Hi all! I know the argument is very famouse, I think I have read all ser.cfg on the net.... The problem is that when I call extension 9916 I don't hear asterisk music-on-old sound...
The scenario:
ser at port 5060 asterisk at port 5090 rtpproxy
I would like that, when an UA (x-lite for example) call 9916 UA can hear "music-on-hold" in asterisk.
*From ser.cfg:*
if (uri=~"sip:9916@") { log("\n\nLOG: 9916 - Asterisk - Test()\n\n");
force_rtp_proxy(); t_relay_to_udp("xxx.xxx.xxx.xxx","5090"); break; }
*From asterisk sip.conf:*
[general]
port = 5090 bindaddr = xxx.xxx.xxx.xxx externip = xxx.xxx.xxx.xxx tos=lowdelay ; disallow = all allow = ulaw allow = alaw ; context = sip-callfrom language = us musicclass = default
at now when I call 9916 from x-lite:
*ser says:*
0(31421) SIP Request: 0(31421) method: <INVITE> 0(31421) uri: sip:9916@xxx.xxx.xxx.xxx 0(31421) version: <SIP/2.0> 0(31421) parse_headers: flags=1 0(31421) Found param type 235, <rport> = <n/a>; state=6 0(31421) Found param type 232, <branch> = <z9hG4bKB3CD5F29F30B4329995057709DFC7500>; state=16 0(31421) end of header reached, state=5 0(31421) parse_headers: Via found, flags=1 0(31421) parse_headers: this is the first via 0(31421) After parse_msg... 0(31421) preparing to run routing scripts... 0(31421) DEBUG : is_maxfwd_present: searching for max_forwards header 0(31421) parse_headers: flags=128 0(31421) end of header reached, state=9 0(31421) DEBUG: get_hdr_field: <To> [26]; uri=[sip:9916@xxx.xxx.xxx.xxx] 0(31421) DEBUG: to body [sip:9916@xxx.xxx.xxx.xxx ] 0(31421) get_hdr_field: cseq <CSeq>: <12362> <INVITE> 0(31421) DEBUG: is_maxfwd_present: value = 70 0(31421) check_via_address(192.168.6.101, 192.168.6.101, 0) 0(31421) parse_headers: flags=64 0(31421) parse_headers: flags=-1 0(31421) DEBUG: get_hdr_body : content_length=267 0(31421) found end of header 0(31421)
LOG: Someone trying to register from private IP, rewriting
0(31421) parse_headers: flags=64 0(31421) parse_headers: flags=-1 0(31421) parse_headers: flags=-1 0(31421) DEBUG:check_content_type: type <application/sdp> found valid 0(31421) DEBUG: add_param: tag=2216232801 0(31421) end of header reached, state=29 0(31421) parse_headers: flags=256 0(31421) find_first_route(): No Route headers found 0(31421) loose_route(): There is no Route HF 0(31421)
LOG: 9916 - Asterisk - Test()
0(31421) parse_headers: flags=-1 0(31421) DEBUG:check_content_type: type <application/sdp> found valid 0(31421) ERROR: extract_mediaip: no `c=' in SDP 0(31421) DEBUG: t_newtran: msg id=7 , global msg id=6 , T on entrance=0xffffffff 0(31421) parse_headers: flags=-1 0(31421) parse_headers: flags=60 0(31421) t_lookup_request: start searching: hash=6960, isACK=0 0(31421) DEBUG: RFC3261 transaction matching failed 0(31421) DEBUG: t_lookup_request: no transaction found 0(31421) SER: new INVITE 0(31421) parse_headers: flags=-1 0(31421) check_via_address(192.168.6.101, 192.168.6.101, 0) 0(31421) WARNING:vqm_resize: resize(0) called 0(31421) DEBUG: reply sent out. buf=0x80f9b98: SIP/2.0 1..., shmem=0x4019d0f0: SIP/2.0 1 0(31421) DEBUG: _reply_light: finished 0(31421) parse_headers: flags=2048 0(31421) parse_headers: flags=-1 0(31421) clen_builder: content-length: 306 (306) 0(31421) check_via_address(192.168.6.101, 192.168.6.101, 0) 0(31421) DEBUG: add_timer_unsafe[4]: 0x4019d520 0(31421) DEBUG: add_timer_unsafe[0]: 0x4019d530 0(31421) SER: new transaction fwd'ed 0(31421) receive_msg: