Hello Paul,<div><br></div><div>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.)</div><div>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:</div>
<div><br></div><div><div>*CLI> sip show settings</div><div><br></div><div><br></div><div>Global Settings:</div><div>----------------</div><div> UDP Bindaddress: [::]:5060</div><div> ** Additional Info:</div><div>
[::] may include IPv4 in addition to IPv6, if such a feature is enabled in the OS.</div><div> TCP SIP Bindaddress: [::]:5060</div><div> TLS SIP Bindaddress: Disabled</div><div> Videosupport: Yes</div>
<div> Textsupport: Yes</div><div> Ignore SDP sess. ver.: Yes</div><div> AutoCreate Peer: No</div><div> Match Auth Username: No</div><div> Allow unknown access: No</div><div> Allow subscriptions: Yes</div>
<div> Allow overlap dialing: No</div><div> Allow promisc. redir: No</div><div> Enable call counters: Yes</div><div> SIP domain support: Yes</div><div> Realm. auth: No</div><div> Our auth realm <a href="http://sip.myhostname.com">sip.myhostname.com</a></div>
<div> Use domains as realms: No</div><div> Call to non-local dom.: Yes</div><div> URI user is phone no: Yes</div><div> Always auth rejects: Yes</div><div> Direct RTP setup: No</div><div> User Agent: asterisk</div>
<div> SDP Session Name: Asterisk PBX 1.8.8.0-rc1</div><div> SDP Owner Name: root</div><div> Reg. context: (not set)</div><div> Regexten on Qualify: No</div><div> Legacy userfield parse: No</div>
<div> Caller ID: asterisk</div><div> From: Domain: <a href="http://sip.myhostname.com">sip.myhostname.com</a></div><div> Record SIP history: On</div><div> Call Events: On</div><div>
Auth. Failure Events: Off</div><div> T.38 support: Yes</div><div> T.38 EC mode: FEC</div><div> T.38 MaxDtgrm: -1</div><div> SIP realtime: Disabled</div><div> Qualify Freq : 60000 ms</div>
<div> Q.850 Reason header: No</div><div> Store SIP_CAUSE: No</div><div><br></div><div>Network QoS Settings:</div><div>---------------------------</div><div> IP ToS SIP: CS3</div><div> IP ToS RTP audio: EF</div>
<div> IP ToS RTP video: AF41</div><div> IP ToS RTP text: AF41</div><div> 802.1p CoS SIP: 3</div><div> 802.1p CoS RTP audio: 5</div><div> 802.1p CoS RTP video: 4</div><div> 802.1p CoS RTP text: 3</div>
<div> Jitterbuffer enabled: Yes</div><div> Jitterbuffer forced: No</div><div> Jitterbuffer max size: 1000</div><div> Jitterbuffer resync: 1000</div><div> Jitterbuffer impl: fixed</div><div> Jitterbuffer log: No</div>
<div><br></div><div>Network Settings:</div><div>---------------------------</div><div> SIP address remapping: Disabled, no localnet list</div><div> Externhost: <none></div><div> Externaddr: (null)</div>
<div> Externrefresh: 10</div><div><br></div><div>Global Signalling Settings:</div><div>---------------------------</div><div> Codecs: 0xe (gsm|ulaw|alaw)</div><div> Codec Order: ulaw:20,alaw:20,gsm:20</div>
<div> Relax DTMF: No</div><div> RFC2833 Compensation: Yes</div><div> Symmetric RTP: No</div><div> Compact SIP headers: No</div><div> RTP Keepalive: 0 (Disabled)</div><div> RTP Timeout: 180</div>
<div> RTP Hold Timeout: 600</div><div> MWI NOTIFY mime type: application/simple-message-summary</div><div> DNS SRV lookup: No</div><div> Pedantic SIP support: No</div><div> Reg. min duration 60 secs</div>
<div> Reg. max duration: 3600 secs</div><div> Reg. default duration: 120 secs</div><div> Outbound reg. timeout: 30 secs</div><div> Outbound reg. attempts: 0</div><div> Notify ringing state: Yes</div><div> Include CID: Yes</div>
<div> Notify hold state: No</div><div> SIP Transfer mode: open</div><div> Max Call Bitrate: 384 kbps</div><div> Auto-Framing: Yes</div><div> Outb. proxy: <not set></div><div>
