Re: [asterisk-users] Dropped calls when all DAHDI lines in use
Andrew Martin wrote: - Original Message - From: "John Novack SCII_U" To: "Asterisk Users Mailing List, Non-Commercial Discussion" , "Andrew Martin" Sent: Monday, October 8, 2018 4:29:41 PM Subject: Re: [asterisk-users] Dropped calls when all DAHDI lines in use Have you given any thought to moving to at least a current supported version 13? Asterisk 11 has been EOL for some time now I doubt you will get a resolution to a version no longer supported. Moving to the latest version 13 should be relatively quick and painless, and if the issue persists you might find more assistance. John Novack Andrew Martin wrote: Hello, I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x analog POTS lines coming into my Asterisk server from the phone company. Internally, I have about 180 SIP clients defined in sip.conf. What appears to be happening is that if existing calls are consuming all 8 external lines and a new SIP client attempts to make a call, an existing call gets dropped. The asterisk log simply shows this as a normal hangup, so I am not able to easily distinguish between a normal hangup and this type of dropped call. In testing, I am able to get a new SIP client to report "service unavailable" when all 8 lines are consumed, yet still drops are reported. I have been unable to find any configuration settings pertaining to prioritizing existing calls over new calls. What else can I look for to attempt to debug and fix this so that existing calls are not dropped? Thanks, Andrew -- Dog is my Co-Pilot John, Thanks for the reply. Yes, I am planning on moving to version 13 but need to find a solution in the interim. If there are any configuration options that pertain to which actions to take with existing calls when new calls come in, I think it is likely that they would be shared between both versions (and I want to make sure I have the correct settings when I switch to version 13 too). Can you advise on any tunables related to handling existing vs new calls? Thanks, Andrew I really can't help with your existing issue(s) I suggest you make the switch to the latest version 13, which should go fairly smoothly, and you may find that you no longer have an issue. JN -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped calls when all DAHDI lines in use
You could use GROUP & GROUP_COUNT to track how many channels you are using before you attempt to dial out and send back a Busy/Congestion/Whatever to your endpoint when you are at your limit. On Mon, Oct 8, 2018 at 4:33 PM Andrew Martin wrote: > Hello, > > I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x > analog > POTS lines coming into my Asterisk server from the phone company. > Internally, I > have about 180 SIP clients defined in sip.conf. What appears to be > happening is > that if existing calls are consuming all 8 external lines and a new SIP > client > attempts to make a call, an existing call gets dropped. The asterisk log > simply > shows this as a normal hangup, so I am not able to easily distinguish > between a > normal hangup and this type of dropped call. In testing, I am able to get > a new > SIP client to report "service unavailable" when all 8 lines are consumed, > yet > still drops are reported. > > I have been unable to find any configuration settings pertaining to > prioritizing > existing calls over new calls. What else can I look for to attempt to > debug and > fix this so that existing calls are not dropped? > > Thanks, > > Andrew > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped calls when all DAHDI lines in use
- Original Message - > From: "John Novack SCII_U" > To: "Asterisk Users Mailing List, Non-Commercial Discussion" > , "Andrew Martin" > > Sent: Monday, October 8, 2018 4:29:41 PM > Subject: Re: [asterisk-users] Dropped calls when all DAHDI lines in use > Have you given any thought to moving to at least a current supported version > 13? > Asterisk 11 has been EOL for some time now > I doubt you will get a resolution to a version no longer supported. > Moving to the latest version 13 should be relatively quick and painless, and > if > the issue persists you might find more assistance. > > John Novack > > > Andrew Martin wrote: >> Hello, >> >> I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x >> analog >> POTS lines coming into my Asterisk server from the phone company. >> Internally, I >> have about 180 SIP clients defined in sip.conf. What appears to be happening >> is >> that if existing calls are consuming all 8 external lines and a new SIP >> client >> attempts to make a call, an existing call gets dropped. The asterisk log >> simply >> shows this as a normal hangup, so I am not able to easily distinguish >> between a >> normal hangup and this type of dropped call. In testing, I am able to get a >> new >> SIP client to report "service unavailable" when all 8 lines are consumed, yet >> still drops are reported. >> >> I have been unable to find any configuration settings pertaining to >> prioritizing >> existing calls over new calls. What else can I look for to attempt to debug >> and >> fix this so that existing calls are not dropped? >> >> Thanks, >> >> Andrew >> > > -- > Dog is my Co-Pilot John, Thanks for the reply. Yes, I am planning on moving to version 13 but need to find a solution in the interim. If there are any configuration options that pertain to which actions to take with existing calls when new calls come in, I think it is likely that they would be shared between both versions (and I want to make sure I have the correct settings when I switch to version 13 too). Can you advise on any tunables related to handling existing vs new calls? Thanks, Andrew -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped calls when all DAHDI lines in use
Have you given any thought to moving to at least a current supported version 13? Asterisk 11 has been EOL for some time now I doubt you will get a resolution to a version no longer supported. Moving to the latest version 13 should be relatively quick and painless, and if the issue persists you might find more assistance. John Novack Andrew Martin wrote: Hello, I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x analog POTS lines coming into my Asterisk server from the phone company. Internally, I have about 180 SIP clients defined in sip.conf. What appears to be happening is that if existing calls are consuming all 8 external lines and a new SIP client attempts to make a call, an existing call gets dropped. The asterisk log simply shows this as a normal hangup, so I am not able to easily distinguish between a normal hangup and this type of dropped call. In testing, I am able to get a new SIP client to report "service unavailable" when all 8 lines are consumed, yet still drops are reported. I have been unable to find any configuration settings pertaining to prioritizing existing calls over new calls. What else can I look for to attempt to debug and fix this so that existing calls are not dropped? Thanks, Andrew -- Dog is my Co-Pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Calls
hi: how about the codecs? Best wishes! Asterisk Support group(sangoma, digium...), providing asterisk conf, pri, ss7, elastix, trixbox support. website:www.cnasterisk.com, www.voip88.com Date: Wed, 31 Mar 2010 17:20:30 -0500 From: br...@texascountrytitle.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dropped Calls On 3/31/2010 12:16 PM, Philipp von Klitzing wrote: Maybe rtptimeout in sip.conf is involved (and not behaving as expected)? I was suspecting something with either rtptimeout or sip registration timeout, but I'm not sure what. -- _ -- 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 _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969-- _ -- 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] Dropped Calls
On 3/31/2010 12:16 PM, Philipp von Klitzing wrote: Maybe rtptimeout in sip.conf is involved (and not behaving as expected)? I was suspecting something with either rtptimeout or sip registration timeout, but I'm not sure what. Hi, I have had similar issue. I have downgraded from 1.6 to 1.4 and issue got solved. Never managed to find what is going on. It was happening only if all were true: - linksys phone or pap - asterisk 1.6 - use certain VOIP provider. Solution: moved to 1.4 I hope thsi helps. Peter -- _ -- 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] Dropped Calls
