[asterisk-users] out of the blue one way audio
Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet link of 1Mbps Dedicated Leased Line. 3- Cisco Router 4- Hosted Asterisk server (Asterisk 1.4.40.1 x64 bit 8 GB ram, Intel(R) Xeon(R) X3210 @ 2.13GHz CPU) 5- additional SIP Soft phones in several locations over the world (Zoiper, X-Lite, Nokia Native Sip). 6- Packet8 Sip trunking for Inbound calls 7- IDT (Net2Phone) Sip Trunk for outbound calls. (two IPs) Network Profile: Cisco Router has a Public IP of 196.XXX.XXX.XXX and a private IP 192.168.100.245 computers have IP addresses : 192.168.100.XXX/24 default gateway: 192.168.100.245 DC: 192.168.100.2 DNS: 192.168.100.2 PROXY Server: 192.168.100.2 (Forced in Internet Explorer) Voip Traffic going directly from 192.168.100.245 Http Traffic goes to 192.168.100.2 then via another internet link (ADSL 8bps connection) Router is preventing any traffic other than VoIP. for example we tried to pass HTTP requests via the internet link .. but did not go through. Asterisk Side: sip.conf sample: [GENERAL] notifyringing=yes notifyhold=yes limitonpeers=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes t38pt_udptl = yes bindport=5070 externip=SERVER_IP rtptimeout=60 session-timers=originate session-expires=600 session-minse=90 session-refresher=uas rtpholdtimeout=120 rtpkeepalive=20 allow=gsm t38pt_udptl=yes sendrpid=yes trustrpid=no directrtpsetup=yes [USERNAME] deny=0.0.0.0/0.0.0.0 type=friend secret=PASSWORD qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes host=dynamic dtmfmode=rfc2833 disallow=all allow=gsm context=from-callcenter canreinvite=no we have a call recording for outbound and inbound calls. the problem is not happening on all calls at once.. it happens on random extensions at random times and random durations however most noticeable durations are around 7 minutes and 20 minutes (most occurring) one additional situation.. the original bind_port for asterisk server is 5060 however after three or four hours of operating on that port the computers unregister and are unable to make calls at all .. or even register we changed the port to 5070 and things are working properly now. although this port issue is only noticeable on the above setup and on that facility only. other internet links are able to provide stable connection over 5060. any additional information can be provided. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] out of the blue one way audio
Check if this problem happening with xlite useres only i remember there is option in xlite causing this problem On May 2, 2011 2:36 PM, Tarek Sawah tareksa...@hotmail.com wrote: -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] out of the blue one way audio
this is happening on all Soft phones are facing the same problem. Zoiper , X=lite , our own pjsip based dialer (CRM). this was not the issue .. it happened suddenly .. we switched internet links even. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Mon, 2 May 2011 14:45:58 +0300 From: hatemm...@gmail.com To: asterisk-users@lists.digium.com CC: yamennaj...@ids-tech.net Subject: Re: [asterisk-users] out of the blue one way audio Check if this problem happening with xlite useres only i remember there is option in xlite causing this problem On May 2, 2011 2:36 PM, Tarek Sawah wrote: -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] out of the blue one way audio
because they are behind a router and using private IP addresses. and the Cisco router is Nating our traffic Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 2 May 2011 08:11:23 -0400 Subject: Re: [asterisk-users] out of the blue one way audio Why nat=yes ? -- Sent from my iPhone On May 2, 2011, at 7:33 AM, Tarek Sawah wrote: Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet link of 1Mbps Dedicated Leased Line. 3- Cisco Router 4- Hosted Asterisk server (Asterisk 1.4.40.1 x64 bit 8 GB ram, Intel (R) Xeon(R) X3210 @ 2.13GHz CPU) 5- additional SIP Soft phones in several locations over the world (Zoiper, X-Lite, Nokia Native Sip). 6- Packet8 Sip trunking for Inbound calls 7- IDT (Net2Phone) Sip Trunk for outbound calls. (two IPs) Network Profile: Cisco Router has a Public IP of 196.XXX.XXX.XXX and a private IP 192.168.100.245 computers have IP addresses : 192.168.100.XXX/24 default gateway: 192.168.100.245 DC: 192.168.100.2 DNS: 192.168.100.2 PROXY Server: 192.168.100.2 (Forced in Internet Explorer) Voip Traffic going directly from 192.168.100.245 Http Traffic goes to 192.168.100.2 then via another internet link (ADSL 8bps connection) Router is preventing any traffic other than VoIP. for example we tried to pass HTTP requests via the internet link .. but did not go through. Asterisk Side: sip.conf sample: [GENERAL] notifyringing=yes notifyhold=yes limitonpeers=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes t38pt_udptl = yes bindport=5070 externip=SERVER_IP rtptimeout=60 session-timers=originate session-expires=600 session-minse=90 session-refresher=uas rtpholdtimeout=120 rtpkeepalive=20 allow=gsm t38pt_udptl=yes sendrpid=yes trustrpid=no directrtpsetup=yes [USERNAME] deny=0.0.0.0/0.0.0.0 type=friend secret=PASSWORD qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes host=dynamic dtmfmode=rfc2833 disallow=all allow=gsm context=from-callcenter canreinvite=no we have a call recording for outbound and inbound calls. the problem is not happening on all calls at once.. it happens on random extensions at random times and random durations however most noticeable durations are around 7 minutes and 20 minutes (most occurring) one additional situation.. the original bind_port for asterisk server is 5060 however after three or four hours of operating on that port the computers unregister and are unable to make calls at all .. or even register we changed the port to 5070 and things are working properly now. although this port issue is only noticeable on the above setup and on that facility only. other internet links are able to provide stable connection over 5060. any additional information can be provided. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
