Re: [Asterisk-Users] Random Disconnects - or ARE they?
Colin Anderson wrote: Having followed this thread, it seems to me that the simplest way to test for the busydetect hangup is to get the guy to make a few phone calls on someone else's phone. If the hangup happens, then it's the guy. If it doesn't, it's his station. The way that I ran into the fix was that I ran Asterisk if full debug mode. When I had a reported hangup, I had them tell me the time. Viewing the full log showed 'Hangup detected". This may be what you'll want to do. Doug ___ --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] Random Disconnects - or ARE they?
Having followed this thread, it seems to me that the simplest way to test for the busydetect hangup is to get the guy to make a few phone calls on someone else's phone. If the hangup happens, then it's the guy. If it doesn't, it's his station. -Original Message- From: Michael Sampson [mailto:[EMAIL PROTECTED] Sent: Thursday, February 16, 2006 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Random Disconnects - or ARE they? Maybe people are just hanging up on him cause he is talking to loud. Doug Lytle wrote: > Brent Torrenga wrote: > >> I have one use on our PBX who has been experiencing seemingly random >> disconnects. The user is on the same LAN as everyone else, using the >> same >> > > Brent, > > The last time I was having random disconnects, it turned out to be > that I had busydetect=yes on my zapata.conf. I changed this to no and > the issues went away. > > Doug > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random Disconnects - or ARE they?
Maybe people are just hanging up on him cause he is talking to loud. Doug Lytle wrote: Brent Torrenga wrote: I have one use on our PBX who has been experiencing seemingly random disconnects. The user is on the same LAN as everyone else, using the same Brent, The last time I was having random disconnects, it turned out to be that I had busydetect=yes on my zapata.conf. I changed this to no and the issues went away. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random Disconnects - or ARE they?
Brent Torrenga wrote: I have one use on our PBX who has been experiencing seemingly random disconnects. The user is on the same LAN as everyone else, using the same Brent, The last time I was having random disconnects, it turned out to be that I had busydetect=yes on my zapata.conf. I changed this to no and the issues went away. Doug ___ --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] Random Disconnects - or ARE they?
Perhaps you can scale him back at the 79XX? Not only might it solve the problem, but I'll bet the people talking to him on the other end would appreciate it as well... On the Aastra 9133i, for example, you can provide gain settings in the (mac).cfg file. Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Torrenga Sent: Wednesday, February 15, 2006 3:36 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Random Disconnects - or ARE they? I have one use on our PBX who has been experiencing seemingly random disconnects. The user is on the same LAN as everyone else, using the same type of phone (79XX loaded with SIP firmware) as everyone else. He had some disconnects a few weeks ago, I suspected the phone, so I swapped his with mine. I have since not had issues with his old phone, however, he has had issues using mine. So, the problem seems to be not with the phone, but with his station. I started thinking maybe the cable is bad. I checked the network stats on his 79XX, and never see any receive errors - perfect network performance. Also, the CLI has no indication of an error whenever a disconnect occurs, it just looks like a normal hangup of the Zap channel (TDM400P). The ONLY difference between this user and everyone else is his extremely loud talking. When I run ztmonitor it is obvious that he simply pegs the meter. Either it reads peaked out or silence, whether he is speaking or being quiet. Is it entirely possible that he is driving the Zap channel so hard that it either hangs up or causes the telco CO to hang up the channel? Is there something else I should look at that might indicate what the problem is? I am kinda pulling my hair out on this one, any help or suggestions would be appreciated. Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random Disconnects - or ARE they?
