Re: [asterisk-users] problem with one way audio
Jason Backshall wrote: Do you have CallProgress=yes in your zapata.conf? This one just bit me in the arse this morning. I set it to no and one-way audio went away. Have heard of issues similar to this - and whilst disabling callprogress may make that symptom disappear, it probably shouldn't be seen as a 'solution', as callprogress has it's place (disconnection detection, etc). Don, have any changed been made to your zapata.conf immediately before this issue started occuring? Jason. I thought that callprogress was highly experiemental according to the wiki. Not sure how recent that information is though. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with one way audio
Do you have CallProgress=yes in your zapata.conf? This one just bit me in the arse this morning. I set it to no and one-way audio went away. Have heard of issues similar to this - and whilst disabling callprogress may make that symptom disappear, it probably shouldn't be seen as a 'solution', as callprogress has it's place (disconnection detection, etc). Don, have any changed been made to your zapata.conf immediately before this issue started occuring? Jason. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with one way audio
One way audio on local network is ALWAY because of RTP ports forwarding problem. Your customer's router's firewall is blocking voice data to pass through from the server to the phones. Voice from phones to the server is not blocked that is why the other party can hear her fine. You can also verify it by typing 'rtp debug' on the asterisk CLI when a call is progress. You'll see that RTP is sending packets but not receiving them. All you need is to open RTP ports 1-2 on the router for Polycom phones IP addresses range. The TDM card, zapata or zaptel has nothing to do in this scenario. If you need help with port forwarding, let me know. Zeeshan A Zakaria On 6/24/07, Jason Backshall [EMAIL PROTECTED] wrote: Do you have CallProgress=yes in your zapata.conf? This one just bit me in the arse this morning. I set it to no and one-way audio went away. Have heard of issues similar to this - and whilst disabling callprogress may make that symptom disappear, it probably shouldn't be seen as a 'solution', as callprogress has it's place (disconnection detection, etc). Don, have any changed been made to your zapata.conf immediately before this issue started occuring? Jason. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with one way audio
My previous mail assumes that sip.conf is configured properly, with the following information in the [general] section: nat = yes externip = (your external IP address) localnet = (your local network and subnet addresses) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with one way audio
I have a company with asterisk 1.2.19 and polycom 501 phones. I get one way audio. A caller from the pstn world hits the tdm400 card, This rings two phones in a ring group. My client answers the phone, the calling party is told the customer here her but she can not here them. The customer hangs up and calls back and the call goes through.. I rolled back to 1.2.14 and the problem is much better but is still there, Are there any ideas Don Briggs 573-614-5667 ext 4037 ___ --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] problem with one way audio
Don Briggs wrote: I have a company with asterisk 1.2.19 and polycom 501 phones. I get one way audio. A caller from the pstn world hits the tdm400 card, This rings two phones in a ring group. My client answers the phone, the calling party is told the customer here her but she can not here them. The customer hangs up and calls back and the call goes through.. I rolled back to 1.2.14 and the problem is much better but is still there, Are there any ideas Don Briggs 573-614-5667 ext 4037 Do you have CallProgress=yes in your zapata.conf? This one just bit me in the arse this morning. I set it to no and one-way audio went away. -- Warm Regards, Lee ___ --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