On 4/7/2010 2:45 AM, asterisk card support wrote: hi: how about the codecs? Best wishes! Asterisk Support group(sangoma, digium...), providing asterisk conf, pri, ss7, elastix, trixbox support. website:www.cnasterisk.com, www.voip88.com I have the phones and asterisk limited to ulaw only. -- _ -- 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] Dropped Calls
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of JR Richardson Sent: Tuesday, March 30, 2010 6:55 PM To: asterisk-users@lists.digium.com Subject: Re: [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 Join 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 Is there a chance that you are using Realtime at all? I am just curious because I was having problems with dropped calls as well and just discovered that it appears to be related to the database server. If for some reason on the database server there is a table lock (which I am investigating why) asterisk drops any PRI calls and SIP calls. Everything looked normal and the error messages never once suggest a problem with the database server or Realtime. I was looking everywhere else but at the Realtime until I stumbled across it. While doing some backups with FLUSH READ LOCKS to a slave machine, which I changed asterisk to use a few months back, I had dropped calls occur. I later confirmed that asterisk seems to hang / freeze during that period but once the database server releases the locks, asterisk continues to function without any problems. This started to occur when we had an increase in call volume and an increase in load on the db server. I was using Realtime for extensions, sip peers and CDR. I had turned off using realtime for CDR (which we don't really use anyway) and started to use a slave server instead of the master when performing some maintenance on the master db server. I left it that way since I was just using it for extensions and sip peers and that had cleared it up over the last few months until I ran my backup. Not sure that helps but it is worth a shot in mentioning to you. Regards, Michael Young (elguero) -- _ -- 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] Dropped Calls
On 3/31/2010 10:38 AM, Michael L. Young wrote: Is there a chance that you are using Realtime at all? I am just curious because I was having problems with dropped calls as well and just discovered that it appears to be related to the database server. If for some reason on the database server there is a table lock (which I am investigating why) asterisk drops any PRI calls and SIP calls. Everything looked normal and the error messages never once suggest a problem with the database server or Realtime. I was looking everywhere else but at the Realtime until I stumbled across it. While doing some backups with FLUSH READ LOCKS to a slave machine, which I changed asterisk to use a few months back, I had dropped calls occur. I later confirmed that asterisk seems to hang / freeze during that period but once the database server releases the locks, asterisk continues to function without any problems. This started to occur when we had an increase in call volume and an increase in load on the db server. I was using Realtime for extensions, sip peers and CDR. I had turned off using realtime for CDR (which we don't really use anyway) and started to use a slave server instead of the master when performing some maintenance on the master db server. I left it that way since I was just using it for extensions and sip peers and that had cleared it up over the last few months until I ran my backup. Not sure that helps but it is worth a shot in mentioning to you. Regards, Michael Young (elguero) In my case, no. All extensions are hard-coded. We only have a handful of phones that don't change. -- _ -- 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] Dropped Calls
Just to get a 100% correct response to last question, are you using the flat CDR or mysql/some other DB? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Davidson Sent: Wednesday, March 31, 2010 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dropped Calls On 3/31/2010 10:38 AM, Michael L. Young wrote: Is there a chance that you are using Realtime at all? I am just curious because I was having problems with dropped calls as well and just discovered that it appears to be related to the database server. If for some reason on the database server there is a table lock (which I am investigating why) asterisk drops any PRI calls and SIP calls. Everything looked normal and the error messages never once suggest a problem with the database server or Realtime. I was looking everywhere else but at the Realtime until I stumbled across it. While doing some backups with FLUSH READ LOCKS to a slave machine, which I changed asterisk to use a few months back, I had dropped calls occur. I later confirmed that asterisk seems to hang / freeze during that period but once the database server releases the locks, asterisk continues to function without any problems. This started to occur when we had an increase in call volume and an increase in load on the db server. I was using Realtime for extensions, sip peers and CDR. I had turned off using realtime for CDR (which we don't really use anyway) and started to use a slave server instead of the master when performing some maintenance on the master db server. I left it that way since I was just using it for extensions and sip peers and that had cleared it up over the last few months until I ran my backup. Not sure that helps but it is worth a shot in mentioning to you. Regards, Michael Young (elguero) In my case, no. All extensions are hard-coded. We only have a handful of phones that don't change. -- _ -- 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 -- _ -- 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] Dropped Calls
Hi! I am just curious because I was having problems with dropped calls as well Maybe rtptimeout in sip.conf is involved (and not behaving as expected)? All extensions are hard-coded. We only have a handful of phones that don't change. This last sentence is a wounderful example of a sentence that can be interpreted in two, and very opposite, ways. :-) Philipp -- _ -- 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] Dropped Calls
On 3/31/2010 12:06 PM, Danny Nicholas wrote: Just to get a 100% correct response to last question, are you using the flat CDR or mysql/some other DB? All sip clients/peers are defined in sip.conf, dial-plan is entirely in extensions.ael. We have one office that uses an Asterisk native database call in the dialplan for the operator extension to see which extension is currently handling operator calls, but other than that there is no no DB used on any of the other systems. -Brent -- _ -- 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] Dropped Calls
On 3/31/2010 12:16 PM, Philipp von Klitzing wrote: Maybe rtptimeout in sip.conf is involved (and not behaving as expected)? I was suspecting something with either rtptimeout or sip registration timeout, but I'm not sure what. -- _ -- 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] Dropped Calls
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-boun...@lists.digium.com [mailto:asterisk-users-boun...@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 -- _ -- 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] Dropped Calls
On 3/30/2010 3:14 PM, Danny Nicholas wrote: 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. Asterisk servers are not on a dedicated network, but our total network utilization is less than 10% max at any time. 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. I have all the phones, the asterisk server and the core switch set to prioritize RTP and SIP packets at top priority. But I never see any indication of dropped or delayed packets in the logs. 3. Have you been able to do a top during one of these failures? Could be a memory leak that comes up randomly. This one is a tough one. When these types of calls occur it is completely random. Sometimes there will be one or two in a row, other times there won't be one for a couple of days. It would take some some serious logging to catch top data at the exact moment one of the calls drops or the one-way audio hits. 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 Cards all get an IRQ on the first try. Other data of interest: Our main office only has 8 incoming analog lines, the other offices all only have 2 incoming lines, and there is no correlation between calls in progress and and either of the problems. Sometimes the main office will have two or three in-progress calls and another incoming or outgoing call will experience one-way audio or a disconnect and the others are unaffected. Not even a glitch in the audio. I have had both problems happen to me after hours when I was the only one in the office so the network was completely idle and my call was the only one active. I've been trying to trace this problem for about two years and still have not been able to make any real progress. I guess I should just update to Dahdi and Asterisk 1.6, but I just hate to change a system that is (mostly) working. -- _ -- 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] 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 Join 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 -- _ -- 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] Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet
Maybe you have a Codec issue? On Tue, Oct 28, 2008 at 11:20 PM, Benny Amorsen [EMAIL PROTECTED][EMAIL PROTECTED] wrote: Lincoln King-Cliby [EMAIL PROTECTED] writes: Periodically I'm seeing calls placed from the 7961s through anything on the PBX that requires digit entry (the Auto Attendant, Voicemail, etc.) 'randomly' drop; extension-to-extension calls extension-to-PSTN, and PSTN-to-extension calls never have any issues whatsoever. Nor have I been able to duplicate the issues hopping around auto attendants on an inbound PSTN call. I am not sure this is relevant in the 1.4.x versions, but here goes anyway: In Asterisk 1.2.x it could sometimes happen that Asterisk believed the path to a server was so good, that it would only allow 1 ms for answers to be received. It would do all its retransmissions in less than 200ms, and then it would complain about no reply to critical packet. Anyway, you can adjust the minimum timer with the configuration option t1min in sip.conf. I would recommend setting it to at least 100 (it is in ms) and perhaps 500 would help for you. It is also highly possible that your issue is completely different. /Benny ___ -- 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] Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet
Benny and Mark, Thank you for your replies. I tried adding t1min=500 to sip.conf per the suggestion below and since doing that haven't been able to reproduce the issue. If it comes back, I'll do the SIP debug per Mark's suggestion and post the results here. (Mark, per your question the Auto Attendant and Voicemail are on the same box) Thanks again for the quick help! Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC http://www.thecontrolworks.com/ Crestron Authorized Independent Programmer -Original Message- From: Benny Amorsen [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 28, 2008 5:20 PM To: Lincoln King-Cliby Cc: 'asterisk-users@lists.digium.com' Subject: Re: Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet snip In Asterisk 1.2.x it could sometimes happen that Asterisk believed the path to a server was so good, that it would only allow 1 ms for answers to be received. It would do all its retransmissions in less than 200ms, and then it would complain about no reply to critical packet. Anyway, you can adjust the minimum timer with the configuration option t1min in sip.conf. I would recommend setting it to at least 100 (it is in ms) and perhaps 500 would help for you. It is also highly possible that your issue is completely different. /Benny ___ -- 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] Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet
Lincoln King-Cliby wrote: Hi All, I've looked through the archives and tried several variations in Google, and I haven't found anything on-point... So I'm hoping someone here may be able to help this relative Asterisk neophyte shed some light on an issue: I have a box running Asterisk 1.4.22 in our lab with several Cisco 7961G phones and an AEX804E card (4 FXO, hardware echo cancellation). The server and all phones are on the same subnet (10.2.0.x/255.255.255.0) of the local LAN with no NAT, routing, firewall, etc., etc. between the server and the phones. Periodically I'm seeing calls placed from the 7961s through anything on the PBX that requires digit entry (the Auto Attendant, Voicemail, etc.) 'randomly' drop; extension-to-extension calls extension-to-PSTN, and PSTN-to-extension calls never have any issues whatsoever. Nor have I been able to duplicate the issues hopping around auto attendants on an inbound PSTN call. When the call drops, the phone still thinks that it is connected, but the audio path is cut off and something similar to the following is dumped to the console -- SIP/1103-b71184e0 Playing 'vm-password' (language 'en') [Oct 28 14:23:07] WARNING[9423]: chan_sip.c:1958 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 28 14:23:07] WARNING[9423]: chan_sip.c:1980 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet (see doc/sip-retransmit.txt). All of the results Google has turned up, and the doc/sip-retransmit.txt file point to problems with things in the middle of the path between the server and the phone (NAT, firewall, SIP middle box, proxy) that simply don't exist in the configuration that we're using. I suspect it's an issue with the way the Cisco phones are dealing with DTMF to Asterisk or Asterisk dealing with the DTMF from Cisco but that's where I go off into unknown territory. (FWIW, until the call drops everything works fine, pressing a button triggers the desired action, and audio quality is fantastic) I've rolled the firmware on the phones up and down with no noticeable change, and I also upgraded to Asterisk 1.4.22 version of Asterisk (I had been running 1.4.21.2, and there are fewer dropped calls with .22 but it's still way too often to be acceptable) Any suggestions are greatly appreciated, but please be explicit... short of editing the configuration files and make install my Asterisk experience is rather limited. Thanks in advance, Lincoln It's hard to diagnose a problem like this without a full SIP trace, but given the problem you are describing, it looks like Asterisk is sending a SIP INVITE that is not being replied to with a 200 OK. It wasn't clear in the scenario you presented why Asterisk would be sending an INVITE out anywhere though, so I'm not sure where this is originating. Is Asterisk dialing out to another box which contains the voicemail and auto-attendant services? If so, and if the box which provides these services is another Asterisk server, be sure that the second Asterisk server has an Answer in the dialplan for these calls. By the way, to see a SIP trace inside Asterisk, you can issue the command sip set debug in the CLI. Then all SIP messages will be written anywhere where you are logging verbose messages. Mark Michelson ___ -- 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] Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet
Lincoln King-Cliby [EMAIL PROTECTED] writes: Periodically I'm seeing calls placed from the 7961s through anything on the PBX that requires digit entry (the Auto Attendant, Voicemail, etc.) 'randomly' drop; extension-to-extension calls extension-to-PSTN, and PSTN-to-extension calls never have any issues whatsoever. Nor have I been able to duplicate the issues hopping around auto attendants on an inbound PSTN call. I am not sure this is relevant in the 1.4.x versions, but here goes anyway: In Asterisk 1.2.x it could sometimes happen that Asterisk believed the path to a server was so good, that it would only allow 1 ms for answers to be received. It would do all its retransmissions in less than 200ms, and then it would complain about no reply to critical packet. Anyway, you can adjust the minimum timer with the configuration option t1min in sip.conf. I would recommend setting it to at least 100 (it is in ms) and perhaps 500 would help for you. It is also highly possible that your issue is completely different. /Benny ___ -- 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] Dropped calls