On Wed, February 15, 2006 22:35, Brent Torrenga said: > I have one use on our PBX who has been experiencing seemingly random > disconnects. The user is on the same LAN as everyone else, using the same > type of phone (79XX loaded with SIP firmware) as everyone else. He had > some > disconnects a few weeks ago, I suspected the phone, so I swapped his with > mine. I have since not had issues with his old phone, however, he has had > issues using mine. So, the problem seems to be not with the phone, but > with > his station. I started thinking maybe the cable is bad. I checked the > network stats on his 79XX, and never see any receive errors - perfect > network performance. Also, the CLI has no indication of an error whenever > a > disconnect occurs, it just looks like a normal hangup of the Zap channel > (TDM400P). > > The ONLY difference between this user and everyone else is his extremely > loud talking. When I run ztmonitor it is obvious that he simply pegs the > meter. Either it reads peaked out or silence, whether he is speaking or > being quiet. > > Is it entirely possible that he is driving the Zap channel so hard that it > either hangs up or causes the telco CO to hang up the channel? Is there > something else I should look at that might indicate what the problem is? I > am kinda pulling my hair out on this one, any help or suggestions would be > appreciated. > > LOL... You could try to explain that he doesn't need to shout to the person on the other side, that the telephone transmits the sound by wire, and not by air, so he doesn't need to shout to be heard on the other side! ;-) But seriously, I am really curious whether there is a connection between voice volume and disconnects... Please do keep us informed... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Random Disconnects - or ARE they?
I have one use on our PBX who has been experiencing seemingly random disconnects. The user is on the same LAN as everyone else, using the same type of phone (79XX loaded with SIP firmware) as everyone else. He had some disconnects a few weeks ago, I suspected the phone, so I swapped his with mine. I have since not had issues with his old phone, however, he has had issues using mine. So, the problem seems to be not with the phone, but with his station. I started thinking maybe the cable is bad. I checked the network stats on his 79XX, and never see any receive errors - perfect network performance. Also, the CLI has no indication of an error whenever a disconnect occurs, it just looks like a normal hangup of the Zap channel (TDM400P). The ONLY difference between this user and everyone else is his extremely loud talking. When I run ztmonitor it is obvious that he simply pegs the meter. Either it reads peaked out or silence, whether he is speaking or being quiet. Is it entirely possible that he is driving the Zap channel so hard that it either hangs up or causes the telco CO to hang up the channel? Is there something else I should look at that might indicate what the problem is? I am kinda pulling my hair out on this one, any help or suggestions would be appreciated. Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com ___ --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] Random Disconnects
On 1/16/06, Thczv F. Thczv <[EMAIL PROTECTED]> wrote: > > > I'm using Sipura 3000 as well, however I will have to wait until > > > Monday about the Switch I'm not sure. So far it looks like Sipura is > > > at fault. In the mean time I would like to hear from others using the > > > Sipura 3000 FXO if they have the same problem. > > > > For now I am experimenting with allowing reinvites between the > > SPA-3000 FXO port and a couple of other extensions. > > Had a disconnect today, even though reinvites were allowed. So now I > am looking for a different solution. Turns out what I thought was this same sort of disconnect was actually my contractor cutting my phone line. :-) Now that it is repaired, I haven't had any of these random disconnects since reinvites were allowed. I am not very satisfied with this, though. I want to use some features (like Park) that apparently don't work well with reinvites. Have any of the rest of you had any luck troubleshooting this problem? Dave ___ --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] Random Disconnects