cleaning up 0(31421) SIP Reply (status): 0(31421) version: <SIP/2.0> 0(31421) status: <100> 0(31421) reason: <Trying> 0(31421) parse_headers: flags=1 0(31421) Found param type 232, <branch> = <z9hG4bK03b1.24325446.0>; state=16 0(31421) end of header reached, state=5 0(31421) parse_headers: Via found, flags=1 0(31421) parse_headers: this is the first via 0(31421) After parse_msg... 0(31421) forward_reply: found module tm, passing reply to it 0(31421) DEBUG: t_check: msg id=8 global id=7 T start=0xffffffff 0(31421) parse_headers: flags=17 0(31421) Found param type 235, <rport> = <5060>; state=6 0(31421) Found param type 232, <branch> = <z9hG4bKB3CD5F29F30B4329995057709DFC7500>; state=16 0(31421) end of header reached, state=5 0(31421) parse_headers: Via found, flags=17 0(31421) parse_headers: this is the second via 0(31421) DEBUG: add_param: tag=as72cacbaf 0(31421) end of header reached, state=29 0(31421) DEBUG: get_hdr_field: <To> [41]; uri=[sip:9916@xxx.xxx.xxx.xxx] 0(31421) DEBUG: to body [sip:9916@xxx.xxx.xxx.xxx] 0(31421) get_hdr_field: cseq <CSeq>: <12362> <INVITE> 0(31421) parse_headers: flags=4 0(31421) DEBUG: t_reply_matching: hash 6960 label 1682252610 branch 0 0(31421) DEBUG: t_reply_matching: reply matched (T=0x4019d408)! 0(31421) DEBUG: t_check: msg id=8 global id=8 T end=0x4019d408 0(31421) DEBUG: reply_received: org. status uas=100, uac[0]=0 local=0 is_invite=1) 0(31421) ->>>>>>>>> T_code=100, new_code=100 0(31421) DEBUG: relay_reply: branch=0, save=0, relay=-1 0(31421) DEBUG: add_timer_unsafe[1]: 0x4019d530 0(31421) receive_msg: cleaning up 0(31421) SIP Reply (status): 0(31421) version: <SIP/2.0> 0(31421) status: <200> 0(31421) reason: <OK> 0(31421) parse_headers: flags=1 0(31421) Found param type 232, <branch> = <z9hG4bK03b1.24325446.0>; state=16 0(31421) end of header reached, state=5 0(31421) parse_headers: Via found, flags=1 0(31421) parse_headers: this is the first via 0(31421) After parse_msg... 0(31421) forward_reply: found module tm, passing reply to it 0(31421) DEBUG: t_check: msg id=9 global id=8 T start=0xffffffff 0(31421) parse_headers: flags=17 0(31421) Found param type 235, <rport> = <5060>; state=6 0(31421) Found param type 232, <branch> = <z9hG4bKB3CD5F29F30B4329995057709DFC7500>; state=16 0(31421) end of header reached, state=5 0(31421) parse_headers: Via found, flags=17 0(31421) parse_headers: this is the second via 0(31421) DEBUG: add_param: tag=as72cacbaf 0(31421) end of header reached, state=29 0(31421) DEBUG: get_hdr_field: <To> [41]; uri=[sip:9916@xxx.xxx.xxx.xxx] 0(31421) DEBUG: to body [sip:9916@xxx.xxx.xxx.xxx] 0(31421) get_hdr_field: cseq <CSeq>: <12362> <INVITE> 0(31421) parse_headers: flags=4 0(31421) DEBUG: t_reply_matching: hash 6960 label 1682252610 branch 0 0(31421) DEBUG: t_reply_matching: reply matched (T=0x4019d408)! 0(31421) parse_headers: flags=4 0(31421) DEBUG: t_check: msg id=9 global id=9 T end=0x4019d408 0(31421) DEBUG: reply_received: org. status uas=100, uac[0]=100 local=0 is_invite=1) 0(31421) ->>>>>>>>> T_code=100, new_code=200 0(31421) DEBUG: relay_reply: branch=0, save=0, relay=0 0(31421) old size: 842, new size: 780 0(31421) build_res_from_sip_res: copied size: orig:78, new: 16, rest: 764 msg= SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.6.101:5060;rport=5060;branch=z9hG4bKB3CD5F29F30B4329995057709DFC7500 Record-Route: sip:xxx.xxx.xxx.xxx;ftag=2216232801;lr=on From: 2001 sip:2001@xxx.xxx.xxx.xxx;tag=2216232801 To: sip:9916@xxx.xxx.xxx.xxx;tag=as72cacbaf Call-ID: 7C0EAF05-D24D-463B-BB2A-F7746D751921@192.168.6.101 CSeq: 12362 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:9916@xxx.xxx.xxx.xxx:5090 Content-Type: application/sdp Content-Length: 265
