flavio
2006-Sep-21 01:26 UTC
[asterisk-users] Unexpected delay: problem with outgoing calls
Hi to all. I've registred my Asterisk 1.2.12.1 to a VoIP Service Provider and I've some problem with outgoing calls: there is a big delay for bidirectional audio flow. Here is mean part of an asterisk trace releted to outgoing calls. (canreinvite=no for both peers). Until SIP 180 ringing signaling is correct...bold highlight time for NOTICE _ _ _ _ Sep 18 16:01:43 [1;33;40mNOTICE[0;37;40m[23098]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m9854[0;37;40m [1;37;40mhandle_response_register[0;37;40m: Outbound Registration: Expiry for 10.28.52.74 is 3599 sec (Scheduling reregistration in 3584 s) [1;30;40m -- [0;37;40mSIP/outgoing-08197388 is ringing Transmitting (no NAT) to 10.28.52.244:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.28.52.244;branch=z9hG4bKc39fdf160cf737ac;received=10.28.52.244 From: <sip:bt102@10.28.52.246;user=phone>;tag=a82e9be13c882482 To: <sip:067202XXXX@10.28.52.246;user=phone>;tag=as2ea0ddd1 Call-ID: a489c3f6ff15e77a@10.28.52.244 CSeq: 829 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:067202XXXX@10.28.52.246> Content-Length: 0 --- [1;30;40m -- [0;37;40mSIP/outgoing-08197388 is making progress passing it to SIP/bt102-08190d90 Sep 18 16:02:37 [1;33;40mNOTICE[0;37;40m[23098]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m11613[0;37;40m [1;37;40msip_poke_noanswer[0;37;40m: Peer 'outgoing' is now UNREACHABLE! Last qualify: 4 <-- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- <-- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- <-- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- <-- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- <-- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- [1;30;40m -- [0;37;40mSIP/outgoing-08197388 answered SIP/bt102-08190d90 We're at 10.28.52.246 port 16274 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 10.28.52.244:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.28.52.244;branch=z9hG4bKc39fdf160cf737ac;received=10.28.52.244 From: <sip:bt102@10.28.52.246;user=phone>;tag=a82e9be13c882482 To: <sip:067202XXXX@10.28.52.246;user=phone>;tag=as2ea0ddd1 Call-ID: a489c3f6ff15e77a@10.28.52.244 CSeq: 829 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:067202XXXX@10.28.52.246> Content-Type: application/sdp Content-Length: 184 v=0 o=root 23109 23110 IN IP4 10.28.52.246 s=session c=IN IP4 10.28.52.246 t=0 0 m=audio 16274 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- [1;30;40m -- [0;37;40mAttempting native bridge of SIP/bt102-08190d90 and SIP/outgoing-08197388 Sep 18 16:03:25 [1;33;40mNOTICE[0;37;40m[23098]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m9882[0;37;40m [1;37;40mhandle_response_peerpoke[0;37;40m: Peer 'outgoing' is now REACHABLE! (6ms / 2000ms) <-- SIP read from 10.28.52.244:5060: ACK sip:067202XXXX@10.28.52.246 SIP/2.0 Via: SIP/2.0/UDP 10.28.52.244;branch=z9hG4bK30db550457acdb99 From: <sip:bt102@10.28.52.246;user=phone>;tag=a82e9be13c882482 To: <sip:067202XXXX28.52.246;user=phone>;tag=as2ea0ddd1 Contact: <sip:bt102@10.28.52.244;user=phone> Call-ID: a489c3f6ff15e77a@10.28.52.244 CSeq: 829 ACK User-Agent: Grandstream BT110 1.0.8.12 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 _ _ _ _>From trace it points out that time gap from 180 Ringing and follow 200Ok is about 1 minute.. and so from 200 OK and ACK Any suggestions? Moreover..when I attempt to make an outgoing call with option canreinvite=yes, Asterisk notifies the follow message? Sep 20 14:13:42 WARNING[2373]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x819b240', 10 retries! Can anyone tell me what it does mean and how to fix it? Thanks 4 all -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ******************************** -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************