Session Timers: Refuse</div><div> Session Refresher: uac</div><div> Session Expires: 180 secs</div><div> Session Min-SE: 90 secs</div><div> Timer T1: 500</div><div> Timer T1 minimum: 100</div>
<div> Timer B: 32000</div><div> No premature media: Yes</div><div> Max forwards: 70</div><div><br></div><div>Default Settings:</div><div>-----------------</div><div> Allowed transports: UDP</div>
<div> Outbound transport: UDP</div><div> Context: default</div><div> Force rport: No</div><div> DTMF: rfc2833</div><div> Qualify: 500</div><div> Use ClientCode: No</div>
<div> Progress inband: Yes</div><div> Language: en</div><div> MOH Interpret: default</div><div> MOH Suggest: default</div><div> Voice Mail Extension: voicemail</div><div><br>
</div><div>----</div></div><div><br></div><div><br></div><div><br><div class="gmail_quote">On Fri, Oct 14, 2011 at 8:51 PM, Paul Belanger <span dir="ltr"><<a href="mailto:pabelanger@digium.com" target="_blank">pabelanger@digium.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div></div><div>On 11-10-14 01:20 PM, Eric Wieling wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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"<br>
<br>
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.<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-dev-bounces@lists.digium.com" target="_blank">asterisk-dev-bounces@lists.<u></u>digium.com</a> [mailto:<a href="mailto:asterisk-dev-bounces@lists.digium.com" target="_blank">asterisk-dev-bounces@<u></u>lists.digium.com</a>] On Behalf Of Catalin S.<br>
Sent: Friday, October 14, 2011 1:13 PM<br>
To: Asterisk Developers Mailing List<br>
Subject: [asterisk-dev] Many sip dialog/ opened channels.<br>
<br>
Hello,<br>
<br>
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.<br>
Is there someone else that en-counted the same problem? Is there any fix to this bug? I have the following settings:<br>
<br>
[snip]<br>
<br>
rr-de*CLI> sip show channels<br>
Peer User/ANR Call ID Format Hold Last Message Expiry Peer<br>
6.6.13.17 (None) 000750d5-411d00 0x0 (nothing) No Rx: REGISTER<guest><br>
6.1.13.17 (None) 7de7064b-6f9f69 0x0 (nothing) No Rx: REGISTER<guest><br>
6.1.18.13 (None) 08a2e79c7f13b73 0x0 (nothing) No Rx: REGISTER<guest><br>
1.2.12.23 (None) 000dbcd9-39db00 0x0 (nothing) No Rx: REGISTER<guest><br>
8.6.13.17 (None) 000750d5-411d00 0x0 (nothing) No Rx: REGISTER<guest><br>
8.1.13.17 (None) ca30cc15-d93e4d 0x0 (nothing) No Rx: REGISTER<guest><br>
6.1.12.17 (None) 226b901d-4bff19 0x0 (nothing) No Rx: REGISTER<guest><br>
9.1.12.20 (None) 2474013819@192_ 0x0 (nothing) No Rx: REGISTER<guest><br>
2.1.14.10 (None) d1bb5072-b6ebcd 0x0 (nothing) No Rx: REGISTER<guest><br>
xxxxx<br>
xxx<br>
xxxx<br>
8.1.13.17 (None) ca30cc15-d93e4d 0x0 (nothing) No Rx: REGISTER<guest><br>
6.1.12.17 (None) 226b901d-4bff19 0x0 (nothing) No Rx: REGISTER<guest><br>
9.1.12.20 (None) 2474013819@192_ 0x0 (nothing) No Rx: REGISTER<guest><br>
2.1.14.10 (None) d1bb5072-b6ebcd 0x0 (nothing) No Rx: REGISTER<guest><br>
<br>
4423 active SIP dialogs<br>
<br>
</blockquote></div></div>
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.<br>
<br>
[1] <a href="https://issues.asterisk.org/jira/browse/ASTERISK-17146" target="_blank">https://issues.asterisk.org/<u></u>jira/browse/ASTERISK-17146</a><br><font color="#888888">
-- <br>
Paul Belanger<br>
Digium, Inc. | Software Developer<br>
twitter: pabelanger | IRC: pabelanger (Freenode)<br>
Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a></font><div><div></div><div><br>
<br>
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