v=0 o=root 31456 31456 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 10100 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -
0(31421) DEBUG: update_totag_set: new totag 0(31421) DEBUG: reply relayed. buf=0x80f9de8: SIP/2.0 2..., shmem=0x4019b240: SIP/2.0 2 0(31421) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(31421) DEBUG: add_timer_unsafe[2]: 0x4019d450 0(31421) receive_msg: cleaning up 0(31421) SIP Request: 0(31421) method: <ACK> 0(31421) uri: sip:9916@xxx.xxx.xxx.xxx:5090 0(31421) version: <SIP/2.0> 0(31421) parse_headers: flags=1 0(31421) Found param type 235, <rport> = <n/a>; state=6 0(31421) Found param type 232, <branch> = <z9hG4bKA83370958AA746AFA673710404627DFC>; state=16 0(31421) end of header reached, state=5 0(31421) parse_headers: Via found, flags=1 0(31421) parse_headers: this is the first via 0(31421) After parse_msg... 0(31421) preparing to run routing scripts... 0(31421) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(31421) DEBUG : is_maxfwd_present: searching for max_forwards header 0(31421) parse_headers: flags=128 0(31421) DEBUG: add_param: tag=as72cacbaf 0(31421) end of header reached, state=29 0(31421) DEBUG: get_hdr_field: <To> [41]; uri=[sip:9916@xxx.xxx.xxx.xxx] 0(31421) DEBUG: to body [sip:9916@xxx.xxx.xxx.xxx] 0(31421) get_hdr_field: cseq <CSeq>: <12362> <ACK> 0(31421) DEBUG: is_maxfwd_present: value = 70 0(31421) check_via_address(192.168.6.101, 192.168.6.101, 0) 0(31421) parse_headers: flags=64 0(31421) parse_headers: flags=-1 0(31421) DEBUG: get_hdr_body : content_length=0 0(31421) found end of header 0(31421)
LOG: Someone trying to register from private IP, rewriting
0(31421) parse_headers: flags=64 0(31421) DEBUG: add_param: tag=2216232801 0(31421) end of header reached, state=29 0(31421) parse_headers: flags=256 0(31421) check_self - checking if host==us: 13==13 && [xxx.xxx.xxx.xxx] == [xxx.xxx.xxx.xxx] 0(31421) check_self - checking if port 5060 matches port 5090 0(31421) check_self - checking if host==us: 13==13 && [xxx.xxx.xxx.xxx] == [xxx.xxx.xxx.xxx] 0(31421) check_self - checking if port 5060 matches port 5090 0(31421) check_self: host != me 0(31421) check_self - checking if host==us: 13==13 && [xxx.xxx.xxx.xxx] == [xxx.xxx.xxx.xxx] 0(31421) check_self - checking if port 5060 matches port 5060 0(31421) ral(): Topmost route URI: 'sip:xxx.xxx.xxx.xxx;ftag=2216232801;lr=on' is me 0(31421) parse_headers: flags=256 0(31421) found end of header 0(31421) fnr(): No next Route HF found 0(31421) ral(): No next URI found 0(31421)
LOG: 9916 - Asterisk - Test()
0(31421) DEBUG: t_newtran: msg id=10 , global msg id=9 , T on entrance=0xffffffff 0(31421) parse_headers: flags=-1 0(31421) parse_headers: flags=60 0(31421) t_lookup_request: start searching: hash=6960, isACK=1 0(31421) parse_headers: flags=28 0(31421) DEBUG: t_lookup_request: e2e proxy ACK found 0(31421) parse_headers: flags=4 0(31421) DEBUG: totag for e2e ACK found: 0 0(31421) SER: forwarding ACK statelessly 0(31421) check_via_address(192.168.6.101, 192.168.6.101, 0) 0(31421) Sending: ACK sip:9916@xxx.xxx.xxx.xxx:5090 SIP/2.0 Record-Route: sip:xxx.xxx.xxx.xxx;ftag=2216232801;lr=on Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=0 Via: SIP/2.0/UDP 192.168.6.101:5060;rport=5060;branch=z9hG4bKA83370958AA746AFA673710404627DFC From: 2001 sip:2001@xxx.xxx.xxx.xxx;tag=2216232801 To: sip:9916@xxx.xxx.xxx.xxx;tag=as72cacbaf Contact: sip:2001@192.168.6.101:5060 Call-ID: 7C0EAF05-D24D-463B-BB2A-F7746D751921@192.168.6.101 CSeq: 12362 ACK Max-Forwards: 69 Content-Length: 0 Voco-dbg: NAThelper
