I've written about this issue several times, but have not yet found any solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones are primarily Snom 300's but I also have a couple of headset phones connected to Grandstream HT286 SIP adapters. I have 8 offices, each has it's own asterisk server all running the same versions of asterisk and Zaptel. Only difference is that one office uses a Digium TDM 8-port card and the other branches use 4-port Rhino cards with only 2 ports in use. What happens is that periodically we will be in a call and the call will just drop. It's usually within the first couple of minutes of the call. The calls can be either incoming or outgoing. The phenomenon affects both the Snoms and the Grandstreams. Along with the dropped call issue, we periodically have a problem where a person we call or a person that calls in cannot hear the person in the our office, but the person in our office can hear the remote person fine. All of the phones are on the same physical network as the asterisk server. There is no NAT, no Firewall, VLAN, etc. between the phones and the server. I have tried running sip debugs on the calls, but on the off chance that my logs catch either a drop or a one-way audio, the sip debug looks like just a normal call. Is there any setting that might cause both one-way audio and dropped calls? Thanks, Brent Davidson
A few thoughts; 1. I assume that the * servers aren't on dedicated networks; Do the dropped or one-way calls occur during high-traffic times or are they concurrent with large downloads? In my shop, we had to get a router that would prioritize voice traffic or we would be dead in the water during client file transmissions. 2. Don't know about the SNOM or GS phones, but my Polycom phones let you establish higher packet priorities for voice traffic as well. 3. Have you been able to do a "top" during one of these failures? Could be a memory leak that comes up randomly. 4. Looking at the startup logs, are the cards having to retry several times to get an IRQ? Digium cards IME can conflict with the Hard Drive (SCSI) controller, causing problems during heavy I/O periods. Hope this helps. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Brent Davidson Sent: Tuesday, March 30, 2010 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped Calls I've written about this issue several times, but have not yet found any solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones are primarily Snom 300's but I also have a couple of headset phones connected to Grandstream HT286 SIP adapters. I have 8 offices, each has it's own asterisk server all running the same versions of asterisk and Zaptel. Only difference is that one office uses a Digium TDM 8-port card and the other branches use 4-port Rhino cards with only 2 ports in use. What happens is that periodically we will be in a call and the call will just drop. It's usually within the first couple of minutes of the call. The calls can be either incoming or outgoing. The phenomenon affects both the Snoms and the Grandstreams. Along with the dropped call issue, we periodically have a problem where a person we call or a person that calls in cannot hear the person in the our office, but the person in our office can hear the remote person fine. All of the phones are on the same physical network as the asterisk server. There is no NAT, no Firewall, VLAN, etc. between the phones and the server. I have tried running sip debugs on the calls, but on the off chance that my logs catch either a drop or a one-way audio, the sip debug looks like just a normal call. Is there any setting that might cause both one-way audio and dropped calls? Thanks, Brent Davidson -- _____________________________________________________________________ -- 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
> I've written about this issue several times, but have not yet found any > solution to it. ?I am using asterisk 1.4.21.2 and zaptel 1.4.12. ?Phones > are primarily Snom 300's but I also have a couple of headset phones > connected to Grandstream HT286 SIP adapters. ?I have 8 offices, each has > it's own asterisk server all running the same versions of asterisk and > Zaptel. ?Only difference is that one office uses a Digium TDM 8-port > card and the other branches use 4-port Rhino cards with only 2 ports in > use. ?What happens is that periodically we will be in a call and the > call will just drop. ?It's usually within the first couple of minutes of > the call. ?The calls can be either incoming or outgoing. ?The phenomenon > affects both the Snoms and the Grandstreams. ?Along with the dropped > call issue, we periodically have a problem where a person we call or a > person that calls in cannot hear the person in the our office, but the > person in our office can hear the remote person fine. > > All of the phones are on the same physical network as the asterisk > server. ?There is no NAT, no Firewall, VLAN, etc. between the phones and > the server. ? I have tried running sip debugs on the calls, but on the > off chance that my logs catch either a drop or a one-way audio, the sip > debug looks like just a normal call. > > Is there any setting that might cause both one-way audio and dropped calls? > > Thanks, > Brent DavidsonJoin the club. I've experienced the same with various strains on 1.4.x above 1.4.21.1 (not an issue with this one that I have seen). This issue is truly random and debugging reveals nothing. I run an all SIP environment with same results. My solution was to downgrade to another version or switch to 1.2 or 1.6 depending on what features I need for the system. Sorry I couldn't be of any help, but I feel your frustration. JR -- JR Richardson Engineering for the Masses