[Asterisk-Users] use of SIP SHOW CHANNELS question

Walker Haddock whaddock at datacrest.com
Tue Oct 14 12:40:31 MST 2003


On Tue, Oct 14, 2003 at 04:20:27PM -0300, CW_ASN - Gus wrote:
> Walker:
> 
> "sip show channel" refers to a Call ID:
> 
> noc2pbx2*CLI> sip show channels
> Peer             User/ANR    Call ID      Seq (Tx/Rx)  Lag      Jitter
> Format
> 172.16.254.62    0341522910  3607139911@  00101/00003  00000ms  0000ms  ALAW
> 1 active SIP channel(s)
> 
> Then, you could see the details:
> 
> noc2pbx2*CLI> sip show channel 3607139911 at 172.16.254.62
> Call-ID: 3607139911 at 172.16.254.62
> Our Codec Capability: 524302
> Non-Codec Capability: 1
> Joint Codec Capability: 12
> Theoretical Address: 172.16.254.62:5060
> Received Address:    172.16.254.62:5060
> NAT Support:         No
> Our Tag:             1906405977
> Their Tag:           1054824328
> Need Destroy:        0
> DTMF Mode: rfc2833
> 
> 
> Hope this helps,
> 
> Gus
Great, Gus!  Also to Martin.  I put 2 and 2 together and figured it out.

The tab key does work when you use the correct channel.  My system has a longer "Call ID" than your example, Gus.  Just for documentation, here's what it looks like on my version:
asterisk*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Lag      Jitter  Format
192.168.0.206    206         79010fe1-5a  00101/61141  00000ms  0000ms  ULAW
...
8 active SIP channel(s)
asterisk*CLI> sip show channel 79010fe1-5a5b-860f-2e7a-ce5403fbcf4b at 192.168.0.206
Call-ID: 79010fe1-5a5b-860f-2e7a-ce5403fbcf4b at 192.168.0.206
...

I typed in the Call ID and hit tab, voila!

Thanks, Walker

> 
> 
> ----- Original Message -----
> From: "Walker Haddock" <whaddock at datacrest.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Tuesday, October 14, 2003 3:37 PM
> Subject: [Asterisk-Users] use of SIP SHOW CHANNELS question
> 
> 
> > I am trying to figure out the correct syntax for the CLI command "SIP SHOW
> CHANNELS".  I have tried
> > SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is
> connected such as:
> >
> >     -- Zap/15-1 is ringing
> >     -- Zap/15-1 answered SIP/206-4299
> > asterisk*CLI> sip show channel SIP/206-4299
> > No such SIP Call ID 'SIP/206-4299'
> >
> >
> > I always get the "No such SIP Call ID ..."
> >
> > Thanks, Walker
> > --
> > ********   DataCrest, Inc. -- Technically Superior   ******************
> > Walker Haddock                       http://www.datacrest.com
> > DataCrest, Inc.                    e-mail:  wh at datacrest.com
> > 1634A Montgomery Hwy.    phone:  1-888-941-3282, 1-205-335-8589
> > Birmingham, AL 35216                  fax:  1-205-823-7838
> > ***********************************************************************
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
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-- 
********   DataCrest, Inc. -- Technically Superior   ******************
Walker Haddock                       http://www.datacrest.com
DataCrest, Inc.                    e-mail:  wh at datacrest.com
1634A Montgomery Hwy.    phone:  1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216                  fax:  1-205-823-7838
***********************************************************************



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