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Summary:ASTERISK-11005: Maximum retries exceeded on transmission
Reporter:Jarl Walker (jwalker)Labels:
Date Opened:2007-12-10 15:31:17.000-0600Date Closed:2007-12-11 01:00:49.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:Good Day,

The issue is that after working currectly Asterisk started the give the errors below.

The call is establish without a problem but there is no Audio

Basically now i can't received nor place calls.

Thanks in Advance

****** ADDITIONAL INFORMATION ******

-- SIP/ivoip-099cda88 answered SIP/7002-099c9b18
   -- SIP/ivoip-099cda88 answered SIP/7002-099c9b18
[Dec 10 15:04:26] WARNING[2535]: chan_sip.c:1939 retrans_pkt: Maximum retries exceeded on transmission Zjk1ODU4ZGNlNTZjNGYxMTRiMWJjN2EyZDRhNzIyMDI. for seqno 2 (Critical Response)
Maximum retries exceeded on transmission Zjk1ODU4ZGNlNTZjNGYxMTRiMWJjN2EyZDRhNzIyMDI. for seqno 2 (Critical Response)
[Dec 10 15:04:26] WARNING[2535]: chan_sip.c:1963 retrans_pkt: Hanging up call Zjk1ODU4ZGNlNTZjNGYxMTRiMWJjN2EyZDRhNzIyMDI. - no reply to our critical packet.
Hanging up call Zjk1ODU4ZGNlNTZjNGYxMTRiMWJjN2EyZDRhNzIyMDI. - no reply to our critical packet.
================================================================================
Sip Debug Output
================================================================================
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '2952822f52d7fdda2adf255454fdea7d@172.21.37.195' Me                                                                              thod: INVITE
Really destroying SIP dialog 'NmM2MzNmM2VjZmI5ZTRkY2E4ZGJkZGU2NDkxNmE1NzQ.' Meth                                                                              od: INVITE

<--- SIP read from 72.55.158.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.21.37.195:5060;branch=z9hG4bK47aaf908;received=200.91.69.15                                                                              8;rport=5060
From: "Jarl Walker" <sip:7002@172.21.37.195>;tag=as12b8ed14
To: <sip:18779268123@72.55.158.172>;tag=as10299ebc
Call-ID: 2952822f52d7fdda2adf255454fdea7d@172.21.37.195
CSeq: 103 BYE
User-Agent: VMU_11ABS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:18779268123@72.55.158.172>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '2952822f52d7fdda2adf255454fdea7d@172.21.37.195' Me                                                                              thod: INVITE
Really destroying SIP dialog 'NmM2MzNmM2VjZmI5ZTRkY2E4ZGJkZGU2NDkxNmE1NzQ.' Meth                                                                              od: INVITE
localhost*CLI>
<--- SIP read from 200.91.69.155:2006 --->



<------------->
--- (0 headers 1 lines) ---

<--- SIP read from 200.91.69.155:2006 --->



<------------->
--- (0 headers 1 lines) ---
localhost*CLI>
<--- SIP read from 201.194.168.46:20850 --->



<------------->
--- (0 headers 1 lines) ---

<--- SIP read from 201.194.168.46:20850 --->



<------------->
--- (0 headers 1 lines) ---
localhost*CLI> [Dec 10 15:23:28] NOTICE[2524]: chan_sip.c:7334 sip_reregister:                                                                                  -- Re-registration for  1507120211@64.34.176.212
[Dec 10 15:23:28] NOTICE[2524]: chan_sip.c:7334 sip_reregister:    -- Re-registr                                                                              ation for  1507120211@64.34.176.212
localhost*CLI> REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 64.34.176.212:5060:
REGISTER sip:64.34.176.212 SIP/2.0
Via: SIP/2.0/UDP 172.21.37.195:5060;branch=z9hG4bK0de31ec0;rport
From: <sip:1507120211@64.34.176.212>;tag=as4b4f5cce
To: <sip:1507120211@64.34.176.212>
Call-ID: 14326d9c7ce344ae4ca2383131bc4ed0@127.0.0.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="xxxxxxxx", realm="did.voip.les.net", algorithm                                                                              =MD5, uri="sip:64.34.176.212", nonce="6a0cdf18", response="341bc9ebb8ea297228cf4                                                                              44703b2ca17", opaque=""
Expires: 120
Contact: <sip:1507120211@172.21.37.195>
Event: registration
Content-Length: 0


---
[Dec 10 15:23:35] NOTICE[2524]: chan_sip.c:7334 sip_reregister:    -- Re-registr                                                                              ation for  xxxxxxx@72.55.158.172
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 64.34.176.212:5060:
REGISTER sip:64.34.176.212 SIP/2.0
Via: SIP/2.0/UDP 172.21.37.195:5060;branch=z9hG4bK0de31ec0;rport
From: <sip:1507120211@64.34.176.212>;tag=as4b4f5cce
To: <sip:1507120211@64.34.176.212>
Call-ID: 14326d9c7ce344ae4ca2383131bc4ed0@127.0.0.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="xxxxxxx", realm="did.voip.les.net", algorithm                                                                              =MD5, uri="sip:64.34.176.212", nonce="6a0cdf18", response="341bc9ebb8ea297228cf4                                                                              44703b2ca17", opaque=""
Expires: 120
Contact: <sip:1507120211@172.21.37.195>
Event: registration
Content-Length: 0


---
[Dec 10 15:23:35] NOTICE[2524]: chan_sip.c:7334 sip_reregister:    -- Re-registr                                                                              ation for  xxxxxxx@72.55.158.172
Comments:By: Olle Johansson (oej) 2007-12-11 01:00:15.000-0600

This is not a support channel, this is a bug tracker. I suggest you use the asterisk-users mailing list or the #asterisk channel on IRC to get help with your configuration.

Thanks.

/O