[asterisk-users] Strange dropped calls
Hi all, I have a user who is reporting dropped calls at his site. We don't have any other users complaining of this. So far, this is what we know: 1. The manager bought all new Polycom phones. (POE) 2. They replaced the network switch with a POE version. 3. It's not just one or two of the phones that have problems. 4. It doesn't matter if they use the headset or the cordless set. 5. The ISP reports a very clean circuit. (Ethernet from the CLEC.) 6. We don't see their phones become unavailable very often. 7. They are the only site that seems to be having trouble. So, where else can/should I look? Mike. -- _ -- 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] Strange dropped calls
I would suggest starting with a packet capture of the SIP messages that will include both call legs (i.e. capture at the Asterisk box). This should tell you who initiated the hangup - the carrier side, the phone side, or Asterisk. On Wed, Mar 26, 2014 at 11:46 AM, Mike Diehl mdiehlena...@gmail.com wrote: Hi all, I have a user who is reporting dropped calls at his site. We don't have any other users complaining of this. So far, this is what we know: 1. The manager bought all new Polycom phones. (POE) 2. They replaced the network switch with a POE version. 3. It's not just one or two of the phones that have problems. 4. It doesn't matter if they use the headset or the cordless set. 5. The ISP reports a very clean circuit. (Ethernet from the CLEC.) 6. We don't see their phones become unavailable very often. 7. They are the only site that seems to be having trouble. So, where else can/should I look? Mike. -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org -- _ -- 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
[Asterisk-Users] Strange dropped calls
Hello List, has anybody seen (and perhaps decoded) this messages in Asterisk debug: ... Jun 30 17:46:23 DEBUG[11033]: Didn't get a frame from channel: SIP/192.168.1.30-ed86 Jun 30 17:46:23 DEBUG[11033]: Bridge stops bridging channels SIP/5602-c545 and SIP/192.168.1.30-ed86 ... This happens always, when a call get's dropped and today we had more than 10 calls which ended this way. The Problem was only with one phone, the others are ok. My first thought, some kind of network error?! Hope, someone can help Regards, Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
