[asterisk-users] Uninstalling Asterisk? No make uninstall?
There does not seem to be any make uninstall for Asterisk 1.2.9.1 and Zaptel 1.2.6... I tried to apply an uninstall patch but got many Hunk errors from both 1.2.9.1 and latest SVN: http://bugs.digium.com/file_download.php?file_id=8805type=bug Is there a reason that there is no make uninstall? And what is the easiest way to completely remove Asterisk and Zaptel from any given system -- cleanly and properly? Thanks in advance. ___ --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] Uninstalling Asterisk? No make uninstall?
On Sat, 2006-07-08 at 10:22 +0300, Tzafrir Cohen wrote: On Sat, Jul 08, 2006 at 04:46:00PM +1000, Carey O'Shea wrote: There does not seem to be any make uninstall for Asterisk 1.2.9.1 and Zaptel 1.2.6... I tried to apply an uninstall patch but got many Hunk errors from both 1.2.9.1 and latest SVN: http://bugs.digium.com/file_download.php?file_id=8805type=bug Next time, patch --dry-run # :-( It's OK, I did the patch on copies of my built source on an isolated server, so I still have my original untouched sources that I am currently running. Is there a reason that there is no make uninstall? And what is the easiest way to completely remove Asterisk and Zaptel from any given system -- cleanly and properly? If you want to reinstall just reinstall on top of the old system. I don't need to reinstall, I need to uninstall. You can't really be sure that the uninstall script you'll be running is using the same options as the one you build with. I should be able to with a proper make uninstall. If you want to allow a clean uninstall, either use asterisk from a decent package or try something like checkinstall . The Asterisk packages built for my systems are always far too out of date. However you are right, since there seems to be no make uninstall, then packages should have been built by hand at compile time instead of make install. But it is too late for that now, make install has already been run, hence my post. Is the only option to manually remove everything? ___ --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] Uninstalling Asterisk? No make uninstall?
On Sat, 2006-07-08 at 10:58 +0300, Tzafrir Cohen wrote: On Sat, Jul 08, 2006 at 05:43:41PM +1000, Carey O'Shea wrote: On Sat, 2006-07-08 at 10:22 +0300, Tzafrir Cohen wrote: On Sat, Jul 08, 2006 at 04:46:00PM +1000, Carey O'Shea wrote: There does not seem to be any make uninstall for Asterisk 1.2.9.1 and Zaptel 1.2.6... I tried to apply an uninstall patch but got many Hunk errors from both 1.2.9.1 and latest SVN: http://bugs.digium.com/file_download.php?file_id=8805type=bug Next time, patch --dry-run # :-( It's OK, I did the patch on copies of my built source on an isolated server, so I still have my original untouched sources that I am currently running. Is there a reason that there is no make uninstall? And what is the easiest way to completely remove Asterisk and Zaptel from any given system -- cleanly and properly? If you want to reinstall just reinstall on top of the old system. I don't need to reinstall, I need to uninstall. You can't really be sure that the uninstall script you'll be running is using the same options as the one you build with. I should be able to with a proper make uninstall. If you want to allow a clean uninstall, either use asterisk from a decent package or try something like checkinstall . The Asterisk packages built for my systems are always far too out of date. However you are right, since there seems to be no make uninstall, then packages should have been built by hand at compile time instead of make install. But it is too late for that now, make install has already been run, hence my post. Is the only option to manually remove everything? Everything is not much. Mostly /etc/asterisk , /var/lib/asterisk , /var/spool/asterisk , /var/run/asterisk , /etc/asterisk , /usr/lib/asterisk (/modules) and /usr/share/asterisk (at least in some cases). There are also a number of binaries in /usr/sbin (most notably asterisk) which may differ a bit, depending on yor installation method. And the many manpages, and zaptel modules, and header include for zaptel, and /dev/zap, and /usr/include/asterisk, and rasterisk, and safe_asterisk, and /var/log/asterisk, and astkeygen, and astman, ztcfg, zttool, and I'm sure there are more. ___ --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] Uninstalling Asterisk? No make uninstall?
