SUGGESTIONS Cisco 7970 Treasure Trove

[walker_jr]

Sorted the backlight behaviour. Just so we have everything in one place, you need :

<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>

in your <vendorConfig> section of the SEPXXXXXXXXXX.cnf.xml file(s), then reboot the phone(s) (press **#** when in the settings section).

Dialling behaviour is still annoying (ie, dial, then wait 5 seconds for things to start happening), even after changing the dialplan.xml (and also copying it to DIALPLAN.XML in case the filename case is important). Googling around some people say that the dialplan.xml file isn't read when in SCCP mode, which would be annoying. Certainly looking at the tftpd port it doesn't seem to get requested.

One posting suggests telnetting to the phone and using "show dialplan" - this only makes sense if you log in as debug/debug and when you do that there is no show dialplan command anyway, which supports the theory it's not used under SCCP?

 

[tabbertmj]

There are timeout values in the sccp.conf file. Under general, add...

firstdigittimeout = 16 ; dialing timeout for the 1st digit
digittimeout = 5 ; more digits
digittimeoutchar = # ; you can force the channel to dial with this char in the dialing state

Then adjust values as you see fit. Then you can also dial pound to dial. I usually dial the number before I pick up or hit dial. Then it calls right away.

 

[walker_jr]

Is it possible to have the phone log into two different Call Managers? The phone itself seems to suggest it can log into up to 5, but I don't know how to do this from SEP<MAC>.conf and I can't edit it from the phone itself.

My employer runs a Cisco Call Manager setup and it would be nice, when I work from home, to log into my work extension.

Of course, then I have to figure out how to get the phone onto my IPSec tunnel, which runs from a client on the PC... hmmm.

 

[jroper]

Hi

I think Windows internet connection sharing wizard and maybe another NIC may do the trick there,

Joe

 

[walker_jr]

I wonder what I'm doing wrong on my display configuration?


The phone's logs say :

ERR 11:12:23.753146 JVM: Startup Module Loader|cip.setg.ScreenSaveProperty:? - Config file field <displayOnDuration> is incorrectly formatted. Defaulting.
ERR 11:12:23.754860 JVM: Startup Module Loader|cip.setg.ScreenSaveProperty:? - Config file field <displayOnTime> is incorrectly formatted. Defaulting.
ERR 11:12:23.756508 JVM: Startup Module Loader|cip.setg.ScreenSaveProperty:? - Config file field <displayIdleTimeout> is incorrectly formatted. Defaulting.

My SEP<MAC>.cnf.xml says :

<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>8:30</displayOnTime>
<displayOnDuration>0:10</displayOnDuration>
<displayIdleTimeout>0:10</displayIdleTimeout>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>

... basically I only want the phone screens on when there's a call, and after 10 minutes of inactivity I would like them to turn back off.

 

[walker_jr]

Hmm, the unwelcome return (or actually, I don't think I fixed it the first time) of the no audio in either direction thing. I've turned iptables back off, but it makes no difference.

I get dialtone and I can dial out, incoming calls ring, but there is no audio in either direction (no ringing tones, if someone answers I can't hear them, they can't hear me).

Looking at the Asterisk console everything looks normal. SIP on another phone is working fine, so I know I have Asterisk configured correctly, it just seems to be SCCP.

The tshark output for port 2000 is fine in that the control traffic (KeepAlives, CallInfo, OffHook etc.) all seems to be fine, but when talking there's no packets exchanged except the KeepAlives.

 

[walker_jr]

Further information : even internal (extension to extension) calls don't get any audio.

These devices are all on the same LAN with the * box and I use a SPA3K for the outside line, so they don't even traverse the firewall. What am I doing wrong?

 

[jroper]

Hi

Audio problems have some fairly common causes, all othr things being equal, and can usually be eliminated one by one.

1. Someone has pressed a mute button on the phone.
2. Someone has turned down the volume on the phone.
3. NAT traversal issues with SIP.
4. Service provider issues.
5. Incompatible codecs, using codecs not supported by Asterisk, or ones that cannot be transcoded.

Given that these are problems within your own internal network, we can discount 3 and 4 straight away.

This leaves 1,2 and 4. I suspect you have probably checked 1 and 2, so I would turn my attention to 4.

Joe

 

[walker_jr]

What would you define as a service provider issue, given that there are no external factors here - just me and my PiaF box?