What version of asterisk you using? On 1/16/06, Kerry Garrison <[EMAIL PROTECTED]> wrote: > Wish I could help, I can tell you I have never been disconnected from a > call on my system that has been running non-stop for months with an > SPA-3000. > > All of my settings are documented here: > http://voipspeak.net/index.php?/content/view/24/27/ > > Kerry Garrison > Publisher - http://GeekGazette.com - http://VOIPSpeak.net > (949) 502-7819 x200 - [EMAIL PROTECTED] > http://www.techdatapros.com > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Thczv F. Thczv > > Sent: Sunday, January 15, 2006 9:09 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Random Disconnects > > > > On 1/14/06, C F <[EMAIL PROTECTED]> wrote: > > > > > I'm using Sipura 3000 as well, however I will have to wait until > > > Monday about the Switch I'm not sure. So far it looks like > > Sipura is > > > at fault. In the mean time I would like to hear from others > > using the > > > Sipura 3000 FXO if they have the same problem. > > > > For now I am experimenting with allowing reinvites between > > the SPA-3000 FXO port and a couple of other extensions. > > > > I sent several inquiries to people who complained of this problem in > > the past (6+ months ago). I haven't heard back from them. > > > > Dave > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random Disconnects
On 1/15/06, Thczv F. Thczv <[EMAIL PROTECTED]> wrote: > > I'm using Sipura 3000 as well, however I will have to wait until > > Monday about the Switch I'm not sure. So far it looks like Sipura is > > at fault. In the mean time I would like to hear from others using the > > Sipura 3000 FXO if they have the same problem. > > For now I am experimenting with allowing reinvites between the > SPA-3000 FXO port and a couple of other extensions. Had a disconnect today, even though reinvites were allowed. So now I am looking for a different solution. Dave ___ --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] Random Disconnects
Wish I could help, I can tell you I have never been disconnected from a call on my system that has been running non-stop for months with an SPA-3000. All of my settings are documented here: http://voipspeak.net/index.php?/content/view/24/27/ Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Thczv F. Thczv > Sent: Sunday, January 15, 2006 9:09 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Random Disconnects > > On 1/14/06, C F <[EMAIL PROTECTED]> wrote: > > > I'm using Sipura 3000 as well, however I will have to wait until > > Monday about the Switch I'm not sure. So far it looks like > Sipura is > > at fault. In the mean time I would like to hear from others > using the > > Sipura 3000 FXO if they have the same problem. > > For now I am experimenting with allowing reinvites between > the SPA-3000 FXO port and a couple of other extensions. > > I sent several inquiries to people who complained of this problem in > the past (6+ months ago). I haven't heard back from them. > > Dave > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random Disconnects
For me allowing reinvites is not an option, as Polycom phones with Sipura 3000 don't work together on this. On 1/16/06, Thczv F. Thczv <[EMAIL PROTECTED]> wrote: > On 1/14/06, C F <[EMAIL PROTECTED]> wrote: > > > I'm using Sipura 3000 as well, however I will have to wait until > > Monday about the Switch I'm not sure. So far it looks like Sipura is > > at fault. In the mean time I would like to hear from others using the > > Sipura 3000 FXO if they have the same problem. > > For now I am experimenting with allowing reinvites between the > SPA-3000 FXO port and a couple of other extensions. > > I sent several inquiries to people who complained of this problem in > the past (6+ months ago). I haven't heard back from them. > > Dave > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random Disconnects
On 1/14/06, C F <[EMAIL PROTECTED]> wrote: > I'm using Sipura 3000 as well, however I will have to wait until > Monday about the Switch I'm not sure. So far it looks like Sipura is > at fault. In the mean time I would like to hear from others using the > Sipura 3000 FXO if they have the same problem. For now I am experimenting with allowing reinvites between the SPA-3000 FXO port and a couple of other extensions. I sent several inquiries to people who complained of this problem in the past (6+ months ago). I haven't heard back from them. Dave ___ --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] Random Disconnects
I'm using Sipura 3000 as well, however I will have to wait until Monday about the Switch I'm not sure. So far it looks like Sipura is at fault. In the mean time I would like to hear from others using the Sipura 3000 FXO if they have the same problem. On 1/13/06, Thczv F. Thczv <[EMAIL PROTECTED]> wrote: > On 1/13/06, C F <[EMAIL PROTECTED]> wrote: > > > OK, I'm using it with Sipura, Polycom, and Asterisk, and only when > > Asterisk is in the media path, do I have this problem. Also I haven't > > noticed this problem with Polycoms alone. So I'm assuming it's a > > Sipura problem. But I might be wrong. > > That makes sense to me. It only seems to happen when I receive calls > through the SPA-3000. I never happens on outbound SIP calls, or > inbound SIP calls. Do you use a SPA-3000 or some other Sipura device? > > > One more question, what network card do you have in that box? > > I believe the network card is a Netgear FA310TX. > > > Yes I was talking about the network hardware switch. > > My asterisk box connects to a switch port on a Linksys WRT54G. Most > of my other devices connect to a Netgear FS608 switch. Some devices > connect to the Netgear switch through an old Asante dumb hub. > > Dave > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random Disconnects