On Sat, 2006-07-08 at 13:00 +0300, Tzafrir Cohen wrote: On Sat, Jul 08, 2006 at 07:27:37PM +1000, Carey O'Shea wrote: On Sat, 2006-07-08 at 10:58 +0300, Tzafrir Cohen wrote: On Sat, Jul 08, 2006 at 05:43:41PM +1000, Carey O'Shea wrote: On Sat, 2006-07-08 at 10:22 +0300, Tzafrir Cohen wrote: On Sat, Jul 08, 2006 at 04:46:00PM +1000, Carey O'Shea wrote: There does not seem to be any make uninstall for Asterisk 1.2.9.1 and Zaptel 1.2.6... I tried to apply an uninstall patch but got many Hunk errors from both 1.2.9.1 and latest SVN: http://bugs.digium.com/file_download.php?file_id=8805type=bug Next time, patch --dry-run # :-( It's OK, I did the patch on copies of my built source on an isolated server, so I still have my original untouched sources that I am currently running. Is there a reason that there is no make uninstall? And what is the easiest way to completely remove Asterisk and Zaptel from any given system -- cleanly and properly? If you want to reinstall just reinstall on top of the old system. I don't need to reinstall, I need to uninstall. You can't really be sure that the uninstall script you'll be running is using the same options as the one you build with. I should be able to with a proper make uninstall. If you want to allow a clean uninstall, either use asterisk from a decent package or try something like checkinstall . The Asterisk packages built for my systems are always far too out of date. However you are right, since there seems to be no make uninstall, then packages should have been built by hand at compile time instead of make install. But it is too late for that now, make install has already been run, hence my post. Is the only option to manually remove everything? Everything is not much. Mostly /etc/asterisk , /var/lib/asterisk , /var/spool/asterisk , /var/run/asterisk , /etc/asterisk , /usr/lib/asterisk (/modules) and /usr/share/asterisk (at least in some cases). There are also a number of binaries in /usr/sbin (most notably asterisk) which may differ a bit, depending on yor installation method. And the many manpages, Right and zaptel modules, This is not asteris. It's zaptel. Take a look at the guessswork done in the install target of zaptel. Do you really want such guesswork be done automatically? Hint: even getting the kernel version may notbe trivial. imply deleting everything under misc/ might deete something else you installed. and header include for zaptel, zaptel.h and libtonezone are part of zaptel, again. and /dev/zap, and /usr/include/asterisk, and rasterisk, and safe_asterisk, and /var/log/asterisk, and astkeygen, and astman, Right ztcfg, zttool, and I'm sure there are more. Zaptel, again. I realise which things I mentioned originated from Zaptel, as my original post asked about removing both Asterisk and Zaptel. I'm not complaining, I'm just saying that removing things manually means that you are likely to miss things. ___ --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] hangup lag causing the answering of already answered calls
Well I've found out what was causing my duplicate logging: it was entirely a NAT issue. Found out it was only happening on some remote endpoints (and not all of them), and that different routers proved to not have duplicate logging. What part of NAT could cause this? Was it really sending all packets twice, or something like that? Just seems kinda strange. Anyway, it's no longer a problem. My original problem, however, remains. Phone doesn't stop ringing when it's meant to. Only happens when call is via my ZapATA. Any ideas/help/input is appreciated! Regards, Carey. On Sun, 2006-06-11 at 15:18 +1000, Carey O'Shea wrote: Does anyone have any ideas as to what can cause this large delay to stop ringing? It's quite a show stopper... imagine ringing a business and being answered by 3 different people, one after the other, all talking over the top of each other. On Fri, 2006-06-09 at 15:12 +1000, Carey O'Shea wrote: Hi Undrhil, A logical idea, but unfortunately adding it didn't change anything. Two important points: (1) When I test this with just IAX endpoints, no Zap, the call is hungup immediately, (2) but the console still shows the user being called twice. So as a wild guess, maybe the console logging twice is OK, and it's my Zap configuration? * extensions.conf: [incoming] exten = s,1,Dial(IAX2/carey) exten = s,2,Hangup(IAX2/carey) * zapata.conf: [channels] usecallerid=no signalling=fxs_ks context=incoming channel = 4 * zaptel.conf loadzone=au defaultzone=au fxsks=4 * ztcfg -vv Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. I'm from Australia so I assume the loadzone and defaultzone is OK as per zaptel.c. Did not post iax.conf due to my SIP phones having the same behaviour, and IAX-to-IAX not exhibiting the problem. On Fri, 2006-06-09 at 04:54 +, [EMAIL PROTECTED] wrote: So, your dialplan for that incoming call is just the one line? exten = s,1,Dial(IAX2/carey) Nothing else? Try adding a Hangup command on the next priority and see if that helps any. exten = s,2,Hangup If you already have a Hangup command in there, then I apologize for wasting your time. :) Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the Asterisk console that my phone is called twice every time there is an incoming call. Is this normal, and could it be causing this behaviour? If not, any ideas as to what could be causing this? I can provide full debug logs and my relevant configuration if needed. Console log: -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- IAX2/carey-1 is ringing -- IAX2/carey-1 is ringing -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' ___ --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 ___ --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] hangup lag causing the answering of already answered calls
