deepak@netandroid.com
2004-May-14 15:46 UTC
[Asterisk-Users] SoftPhone to SoftPhone with No Voice
Hello
I Installed Asterisk on RedHat 9. I am currently try to configure minimum with
two softphone talking to each other over the LAN. I am using X-Lite softphones
from xten.com site. I defined 3 phones in sip.conf and also specifies in
extensions.conf file. I am able to ring other computer but there is no voice
exchange ( i can't hear any think except ring). Here is the portion of
sip.conf
and extensions.conf.
Let me know if i missed something.
Thanks
Deepak
sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind SIP channel to
context = from-sip ; Default context for incoming calls
;srvlookup = yes ; Enable DNS SRV lookups on outbound calls
; Asterisk only uses the first host in SRV
records
;pedantic = yes ; Enable slow, pedantic checking for Pingtel
; and multiline formatted headers for strict
; SIP compatibility
;tos=lowdelay ; IP QoS parameter, either keyword or value
; like tos=184
;maxexpirey=3600 ; Max length of incoming registration we allow
realm=asterisk ; Our global authentication realm
;defaultexpirey=120 ; Default length of incoming/outoing
registration
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
;videosupport=yes ; Turn on support for SIP video
;disallow=all ; Disallow all codecs
allow=all ; Allow codecs in order of preference
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc
.....
[Phone1]
type=friend
host=dynamic
defaultip=192.168.3.103
dtmfmode=rfc2833
context=from-sip
callerid=" Win box " <1>
[Phone2]
type=friend
host=dynamic
defaultip=192.168.3.119
dtmfmode=rfc2833
context=from-sip
callerid=" Deepak" <2>
[Phone3]
type=friend
host=dynamic
defaultip=192.168.3.106
dtmfmode=rfc2833
context=from-sip
callerid=" Ravi " <3>
[extensions.conf]
[from-sip]
exten=>1,1,Dial(SIP/Phone1,20,tr)
exten=>2,1,Dial(SIP/Phone2,20,tr)
exten=>3,1,Dial(SIP/Phone3,20,tr)
----------------------------------------------------------------
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Do you have iptables turned on with rules that restrict packets to the RTP ports? Try doing an iptables --flush then see if it works. If so, you'll need to open up the UDP ports that RTP is configured to use. Normally Asterisk would open up the firewall by sending packets out those ports, but when both sides of the call live outside the firewall (iptables in this case) and they're both SIP you'll have this problem. See my previous posting for a more detailed explanation (or email me directly). -brian deepak@netandroid.com wrote:>Hello > >I Installed Asterisk on RedHat 9. I am currently try to configure minimum with >two softphone talking to each other over the LAN. I am using X-Lite softphones >from xten.com site. I defined 3 phones in sip.conf and also specifies in >extensions.conf file. I am able to ring other computer but there is no voice >exchange ( i can't hear any think except ring). Here is the portion of sip.conf >and extensions.conf. > >Let me know if i missed something. > >Thanks > >Deepak > >sip.conf >[general] >port = 5060 ; Port to bind to >bindaddr = 0.0.0.0 ; Address to bind SIP channel to >context = from-sip ; Default context for incoming calls >;srvlookup = yes ; Enable DNS SRV lookups on outbound calls > ; Asterisk only uses the first host in SRV >records >;pedantic = yes ; Enable slow, pedantic checking for Pingtel > ; and multiline formatted headers for strict > ; SIP compatibility >;tos=lowdelay ; IP QoS parameter, either keyword or value > ; like tos=184 >;maxexpirey=3600 ; Max length of incoming registration we allow >realm=asterisk ; Our global authentication realm >;defaultexpirey=120 ; Default length of incoming/outoing >registration >;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY >;videosupport=yes ; Turn on support for SIP video > >;disallow=all ; Disallow all codecs >allow=all ; Allow codecs in order of preference >;allow=ulaw ; Allow codecs in order of preference >;allow=ilbc > >..... >[Phone1] >type=friend >host=dynamic >defaultip=192.168.3.103 >dtmfmode=rfc2833 >context=from-sip >callerid=" Win box " <1> >[Phone2] >type=friend >host=dynamic >defaultip=192.168.3.119 >dtmfmode=rfc2833 >context=from-sip >callerid=" Deepak" <2> >[Phone3] >type=friend >host=dynamic >defaultip=192.168.3.106 >dtmfmode=rfc2833 >context=from-sip >callerid=" Ravi " <3> > >[extensions.conf] >[from-sip] >exten=>1,1,Dial(SIP/Phone1,20,tr) >exten=>2,1,Dial(SIP/Phone2,20,tr) >exten=>3,1,Dial(SIP/Phone3,20,tr) > > >---------------------------------------------------------------- >This message was sent using IMP, the Internet Messaging Program. > >_______________________________________________ >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 > >
Suggestions
disallow = all
allow = gsm
allow = ulaw
etc.
X-Lite/X-Pro from past experience does not do well unless this is done. You
might try other codecs besides gsm but I am betting this will get you started on
your way. I am by no means an expert just giving you something to try.
----- Original Message -----
From: <deepak@netandroid.com>
To: <asterisk-users@lists.digium.com>
Sent: Friday, May 14, 2004 3:46 PM
Subject: [Asterisk-Users] SoftPhone to SoftPhone with No Voice
Hello
I Installed Asterisk on RedHat 9. I am currently try to configure minimum with
two softphone talking to each other over the LAN. I am using X-Lite softphones
from xten.com site. I defined 3 phones in sip.conf and also specifies in
extensions.conf file. I am able to ring other computer but there is no voice
exchange ( i can't hear any think except ring). Here is the portion of
sip.conf
and extensions.conf.
Let me know if i missed something.
Thanks
Deepak
sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind SIP channel to
context = from-sip ; Default context for incoming calls
;srvlookup = yes ; Enable DNS SRV lookups on outbound calls
; Asterisk only uses the first host in SRV
records
;pedantic = yes ; Enable slow, pedantic checking for Pingtel
; and multiline formatted headers for strict
; SIP compatibility
;tos=lowdelay ; IP QoS parameter, either keyword or value
; like tos=184
;maxexpirey=3600 ; Max length of incoming registration we allow
realm=asterisk ; Our global authentication realm
;defaultexpirey=120 ; Default length of incoming/outoing
registration
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
;videosupport=yes ; Turn on support for SIP video
;disallow=all ; Disallow all codecs
allow=all ; Allow codecs in order of preference
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc
.....
[Phone1]
type=friend
host=dynamic
defaultip=192.168.3.103
dtmfmode=rfc2833
context=from-sip
callerid=" Win box " <1>
[Phone2]
type=friend
host=dynamic
defaultip=192.168.3.119
dtmfmode=rfc2833
context=from-sip
callerid=" Deepak" <2>
[Phone3]
type=friend
host=dynamic
defaultip=192.168.3.106
dtmfmode=rfc2833
context=from-sip
callerid=" Ravi " <3>
[extensions.conf]
[from-sip]
exten=>1,1,Dial(SIP/Phone1,20,tr)
exten=>2,1,Dial(SIP/Phone2,20,tr)
exten=>3,1,Dial(SIP/Phone3,20,tr)
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On Fri, 2004-05-14 at 17:46, deepak@netandroid.com wrote:> allow=all ; Allow codecs in order of preferenceI think the option allow=all should be renamed ScrewUpMyAudio=yes. Don't use allow=all. Use disallow=all and allow=ulaw for testing. For production use disallow=all and allow=thecodecyouwant. I do NOT mean disallow=all, allow=ulaw and allow=alaw. I mean disallow=all and allow=ulaw. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."