I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk box. Sorry for the long logs. Sip read: INVITE sip:99826816@10.1.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.11:5060;rport;branch=z9hG4bK493E94339A0F46AAACD2EF6C557922C2 From: damian <sip:damian@10.1.1.2>;tag=2667644054 To: <sip:99826816@10.1.1.2> Contact: <sip:damian@10.1.1.11:5060> Call-ID: 912EEDDD-2BDC-4CF8-A627-DECC35793EA5@10.1.1.11 CSeq: 42510 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103a Content-Length: 291 v=0 o=damian 23894728 23894788 IN IP4 10.1.1.11 s=X-Lite c=IN IP4 10.1.1.11 t=0 0 m=audio 8000 RTP/AVP 0 8 3 97 110 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 13 lines Using latest request as basis request Sending to 10.1.1.11 : 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 110 Found RTP audio format 101 Peer RTP is at port 10.1.1.11:0 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x60e(GSM|ULAW|ALAW|SPEEX|ILBC)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found peer 'phone1010' Jul 8 16:47:21 DEBUG[65541]: chan_sip.c:4851 check_user: Setting NAT on RTP to 0 Jul 8 16:47:21 DEBUG[65541]: chan_sip.c:6424 handle_request: Check for res for damian Jul 8 16:47:21 DEBUG[65541]: chan_sip.c:1386 update_user_counter: damian is not a local user Looking for 99826816 in default Jul 8 16:47:21 DEBUG[65541]: chan_sip.c:4115 build_route: build_route: Contact hop: <sip:damian@10.1.1.11:5060> list_route: hop: <sip:damian@10.1.1.11:5060> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.11:5060;rport;branch=z9hG4bK493E94339A0F46AAACD2EF6C557922C2 From: damian <sip:damian@10.1.1.2>;tag=2667644054 To: <sip:99826816@10.1.1.2>;tag=as5b6158bb Call-ID: 912EEDDD-2BDC-4CF8-A627-DECC35793EA5@10.1.1.11 CSeq: 42510 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:99826816@10.1.1.2:0> Content-Length: 0 to 10.1.1.11:5060 -- Executing Dial("SIP/damian-ff45", "Zap/4/9826816") in new stack Jul 8 16:47:21 DEBUG[262159]: chan_zap.c:1576 zt_call: Dialing '9826816' Jul 8 16:47:21 DEBUG[262159]: chan_zap.c:1633 zt_call: Deferring dialing... -- Called 4/9826816 Jul 8 16:47:21 DEBUG[262159]: chan_zap.c:3596 __zt_exception: Exception on 21, channel 4 Jul 8 16:47:21 DEBUG[262159]: chan_zap.c:2944 zt_handle_event: Got event Hook Transition Complete(12) on channel 4 (index 0) Jul 8 16:47:23 DEBUG[262159]: chan_zap.c:3596 __zt_exception: Exception on 21, channel 4 Jul 8 16:47:23 DEBUG[262159]: chan_zap.c:2944 zt_handle_event: Got event Dial Complete(9) on channel 4 (index 0) Jul 8 16:47:23 DEBUG[262159]: chan_zap.c:1169 zt_enable_ec: No echocancellation requested Jul 8 16:47:23 DEBUG[262159]: chan_zap.c:1185 zt_train_ec: No echo training requested Jul 8 16:47:24 DEBUG[262159]: chan_zap.c:3596 __zt_exception: Exception on 21, channel 4 Jul 8 16:47:24 DEBUG[262159]: chan_zap.c:2944 zt_handle_event: Got event Dial Complete(9) on channel 4 (index 0) Jul 8 16:47:24 DEBUG[262159]: chan_zap.c:1169 zt_enable_ec: No echocancellation requested Jul 8 16:47:24 DEBUG[262159]: chan_zap.c:3007 zt_handle_event: Done dialing, but waiting for progress detection before doing more... We're at 10.1.1.2 port 10524 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) Transmitting (no NAT): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.1.1.11:5060;rport;branch=z9hG4bK493E94339A0F46AAACD2EF6C557922C2 From: damian <sip:damian@10.1.1.2>;tag=2667644054 To: <sip:99826816@10.1.1.2>;tag=as5b6158bb Call-ID: 912EEDDD-2BDC-4CF8-A627-DECC35793EA5@10.1.1.11 CSeq: 42510 