http://www.voip-info.org/wiki/index.php?page=Australia%20Asterisk% 20Details Stumbled across this Reverse On Idle Condition (ROIC) 'feature' that sounds very promising. Will get it enabled later today and give it a go. On Tue, 2006-06-20 at 23:35 +1000, Carey O'Shea wrote: Well I've found out what was causing my duplicate logging: it was entirely a NAT issue. Found out it was only happening on some remote endpoints (and not all of them), and that different routers proved to not have duplicate logging. What part of NAT could cause this? Was it really sending all packets twice, or something like that? Just seems kinda strange. Anyway, it's no longer a problem. My original problem, however, remains. Phone doesn't stop ringing when it's meant to. Only happens when call is via my ZapATA. Any ideas/help/input is appreciated! Regards, Carey. On Sun, 2006-06-11 at 15:18 +1000, Carey O'Shea wrote: Does anyone have any ideas as to what can cause this large delay to stop ringing? It's quite a show stopper... imagine ringing a business and being answered by 3 different people, one after the other, all talking over the top of each other. On Fri, 2006-06-09 at 15:12 +1000, Carey O'Shea wrote: Hi Undrhil, A logical idea, but unfortunately adding it didn't change anything. Two important points: (1) When I test this with just IAX endpoints, no Zap, the call is hungup immediately, (2) but the console still shows the user being called twice. So as a wild guess, maybe the console logging twice is OK, and it's my Zap configuration? * extensions.conf: [incoming] exten = s,1,Dial(IAX2/carey) exten = s,2,Hangup(IAX2/carey) * zapata.conf: [channels] usecallerid=no signalling=fxs_ks context=incoming channel = 4 * zaptel.conf loadzone=au defaultzone=au fxsks=4 * ztcfg -vv Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. I'm from Australia so I assume the loadzone and defaultzone is OK as per zaptel.c. Did not post iax.conf due to my SIP phones having the same behaviour, and IAX-to-IAX not exhibiting the problem. On Fri, 2006-06-09 at 04:54 +, [EMAIL PROTECTED] wrote: So, your dialplan for that incoming call is just the one line? exten = s,1,Dial(IAX2/carey) Nothing else? Try adding a Hangup command on the next priority and see if that helps any. exten = s,2,Hangup If you already have a Hangup command in there, then I apologize for wasting your time. :) Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the Asterisk console that my phone is called twice every time there is an incoming call. Is this normal, and could it be causing this behaviour? If not, any ideas as to what could be causing this? I can provide full debug logs and my relevant configuration if needed. Console log: -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- IAX2/carey-1 is ringing -- IAX2/carey-1 is ringing -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' ___ --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
[Asterisk-Users] TDM400P bad echo problem, tried lots of things
I have a bad echo problem on my TDM400P with one FXO module installed. I have tried a few things, such as: * setting rxgain and txgain to 0 * setting echocancelwhenbridged to no / yes * settting echocancel to 64 / no / yes * setting echocanceltraining to 800 / no / yes * MG2 echo cancellation * MARK2 echo cancellation * KB1 echo cancellation * AGGRESSIVE_SUPPRESSOR option of MARK2 Each time restarting Asterisk, then opening the Zap channel, and then speaking...only to hear my self played back almost instantly. None of these options changed the echo for me, it always sounded the same -- except for the AGGRESSIVE_SUPPRESSOR option, in which every time I spoke it made the other end a very low volume, so much that I couldn't hear the other end (ie: not useful). I don't have this problem with pure IP calls, it's only with my TDM400P and FXO that I have this echo problem. This means my headset and IP phones are fine (of course). So, what else can I try? :-) Any ideas why this is so consistent and persistent? Maybe it's something to do with my phone cable or something of that nature (hmm?)? Any input appreciated. Thanks, Carey O'Shea. ___ --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] Which application to open Zap channel?
I'm using Dial(Zap/X/) as suggested. However, Dial(Zap/X) does indeed work for me. So I'm curious, what's the difference between them, and when wouldn't just Zap/X work? On Wed, 2006-06-14 at 11:14 -0500, Eric ManxPower Wieling wrote: Carey O'Shea wrote: I swear Dial(Zap/X) was the first thing I tried and it didn't work, but now it works fine... hmmm maybe I forgot to reload my extensions or something like that. Don't expect Dial(Zap/X) to work. Expect Dial(Zap/X/) to work. ___ --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] Which application to open Zap channel?
