[Asterisk-Users] audio problems between asterisk and Cisco 7910 using SCCP - SOLVED

[Mark Mills]

Hi,

We must have been tired last night when we were trying to get this 
working, the problem has now been solved.

Just in case anyone has a similar problem in future and are searching the 
archives, the problem was caused by the /etc/hosts file.

*DO NOT* have the servers name listed as 127.0.0.1 in /etc/hosts !   Put 
it against one of the ethernet interfaces.

This is actually listed on www.voip-info.org and we still missed it   :)

Cheers,
   Mark


On Tue, 1 Jun 2004, Mark Mills wrote:

> 
> Hi,
> 
> I am working with a friend to setup two Asterisk servers over the 
> internet, one at each location and using IAX2 for trunking calls, using 
> Cisco 7910 phones and chan_sccp.   The phones are all the same hardware 
> and firmware revisions.
> 
> Lets call the sites AsteriskA and AsteriskB.   PhoneA is at AsteriskA, 
> PhoneB is at AsteriskB.
> 
> PhoneA has problems, when calling the local voice mail service at
> AsteriskA, the prompts are heard, button presses work, but audio does not
> appear to reach the asterisk server.  The following error message appears
> within the asterisk console:
> 
> Jun  1 08:43:01 WARNING[13326]: app_voicemail.c:1222 play_and_record: No 
> audio available on SCCP/201-00000001
> 
> The voice mail files that are created are empty.   Performing a packet 
> dump I do see packets going to the Asterisk server.
> 
> Now also IAX2 is setup between AsteriskA and AsteriskB, and that seems to 
> be functioning.   PhoneA and PhoneB can call each other from either 
> direction, but once again there is no sound coming from PhoneA, its only 
> one way.   If PhoneA is not answered, voicemail works and PhoneB can leave 
> messages that PhoneA can retrieve, but not the other way around.
> 
> We performed a packet dump When making calls between the two locations, 
> PhoneA sends data to AsteriskA, but AsteriskA doesnt forward it to 
> AsteriskB.   It seems that the voice traffic is going from PhoneA is not 
> being accepted at all?
> 
> Below is the config files that are in use for this setup. This has been 
> compiled from source using asterisk-0.9.0.tar.gz and 
> chan_sccp.02-easter.tar, on a Redhat 9 box running kernel 2.4.20-8.     
> 
> Does anyone have any idea what could be the problem and what we have 
> missed?
> 
> Thanks,
>   Mark
> 
> 
> /etc/asterisk/sccp.conf
> ==========================
> [general]
> 
> keepalive = 300
> context = default
> dateFormat = D/M/Y  
> 
> [SEP000427E8CD80]
> type            = 7910
> autologin       = 201
> description     = Extension 201
> 
> [201]
> id              = 201
> pin             = 1234
> label           = Mark Mills <201>
> description     = Mark Cisco 7910 Phone
> callwaiting     = 1
> mailbox         = 201
> callerid        = "Mark Mills", <201>
> 
> 
> 
> 
> /etc/asterisk/extensions.conf
> ==========================
> [general]
> static=yes
> writeprotect=no
> 
> [globals]
> CONSOLE=Console/dsp                             ; Console interface for 
> demo
> 
> [unknown]
> 
> exten => _.,1,Congestion
> 
> [default]
> 
> exten => 201,1,Macro(std-exten,SCCP/201,40)
> exten => _1XX,1,Dial(IAX2/asterisk:1945 at 150.101.55.194/${EXTEN}@default) 
> 
> exten => 999,1,wait(1)
> exten => 999,2,VoicemailMain(${CALLERIDNUM})
> exten => 999,3,Hangup
> 
> [macro-std-exten]
> exten => s,1,Dial(${ARG1},${ARG2})
> exten => s,2,Voicemail(u${MACRO_EXTEN})
> exten => s,3,Hangup
> exten => s,102,Voicemail(b${MACRO_EXTEN})
> exten => s,103,Hangup
> 
> 
> 
> 
> 
> 
> /etc/asterisk/modules.conf
> ==========================
> [modules]
> autoload=yes
> noload => pbx_gtkconsole.so
> noload => pbx_kdeconsole.so
> noload => app_intercom.so
> load => chan_modem.so
> load => res_musiconhold.so
> noload => chan_alsa.so
> noload => chan_skinny.so
> load => chan_sccp.so
> noload => chan_oh323.so
> 
> [global]
> chan_modem.so=yes
> 
> 
> 
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-- 
Mark Mills
phone:   +61 421 707019
email:   markmill at ravey.org
www:     www.ravey.org
icq:     769320