On 1/13/06, C F <[EMAIL PROTECTED]> wrote: > OK, I'm using it with Sipura, Polycom, and Asterisk, and only when > Asterisk is in the media path, do I have this problem. Also I haven't > noticed this problem with Polycoms alone. So I'm assuming it's a > Sipura problem. But I might be wrong. That makes sense to me. It only seems to happen when I receive calls through the SPA-3000. I never happens on outbound SIP calls, or inbound SIP calls. Do you use a SPA-3000 or some other Sipura device? > One more question, what network card do you have in that box? I believe the network card is a Netgear FA310TX. > Yes I was talking about the network hardware switch. My asterisk box connects to a switch port on a Linksys WRT54G. Most of my other devices connect to a Netgear FS608 switch. Some devices connect to the Netgear switch through an old Asante dumb hub. Dave ___ --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] Random Disconnects
OK, I'm using it with Sipura, Polycom, and Asterisk, and only when Asterisk is in the media path, do I have this problem. Also I haven't noticed this problem with Polycoms alone. So I'm assuming it's a Sipura problem. But I might be wrong. One more question, what network card do you have in that box? Yes I was talking about the network hardware switch. On 1/13/06, Thczv F. Thczv <[EMAIL PROTECTED]> wrote: > On 1/13/06, Thczv F. Thczv <[EMAIL PROTECTED]> wrote: > > > > 2. Is asterisk in the media path? > > > > I don't think I understand this question. > > I learned a little about this. Assuming I understand this correctly, > I think asterisk is in the media path. canreinvite=no for all my > devices. > > Sorry for being slow. :-) > > Dave > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random Disconnects
On 1/13/06, Thczv F. Thczv <[EMAIL PROTECTED]> wrote: > > 2. Is asterisk in the media path? > > I don't think I understand this question. I learned a little about this. Assuming I understand this correctly, I think asterisk is in the media path. canreinvite=no for all my devices. Sorry for being slow. :-) Dave ___ --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] Random Disconnects
On 1/13/06, C F <[EMAIL PROTECTED]> wrote: > I have this exact same problem, so far I have not been able to figure > out whats going on. I also get the follwoing errors: > == > Didn't get a frame from channel: SIP/119-41c6 > Bridge stops bridging channels SIP/119-41c6 and SIP/101-264e > == > Which tells us that a frame was dropped. That seems to be the common problem for others as well. > Lets try compring setups: > 1. What phones are you using? I use vanilla analog phones with Sipura ATAs. > 2. Is asterisk in the media path? I don't think I understand this question. > 3. Are they all local without NAT? Yes, they are all local. They are behind my NAT, but my asterisk box and my inbound POTS line are also behind my NAT, so I don't think NAT is a problem. > 4. What phone is SIP/101? > 5. What phone is SIP/119? Both of these are extensions on my SPA-3000. SIP/119 is the extension I use for incoming calls on the FXO port. POTS calls hit the FXO port, the SPA-3000 sends them to Asterisk, which sees them as coming from extension SIP/119. Asterisk then rings SIP/101 (which is the FXS port on my SPA-3000), AND SIP/109, which is a different SPA-2002 device. > 6. What platform is Asterisk running on (Intel? AMD? etc.)? I believe it is an AMD processor. > 7. What switch are you using? I'm not sure I understand this question. Are you asking about my network switch hardware? > 8. What version of Asterisk are you running? I started with [EMAIL PROTECTED] 1.5. But I stripped everything out of the sip.conf, extensions.conf, and related files and rebuilt them myself. [EMAIL PROTECTED] 1.5 runs Asterisk 1.0.9. Thanks, Dave ___ --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] Random Disconnects
I have this exact same problem, so far I have not been able to figure
out whats going on. I also get the follwoing errors:
==
Didn't get a frame from channel: SIP/119-41c6
Bridge stops bridging channels SIP/119-41c6 and SIP/101-264e
==
Which tells us that a frame was dropped.