I'm sure this a very common and easy thing to do with Asterisk, but for the life of me I can't find the application that will allow me to open a Zap channel. Real world example: To be able to connect to an open Zap channel, so it would allow me to say, join in on a call that was originally answered by a PSTN phone (ie. just like you would by simply picking up another PSTN phone..!). There is ZapBarge, but allows no speaking, which is useless for this situation. Maybe I just have to use Dial in some way? Thanks. ___ --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] Which application to open Zap channel?
I swear Dial(Zap/X) was the first thing I tried and it didn't work, but now it works fine... hmmm maybe I forgot to reload my extensions or something like that. Thanks though. On Wed, 2006-06-14 at 10:03 -0400, Mailing List wrote: This will just pick up the line exten = *01,1,Dial(ZAP/1/) _ Mobilcom http://www.mobilcom.net - Original Message - From: Carey O'Shea [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, June 14, 2006 9:48 AM Subject: [Asterisk-Users] Which application to open Zap channel? I'm sure this a very common and easy thing to do with Asterisk, but for the life of me I can't find the application that will allow me to open a Zap channel. Real world example: To be able to connect to an open Zap channel, so it would allow me to say, join in on a call that was originally answered by a PSTN phone (ie. just like you would by simply picking up another PSTN phone..!). There is ZapBarge, but allows no speaking, which is useless for this situation. Maybe I just have to use Dial in some way? Thanks. ___ --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] hangup lag causing the answering of already answered calls
Does anyone have any ideas as to what can cause this large delay to stop ringing? It's quite a show stopper... imagine ringing a business and being answered by 3 different people, one after the other, all talking over the top of each other. On Fri, 2006-06-09 at 15:12 +1000, Carey O'Shea wrote: Hi Undrhil, A logical idea, but unfortunately adding it didn't change anything. Two important points: (1) When I test this with just IAX endpoints, no Zap, the call is hungup immediately, (2) but the console still shows the user being called twice. So as a wild guess, maybe the console logging twice is OK, and it's my Zap configuration? * extensions.conf: [incoming] exten = s,1,Dial(IAX2/carey) exten = s,2,Hangup(IAX2/carey) * zapata.conf: [channels] usecallerid=no signalling=fxs_ks context=incoming channel = 4 * zaptel.conf loadzone=au defaultzone=au fxsks=4 * ztcfg -vv Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. I'm from Australia so I assume the loadzone and defaultzone is OK as per zaptel.c. Did not post iax.conf due to my SIP phones having the same behaviour, and IAX-to-IAX not exhibiting the problem. On Fri, 2006-06-09 at 04:54 +, [EMAIL PROTECTED] wrote: So, your dialplan for that incoming call is just the one line? exten = s,1,Dial(IAX2/carey) Nothing else? Try adding a Hangup command on the next priority and see if that helps any. exten = s,2,Hangup If you already have a Hangup command in there, then I apologize for wasting your time. :) Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the Asterisk console that my phone is called twice every time there is an incoming call. Is this normal, and could it be causing this behaviour? If not, any ideas as to what could be causing this? I can provide full debug logs and my relevant configuration if needed. Console log: -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- IAX2/carey-1 is ringing -- IAX2/carey-1 is ringing -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' ___ --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 ___ --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] hangup lag causing the answering of already answered calls
I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the Asterisk console that my phone is called twice every time there is an incoming call. Is this normal, and could it be causing this behaviour? If not, any ideas as to what could be causing this? I can provide full debug logs and my relevant configuration if needed. Console log: -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- IAX2/carey-1 is ringing -- IAX2/carey-1 is ringing -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' ___ --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] hangup lag causing the answering of already answered calls
Hi Undrhil, A logical idea, but unfortunately adding it didn't change anything. Two important points: (1) When I test this with just IAX endpoints, no Zap, the call is hungup immediately, (2) but the console still shows the user being called twice. So as a wild guess, maybe the console logging twice is OK, and it's my Zap configuration? * extensions.conf: [incoming] exten = s,1,Dial(IAX2/carey) exten = s,2,Hangup(IAX2/carey) * zapata.conf: [channels] usecallerid=no signalling=fxs_ks context=incoming channel = 4 * zaptel.conf loadzone=au defaultzone=au fxsks=4 * ztcfg -vv Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. I'm from Australia so I assume the loadzone and defaultzone is OK as per zaptel.c. Did not post iax.conf due to my SIP phones having the same behaviour, and IAX-to-IAX not exhibiting the problem. On Fri, 2006-06-09 at 04:54 +, [EMAIL PROTECTED] wrote: So, your dialplan for that incoming call is just the one line? exten = s,1,Dial(IAX2/carey) Nothing else? Try adding a Hangup command on the next priority and see if that helps any. exten = s,2,Hangup If you already have a Hangup command in there, then I apologize for wasting your time. :) Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the Asterisk console that my phone is called twice every time there is an incoming call. Is this normal, and could it be causing this behaviour? If not, any ideas as to what could be causing this? I can provide full debug logs and my relevant configuration if needed. Console log: -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- IAX2/carey-1 is ringing -- IAX2/carey-1 is ringing -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' ___ --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] Re: Received VNAK: resending outstanding frames?
