[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) This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoftPhone to SoftPhone with No Voice
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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoftPhone to SoftPhone with No Voice
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: [EMAIL PROTECTED] To: [EMAIL PROTECTED] 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) This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoftPhone to SoftPhone with No Voice
On Fri, 2004-05-14 at 17:46, [EMAIL PROTECTED] wrote: allow=all ; Allow codecs in order of preference I 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. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