Lets try compring setups:
1. What phones are you using?
2. Is asterisk in the media path?
3. Are they all local without NAT?
4. What phone is SIP/101?
5. What phone is SIP/119?
6. What platform is Asterisk running on (Intel? AMD? etc.)?
7. What switch are you using?
8. What version of Asterisk are you running?
On 1/12/06, Thczv F. Thczv <[EMAIL PROTECTED]> wrote:
> I am hoping some of you can help me troubleshoot this problem I am
> having with my home asterisk machine. I have incoming POTS service
> using a SPA-3000 (extension 119). Calls on that line go to an
> attendant recording that offers a menu choice: press 1 for Nancy,
> press 2 for the rest of us. In reality, pressing anything other than
> 1 sends the call to the rest of us by dialing both extensions 101 and
> 109. Tonight we received a call on this POTS line. The caller
> pressed 2. We answered extension 101. Several seconds later, the
> call was cut off. However, I noticed that extension 109 started (or
> was still) ringing. But the caller wasn't completely disconnected.
> He ended up leaving a voicemail message for extension 101 (which is
> where calls to 101 or 109 go).
>
> Any ideas what might be going on here? I have included logs from this
> call below.
>
> I would appreciate any advice. This Asterisk hobby of mine is just
> barely passing the wife test. :-) I'm not sure how long it will
> last.
>
> Thanks,
>
> Dave
>
> **
> Jan 12 19:38:36 DEBUG[1546]: Setting NAT on RTP to 0
> Jan 12 19:38:36 DEBUG[1546]: Stopping retransmission on
> '[EMAIL PROTECTED]' of Response 101: Found
> Jan 12 19:38:36 DEBUG[1546]: Setting NAT on RTP to 0
> Jan 12 19:38:36 DEBUG[1546]: Check for res for 119
> Jan 12 19:38:36 DEBUG[1546]: Call from user '119' is 1 out of 0
> Jan 12 19:38:36 DEBUG[1546]: build_route: Contact hop: SureWest
> Jan 12 19:38:36 VERBOSE[1546]: -- Executing Goto("SIP/119-41c6",
> "auto-attendant|115|1") in new stack
> Jan 12 19:38:36 VERBOSE[1546]: -- Goto (auto-attendant,115,1)
> Jan 12 19:38:36 VERBOSE[1546]: -- Executing Answer("SIP/119-41c6", "")
> in new stack
> Jan 12 19:38:36 VERBOSE[1546]: -- Executing Wait("SIP/119-41c6", "1")
> in new stack
> Jan 12 19:38:36 DEBUG[1546]: Stopping retransmission on
> '[EMAIL PROTECTED]' of Response 102: Found
> Jan 12 19:38:37 VERBOSE[1546]: -- Executing
> ResponseTimeout("SIP/119-41c6", "7") in new stack
> Jan 12 19:38:37 VERBOSE[1546]: -- Set Response Timeout to 7
> Jan 12 19:38:37 VERBOSE[1546]: -- Executing BackGround("SIP/119-41c6",
> "/var/lib/asterisk/sounds/custom/haddock-main-menu") in new stack
> Jan 12 19:38:37 DEBUG[1546]: Ooh, format changed from unknown to ulaw
> Jan 12 19:38:37 DEBUG[1546]: Scheduling timer at 160 sample intervals
> Jan 12 19:38:37 VERBOSE[1546]: -- Playing
> '/var/lib/asterisk/sounds/custom/haddock-main-menu' (language 'en')
