Re: [asterisk-users] Suddenly the voice became garbage (likerobot)using Asterisk 1.4.19.2
We had the problem between 2 extensions on same PBX. Both ends using Aastra phones. Not sure where to point the finger... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina Sent: Monday, June 01, 2009 2:15 AM To: Asterisk Users List Subject: Re: [asterisk-users] Suddenly the voice became garbage (likerobot)using Asterisk 1.4.19.2 Michelle Dupuis escribió: You're not alone...we never found the cause of this (rare) occurance... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Sunday, May 31, 2009 8:58 PM To: Asterisk Users List Subject: [asterisk-users] Suddenly the voice became garbage (like robot)using Asterisk 1.4.19.2 Hi All; I was using since one year Asterisk 1.4.19.2 and zaptel 1.4.10.1 and they were working fine via SIP, IAX and Digium fxo and fxs ports. Suddenly just before 2 or 3 days, the voice become garbage like robot when I place a call from the SIP Phone (which is in a country and the Asterisk box in another country). I am surprise what is the reason that let rtp become like this ! The sound now like robot (human machine:)-- What could be? Actually before about 10 days, we added one Polycom SIP Phone and added qualify=yes for the SIP entities, also I fixed the externip to be the public IP address of the machine, and I fixed the localnet (I was not using externip and localnet, because I was not need them for NAT issues, but I used it when I start have NATed Polycom devices). But even, the voice was fine after these changes, so I do not think it has any relation. Could it be related the Asterisk 1.4.19.2 and how it manipulated the rtp, packets, so maybe a small changes in the Internet provider effected and let things becoming bad??!! It sounds like an ISP problem. Is your bandwidth ok? Do you share the connection with another services? In my case, usually robot calls are related to upload congestion, and implementing egress QoS policies on the router helped a lot to maintain the good quality of the calls. This may be your case. If your asterisk worked fine for one entire year flawlessly, you surely can't blame it for bad call quality. Cheers, I used gsm and g729 codecs, and both are bad. g729 give little bit better quality than gsm (and both are bad now .. garbage). The voice becoming bad even if we do a call using IAX Trunk or from SIP Phone (Polycom). Any advise what can I do? Regards Bilal ___ -- 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 ___ -- 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 -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- 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 ___ -- 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] Suddenly the voice became garbage(likerobot)using Asterisk 1.4.19.2
Since this is internal SIP, I'd probably vote for a memory leak, bandwidth problem or hardware hiccup. I've had a similar situation when a grep caused pounding of a bad disk sector. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Monday, June 01, 2009 8:27 AM To: 'Asterisk Users List' Subject: Re: [asterisk-users] Suddenly the voice became garbage(likerobot)using Asterisk 1.4.19.2 We had the problem between 2 extensions on same PBX. Both ends using Aastra phones. Not sure where to point the finger... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina Sent: Monday, June 01, 2009 2:15 AM To: Asterisk Users List Subject: Re: [asterisk-users] Suddenly the voice became garbage (likerobot)using Asterisk 1.4.19.2 Michelle Dupuis escribió: You're not alone...we never found the cause of this (rare) occurance... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Sunday, May 31, 2009 8:58 PM To: Asterisk Users List Subject: [asterisk-users] Suddenly the voice became garbage (like robot)using Asterisk 1.4.19.2 Hi All; I was using since one year Asterisk 1.4.19.2 and zaptel 1.4.10.1 and they were working fine via SIP, IAX and Digium fxo and fxs ports. Suddenly just before 2 or 3 days, the voice become garbage like robot when I place a call from the SIP Phone (which is in a country and the Asterisk box in another country). I am surprise what is the reason that let rtp become like this ! The sound now like robot (human machine:)-- What could be? Actually before about 10 days, we added one Polycom SIP Phone and added qualify=yes for the SIP entities, also I fixed the externip to be the public IP address of the machine, and I fixed the localnet (I was not using externip and localnet, because I was not need them for NAT issues, but I used it when I start have NATed Polycom devices). But even, the voice was fine after these changes, so I do not think it has any relation. Could it be related the Asterisk 1.4.19.2 and how it manipulated the rtp, packets, so maybe a small changes in the Internet provider effected and let things becoming bad??!! It sounds like an ISP problem. Is your bandwidth ok? Do you share the connection with another services? In my case, usually robot calls are related to upload congestion, and implementing egress QoS policies on the router helped a lot to maintain the good quality of the calls. This may be your case. If your asterisk worked fine for one entire year flawlessly, you surely can't blame it for bad call quality. Cheers, I used gsm and g729 codecs, and both are bad. g729 give little bit better quality than gsm (and both are bad now .. garbage). The voice becoming bad even if we do a call using IAX Trunk or from SIP Phone (Polycom). Any advise what can I do? Regards Bilal ___ -- 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 ___ -- 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 -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- 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 ___ -- 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 ___ -- 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] Suddenly the voice became garbage(likerobot)using Asterisk 1.4.19.2
Danny, Just out of curiosity, can you elaborate? Anything in use for asterisk should be in cache by the time it's needed for a SIP stream. And nothing related to a SIP stream should ever be read directly from the disk... Unless I'm mistaken. Thanks Dave snip Since this is internal SIP, I'd probably vote for a memory leak, bandwidth problem or hardware hiccup. I've had a similar situation when a grep caused pounding of a bad disk sector. /snip ___ -- 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