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:99826816@10.1.1.2:0> Content-Type: application/sdp Content-Length: 251 v=0 o=root 586 586 IN IP4 10.1.1.2 s=session c=IN IP4 10.1.1.2 t=0 0 m=audio 10524 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 10.1.1.11:5060 Jul 8 16:47:24 DEBUG[262159]: rtp.c:1123 ast_rtp_write: Ooh, format changed from UNKN to ULAW Jul 8 16:47:24 DEBUG[262159]: chan_sip.c:1976 sip_rtp_read: Oooh, format changed to 2 Jul 8 16:47:24 DEBUG[262159]: rtp.c:1123 ast_rtp_write: Ooh, format changed from ULAW to GSM 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:10.1.1.11 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.2:0;branch=z9hG4bK6eac0d88 From: "asterisk" <sip:asterisk@10.1.1.2:0>;tag=as5fdf9f82 To: <sip:10.1.1.11> Contact: <sip:asterisk@10.1.1.2:0> Call-ID: 2b4d7a39423d1c805053483b6fb5367a@10.1.1.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 08 Jul 2004 13:47:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.1.11:5060 voipgw*CLI> Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 10.1.1.2:0;branch=z9hG4bK6eac0d88 From: "asterisk" <sip:asterisk@10.1.1.2:0>;tag=as5fdf9f82 To: <sip:10.1.1.11>;tag=2355563749 Contact: <sip:damian@10.1.1.11:5060> Call-ID: 2b4d7a39423d1c805053483b6fb5367a@10.1.1.2 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1103a Content-Length: 0 10 headers, 0 lines Jul 8 16:47:31 DEBUG[65541]: chan_sip.c:752 __sip_ack: Stopping retransmission on '2b4d7a39423d1c805053483b6fb5367a@10.1.1.2' of Request 102: Found Destroying call '2b4d7a39423d1c805053483b6fb5367a@10.1.1.2' voipgw*CLI> Sip read: CANCEL sip:99826816@10.1.1.2 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.11:5060;rport;branch=z9hG4bK493E94339A0F46AAACD2EF6C557922C2 From: damian <sip:damian@10.1.1.2>;tag=2667644054 To: <sip:99826816@10.1.1.2> Contact: <sip:damian@10.1.1.11:5060> Call-ID: 912EEDDD-2BDC-4CF8-A627-DECC35793EA5@10.1.1.11 CSeq: 42510 CANCEL Max-Forwards: 70 User-Agent: X-Lite release 1103a Content-Length: 0 10 headers, 0 lines Sending to 10.1.1.11 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.11:5060;rport;branch=z9hG4bK493E94339A0F46AAACD2EF6C557922C2 From: damian <sip:damian@10.1.1.2>;tag=2667644054 To: <sip:99826816@10.1.1.2>;tag=as5b6158bb Call-ID: 912EEDDD-2BDC-4CF8-A627-DECC35793EA5@10.1.1.11 CSeq: 42510 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:99826816@10.1.1.2:0> Content-Length: 0 to 10.1.1.11:5060 Jul 8 16:47:36 DEBUG[262159]: chan_zap.c:1876 zt_hangup: Hangup: channel: 4 index = 0, normal = 21, callwait = -1, thirdcall = -1 Jul 8 16:47:36 DEBUG[262159]: chan_zap.c:2272 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/4-1 Jul 8 16:47:36 DEBUG[262159]: chan_zap.c:1141 update_conf: Updated conferencing on 4, with 0 conference users -- Hungup 'Zap/4-1' == Spawn extension (default, 99826816, 1) exited non-zero on 'SIP/damian-ff45' Jul 8 16:47:36 DEBUG[262159]: cdr_addon_mysql.c:181 mysql_log: cdr_mysql: inserting a CDR record. Jul 8 16:47:36 DEBUG[262159]: cdr_addon_mysql.c:200 mysql_log: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2004-07-08 16:47:36','\"damian\" <damian>','damian','99826816','default', 'SIP/damian-ff45','Zap/4-1','Dial','Zap/4/9826816',15,0,'NO ANSWER',3,'') Jul 8 16:47:36 DEBUG[262159]: chan_sip.c:1508 sip_hangup: update_user_counter(damian) - decrement inUse counter Jul 8 16:47:36 DEBUG[262159]: chan_sip.c:1386 update_user_counter: damian is not a local user Destroying call '912EEDDD-2BDC-4CF8-A627-DECC35793EA5@10.1.1.11' voipgw*CLI>