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 200 Ok 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