On Jan 31, 2008 6:45 AM, mccoy silva [EMAIL PROTECTED] wrote: I have a very serious problem with calls between PAP2-NA and a TDM2400 (8 FXO). Almost every call dropped after between 20 and 30 seconds with conversation. I disable the sound card, serial and other things on my server, but the problem still continues. I've changed the RPT Packet Size to .20 on PAP2-NA, but nothing. Here a piece of my log: [Jan 31 07:10:43] DEBUG[3131] channel.c: Hanging up channel 'Zap/17-1' [Jan 31 07:10:43] DEBUG[3131] chan_zap.c: zt_hangup(Zap/17-1) [Jan 31 07:10:43] DEBUG[3131] chan_zap.c: Hangup: channel: 17 index = 0, normal = 11, callwait = -1, thirdcall = -1 [Jan 31 07:10:43] DEBUG[3131] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/17-1 [Jan 31 07:10:43] DEBUG[3131] chan_zap.c: Updated conferencing on 17, with 0 conference users [Jan 31 07:10:43] VERBOSE[3131] logger.c: -- Hungup 'Zap/17-1' [Jan 31 07:10:43] DEBUG[3131] devicestate.c: Notification of state change to be queued on device/channel Zap/17-1 [Jan 31 07:10:43] DEBUG[3131] app_dial.c: Exiting with DIALSTATUS=NOANSWER. [Jan 31 07:10:43] DEBUG[2695] devicestate.c: No provider found, checking channel drivers for Zap - 17 [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] VERBOSE[3131] logger.c: == Auto fallthrough, channel 'SIP/dep2_1154-08202968' status is 'NOANSWER' [Jan 31 07:10:43] DEBUG[3131] channel.c: Soft-Hanging up channel 'SIP/dep2_1154-08202968' [Jan 31 07:10:43] DEBUG[3131] channel.c: Hanging up channel 'SIP/dep2_1154-08202968' [Jan 31 07:10:43] DEBUG[3131] chan_sip.c: Hangup call SIP/dep2_1154-08202968, SIP callid [EMAIL PROTECTED]) [Jan 31 07:10:43] DEBUG[3131] chan_sip.c: Hanging up channel in state Ring (not UP) [Jan 31 07:10:43] DEBUG[3131] devicestate.c: Notification of state change to be queued on device/channel SIP/dep2_1154-08202968 [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for channel '0x82042e8' [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag c136d668-768786 Our tag: as0bc591fc [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 2b4f6f33-768786 Our tag: as496fd97d [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 73176828-768785 Our tag: as1ab79f58 [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag eae1f94d-768783 Our tag: as1b0024a8 [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag f0629993-768783 Our tag: as3f520446 [Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 728b9929-768782 Our tag: as222bab2d Regards, McCoy You need to Answer() the call in your dialplan, that is my guess without seeing your dialplan. Try adding EXTEN,1,Answer() before the rest of the stuff in your dialplan in the context that handles your inbound calls. Thanks, Steve Totaro ___ -- 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] Dropped Calls
On Tue, 2007-12-18 at 17:43 +0100, Administrateur www.jeremy-salmon.org wrote: Randomly I have dropped calls during communication. No absolutetimeout or other calling limitation options. Any ideas on how to solve this problem? The first place I'd look would be the Asterisk CLI. Make sure you turn up the CLI verbosity first by typing core set verbose 5 before the call. If that doesn't offer any clues, I'd next look at the SIP signaling. You can see that by typing sip set debug at the Asterisk CLI, or by using a network tool such as Wireshark, tcpdump, or ngrep. --- Jared Smith Community Relations Manager Digium, Inc. ___ --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] Dropped Calls
Jared Smith wrote: On Tue, 2007-12-18 at 17:43 +0100, Administrateur www.jeremy-salmon.org wrote: Randomly I have dropped calls during communication. No absolutetimeout or other calling limitation options. Any ideas on how to solve this problem? The first place I'd look would be the Asterisk CLI. Make sure you turn up the CLI verbosity first by typing core set verbose 5 before the call. If that doesn't offer any clues, I'd next look at the SIP signaling. You can see that by typing sip set debug at the Asterisk CLI, or by using a network tool such as Wireshark, tcpdump, or ngrep. --- Jared Smith Community Relations Manager Digium, Inc. I would also ask that all user's keep a log or send an email to you with their extension, if the call was internal or external, time, and how long into the call that it dropped. Collecting this data might help you figure out a trend. I would open an SSH session with txt logging and ask everyone to submit a dropped call report and see if you can link up some common events or errors. You may find it is only happening on external calls which may look like a normal hangup and could indicate a problem with your PSTN connectivity. Thanks, Steve Totaro ___ --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] Dropped calls in Asterisk - A general question
I've got 415 phones, mostly Cisco 7960's. The only time I see dropped calls is when either end hangs up, or I restart asterisk. Using all T1 PRI. HW mainly: Dell 1750 w/2GB, Digium TE410 or TE412P's. Raid1 w/PERC. I use Dell 1950's for the VM servers, but anything with a Digium card is a Dell 1750. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J. Oquendo Sent: Thursday, March 15, 2007 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped calls in Asterisk - A general question Hey all, I have a question for those administrating/building out systems with over 30 users on them. How often do you experience the dropped call phenomena. Would you care to share your experiences including what versions of * you were using, what kind of connectivity was present (T1, Fractional T, Intergrated T, DSL, Cable). Echo? Solutions? (e.g. we bought an X_Brand Echo Canceller). Also, which phones most found favorable with Asterisk on a full functional level. Not Polycoms because they're so neat! Or: Cisco rocks!. Something more to the tune of X_Brand phones worked well with Asterisk 1.2.xx for 70 users on a Data T. We had an X_Brand switch which did/didn't do PoE running Asterisk on a SuperX_Brand server with X amount of memory. Any response is appreciated as long as its something productive. No My SuperX_Brand system has a new logo and a shiny silver box that the vendor states `surpasses unforseen functionality due to hyperbolic hooplah blah blah`. Short, sweet effective. Thanks. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --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] Dropped calls
On Fri, Jan 12, 2007 at 08:23:27PM -0600, Carlos Chavez wrote: I have an Asterisk server with 3 TDM400P cards. 9 FXO and 3 FXS ports. It also has 2 Astribank-8 units connected. The customer is having calls dropped at random intervals but several times a day. Could this be an issue with Interrupts with the 3 cards? For starters, set asterisk to debug mode (set debug 10). You should be able to see where this hangup came from. I am also having a problem sending and receiving faxes when they are either connected to the Astribank or to an FXS port on the TDM card. I know there are issues with Asterisk and faxes but I have never had so many problems in a single installation. Some faxes go through (mostly to local numbers) but long distance calls always give a transmission error on the fax machine. echo cancelling? Do you guarantee that faxes go on a channel with no echo cancelling? If you want to fak from the ports of the Astribank, you'll probably need to use adj_clock . Also make sure that you set the kernel parameter prefmaster of xpp to 0, so it won't force itself t become the zaptel master. Any tips? I am using Asterisk 1.2.13 and Zaptel 1.2.11 on a CentOS 4.4 dual core server. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Dropped Calls on TDM400p
They need to document the exact day and time so you can look in the logs. Is this a T1 or POTS? Customers always complain and threaten to go back to their old PBX. First, calm down, then calm them down and make sure they know you are working on it. Every new install is going to have issues that will take time to resolve. Remember that in your pricing or you will soon be out of business. Get exact times and frequency then check your logs to see if anything matches that may be an issue. Thanks, Steve From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Barratt Sent: Wednesday, September 13, 2006 1:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped Calls on TDM400p These are just PSTN calls, and I have set busydetect=no and callprogress=no in zapata.conf as per voip-info guidance, but problem persists. CPU load never breaks 20, so that doesn't seem to be the problem, but it's a 1.2Ghz Athlon with 768MB RAM. Power supply to system is clean, there's no heavy network traffic going on besides Asterisk to the phones (Aastra 480i's). What other factors can I investigate? This client is so unhappy they are ready to go back to their old PBX system. I am desperate, please help!! Thanks in advance, Jonathan ___ --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] Dropped Calls on TDM400p
