Hi all, I have configured a connection my sip voip provider. I can make outbound call without trouble. But I cannot recieve voip calls. The sip negociation seams to start well but at some point during the rtcp dialog, things seems to block. As you can see on the above log sample, I recieve some rtcp packets as long I can ear the ring tone on my soft phone (a tcpdump confirms the reception only, nothing goes out). But no sip channel for my voip provider ever starts on my * server. Is there anything you would suggest I could make or test to change this beahviour ? Is there something in this log that catches your attention more than me ? Thanks for your help, comments and suggestions. Vincent ****** 192.168.0.30 is the LAN interface of my asterisk server, 62.147.170.96 its public IP. freephonie.net is my voip provider. Jul 19 19:41:00 VERBOSE[23199] logger.c: -- Zap/2-1 answered SIP/220-081c4618 Jul 19 19:41:00 DEBUG[23199] channel.c: Set channel SIP/220-081c4618 to read format slin Jul 19 19:41:00 DEBUG[23199] channel.c: Set channel Zap/2-1 to write format slin Jul 19 19:41:00 DEBUG[23199] channel.c: Set channel Zap/2-1 to read format slin Jul 19 19:41:00 DEBUG[23199] channel.c: Set channel SIP/220-081c4618 to write format slin Jul 19 19:41:00 DEBUG[23199] chan_sip.c: sip_answer(SIP/220-081c4618) Jul 19 19:41:00 DEBUG[15342] devicestate.c: Changing state for Zap/2 - state 2 (In use) Jul 19 19:41:00 DEBUG[15342] chan_sip.c: Checking device state for peer 220 Jul 19 19:41:00 DEBUG[15342] devicestate.c: Changing state for SIP/220 - state 2 (In use) Jul 19 19:41:00 DEBUG[15342] chan_iax2.c: Checking device state for device 200 Jul 19 19:41:00 DEBUG[15342] chan_iax2.c: iax2_devicestate: Found peer. What's device state of 200? addr=0, defaddr=0 maxms=0, lastms=0 Jul 19 19:41:00 DEBUG[15342] chan_sip.c: Checking device state for peer 220 Jul 19 19:41:00 DEBUG[23205] app_queue.c: Device 'Zap/2' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jul 19 19:41:00 DEBUG[23206] app_queue.c: Device 'SIP/220' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jul 19 19:41:00 DEBUG[23199] rtp.c: Ooh, format changed from unknown to alaw Jul 19 19:41:00 DEBUG[15347] chan_sip.c: = No match Their Call ID: xnrpwvawplnvjof@192.168.1.160 Their Tag gxqny Our tag: as324b2206 Jul 19 19:41:00 DEBUG[15347] chan_sip.c: Allocating new SIP dialog for 01-00561-0180cd86-4232bf2a1@freephonie.net - INVITE (With RTP) Jul 19 19:41:00 DEBUG[15347] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Jul 19 19:41:00 DEBUG[15347] chan_sip.c: Setting NAT on RTP to 0 Jul 19 19:41:00 DEBUG[15347] chan_sip.c: = No match Their Call ID: 01-00561-0180cd86-4232bf2a1@freephonie.net Their Tag 01-00561-0180cd87-27cdfef30 Our tag: as4b0f6c15 Jul 19 19:41:00 DEBUG[15347] chan_sip.c: = Found Their Call ID: xnrpwvawplnvjof@192.168.1.160 Their Tag gxqny Our tag: as324b2206 Jul 19 19:41:00 DEBUG[15347] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Jul 19 19:41:00 DEBUG[15347] chan_sip.c: Stopping retransmission on 'xnrpwvawplnvjof@192.168.1.160' of Response 638: Match Found Jul 19 19:41:01 DEBUG[23199] rtp.c: Got RTCP report of 52 bytes Jul 19 19:41:01 DEBUG[15347] chan_sip.c: = Found Their Call ID: 01-00561-0180cd86-4232bf2a1@freephonie.net Their Tag 01-00561-0180cd87-27cdfef30 Our tag: as4b0f6c15 Jul 19 19:41:01 DEBUG[15347] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Jul 19 19:41:01 DEBUG[15347] chan_sip.c: Stopping retransmission on '01-00561-0180cd86-4232bf2a1@freephonie.net' of Response 25491653: Match Found Jul 19 19:41:07 DEBUG[23199] rtp.c: Got RTCP report of 96 bytes Jul 19 19:41:11 DEBUG[15347] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Jul 19 19:41:11 DEBUG[15347] chan_sip.c: = Found Their Call ID: 262604ac7c68bdcb0f8b009767da7f6c@192.168.0.30 Their Tag Our tag: as2e11d2f1 Jul 19 19:41:11 DEBUG[15347] chan_sip.c: Stopping retransmission on '262604ac7c68bdcb0f8b009767da7f6c@192.168.0.30' of Request 102: Match Found Jul 19 19:41:12 DEBUG[15347] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Jul 19 19:41:12 DEBUG[15347] chan_sip.c: = Found Their Call ID: 68d37ecc373ad3c43d6961007abecd8d@62.147.170.96 Their Tag Our tag: as5cc60643 Jul 19 19:41:12 DEBUG[15347] chan_sip.c: Stopping retransmission on '68d37ecc373ad3c43d6961007abecd8d@62.147.170.96' of Request 102: Match Found Jul 19 19:41:12 DEBUG[23199] rtp.c: Got RTCP report of 96 bytes Jul 19 19:41:15 DEBUG[15347] chan_sip.c: Auto destroying call '01-00561-0180cd86-4232bf2a1@freephonie.net' Jul 19 19:41:19 DEBUG[23199] rtp.c: Got RTCP report of 96 bytes Jul 19 19:41:24 DEBUG[23199] rtp.c: Got RTCP report of 96 bytes Jul 19 19:41:31 DEBUG[23199] rtp.c: Got RTCP report of 96 bytes Jul 19 19:41:34 DEBUG[23199] rtp.c: Got RTCP report of 96 bytes Jul 19 19:41:34 DEBUG[15347] chan_sip.c: = Found Their Call ID: xnrpwvawplnvjof@192.168.1.160 Their Tag gxqny Our tag: as324b2206 Jul 19 19:41:34 DEBUG[15347] chan_sip.c: **** Received BYE (8) - Command in SIP BYE Jul 19 19:41:34 DEBUG[23199] channel.c: Didn't get a frame from channel: SIP/220-081c4618 Jul 19 19:41:34 DEBUG[23199] channel.c: Bridge stops bridging channels SIP/220-081c4618 and Zap/2-1 Jul 19 19:41:34 DEBUG[23199] channel.c: Hanging up channel 'Zap/2-1' My sip config is as follows (sip register is ok): [08XXXXXX] username=08XXXXXX type=user canreinvite=no secret=XXXXXX qualify=yes nat=never permit=212.27.52.5/255.255.255.255 deny=0.0.0.0/0.0.0.0 context=from-pstn allow=alaw [freephonie_outbound] username=08XXXXXX type=peer secret=XXXXXX qualify=yes host=freephonie.net fromuser=08XXXXXX fromdomain=freephonie.net context=from-internal nat=never canreinvite=no allow=alaw