> Jan 12 19:38:42 DEBUG[1546]: Scheduling timer at 138 sample intervals
> Jan 12 19:38:42 DEBUG[1546]: Scheduling timer at 0 sample intervals
> Jan 12 19:38:42 DEBUG[1546]: Scheduling timer at 0 sample intervals
> Jan 12 19:38:44 DEBUG[1546]: Sending dtmf: 50 (2), at 192.168.1.150
> Jan 12 19:38:44 VERBOSE[1546]: -- Invalid extension '2' in context
> 'auto-attendant' on SIP/119-41c6
> Jan 12 19:38:44 VERBOSE[1546]: == CDR updated on SIP/119-41c6
> Jan 12 19:38:44 VERBOSE[1546]: -- Executing Goto("SIP/119-41c6",
> "from-pstn|main|1") in new stack
> Jan 12 19:38:44 VERBOSE[1546]: -- Goto (from-pstn,main,1)
> Jan 12 19:38:44 VERBOSE[1546]: -- Executing Dial("SIP/119-41c6",
> "SIP/101&SIP/109|20") in new stack
> Jan 12 19:38:44 DEBUG[1546]: Setting NAT on RTP to 0
> Jan 12 19:38:44 DEBUG[1546]: Outgoing Call for 101
> Jan 12 19:38:44 DEBUG[1546]: Call from user '101' is 1 out of 0
> Jan 12 19:38:44 VERBOSE[1546]: -- Called 101
> Jan 12 19:38:44 DEBUG[1546]: Setting NAT on RTP to 0
> Jan 12 19:38:44 DEBUG[1546]: Outgoing Call for 109
> Jan 12 19:38:44 DEBUG[1546]: Call from user '109' is 1 out of 0
> Jan 12 19:38:44 VERBOSE[1546]: -- Called 109
> Jan 12 19:38:44 DEBUG[1546]: (Provisional) Stopping retransmission
> (but retaining packet) on
> '[EMAIL PROTECTED]' Request 102: Found
> Jan 12 19:38:44 DEBUG[1546]: (Provisional) Stopping retransmission
> (but retaining packet) on
> '[EMAIL PROTECTED]' Request 102: Found
> Jan 12 19:38:44 DEBUG[1546]: (Provisional) Stopping retransmission
> (but retaining packet) on
> '[EMAIL PROTECTED]' Request 102: Found
> Jan 12 19:38:44 VERBOSE[1546]: -- SIP/101-264e is ringing
> Jan 12 19:38:44 DEBUG[1546]: (Provisional) Stopping retransmission
> (but retaining packet) on
> '[EMAIL PROTECTED]' Request 102: Found
> Jan 12 19:38:44 DEBUG[1546]: Driver for channel 'SIP/119-41c6' does
> not support indication 3, e
[Asterisk-Users] Random Disconnects
I am hoping some of you can help me troubleshoot this problem I am
having with my home asterisk machine. I have incoming POTS service
using a SPA-3000 (extension 119). Calls on that line go to an
attendant recording that offers a menu choice: press 1 for Nancy,
press 2 for the rest of us. In reality, pressing anything other than
1 sends the call to the rest of us by dialing both extensions 101 and
109. Tonight we received a call on this POTS line. The caller
pressed 2. We answered extension 101. Several seconds later, the
call was cut off. However, I noticed that extension 109 started (or
was still) ringing. But the caller wasn't completely disconnected.
He ended up leaving a voicemail message for extension 101 (which is
where calls to 101 or 109 go).
Any ideas what might be going on here? I have included logs from this
call below.
I would appreciate any advice. This Asterisk hobby of mine is just
barely passing the wife test. :-) I'm not sure how long it will
last.