Sorry, see now that it is pots. How do your interrupts look? What is the hardware platform or more specifically the MB? Is the platform listed on Digium's site as approved or listed as having issues? Thanks, Steve Steven Totaro wrote: They need to document the exact day and time so you can look in the logs. Is this a T1 or POTS? Customers always complain and threaten to go back to their old PBX. First, calm down, then calm them down and make sure they know you are working on it. Every new install is going to have issues that will take time to resolve. Remember that in your pricing or you will soon be out of business. Get exact times and frequency then check your logs to see if anything matches that may be an issue. Thanks, Steve *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Jonathan Barratt *Sent:* Wednesday, September 13, 2006 1:02 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Dropped Calls on TDM400p These are just PSTN calls, and I have set busydetect=no and callprogress=no in zapata.conf as per voip-info guidance, but problem persists. CPU load never breaks 20, so that doesn't seem to be the problem, but it's a 1.2Ghz Athlon with 768MB RAM. Power supply to system is clean, there's no heavy network traffic going on besides Asterisk to the phones (Aastra 480i's). What other factors can I investigate? This client is so unhappy they are ready to go back to their old PBX system. I am desperate, please help!! Thanks in advance, Jonathan ___ --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] Dropped Calls on TDM400p
Thanks for the reply Steve. I am calm now. :) I've been getting the exact time and number of the dropped calls for the last couple weeks, and there was nothing in system or asterisk logs at those times. So I've spent the last three days sitting at the server, in their office. I can see nothing out of the ordinary going on when the calls are dropped. It seems to happen mostly on outbound calls. The TDM400p is on the same interrupt as the graphics card but nothing else. I didn't think this would be an issue as the box was running headless. So I guess first up is changing PCI slots. MB is an EPoX EP-8KTA2L. I haven't yet found the Hardware Compatibility List on digium.com to determine if it's supported or has known issues, but will keep looking. I was expecting issues, and we've had many of them (esp. echo), but I've been able to resolve them all myself with research and experimentation, except for this persistent intermittent dropped call problem... I'm really grateful for your input Steve, please keep it coming! Thanks very much! Jonathan On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote: Sorry, see now that it is pots.How do your interrupts look?What is the hardware platform or morespecifically the MB?Is the platform listed on Digium's site asapproved or listed as having issues? Thanks,SteveSteven Totaro wrote: They need to document the exact day and time so you can look in the logs.Is this a T1 or POTS? Customers always complain and threaten to go back to their old PBX. First, calm down, then calm them down and make sure they know you are working on it.Every new install is going to have issues that will take time to resolve.Remember that in your pricing or you will soon be out of business. Get exact times and frequency then check your logs to see if anything matches that may be an issue. Thanks, Steve *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] *On Behalf Of *Jonathan Barratt *Sent:* Wednesday, September 13, 2006 1:02 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Dropped Calls on TDM400p These are just PSTN calls, and I have set busydetect=no and callprogress=no in zapata.conf as per voip-info guidance, but problem persists. CPU load never breaks 20, so that doesn't seem to be the problem, but it's a 1.2Ghz Athlon with 768MB RAM. Power supply to system is clean, there's no heavy network traffic going on besides Asterisk to the phones (Aastra 480i's). What other factors can I investigate? This client is so unhappy they are ready to go back to their old PBX system. I am desperate, please help!! Thanks in advance, Jonathan___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Dropped Calls on TDM400p
And of course I find the HCL right after clicking send. My MB is not listed as having any known issues. I still haven't found an approved list, so I can't say that it's approved either. But if swapping the TDM400P's PCI slot doesn't fix the problem I am down to replace the MB and do a full re-install (again). Gratefully, JonathanOn 9/13/06, Jonathan Barratt [EMAIL PROTECTED] wrote: Thanks for the reply Steve. I am calm now. :) I've been getting the exact time and number of the dropped calls for the last couple weeks, and there was nothing in system or asterisk logs at those times. So I've spent the last three days sitting at the server, in their office. I can see nothing out of the ordinary going on when the calls are dropped. It seems to happen mostly on outbound calls. The TDM400p is on the same interrupt as the graphics card but nothing else. I didn't think this would be an issue as the box was running headless. So I guess first up is changing PCI slots. MB is an EPoX EP-8KTA2L. I haven't yet found the Hardware Compatibility List on digium.com to determine if it's supported or has known issues, but will keep looking. I was expecting issues, and we've had many of them (esp. echo), but I've been able to resolve them all myself with research and experimentation, except for this persistent intermittent dropped call problem... I'm really grateful for your input Steve, please keep it coming! Thanks very much! Jonathan On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote: Sorry, see now that it is pots.How do your interrupts look?What is the hardware platform or morespecifically the MB?Is the platform listed on Digium's site asapproved or listed as having issues? Thanks,SteveSteven Totaro wrote: They need to document the exact day and time so you can look in the logs.Is this a T1 or POTS? Customers always complain and threaten to go back to their old PBX. First, calm down, then calm them down and make sure they know you are working on it.Every new install is going to have issues that will take time to resolve.Remember that in your pricing or you will soon be out of business. Get exact times and frequency then check your logs to see if anything matches that may be an issue. Thanks, Steve *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] *On Behalf Of *Jonathan Barratt *Sent:* Wednesday, September 13, 2006 1:02 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Dropped Calls on TDM400p These are just PSTN calls, and I have set busydetect=no and callprogress=no in zapata.conf as per voip-info guidance, but problem persists. CPU load never breaks 20, so that doesn't seem to be the problem, but it's a 1.2Ghz Athlon with 768MB RAM. Power supply to system is clean, there's no heavy network traffic going on besides Asterisk to the phones (Aastra 480i's). What other factors can I investigate? This client is so unhappy they are ready to go back to their old PBX system. I am desperate, please help!! Thanks in advance, Jonathan___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Dropped calls
Chris Mason (Lists) wrote: I have been experiencing dropped calls on my iax2 connections between my Asterisk server and my ITSP providers, I use Teliax and Voxee but it seems to happen on both so I don't think it is the provider. I don't see any packet loss at the time so I don't think it is poor Internet connectivity, what else can I look for? Using Asterisk 1.2.6 but had this problem on 1.2.5 also. Apparently differing Asterisk versions can have an impact on this, however we ended up ditching IAX2 completely and moving to SIP. ___ --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] dropped calls when g729 is used on sip leg
