SV: [asterisk-users] lost packets when bridging zap and iax
Hi all, I have got same problem - bridging between IAX and IAX goes fine without lost packets. ZAP to IAX - one lag show lost packets. Any ideas and/or solutions? Best regards, Dmitry -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Simone Cittadini Sendt: 28. august 2006 13:58 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [asterisk-users] lost packets when bridging zap and iax We have a machine with a TE410P in it acting as a client to route calls via iax2 to our central server, caller -- ( zap - iax ) --- ( iax - whatever ) -- called client server often the called can't hear the caller (both machines on public ip) 'iax2 show netstats on client machine shows more and more dropped packets on the local side if we use sip as the entering point for the calls all works well : caller -- ( sip - iax ) --- ( iax - whatever ) -- called client server seems something in the bridging between zap and iax screws up, but I don't know if it's a bug or a misconfiguration, my conf files follows, someone has similar experiences to share ? /etc/asterisk# cat iax.conf [general] bindport=4569 bindaddr=xxx.xx.xx.xxx disallow=all allow=alaw jitterbuffer=yes forcejitterbuffer=no tos=lowdelay autokill=yes language=it notransfer=yes /etc/asterisk# cat sip.conf [general] context=invalid bindport=5060 bindaddr=xxx.xx.xx.xxx srvlookup=no disallow=all allow=alaw progressinband=no canreinvite=no language=it [authentication] [some-ip] type=friend context=ip host=some-ip /etc/asterisk# cat zapata.conf [channels] language=it context=default switchtype=euroisdn pridialplan=national prilocaldialplan=national signalling=pri_cpe immediate=no callerid=asreceived usecallingpres=yes echocancel=yes echocancelwhenbridged=no ;echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 group = 1 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 /etc/asterisk# cat /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 span=3,1,0,ccs,hdb3 bchan=63-77 dchan=78 bchan=79-93 span=4,1,0,ccs,hdb3 bchan=94-108 dchan=109 bchan=110-124 loadzone=it defaultzone=it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] lost packets when bridging zap and iax
We have a machine with a TE410P in it acting as a client to route calls via iax2 to our central server, caller -- ( zap - iax ) --- ( iax - whatever ) -- called client server often the called can't hear the caller (both machines on public ip) 'iax2 show netstats on client machine shows more and more dropped packets on the local side if we use sip as the entering point for the calls all works well : caller -- ( sip - iax ) --- ( iax - whatever ) -- called client server seems something in the bridging between zap and iax screws up, but I don't know if it's a bug or a misconfiguration, my conf files follows, someone has similar experiences to share ? /etc/asterisk# cat iax.conf [general] bindport=4569 bindaddr=xxx.xx.xx.xxx disallow=all allow=alaw jitterbuffer=yes forcejitterbuffer=no tos=lowdelay autokill=yes language=it notransfer=yes /etc/asterisk# cat sip.conf [general] context=invalid bindport=5060 bindaddr=xxx.xx.xx.xxx srvlookup=no disallow=all allow=alaw progressinband=no canreinvite=no language=it [authentication] [some-ip] type=friend context=ip host=some-ip /etc/asterisk# cat zapata.conf [channels] language=it context=default switchtype=euroisdn pridialplan=national prilocaldialplan=national signalling=pri_cpe immediate=no callerid=asreceived usecallingpres=yes echocancel=yes echocancelwhenbridged=no ;echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 group = 1 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 /etc/asterisk# cat /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 span=3,1,0,ccs,hdb3 bchan=63-77 dchan=78 bchan=79-93 span=4,1,0,ccs,hdb3 bchan=94-108 dchan=109 bchan=110-124 loadzone=it defaultzone=it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lost packets when bridging zap and iax
Simone Cittadini wrote: We have a machine with a TE410P in it acting as a client to route calls via iax2 to our central server, caller -- ( zap - iax ) --- ( iax - whatever ) -- called client server often the called can't hear the caller (both machines on public ip) 'iax2 show netstats on client machine shows more and more dropped packets on the local side if we use sip as the entering point for the calls all works well : caller -- ( sip - iax ) --- ( iax - whatever ) -- called client server seems something in the bridging between zap and iax screws up, but I don't know if it's a bug or a misconfiguration, my conf files follows, someone has similar experiences to share ? /etc/asterisk# cat iax.conf [general] bindport=4569 bindaddr=xxx.xx.xx.xxx disallow=all allow=alaw jitterbuffer=yes forcejitterbuffer=no tos=lowdelay autokill=yes language=it notransfer=yes /etc/asterisk# cat sip.conf [general] context=invalid bindport=5060 bindaddr=xxx.xx.xx.xxx srvlookup=no disallow=all allow=alaw progressinband=no canreinvite=no language=it [authentication] [some-ip] type=friend context=ip host=some-ip /etc/asterisk# cat zapata.conf [channels] language=it context=default switchtype=euroisdn pridialplan=national prilocaldialplan=national signalling=pri_cpe immediate=no callerid=asreceived usecallingpres=yes echocancel=yes echocancelwhenbridged=no ;echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 group = 1 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 /etc/asterisk# cat /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 span=3,1,0,ccs,hdb3 bchan=63-77 dchan=78 bchan=79-93 span=4,1,0,ccs,hdb3 bchan=94-108 dchan=109 bchan=110-124 loadzone=it defaultzone=it Only one of the above four span entries should have a 1 as the second digit. That second digit is telling the digium card which span to sync its on-board clock to. Pick the span that goes to a central office and specify it as 1 and all other spans should be either 0 or increasing numerical digits (eg, 2,3,4). If none of the spans go to a central office, its still a problem. You'll have to reload the drivers for the change to take effect. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
