[OpenSIPS-Users] Strange errors forwarding requests
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Thu May 6 11:35:50 CEST 2010
What is the state of the dialog stuck in memory? (use dlg_list MI
command to see them).
Also check that the sequential requests (ACK, re-INVITE, BYE) do go
through your proxy (via loose_route).
Regards,
Bogdan
Erik Versaevel - InfoPact Netwerkdiensten wrote:
> That might be :)
>
> I'm now running into problems with the dialog module (which i use to limit concurrent calls).
> Calls seem to stick in the dialog module (thus denying additional calls) while the endpoint isn't
> listing the same amount of calls :/
>
> Regards,
> Erik
>
>
> Op 6-5-2010 11:02, Bogdan-Andrei Iancu schreef:
>
>> So, after all, it was a network layer configuration issue... :)
>>
>> Regards,
>> Bogdan
>>
>> Erik Versaevel wrote:
>>
>>> The destination (in this case) is the 1st server in the loadbalancer
>>> list (as there are no other calls).
>>> I've upgraded this machine to ubuntu 10 (from 8) and started getting
>>> Connection Tracking drop messages in my
>>> syslog. I've disabled connection tracking and the issue hasn't
>>> appeared since...
>>>
>>>
>>> Op 5-5-2010 12:24, Bogdan-Andrei Iancu schreef:
>>>
>>>
>>>> Hi Erik,
>>>>
>>>> have you tried to print the destination of the requests that fail?
>>>>
>>>> regards,
>>>> Bogdan
>>>>
>>>> Erik Versaevel wrote:
>>>>
>>>>
>>>>> Hi All,
>>>>>
>>>>> I attempted an migration last night (from our current environment to
>>>>> this new setup) but i ran into this
>>>>> problem as soon as i tried to make some test calls, funny thing is i
>>>>> can't get it reproduced :/ Any clues
>>>>> on how to debug this any further?
>>>>>
>>>>> Kind regards,
>>>>>
>>>>> Erik
>>>>>
>>>>> Op 27-4-2010 15:17, Erik Versaevel schreef:
>>>>>
>>>>>
>>>>>> Hi all,
>>>>>>
>>>>>> I'm building a setup in which opensips is acting as registar for my
>>>>>> endpoints and loadbalancing
>>>>>> calls made by those endpoint over an cluster of asterisk machines.
>>>>>> (so that if we need more asterisk
>>>>>> power, we just have to add another destination to the loadbalancer
>>>>>> module)
>>>>>> Opensips is listening on multiple IP addresses and uses the
>>>>>> loadbalancer module to poll my asterisk
>>>>>> machines and select the destination.
>>>>>> My problem is that every now and then opensips fails to forward an
>>>>>> invite to my asterisk cluster and
>>>>>> generates
>>>>>>
>>>>>> "ERROR:core:udp_send:
>>>>>> sendto(sock,0x77b81280,1353,0,0x77b81b04,16): Operation not
>>>>>> permitted(1)"
>>>>>>
>>>>>> there is some iptables filtering on this machine, however it is not
>>>>>> showing drops in the logfile (and it keeps
>>>>>> occuring even without any iptable rules).
>>>>>> I tried stracing opensips but all i get is:
>>>>>>
>>>>>> opensipstrace.7423:sendto(6, "INVITE
>>>>>> sip:E164_DST_PHONE_NR at OPENSIPS_IP_ADDRESS SIP/2.0
>>>>>> Record-Route:
>>>>>> <sip:OPENSIPS_IP_ADDRESS;lr=on;ftag=AI05ED431A05432EB8;nat=yes;did=fd6.e1f16fe3;vsf=AAAAAAMIBgl3AggLFgF5HAAFGhwBHzE3NC44MQ-->
>>>>>>
>>>>>> Via: SIP/2.0/UDP OPENSIPS_IP_ADDRESS;branch=z9hG4bK3177.1e0e38b7.0
>>>>>> Via: SIP/2.0/UDP
>>>>>> 192.168.178.44:5060;received=CPE_IP_ADDRESS;rport=61008;branch=z9hG4bK2010Apr222938466E164_DST_PHONE_NR
>>>>>>
>>>>>> To: <sip:E164_DST_PHONE_NR at OPENSIPS_IP_ADDRESS>
>>>>>> From: \"3961\"
>>>>>> <sip:3961 at OPENSIPS_IP_ADDRESS>;tag=AI05ED431A05432EB8
>>>>>> Call-ID: AIF001C45E85F7921C at 192.168.178.44
>>>>>> CSeq: 2 INVITE
>>>>>> Max-Forwards: 69
>>>>>> Contact:
>>>>>> <sip:E164PHONE_NR at CPE_IP_ADDRESS:61008;line=AIF8F01E8DF866D7CB>
>>>>>> Accept: application/sdp
>>>>>> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER
>>>>>> Allow-Events: dialog,message-summary
>>>>>> P-Preferred-Identity: <sip:E164PHONE_NR at OPENSIPS_IP_ADDRESS>
>>>>>> Privacy: none
>>>>>> User-Agent: SomeStrangeDude
>>>>>> Content-Type: application/sdp
>>>>>> Content-Length: 324
>>>>>> I-FromDisp: <null>
>>>>>> I-FromUri: E164PHONE_NR
>>>>>> I-CustId: 3961
>>>>>>
>>>>>> v=0
>>>>>> o=intelligate 1133701155 1133701155 IN IP4 192.168.178.44
>>>>>> s=call
>>>>>> c=IN IP4 CPE_IP_ADDRESS
>>>>>> t=0 0
>>>>>> m=audio 5004 RTP/AVP 18 8 101
>>>>>> a=rtpmap:18 G729/8000
>>>>>> a=fmtp:18 annexb=no
>>>>>> a=rtpmap:8 PCMA/8000
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>> a=fmtp:101 0-15
>>>>>> a=sendrecv
>>>>>> a=ptime:20
>>>>>> a=direction:active
>>>>>> a=oldmediaip:192.168.178.44
>>>>>> ", 1253, 0, {sa_family=AF_INET, sin_port=htons(5060),
>>>>>> sin_addr=inet_addr("ASTERISK_IP_ADDRESS")}, 16) = -1 EPERM
>>>>>> (Operation not permitted)
>>>>>>
>>>>>> I also use the uac_replace_from() to mangle the from header so
>>>>>> asterisk uses the correct user/peer/client to connect the call
>>>>>> (codec/dialplan etc).
>>>>>> I'm having trouble reproducing the error as it's not allways
>>>>>> occuring, the errors i straced where mainly the initial invite
>>>>>> towards my asterisk
>>>>>> cluster and a few 200 OK's which didn't get processed correctly.
>>>>>>
>>>>>> Any clues on how to debug this further?
>>>>>>
>>>>>> Kind regards,
>>>>>>
>>>>>> Erik Versaevel
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>
>>> Erik Versaevel
>>>
>>>
>>>
>>
>
>
>
> Erik Versaevel
>
>
--
Bogdan-Andrei Iancu
www.voice-system.ro
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