I don't know what to look for in my sip debug logs, can anybody suggest what sorts of messages my phones might unexpectedly give asterisk causing it to drop the zap leg? Mojo with Horan Company, LLC wrote: Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to my extension's sip.conf entry, where existed before disallow=all and allow=ulaw. told asterisk to do a reload, and tried dialing out on a zap line. It was obvious from the call quality that g729 had been selected, and I double-checked and triple-checked by 1) a sip show peer 112 shows: Codecs : 0x104 (ulaw|g729) Codec Order : (g729,ulaw) and 2) the status of the current call as reported by the phone's menu system shows it using g729 as well. So, great. lower network usage, and the quality is good. And if I call another polycom configured the same way, they drop asterisk per canreinvite=yes, and continue their happy g729 way. After an indeterminate amount of time, sometimes 30 seconds and sometimes 5 minutes, one of two things happens: first, sometimes, the zap leg just disappears. I don't get any messages on the CLI at verbose level 30 and debug level 30. The SIP leg stays connected, but the audio trails out into a lovely mash of codec ether before silence. The phone remains off-hook when this happens, and it just remains silent. So I didn't think sip debug logs would help, but I will post them if someone thinks it might help. Secondly, sometimes, the zap leg doesn't disappear, but audio is not delivered from the g729-using polycoms to the zap callee. I hear them but they are just hello? hello?. Neither of these things happen when the phones runs in ulaw. Does anyone have any idea where to look? I'll post whatever logs anyone thinks might help. I'm using 1.2.0b1, but this occurred with my CVS HEAD of around 7/20/2005 as well. Thanks! Mojo -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] dropped calls when g729 is used on sip leg
With verbose and debug both on 255, here's all I get at the CLI. The X is during the call, at the instant the Zap leg seems to drop, almost concurrently with the 'Hungup Zap/1-1'. -- Executing Macro(SIP/112-a88a, internaldialout|7476011) in new stack -- Executing ChanIsAvail(SIP/112-a88a, ZAP/1ZAP/2ZAP/3) in new stack -- Hungup 'Zap/1-1' -- Executing Cut(SIP/112-a88a, theChannel=AVAILCHAN||1) in new stack -- Executing Dial(SIP/112-a88a, Zap/1/7476011||TW) in new stack -- Called 1/7476011 -- Zap/1-1 answered SIP/112-a88a X -- Hungup 'Zap/1-1' Oct 7 11:07:15 WARNING[5895]: channel.c:709 channel_find_locked: Avoided initial deadlock for '0x853ae28', 10 retries! Mojo with Horan Company, LLC wrote: Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to my extension's sip.conf entry, where existed before disallow=all and allow=ulaw. told asterisk to do a reload, and tried dialing out on a zap line. It was obvious from the call quality that g729 had been selected, and I double-checked and triple-checked by 1) a sip show peer 112 shows: Codecs : 0x104 (ulaw|g729) Codec Order : (g729,ulaw) and 2) the status of the current call as reported by the phone's menu system shows it using g729 as well. So, great. lower network usage, and the quality is good. And if I call another polycom configured the same way, they drop asterisk per canreinvite=yes, and continue their happy g729 way. After an indeterminate amount of time, sometimes 30 seconds and sometimes 5 minutes, one of two things happens: first, sometimes, the zap leg just disappears. I don't get any messages on the CLI at verbose level 30 and debug level 30. The SIP leg stays connected, but the audio trails out into a lovely mash of codec ether before silence. The phone remains off-hook when this happens, and it just remains silent. So I didn't think sip debug logs would help, but I will post them if someone thinks it might help. Secondly, sometimes, the zap leg doesn't disappear, but audio is not delivered from the g729-using polycoms to the zap callee. I hear them but they are just hello? hello?. Neither of these things happen when the phones runs in ulaw. Does anyone have any idea where to look? I'll post whatever logs anyone thinks might help. I'm using 1.2.0b1, but this occurred with my CVS HEAD of around 7/20/2005 as well. Thanks! Mojo -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] dropped calls when g729 is used on sip leg
This post is exactly my problem: http://lists.digium.com/pipermail/asterisk-users/2005-July/117988.html Has anybody encountered this and been able to solve it and use g729 successfully? Are there other g729 implementations available as a codec for asterisk? Mojo Mojo with Horan Company, LLC wrote: Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to my extension's sip.conf entry, where existed before disallow=all and allow=ulaw. told asterisk to do a reload, and tried dialing out on a zap line. It was obvious from the call quality that g729 had been selected, and I double-checked and triple-checked by 1) a sip show peer 112 shows: Codecs : 0x104 (ulaw|g729) Codec Order : (g729,ulaw) and 2) the status of the current call as reported by the phone's menu system shows it using g729 as well. So, great. lower network usage, and the quality is good. And if I call another polycom configured the same way, they drop asterisk per canreinvite=yes, and continue their happy g729 way. After an indeterminate amount of time, sometimes 30 seconds and sometimes 5 minutes, one of two things happens: first, sometimes, the zap leg just disappears. I don't get any messages on the CLI at verbose level 30 and debug level 30. The SIP leg stays connected, but the audio trails out into a lovely mash of codec ether before silence. The phone remains off-hook when this happens, and it just remains silent. So I didn't think sip debug logs would help, but I will post them if someone thinks it might help. Secondly, sometimes, the zap leg doesn't disappear, but audio is not delivered from the g729-using polycoms to the zap callee. I hear them but they are just hello? hello?. Neither of these things happen when the phones runs in ulaw. Does anyone have any idea where to look? I'll post whatever logs anyone thinks might help. I'm using 1.2.0b1, but this occurred with my CVS HEAD of around 7/20/2005 as well. Thanks! Mojo -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] dropped calls when g729 is used on sip leg
Is it at all possible asterisk is receiving a SIP message from the phone causing it to drop the zap channel? I've got vad turned off in the polycom configs. Guess I'll comb the sip debug logs. I've got callprogress turned off. I'll try verbosity and debug levels greater than 30 to see if anything gives. Thanks for any suggestions you all might have :) Moj Mojo with Horan Company, LLC wrote: Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to my extension's sip.conf entry, where existed before disallow=all and allow=ulaw. told asterisk to do a reload, and tried dialing out on a zap line. It was obvious from the call quality that g729 had been selected, and I double-checked and triple-checked by 1) a sip show peer 112 shows: Codecs : 0x104 (ulaw|g729) Codec Order : (g729,ulaw) and 2) the status of the current call as reported by the phone's menu system shows it using g729 as well. So, great. lower network usage, and the quality is good. And if I call another polycom configured the same way, they drop asterisk per canreinvite=yes, and continue their happy g729 way. After an indeterminate amount of time, sometimes 30 seconds and sometimes 5 minutes, one of two things happens: first, sometimes, the zap leg just disappears. I don't get any messages on the CLI at verbose level 30 and debug level 30. The SIP leg stays connected, but the audio trails out into a lovely mash of codec ether before silence. The phone remains off-hook when this happens, and it just remains silent. So I didn't think sip debug logs would help, but I will post them if someone thinks it might help. Secondly, sometimes, the zap leg doesn't disappear, but audio is not delivered from the g729-using polycoms to the zap callee. I hear them but they are just hello? hello?. Neither of these things happen when the phones runs in ulaw. Does anyone have any idea where to look? I'll post whatever logs anyone thinks might help. I'm using 1.2.0b1, but this occurred with my CVS HEAD of around 7/20/2005 as well. Thanks! Mojo -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] Dropped calls