I'm also concerned by the phone log, which says :

WRN 14:48:07.197511 JVM: Startup Module Loader|cip.mmgr.ds:? - [MediaMgrSM]: Unhandled Event, State = StateSpeakerOffHook Event = EventSetSpeakerMode
ERR 14:48:07.714999 DSP: wcTrans*** Invalid response 5
ERR 14:48:07.715827 DSP: Tone*** Connect/disconnect fails: -1
ERR 14:48:07.788178 DSP: wcTrans*** Invalid response 5
ERR 14:48:07.789042 DSP: Tone*** Connect/disconnect fails: -1
ERR 14:48:08.289134 DSP: wcTrans*** Invalid response 5
ERR 14:48:08.289935 DSP: Tone*** Connect/disconnect fails: -1
ERR 14:48:08.686084 DSP: wcTrans*** Invalid response 5
ERR 14:48:08.686912 DSP: Tone*** Connect/disconnect fails: -1
ERR 14:48:19.508749 DSP: wcTrans*** Invalid response 5
ERR 14:48:19.509611 DSP: Tone*** Connect/disconnect fails: -1
WRN 14:48:19.726487 JVM: Startup Module Loader|cip.sccp.ck:? - Invalid SCCP message! : ID :105: readInt() underrun, length = 0 - close connection and alarm in future
ERR 14:48:19.731921 JVM: 14:48:19|cip.sccp.ba: ID :105: readInt() underrun, length = 0
at cip.sccp.av.readInt(Unknown Source)
at cip.sccp.bu.b(Unknown Source)
at cip.sccp.br.a(Unknown Source)
at cip.sccp.av.a(Unknown Source)
at cip.sccp.ck.e(Unknown Source)

 

[walker_jr]

Ah, sorry, you meant 5.

Well, the sccp.conf is allowing ulaw and alaw, in that order. I'm not sure what else to check?

 

[tabbertmj]

I think it should be four digits...

<displayOnTime>08:30</displayOnTime>
<displayOnDuration>00:10</displayOnDuration>
<displayIdleTimeout>00:10</displayIdleTimeout>

 

[walker_jr]

I looooove talking to myself.

Further digging found this in the * log :

[2009-02-02 17:48:42] WARNING[3309] config.c: No '=' (equal sign) in line 103 of /etc/asterisk/sccp.conf
[2009-02-02 17:48:42] WARNING[3309] config.c: No '=' (equal sign) in line 127 of /etc/asterisk/sccp.conf
[2009-02-02 17:48:42] WARNING[3309] config.c: No '=' (equal sign) in line 178 of /etc/asterisk/sccp.conf
[2009-02-02 17:48:42] WARNING[3309] chan_sccp.c: SCCP: Unknown param at line 73: context = from-internal
[2009-02-02 17:48:42] WARNING[3309] chan_sccp.c: SCCP: Unknown param at line 106: context = from-internal

Apart from the fact that lines 103, 127 and 187 do have equals signs in them, I'm most concerned about it not understanding what the from-internal context is. Unless that is only supposed to appear once, at the top of the file, and not in the individual line declarations?

 

[walker_jr]

Mike,

You may well be right, that's what I had to start. Cisco disagreed in one piece of documentation, but I'll switch back.

 

[walker_jr]

sccp.conf file attached.

 

[walker_jr]

Asterisk now loads sccp.conf without complaint, but still no audio in either direction from the 7965s, whether on an internal or external call. But they can phone numbers and ring when called.

 

[tabbertmj]

Do you get audio when you call your other SIP phone if the Cisco is on speakerphone?

 

[walker_jr]

No, no audio - no sound, either on speaker or on handset, on either phone, either phoning each other or a SIP endpoint. I do however get ringing tones when calling an internal extension, which I don't get if I phone an outside number.

As I said the phones do ring when called, and can dial fine.

I'm (clearly) no expert, but it feels like either a NAT (the phones or * are expecting NAT, which isn't there) or a firewall problem. I have iptables turned off for now, but I don't really know where to look to identify the problem.

I see that when I make a Cisco to Cisco call, the logs indicate it's expecting to be able to do RTP, but isn't able to. I don't know if this is expected behaviour.

 

[jroper]

Hi

In the extension config in FreePBX, have you tried changing NAT=Yes to NAT=no?

Joe

 

[walker_jr]

No, because when you make a custom extension for SCCP, there's no option to do so.

 

[tabbertmj]

Can you paste in to notepad the console output when you call a SIP phone? It might be a bit easier to look at then if you call a second sccp phone. I'm at a loss on this one. Your sccp.conf file looks good. I just set up a second 7965 today and it worked fine.