Thanks,
Dave
**
Jan 12 19:38:36 DEBUG[1546]: Setting NAT on RTP to 0
Jan 12 19:38:36 DEBUG[1546]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 101: Found
Jan 12 19:38:36 DEBUG[1546]: Setting NAT on RTP to 0
Jan 12 19:38:36 DEBUG[1546]: Check for res for 119
Jan 12 19:38:36 DEBUG[1546]: Call from user '119' is 1 out of 0
Jan 12 19:38:36 DEBUG[1546]: build_route: Contact hop: SureWest
Jan 12 19:38:36 VERBOSE[1546]: -- Executing Goto("SIP/119-41c6",
"auto-attendant|115|1") in new stack
Jan 12 19:38:36 VERBOSE[1546]: -- Goto (auto-attendant,115,1)
Jan 12 19:38:36 VERBOSE[1546]: -- Executing Answer("SIP/119-41c6", "")
in new stack
Jan 12 19:38:36 VERBOSE[1546]: -- Executing Wait("SIP/119-41c6", "1")
in new stack
Jan 12 19:38:36 DEBUG[1546]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 102: Found
Jan 12 19:38:37 VERBOSE[1546]: -- Executing
ResponseTimeout("SIP/119-41c6", "7") in new stack
Jan 12 19:38:37 VERBOSE[1546]: -- Set Response Timeout to 7
Jan 12 19:38:37 VERBOSE[1546]: -- Executing BackGround("SIP/119-41c6",
"/var/lib/asterisk/sounds/custom/haddock-main-menu") in new stack
Jan 12 19:38:37 DEBUG[1546]: Ooh, format changed from unknown to ulaw
Jan 12 19:38:37 DEBUG[1546]: Scheduling timer at 160 sample intervals
Jan 12 19:38:37 VERBOSE[1546]: -- Playing
'/var/lib/asterisk/sounds/custom/haddock-main-menu' (language 'en')
Jan 12 19:38:42 DEBUG[1546]: Scheduling timer at 138 sample intervals
Jan 12 19:38:42 DEBUG[1546]: Scheduling timer at 0 sample intervals
Jan 12 19:38:42 DEBUG[1546]: Scheduling timer at 0 sample intervals
Jan 12 19:38:44 DEBUG[1546]: Sending dtmf: 50 (2), at 192.168.1.150
Jan 12 19:38:44 VERBOSE[1546]: -- Invalid extension '2' in context
'auto-attendant' on SIP/119-41c6
Jan 12 19:38:44 VERBOSE[1546]: == CDR updated on SIP/119-41c6
Jan 12 19:38:44 VERBOSE[1546]: -- Executing Goto("SIP/119-41c6",
"from-pstn|main|1") in new stack
Jan 12 19:38:44 VERBOSE[1546]: -- Goto (from-pstn,main,1)
Jan 12 19:38:44 VERBOSE[1546]: -- Executing Dial("SIP/119-41c6",
"SIP/101&SIP/109|20") in new stack
Jan 12 19:38:44 DEBUG[1546]: Setting NAT on RTP to 0
Jan 12 19:38:44 DEBUG[1546]: Outgoing Call for 101
Jan 12 19:38:44 DEBUG[1546]: Call from user '101' is 1 out of 0
Jan 12 19:38:44 VERBOSE[1546]: -- Called 101
Jan 12 19:38:44 DEBUG[1546]: Setting NAT on RTP to 0
Jan 12 19:38:44 DEBUG[1546]: Outgoing Call for 109
Jan 12 19:38:44 DEBUG[1546]: Call from user '109' is 1 out of 0
Jan 12 19:38:44 VERBOSE[1546]: -- Called 109
Jan 12 19:38:44 DEBUG[1546]: (Provisional) Stopping retransmission
(but retaining packet) on
'[EMAIL PROTECTED]' Request 102: Found
Jan 12 19:38:44 DEBUG[1546]: (Provisional) Stopping retransmission
(but retaining packet) on
'[EMAIL PROTECTED]' Request 102: Found
Jan 12 19:38:44 DEBUG[1546]: (Provisional) Stopping retransmission
(but retaining packet) on
'[EMAIL PROTECTED]' Request 102: Found
Jan 12 19:38:44 VERBOSE[1546]: -- SIP/101-264e is ringing
Jan 12 19:38:44 DEBUG[1546]: (Provisional) Stopping retransmission
(but retaining packet) on
'[EMAIL PROTECTED]' Request 102: Found
Jan 12 19:38:44 DEBUG[1546]: Driver for channel 'SIP/119-41c6' does
not support indication 3, emulating it
Jan 12 19:38:44 DEBUG[1546]: Scheduling timer at 160 sample intervals
Jan 12 19:38:44 VERBOSE[1546]: -- SIP/109-88dd is ringing
Jan 12 19:38:44 DEBUG[1546]: Generator got voice, switching to phase locked mode
Jan 12 19:38:44 DEBUG[1546]: Scheduling timer at 0 sample intervals
Jan 12 19:38:44 DEBUG[1546]: Difference is 15304, ms is 1933
Jan 12 19:38:49 DEBUG[1546]: Acked pending invite 102
Jan 12 19:38:49 DEBUG[1546]: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Found
Jan 12 19:38:49 DEBUG[1546]: build_route: Contact hop:
Jan 12 19:38:49 VERBOSE[1546]: -- SIP/101-264e answered SIP/119-41c6
Jan 12 19:38:49 DEBUG[1546]: update_user_counter(109) - decrement outUse counter
