 SIP/2.0 Via: SIP/2.0/UDP 172.17.3.250:5060;branch=z9hG4bK1555d74f;rport Max-Forwards: 70 From: "Unknown" <sip:Unknown@172.17.3.250>;tag=as1bb1be7a To: <sip:172.17.3.241> Contact: <sip:Unknown@172.17.3.250:5060> Call-ID: 3bc7d30056bbf39c7550c13b1d6f9b97@172.17.3.250:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.9.0(1.8.15.1) Date: Thu, 25 Apr 2013 06:58:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0