do a 'sip debug' and make sure all looks good. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bruce Komito Sent: Monday, May 17, 2004 10:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dropped calls I'm having a problem with outgoing dropped calls. They symptom is, when I place a call from a sip extension to the outside, the call is connected properly, but then abruptly disconnects anywhere from 10 to 60 seconds later. This happens when the outgoing call is through a POTS line (TDM) as well as over a sip gateway. Calls between sip extensions do not have this problem. Has anyone ever experienced this? Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dropped calls
Ok, I'll watch for that as well since I upgraded my desk's Grandstream to 1.0.4.54 an hour ago (previously I had either 4.26 or 4.17). It appears the hangup is triggered by a SIP ACK with CSeq set to 0. Some Grandstream UAs happen to pick 0 as CSeq. chan_sip.c contains if (!p-lastinvite !strlen(p-randdata)) p-needdestroy = 1; where p-lastinvite hold the matching CSeq from the last INVITE. 0 in this case... Fixing this does bring downs the number of hangups, but does not entirely solve the problem. We are still looking. Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dropped calls
I see this very same effect rather often in the following setup: SIP (GS101) -- * -- IAX2 -- * -- MGCP (ip10) In fact I think I've seen it also with SIP instead of MGCP at the end. The first client is behind NAT, by the way. That must be it. I have seen this happening with sip -- * -- IAX as well. I take it you don't know a cure? Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dropped calls
Hi! I see this very same effect rather often in the following setup: SIP (GS101) -- *1 -- IAX2 -- *2 -- MGCP (ip10) In fact I think I've seen it also with SIP instead of MGCP at the end. The first client is behind NAT, by the way. That must be it. I have seen this happening with sip -- * -- IAX as well. I take it you don't know a cure? Unfortunately not, no. By the way I am not on latest CVS as that would disable my MGCP phones. And so far I didn't even get a chance to debug this since it happens approx 1 out of 10 calls only. By the way, I can now conirm that it can be both MGCP or SIP at the end, it doesn't matter. So to me it looks like IAX2 is involved as well, not just SIP. *1: CVS-02/10/04-16:49:37 *2: CVS-03/05/04-00:50:56 Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dropped calls
On Thu, 2004-04-15 at 18:21, Philipp von Klitzing wrote: So to me it looks like IAX2 is involved as well, not just SIP. Are you sure? I did some analysis of my traffic. Here is what I found so far: Only Grandstream phones appear to be affected. All phones affected have been behind a coned NAT, running firmware 1.0.4.39 with STUN enabled. The hangup only occurs in dialogs with CSeq set to '0'. I will test whether another firmware will solve this issue. Let's hoep the best. Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dropped calls
We also are having randomly dropped calls with our IAX2 connections, we have tried IAX2 with and without trunking enabled. the phones are snom 200's with SIP and there is an asterisk box at each site so no sip nat problems. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Philipp von Klitzing Sent: Thursday, April 15, 2004 11:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dropped calls Hi! I see this very same effect rather often in the following setup: SIP (GS101) -- *1 -- IAX2 -- *2 -- MGCP (ip10) In fact I think I've seen it also with SIP instead of MGCP at the end. The first client is behind NAT, by the way. That must be it. I have seen this happening with sip -- * -- IAX as well. I take it you don't know a cure? Unfortunately not, no. By the way I am not on latest CVS as that would disable my MGCP phones. And so far I didn't even get a chance to debug this since it happens approx 1 out of 10 calls only. By the way, I can now conirm that it can be both MGCP or SIP at the end, it doesn't matter. So to me it looks like IAX2 is involved as well, not just SIP. *1: CVS-02/10/04-16:49:37 *2: CVS-03/05/04-00:50:56 Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dropped calls
Hi! Only Grandstream phones appear to be affected. All phones affected have been behind a coned NAT, running firmware 1.0.4.39 with STUN enabled. The hangup only occurs in dialogs with CSeq set to '0'. Ok, I'll watch for that as well since I upgraded my desk's Grandstream to 1.0.4.54 an hour ago (previously I had either 4.26 or 4.17). Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dropped calls
Hi! Lately, I have been experiencing unexpected hangups just when the a call has been established. This effects a small percentage of all calls coming from sip phone which are terminated on a zap pri channel. I see this very same effect rather often in the following setup: SIP (GS101) -- * -- IAX2 -- * -- MGCP (ip10) In fact I think I've seen it also with SIP instead of MGCP at the end. The first client is behind NAT, by the way. Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dropped calls from queue
I just updated to latest cvs and the problem remains. I did also notice that when the call coming in on the queue is through a Zap line (from an adtran 750 to an x100p) it logs the following just before the warnings below: pr 7 14:21:21 VERBOSE[60194841]: -- SIP/hrutter-432b answered Zap/13-1 Apr 7 14:21:21 DEBUG[60194841]: Set option TONE VERIFY, mode: MUTECONF/MAX(2) on Zap/13-1 Apr 7 14:21:21 VERBOSE[60194841]: -- Stopped music on hold on Zap/13-1 Tony Buser wrote: We're having a strange problem with our receptionist. She runs an xpro softphone and we're using a queue to handle incoming calls. It seems nearly all of the calls that come in through the queue get dropped. At first we thought it might have been human error (clicking the wrong button in xpro or something) or that the person waiting in the queue just gave up and hungup, however it seems to happen when the following gets logged: Apr 7 14:53:35 WARNING[60424217]: File 10 does not exist in any format Apr 7 14:53:35 WARNING[60424217]: Unable to open 10 (format G729A): No such file or directory Apr 7 14:53:35 WARNING[60424217]: Agent on SIP/hrutter-c6fa hungup on the customer. They're going to be pissed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dropped calls from queue
Hate to reply to my own message again, but I just figured it out. Nothing wrong with asterisk really, just a bad configuration. Somehow the queue line in extensions conf got changed by someone to: exten = 81003,3,Queue(receptionistq|tTH||10) Thats where the 10 was coming from. :) Could this be considered a bug? It shouldn't hang up on someone just because the wav file for an announcement can't be found? All this time we were blaming the poor receptionist. Tony Buser wrote: Apr 7 14:53:35 WARNING[60424217]: File 10 does not exist in any format Apr 7 14:53:35 WARNING[60424217]: Unable to open 10 (format G729A): No such file or directory Apr 7 14:53:35 WARNING[60424217]: Agent on SIP/hrutter-c6fa hungup on the customer. They're going to be pissed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dropped calls
Well, after days chasing this ghost, the problem seems to be solved: - There was a problem with the NTP server (not responding to clients due to config restrictions) - upgrade gs firmware to latest available: 1.4.50. If the NTP server is down gs phones keep trying to reach it and drop some of the packets received from * during a call. * thinks the phone is dead and drops the call. Anyone can confirm this? --- Paulo Loureiro. On Fri, 2004-03-05 at 19:24, Bartosz Jozwiak wrote: I have couple of GS phone and CISCO 7960. The funny thing is that two of that GS phone keep disconnecting and also CISCO 7960 phone keeps disconnecting. But the problem appear month ago! This is really strange! Bart Hello, I'll try that, but why on earth gs phones with the same firmware on another * server, work with no problem? I've failed to state I'm using zaprtc, since there is no digium hardware on the server. Does it matter? Thanks, --- Paulo. On Fri, 2004-03-05 at 18:49, Ross Donaldson wrote: There is new firmware that may help http://www.grandstream.com/BETATEST/. Grandstream acknowledges this problem. They say it is a codec issue with asterisk. I don't know if this update addresses this problem but it may be worth a try. