Mark Mills
2004-May-31 17:03 UTC
[Asterisk-Users] audio problems between asterisk and Cisco 7910 using SCCP
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@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
Mark Mills
2004-Jun-01 04:06 UTC
[Asterisk-Users] audio problems between asterisk and Cisco 7910 using SCCP - SOLVED
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@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 > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Mark Mills phone: +61 421 707019 email: markmill@ravey.org www: www.ravey.org icq: 769320