Hi all, I have next scenario: One E1 Line connected to my asterisk through TE412P. One analog modem connected to an PAP2. I want to make internet conecction with this modem through E1 Primary Line but I obtain more thatn 95% of errors in connections. Must I do anything to permit data connection with an analog modem through E1 ISDN Primary Line? Regards, Tron -----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] En nombre de flavio Enviado el: jueves, 21 de septiembre de 2006 10:26 Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] Unexpected delay: problem with outgoing calls Hi to all. I've registred my Asterisk 1.2.12.1 to a VoIP Service Provider and I've some problem with outgoing calls: there is a big delay for bidirectional audio flow. Here is mean part of an asterisk trace releted to outgoing calls. (canreinvite=no for both peers). Until SIP 180 ringing signaling is correct...bold highlight time for NOTICE _ _ _ _ Sep 18 16:01:43 [1;33;40mNOTICE[0;37;40m[23098]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m9854[0;37;40m [1;37;40mhandle_response_register[0;37;40m: Outbound Registration: Expiry for 10.28.52.74 is 3599 sec (Scheduling reregistration in 3584 s) [1;30;40m -- [0;37;40mSIP/outgoing-08197388 is ringing Transmitting (no NAT) to 10.28.52.244:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.28.52.244;branch=z9hG4bKc39fdf160cf737ac;received=10.28.52.244 From: <sip:bt102@10.28.52.246;user=phone>;tag=a82e9be13c882482 To: <sip:067202XXXX@10.28.52.246;user=phone>;tag=as2ea0ddd1 Call-ID: a489c3f6ff15e77a@10.28.52.244 CSeq: 829 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:067202XXXX@10.28.52.246> Content-Length: 0 --- [1;30;40m -- [0;37;40mSIP/outgoing-08197388 is making progress passing it to SIP/bt102-08190d90 Sep 18 16:02:37 [1;33;40mNOTICE[0;37;40m[23098]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m11613[0;37;40m [1;37;40msip_poke_noanswer[0;37;40m: Peer 'outgoing' is now UNREACHABLE! Last qualify: 4 <-- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- <-- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- <-- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- <-- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- <-- SIP read from 10.28.52.244:5060: --- (0 headers 0 lines) Nat keepalive --- [1;30;40m -- [0;37;40mSIP/outgoing-08197388 answered SIP/bt102-08190d90 We're at 10.28.52.246 port 16274 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 10.28.52.244:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.28.52.244;branch=z9hG4bKc39fdf160cf737ac;received=10.28.52.244 From: <sip:bt102@10.28.52.246;user=phone>;tag=a82e9be13c882482 To: <sip:067202XXXX@10.28.52.246;user=phone>;tag=as2ea0ddd1 Call-ID: a489c3f6ff15e77a@10.28.52.244 CSeq: 829 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:067202XXXX@10.28.52.246> Content-Type: application/sdp Content-Length: 184 v=0 o=root 23109 23110 IN IP4 10.28.52.246 s=session c=IN IP4 10.28.52.246 t=0 0 m=audio 16274 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- [1;30;40m -- [0;37;40mAttempting native bridge of SIP/bt102-08190d90 and SIP/outgoing-08197388 Sep 18 16:03:25 [1;33;40mNOTICE[0;37;40m[23098]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m9882[0;37;40m [1;37;40mhandle_response_peerpoke[0;37;40m: Peer 'outgoing' is now REACHABLE! (6ms / 2000ms) <-- SIP read from 10.28.52.244:5060: ACK sip:067202XXXX@10.28.52.246 SIP/2.0 Via: SIP/2.0/UDP 10.28.52.244;branch=z9hG4bK30db550457acdb99 From: <sip:bt102@10.28.52.246;user=phone>;tag=a82e9be13c882482 To: <sip:067202XXXX28.52.246;user=phone>;tag=as2ea0ddd1 Contact: <sip:bt102@10.28.52.244;user=phone> Call-ID: a489c3f6ff15e77a@10.28.52.244 CSeq: 829 ACK User-Agent: Grandstream BT110 1.0.8.12 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 _ _ _ _>From trace it points out that time gap from 180 Ringing and follow 200Ok is about 1 minute.. and so from 200 OK and ACK Any suggestions? Moreover..when I attempt to make an outgoing call with option canreinvite=yes, Asterisk notifies the follow message? Sep 20 14:13:42 WARNING[2373]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x819b240', 10 retries! Can anyone tell me what it does mean and how to fix it? Thanks 4 all -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ******************************** -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ******************************** _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users