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 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

Ho to all,

I’ve recompiled asterisk, re-typed ‘make samples’ and re-edited conf file.
With canreinvite=no option the follow message disappears.

Sep 20 14:13:42 WARNING[2373]: channel.c:787 channel_find_locked: Avoided initial deadlock for ‘0x819b240’, 10 retries!

By traces, I’ve observed that several 200 OK SIP messages are sent by my SIP Provider until ACK is riceved. Maybe the 200 OK messages sequens freezes Asterisk introducing delay for biderctional audio flow.
Can anyone tell me if there is some option to set in order to manage sip messages time or similar?

Thanks for all