. 0(31421) orig. len=435, new_len=509, proto=1 0(31421) receive_msg: cleaning up 1(31422) DEBUG: timer routine:1,tl=0x4019d530 next=(nil) 1(31422) DEBUG: timer routine:4,tl=0x4019d520 next=(nil)
*rtpproxy -f says:*
rtpproxy: new session 7C0EAF05-D24D-463B-BB2A-F7746D751921@192.168.6.101, tag 2216232801 requested rtpproxy: new session on a port 10018 created, tag 2216232801 rtpproxy: pre-filling caller's address with 192.168.6.101:8000
*asterisk sip debug says:*
Sip read: INVITE sip:9916@xxx.xxx.xxx.xxx:5090 SIP/2.0 Record-Route: sip:xxx.xxx.xxx.xxx;ftag=2216232801;lr=on Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bK03b1.24325446.0 Via: SIP/2.0/UDP 192.168.6.101:5060;rport=5060;branch=z9hG4bKB3CD5F29F30B4329995057709DFC7500 From: 2001 sip:2001@xxx.xxx.xxx.xxx;tag=2216232801 To: sip:9916@xxx.xxx.xxx.xxx Contact: sip:2001@192.168.6.101:5060 Call-ID: 7C0EAF05-D24D-463B-BB2A-F7746D751921@192.168.6.101 CSeq: 12362 INVITE Max-Forwards: 69 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 306 Voco-dbg: NAThelper Voco-dbg: SDP rewrited
v=0 o=2001 8220109 8220125 IN IP4 192.168.6.101 s=X-Lite c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 10018 RTP/AVP 0 8 3 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=direction:active a=nortpproxy:yes
15 headers, 14 lines Using latest request as basis request Sending to xxx.xxx.xxx.xxx : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 101 Peer RTP is at port xxx.xxx.xxx.xxx:0 Found description format pcmu Found description format pcma Found description format gsm Found description format speex Found description format telephone-event Capabilities: us - 0x40e(GSM|ULAW|ALAW|ILBC), peer - audio=0x20e(GSM|ULAW|ALAW|SPEEX)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found no matching peer or user for 'xxx.xxx.xxx.xxx:5060' Looking for 9916 in sip-callfrom list_route: hop: sip:xxx.xxx.xxx.xxx;ftag=2216232801;lr=on list_route: hop: sip:2001@192.168.6.101:5060 Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bK03b1.24325446.0 Via: SIP/2.0/UDP 192.168.6.101:5060;rport=5060;branch=z9hG4bKB3CD5F29F30B4329995057709DFC7500 From: 2001 sip:2001@xxx.xxx.xxx.xxx;tag=2216232801 To: sip:9916@xxx.xxx.xxx.xxx;tag=as72cacbaf Call-ID: 7C0EAF05-D24D-463B-BB2A-F7746D751921@192.168.6.101 CSeq: 12362 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:9916@xxx.xxx.xxx.xxx:5090 Content-Length: 0
to xxx.xxx.xxx.xxx:5060 -- Executing Answer("SIP/xxx.xxx.xxx.xxx-081346f0", "") in new stack We're at xxx.xxx.xxx.xxx port 10100 Answering with preferred capability 0x2(GSM) Answering with preferred capability 0x4(ULAW) Answering with preferred capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bK03b1.24325446.0 Via: SIP/2.0/UDP 192.168.6.101:5060;rport=5060;branch=z9hG4bKB3CD5F29F30B4329995057709DFC7500 Record-Route: sip:xxx.xxx.xxx.xxx;ftag=2216232801;lr=on From: 2001 sip:2001@xxx.xxx.xxx.xxx;tag=2216232801 To: sip:9916@xxx.xxx.xxx.xxx;tag=as72cacbaf Call-ID: 7C0EAF05-D24D-463B-BB2A-F7746D751921@192.168.6.101 CSeq: 12362 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:9916@xxx.xxx.xxx.xxx:5090 Content-Type: application/sdp Content-Length: 265