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Loureiro Sent: Friday, March 05, 2004 10:26 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] dropped calls Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message Didn't get a frame from channel: SIP/3805-df43, but I can't figure why. asterisk logs: - Mar 5 15:57:26 DEBUG[1116957488]: build_route: Contact hop: sip:192.168.60.106 Mar 5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered SIP/-08122450 Mar 5 15:57:26 VERBOSE[1217669936]: -- Attempting native bridge of SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:26 DEBUG[1116957488]: Stopping retransmission on '[EMAIL PROTECTED]' of Response\ 25663: Found Mar 5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524 Mar 5 15:57:26 DEBUG[1217669936]: Ooh, format changed from UNKN to ULAW Mar 5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel: SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse counter Mar 5 15:57:38 DEBUG[1217669936]: is not a local user Mar 5 15:57:38 VERBOSE[1217669936]: == Spawn extension (local, 3805, 1) exited non-zero on 'SIP/-0812245\0' - The scenario: 1 server (redhat 9), asterisk (stable) and a 2 x hisax ISDN BRI. One of the BRI boards is used to dial out (ppp) on one channel and a mgetty on the other channel. The other board is in ptp and used by *. The phones are Grandstream BT101 and Handytone and are all on a switched network (3 procurve switches, stacked). The configs are ok, since the same files on another server work ok (no dropped calls), but I can post them if needed. Any help will be greatly appreciated. Thanks in advance, --- Paulo Loureiro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cumprimentos, --- Paulo Loureiro Netmania ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cumprimentos, --- Paulo Loureiro Netmania ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dropped calls
There is new firmware that may help http://www.grandstream.com/BETATEST/. Grandstream acknowledges this problem. They say it is a codec issue with asterisk. I don't know if this update addresses this problem but it may be worth a try. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Loureiro Sent: Friday, March 05, 2004 10:26 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] dropped calls Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message Didn't get a frame from channel: SIP/3805-df43, but I can't figure why. asterisk logs: - Mar 5 15:57:26 DEBUG[1116957488]: build_route: Contact hop: sip:192.168.60.106 Mar 5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered SIP/-08122450 Mar 5 15:57:26 VERBOSE[1217669936]: -- Attempting native bridge of SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:26 DEBUG[1116957488]: Stopping retransmission on '[EMAIL PROTECTED]' of Response\ 25663: Found Mar 5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524 Mar 5 15:57:26 DEBUG[1217669936]: Ooh, format changed from UNKN to ULAW Mar 5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel: SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse counter Mar 5 15:57:38 DEBUG[1217669936]: is not a local user Mar 5 15:57:38 VERBOSE[1217669936]: == Spawn extension (local, 3805, 1) exited non-zero on 'SIP/-0812245\0' - The scenario: 1 server (redhat 9), asterisk (stable) and a 2 x hisax ISDN BRI. One of the BRI boards is used to dial out (ppp) on one channel and a mgetty on the other channel. The other board is in ptp and used by *. The phones are Grandstream BT101 and Handytone and are all on a switched network (3 procurve switches, stacked). The configs are ok, since the same files on another server work ok (no dropped calls), but I can post them if needed. Any help will be greatly appreciated. Thanks in advance, --- Paulo Loureiro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dropped calls
Hello, I'll try that, but why on earth gs phones with the same firmware on another * server, work with no problem? I've failed to state I'm using zaprtc, since there is no digium hardware on the server. Does it matter? Thanks, --- Paulo. On Fri, 2004-03-05 at 18:49, Ross Donaldson wrote: There is new firmware that may help http://www.grandstream.com/BETATEST/. Grandstream acknowledges this problem. They say it is a codec issue with asterisk. I don't know if this update addresses this problem but it may be worth a try. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Loureiro Sent: Friday, March 05, 2004 10:26 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] dropped calls Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message Didn't get a frame from channel: SIP/3805-df43, but I can't figure why. asterisk logs: - Mar 5 15:57:26 DEBUG[1116957488]: build_route: Contact hop: sip:192.168.60.106 Mar 5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered SIP/-08122450 Mar 5 15:57:26 VERBOSE[1217669936]: -- Attempting native bridge of SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:26 DEBUG[1116957488]: Stopping retransmission on '[EMAIL PROTECTED]' of Response\ 25663: Found Mar 5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524 Mar 5 15:57:26 DEBUG[1217669936]: Ooh, format changed from UNKN to ULAW Mar 5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel: SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse counter Mar 5 15:57:38 DEBUG[1217669936]: is not a local user Mar 5 15:57:38 VERBOSE[1217669936]: == Spawn extension (local, 3805, 1) exited non-zero on 'SIP/-0812245\0' - The scenario: 1 server (redhat 9), asterisk (stable) and a 2 x hisax ISDN BRI. One of the BRI boards is used to dial out (ppp) on one channel and a mgetty on the other channel. The other board is in ptp and used by *. The phones are Grandstream BT101 and Handytone and are all on a switched network (3 procurve switches, stacked). The configs are ok, since the same files on another server work ok (no dropped calls), but I can post them if needed. Any help will be greatly appreciated. Thanks in advance, --- Paulo Loureiro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cumprimentos, --- Paulo Loureiro Netmania ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dropped calls
I have couple of GS phone and CISCO 7960. The funny thing is that two of that GS phone keep disconnecting and also CISCO 7960 phone keeps disconnecting. But the problem appear month ago! This is really strange! Bart Hello, I'll try that, but why on earth gs phones with the same firmware on another * server, work with no problem? I've failed to state I'm using zaprtc, since there is no digium hardware on the server. Does it matter? Thanks, --- Paulo. On Fri, 2004-03-05 at 18:49, Ross Donaldson wrote: There is new firmware that may help http://www.grandstream.com/BETATEST/. Grandstream acknowledges this problem. They say it is a codec issue with asterisk. I don't know if this update addresses this problem but it may be worth a try. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Loureiro Sent: Friday, March 05, 2004 10:26 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] dropped calls Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message Didn't get a frame from channel: SIP/3805-df43, but I can't figure why. asterisk logs: - Mar 5 15:57:26 DEBUG[1116957488]: build_route: Contact hop: sip:192.168.60.106 Mar 5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered SIP/-08122450 Mar 5 15:57:26 VERBOSE[1217669936]: -- Attempting native bridge of SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:26 DEBUG[1116957488]: Stopping retransmission on '[EMAIL PROTECTED]' of Response\ 25663: Found Mar 5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524 Mar 5 15:57:26 DEBUG[1217669936]: Ooh, format changed from UNKN to ULAW Mar 5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel: SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse counter Mar 5 15:57:38 DEBUG[1217669936]: is not a local user Mar 5 15:57:38 VERBOSE[1217669936]: == Spawn extension (local, 3805, 1) exited non-zero on 'SIP/-0812245\0' - The scenario: 1 server (redhat 9), asterisk (stable) and a 2 x hisax ISDN BRI. One of the BRI boards is used to dial out (ppp) on one channel and a mgetty on the other channel. The other board is in ptp and used by *. The phones are Grandstream BT101 and Handytone and are all on a switched network (3 procurve switches, stacked). The configs are ok, since the same files on another server work ok (no dropped calls), but I can post them if needed. Any help will be greatly appreciated. Thanks in advance, --- Paulo Loureiro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cumprimentos, --- Paulo Loureiro Netmania ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