Jan 12 19:38:49 DEBUG[1546]: Scheduling timer at 0 sample intervals
Jan 12 19:38:49 VERBOSE[1546]: -- Attempti
Re: [Asterisk-Users] Random Disconnects
| - Original Message - | | Hi Matt, | | | | Increase your busycount to 6 or 7. I had that problem also with an | | X100P, and it went away increasing the busycount parameter. | | Now that I check it, I don't have a busycount...does this really need to be | set in dsp.c? | | If so how would I compile it and install it with the machine running? | According to http://www.automated.it/guidetoasterisk.htm it goes in the zapata.conf file after busydetect=yes. Soz for mail list spam... Kind regards, Matt Riddell | | On Mon, 2004-04-19 at 20:28, Matt Riddell wrote: | | > I am getting random disconnects about 5-10 times a day. The logs show | | > nothing except that the call was hung up. The calls are from | | > X100P->*->digium T1 card->carrier access channel bank II->analogue | | > phone. It is happening to all users. Is it possible that this is | | > coming from busydetect=yes? | | > | | > Does busydetect detect cadences etc for the hangup frequencies? I | | > have busycount=3... | | > | | > Any ideas? Any more information I could provide? | | > | | > Kind regards, | | > | | > | | > Matt Riddell | | -- | | Nicolas Gudino <[EMAIL PROTECTED]> | | House Internet S.R.L. | | | | ___ | | 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 ___ 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] Random Disconnects
- Original Message - | Hi Matt, | | Increase your busycount to 6 or 7. I had that problem also with an | X100P, and it went away increasing the busycount parameter. Now that I check it, I don't have a busycount...does this really need to be set in dsp.c? If so how would I compile it and install it with the machine running? Cheers, Matt | On Mon, 2004-04-19 at 20:28, Matt Riddell wrote: | > I am getting random disconnects about 5-10 times a day. The logs show | > nothing except that the call was hung up. The calls are from | > X100P->*->digium T1 card->carrier access channel bank II->analogue | > phone. It is happening to all users. Is it possible that this is | > coming from busydetect=yes? | > | > Does busydetect detect cadences etc for the hangup frequencies? I | > have busycount=3... | > | > Any ideas? Any more information I could provide? | > | > Kind regards, | > | > | > Matt Riddell | -- | Nicolas Gudino <[EMAIL PROTECTED]> | House Internet S.R.L. | | ___ | 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] Random Disconnects
Hi Matt, Increase your busycount to 6 or 7. I had that problem also with an X100P, and it went away increasing the busycount parameter. On Mon, 2004-04-19 at 20:28, Matt Riddell wrote: > I am getting random disconnects about 5-10 times a day. The logs show > nothing except that the call was hung up. The calls are from > X100P->*->digium T1 card->carrier access channel bank II->analogue > phone. It is happening to all users. Is it possible that this is > coming from busydetect=yes? > > Does busydetect detect cadences etc for the hangup frequencies? I > have busycount=3... > > Any ideas? Any more information I could provide? > > Kind regards, > > > Matt Riddell -- Nicolas Gudino <[EMAIL PROTECTED]> House Internet S.R.L. ___ 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] Random Disconnects
I am getting random disconnects about 5-10 times a day. The logs show nothing except that the call was hung up. The calls are from X100P->*->digium T1 card->carrier access channel bank II->analogue phone. It is happening to all users. Is it possible that this is coming from busydetect=yes? Does busydetect detect cadences etc for the hangup frequencies? I have busycount=3... Any ideas? Any more information I could provide? Kind regards, Matt Riddell
