Re: [asterisk-users] Asterisk 1.4 and g723
What G723 codec do you have on Asterisk? What is your SIP.CONF? What ATA/Phone is being used and what are the exact settings, especially for the codec? On 1/19/07, Phil French [EMAIL PROTECTED] wrote: I am setting up Asterisk for use in a low bandwidth environment. As bandwidth is precious and our ATA's support it, the decision was made to use the g723 codec. I have been working on this for a few days and have not been successful. The issue that I am having is garbled noise at the client on calls whose RTP streams are terminated by Asterisk system. This is the case for all the dialplan applications I have tested except for Echo. The critical application for us is Voicemail. When a call to voicemail extension is initiated the Asterisk console does not indicate any error. Packet captures indicate the call is active and streaming g723 data. Everything seems well but is not. The audio at the client is unrecognizable. One thing that I have noticed is that the bitrates in the upstream and downstream direction differ. From Asterisk to ATA the g723 bitrate switches between 5.3 kb/s and 6.3 kb/s. From ATA to Asterisk the bitrate is a constant 6.3 kb/s. I don't think this is a problem but seems odd. As a comparison I captured packets from a call to the echo application and found that the bitrate was 6.3 kb/s in both upstream and downstream packets. Additionally, all prompts are g723 format. Voicemail is saved as g723sf. As a parrallel task a co-worker has gotten 1.2 to work with g723. However we require 1.4 for t.38 pass-through. The end-to-end system is illustrated below. [Asterisk] / \ ip ip / \ [PSTN]--pri--[GATEWAY]--ip--[ATA]--2pr--[Phone] System details -Asterisk server version 1.4 - compiled from source - Fedora Core 6 -Gateway - Cisco 2811 -ATA - Linksys 2102 I would appreciate any advice or suggestions. It should be noted that the calls to the PSTN through the gateway and calls between ATA's are working fine. Regards, Phil French Phil French Systems Engineer --- CapRock Communications 4400 S. Sam Houston Parkway E. Houston, Texas 77048 Office: 832 668 2643 [EMAIL PROTECTED] www.caprock.com NOTICE OF CONFIDENTIALITY: This e-mail message may contain confidential information and is intended only for the person(s) named above. Any review, use, disclosure or distribution by any other person is prohibited. If you are not the intended recipient, please contact the sender by e-mail and destroy all copies of this message. ___ --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
RE: [asterisk-users] Asterisk 1.4 and g723
is prohibited. If you are not the intended recipient, please contact the sender by e-mail and destroy all copies of this message. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: Saturday, January 20, 2007 2:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 and g723 What G723 codec do you have on Asterisk? What is your SIP.CONF? What ATA/Phone is being used and what are the exact settings, especially for the codec? On 1/19/07, Phil French [EMAIL PROTECTED] wrote: I am setting up Asterisk for use in a low bandwidth environment. As bandwidth is precious and our ATA's support it, the decision was made to use the g723 codec. I have been working on this for a few days and have not been successful. The issue that I am having is garbled noise at the client on calls whose RTP streams are terminated by Asterisk system. This is the case for all the dialplan applications I have tested except for Echo. The critical application for us is Voicemail. When a call to voicemail extension is initiated the Asterisk console does not indicate any error. Packet captures indicate the call is active and streaming g723 data. Everything seems well but is not. The audio at the client is unrecognizable. One thing that I have noticed is that the bitrates in the upstream and downstream direction differ. From Asterisk to ATA the g723 bitrate switches between 5.3 kb/s and 6.3 kb/s. From ATA to Asterisk the bitrate is a constant 6.3 kb/s. I don't think this is a problem but seems odd. As a comparison I captured packets from a call to the echo application and found that the bitrate was 6.3 kb/s in both upstream and downstream packets. Additionally, all prompts are g723 format. Voicemail is saved as g723sf. As a parrallel task a co-worker has gotten 1.2 to work with g723. However we require 1.4 for t.38 pass-through. The end-to-end system is illustrated below. [Asterisk] / \ ip ip / \ [PSTN]--pri--[GATEWAY]--ip--[ATA]--2pr--[Phone] System details -Asterisk server version 1.4 - compiled from source - Fedora Core 6 -Gateway - Cisco 2811 -ATA - Linksys 2102 I would appreciate any advice or suggestions. It should be noted that the calls to the PSTN through the gateway and calls between ATA's are working fine. Regards, Phil French ___ --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 ___ --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] Asterisk 1.4 and g723
I am setting up Asterisk for use in a low bandwidth environment. As bandwidth is precious and our ATA's support it, the decision was made to use the g723 codec. I have been working on this for a few days and have not been successful. The issue that I am having is garbled noise at the client on calls whose RTP streams are terminated by Asterisk system. This is the case for all the dialplan applications I have tested except for Echo. The critical application for us is Voicemail. When a call to voicemail extension is initiated the Asterisk console does not indicate any error. Packet captures indicate the call is active and streaming g723 data. Everything seems well but is not. The audio at the client is unrecognizable. One thing that I have noticed is that the bitrates in the upstream and downstream direction differ. From Asterisk to ATA the g723 bitrate switches between 5.3 kb/s and 6.3 kb/s. From ATA to Asterisk the bitrate is a constant 6.3 kb/s. I don't think this is a problem but seems odd. As a comparison I captured packets from a call to the echo application and found that the bitrate was 6.3 kb/s in both upstream and downstream packets. Additionally, all prompts are g723 format. Voicemail is saved as g723sf. As a parrallel task a co-worker has gotten 1.2 to work with g723. However we require 1.4 for t.38 pass-through. The end-to-end system is illustrated below. [Asterisk] / \ ip ip / \ [PSTN]--pri--[GATEWAY]--ip--[ATA]--2pr--[Phone] System details -Asterisk server version 1.4 - compiled from source - Fedora Core 6 -Gateway - Cisco 2811 -ATA - Linksys 2102 I would appreciate any advice or suggestions. It should be noted that the calls to the PSTN through the gateway and calls between ATA's are working fine. Regards, Phil French Phil French Systems Engineer --- CapRock Communications 4400 S. Sam Houston Parkway E. Houston, Texas 77048 Office: 832 668 2643 [EMAIL PROTECTED] www.caprock.com NOTICE OF CONFIDENTIALITY: This e-mail message may contain confidential information and is intended only for the person(s) named above. Any review, use, disclosure or distribution by any other person is prohibited. If you are not the intended recipient, please contact the sender by e-mail and destroy all copies of this message. ___ --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
