[asterisk-dev] Many sip dialog/ opened channels.
Catalin S.
jonsonplayer at gmail.com
Sat Oct 15 11:24:06 CDT 2011
Hello Paul,
Can you suggest me some settings? I think is something related to my timers
(Reg. min duration , Reg. max duration: , Reg. default duration:
, Outbound reg. timeout:, etc.)
Also I want to use TCP for a brand new Cisco Phone (Cisco-CP9971/9.2.1) and
IPv6 for terminals that comes from IPv6 networks. Right now i had the
following times/settings:
*CLI> sip show settings
Global Settings:
----------------
UDP Bindaddress: [::]:5060
** Additional Info:
[::] may include IPv4 in addition to IPv6, if such a feature is enabled
in the OS.
TCP SIP Bindaddress: [::]:5060
TLS SIP Bindaddress: Disabled
Videosupport: Yes
Textsupport: Yes
Ignore SDP sess. ver.: Yes
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promisc. redir: No
Enable call counters: Yes
SIP domain support: Yes
Realm. auth: No
Our auth realm sip.myhostname.com
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: Yes
Always auth rejects: Yes
Direct RTP setup: No
User Agent: asterisk
SDP Session Name: Asterisk PBX 1.8.8.0-rc1
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Legacy userfield parse: No
Caller ID: asterisk
From: Domain: sip.myhostname.com
Record SIP history: On
Call Events: On
Auth. Failure Events: Off
T.38 support: Yes
T.38 EC mode: FEC
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
---------------------------
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: AF41
802.1p CoS SIP: 3
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 4
802.1p CoS RTP text: 3
Jitterbuffer enabled: Yes
Jitterbuffer forced: No
Jitterbuffer max size: 1000
Jitterbuffer resync: 1000
Jitterbuffer impl: fixed
Jitterbuffer log: No
Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externaddr: (null)
Externrefresh: 10
Global Signalling Settings:
---------------------------
Codecs: 0xe (gsm|ulaw|alaw)
Codec Order: ulaw:20,alaw:20,gsm:20
Relax DTMF: No
RFC2833 Compensation: Yes
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 180
RTP Hold Timeout: 600
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 30 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: Yes
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: Yes
Outb. proxy: <not set>
Session Timers: Refuse
Session Refresher: uac
Session Expires: 180 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: default
Force rport: No
DTMF: rfc2833
Qualify: 500
Use ClientCode: No
Progress inband: Yes
Language: en
MOH Interpret: default
MOH Suggest: default
Voice Mail Extension: voicemail
----
On Fri, Oct 14, 2011 at 8:51 PM, Paul Belanger <pabelanger at digium.com>wrote:
> On 11-10-14 01:20 PM, Eric Wieling wrote:
>
>> SIP channels are created of a variety of reasons. The ones below are
>> created because of a registration this is normal. To see a list of calls
>> (channels associated with a call), use "core show channels"
>>
>> I find it odd the peer is shown as<guest>. Are those IPs listed actual
>> real IPs or did you mask them? If they are real, I'd say your server is
>> under attack.
>>
>> -----Original Message-----
>> From: asterisk-dev-bounces at lists.**digium.com<asterisk-dev-bounces at lists.digium.com>[mailto:
>> asterisk-dev-bounces@**lists.digium.com<asterisk-dev-bounces at lists.digium.com>]
>> On Behalf Of Catalin S.
>> Sent: Friday, October 14, 2011 1:13 PM
>> To: Asterisk Developers Mailing List
>> Subject: [asterisk-dev] Many sip dialog/ opened channels.
>>
>> Hello,
>>
>> I'm using asterisk with 84 extensions (aprox 45 always connected). When i
>> look to the opened channels i sow many channels opened without reason even i
>> don't have any active calls.
>> Is there someone else that en-counted the same problem? Is there any fix
>> to this bug? I have the following settings:
>>
>> [snip]
>>
>> rr-de*CLI> sip show channels
>> Peer User/ANR Call ID Format Hold
>> Last Message Expiry Peer
>> 6.6.13.17 (None) 000750d5-411d00 0x0 (nothing) No
>> Rx: REGISTER<guest>
>> 6.1.13.17 (None) 7de7064b-6f9f69 0x0 (nothing) No
>> Rx: REGISTER<guest>
>> 6.1.18.13 (None) 08a2e79c7f13b73 0x0 (nothing) No
>> Rx: REGISTER<guest>
>> 1.2.12.23 (None) 000dbcd9-39db00 0x0 (nothing) No
>> Rx: REGISTER<guest>
>> 8.6.13.17 (None) 000750d5-411d00 0x0 (nothing) No
>> Rx: REGISTER<guest>
>> 8.1.13.17 (None) ca30cc15-d93e4d 0x0 (nothing) No
>> Rx: REGISTER<guest>
>> 6.1.12.17 (None) 226b901d-4bff19 0x0 (nothing) No
>> Rx: REGISTER<guest>
>> 9.1.12.20 (None) 2474013819 at 192_ 0x0 (nothing) No Rx:
>> REGISTER<guest>
>> 2.1.14.10 (None) d1bb5072-b6ebcd 0x0 (nothing) No
>> Rx: REGISTER<guest>
>> xxxxx
>> xxx
>> xxxx
>> 8.1.13.17 (None) ca30cc15-d93e4d 0x0 (nothing) No
>> Rx: REGISTER<guest>
>> 6.1.12.17 (None) 226b901d-4bff19 0x0 (nothing) No
>> Rx: REGISTER<guest>
>> 9.1.12.20 (None) 2474013819 at 192_ 0x0 (nothing) No Rx:
>> REGISTER<guest>
>> 2.1.14.10 (None) d1bb5072-b6ebcd 0x0 (nothing) No
>> Rx: REGISTER<guest>
>>
>> 4423 active SIP dialogs
>>
>> If I had to guess this looks to be the regression[1] in netsock2 when we
> converted to IPv6. It basically happens when asterisk dials an unknown SIP
> peer.
>
> [1] https://issues.asterisk.org/**jira/browse/ASTERISK-17146<https://issues.asterisk.org/jira/browse/ASTERISK-17146>
> --
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
>
>
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