v=0 o=root 31456 31456 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 10100 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -
to xxx.xxx.xxx.xxx:5060 -- Executing MusicOnHold("SIP/xxx.xxx.xxx.xxx-081346f0", "default") in new stack -- Started music on hold, class 'default', on SIP/xxx.xxx.xxx.xxx-081346f0
Sip read: ACK sip:9916@xxx.xxx.xxx.xxx:5090 SIP/2.0 Record-Route: sip:xxx.xxx.xxx.xxx;ftag=2216232801;lr=on Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=0 Via: SIP/2.0/UDP 192.168.6.101:5060;rport=5060;branch=z9hG4bKA83370958AA746AFA673710404627DFC From: 2001 sip:2001@xxx.xxx.xxx.xxx;tag=2216232801 To: sip:9916@xxx.xxx.xxx.xxx;tag=as72cacbaf Contact: sip:2001@192.168.6.101:5060 Call-ID: 7C0EAF05-D24D-463B-BB2A-F7746D751921@192.168.6.101 CSeq: 12362 ACK Max-Forwards: 69 Content-Length: 0 Voco-dbg: NAThelper
12 headers, 0 lines
Aldo Armiento wrote:
Hi all! I know the argument is very famouse, I think I have read all ser.cfg on the net.... The problem is that when I call extension 9916 I don't hear asterisk music-on-old sound...
I have analyzed packet with ethereal:
X-lite transmit on UDP src port: 8000 dst: 12976 (asterisk port) Asterisk transmit on UDP src: 12976 dst: 10022 (rtpproxy port)
Why X-lite don't transmit to rtpproxy port 10022 instead transmit direct to asterisk port?
Thanks you, Aldo.
You must have noticed I have been asking this question for last 4 days. Lets work together. I do not force_rtp_proxy before t_relay_to_udp because it breaks a lot more things than just NAT'd clients. When I dont use force_rtp_proxy, everything works, I can talk from NAT'd cleints to non-NAT'd, but asterisk doenst work for clients through NAT. I have noticed that the rtpproxy is still not getting engaged.
So t_relay_to_udp is not what we need, it may be a B2BUA. I wish someone could give us a simple answer before we dive into it ourselves.
Regards
On Thu, 1 Jul 2004, Aldo Armiento wrote:
Hi all! I know the argument is very famouse, I think I have read all ser.cfg on the net.... The problem is that when I call extension 9916 I don't hear asterisk music-on-old sound...
The scenario:
ser at port 5060 asterisk at port 5090 rtpproxy
I would like that, when an UA (x-lite for example) call 9916 UA can hear "music-on-hold" in asterisk.
On Jul 01, 2004 at 23:15, Aldo Armiento aldo@armiento.com wrote:
Hi all! I know the argument is very famouse, I think I have read all ser.cfg on the net.... The problem is that when I call extension 9916 I don't hear asterisk music-on-old sound...
[...]
if (uri=~"sip:9916@") { log("\n\nLOG: 9916 - Asterisk - Test()\n\n"); force_rtp_proxy();
With this setup you won't catch the reply. So the reply won't be modified and you UA will try to send media directly to asterisk. Set a t_on_reply route (e.g. t_on_reply("5")) and on the onreply route if the reply status is 183 or 2xx force_rtp_proxy on it.
A nicer way would be to do the nat test at the beginning of the cfg, set a flag if a request was nated, and also an onreply route. In the rest of the cfg you can check only for the flag.
t_relay_to_udp("xxx.xxx.xxx.xxx","5090"); break; }
Andrei
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Aldo Armiento -
Andrei Pelinescu-Onciul -
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