I've recently swapped my router out for a slightly different router, and now everything works fine. I guess my other router must not be very good or have some issue with this protocol/setup/something. Thanks for the help. Regards, Carey O'Shea. On Wed, 2006-04-12 at 20:16 +1000, Carey O'Shea wrote: On Wed, 2006-04-12 at 07:58 +0100, Tim Panton wrote: There is a manual at: http://www.centralitycomm.com/solutions/Download/documents/product/ PA168SUserguideEng.pdf Tim Panton [EMAIL PROTECTED] I'm now outside the network again and have run iax2 debug. Below are the results. Notice how after the Raw Hangup there is a 30 second pause, then it retries, and when it gets to the VNAK then it repeats the same message constantly for another 30 seconds (snipped the 5000+ lines of course), and then gets the Raw Hangup again. Ad infinitum. I have uploaded the log here: http://www.users.on.net/~lncoshea/carey/asterisk-log.txt Does the log help? Anyone have any ideas going from the log? Regards, Carey O'Shea. PS: Thanks Tim, I worked out how to reset the phone a few hours ago, the manual was wrong for my particular model, I had to press hash (#) _before_ power on, see here: http://forums.whirlpool.net.au/forum-replies.cfm?t=504889 ___ --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] Re: Received VNAK: resending outstanding frames?
On Wed, 2006-04-12 at 07:58 +0100, Tim Panton wrote: There is a manual at: http://www.centralitycomm.com/solutions/Download/documents/product/ PA168SUserguideEng.pdf Tim Panton [EMAIL PROTECTED] I'm now outside the network again and have run iax2 debug. Below are the results. Notice how after the Raw Hangup there is a 30 second pause, then it retries, and when it gets to the VNAK then it repeats the same message constantly for another 30 seconds (snipped the 5000+ lines of course), and then gets the Raw Hangup again. Ad infinitum. I have uploaded the log here: http://www.users.on.net/~lncoshea/carey/asterisk-log.txt Does the log help? Anyone have any ideas going from the log? Regards, Carey O'Shea. PS: Thanks Tim, I worked out how to reset the phone a few hours ago, the manual was wrong for my particular model, I had to press hash (#) _before_ power on, see here: http://forums.whirlpool.net.au/forum-replies.cfm?t=504889 ___ --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] Re: Received VNAK: resending outstanding frames?
Some more info: Just tried this on a server without using any NAT and no port forwarding, no masquerading, and I still have the same problem. So there goes that idea. I do not know what this VNAK error means. By the way, I am using the latest version (1.2.6) of asterisk, have also tried other versions with the same problem [1.0.9 (Ubuntu Breezy) and 1.0.7 (Debian Sarge) and 1.2.1 (Ubuntu Dapper)]. Wonder if this is something simple or not? On Tue, 2006-04-11 at 23:00 +1000, Carey O'Shea wrote: I only receive 4 google results on my error. So some help would be appreciated. I could not even determine what VNAK was. Let me describe my problem. I have an IAX hardware phone here that connects and operates fine within my internal network. However, outside my internal network, the hardware phone fails to register. The plot thickens: outside my internal network I've tested numerous IAX softphones and strangely enough they function fine, where the hardware phone does not. Of course (seeing as how the softphones work externally) I have both TCP and UDP 4569 port forwarded to Asterisk server and there is no firewall on the Asterisk server. So I am guessing there is some NAT issue or some configuration issue with this hardware IAX phone I have. Below are the messages I receieve in my full log (about 20 or 30 of them each second for many seconds, then after the flood of messages it reports Raw Hangup, and then soon enough it starts again). I'm using a PA1686 IAX hardware phone. Any ideas please? snip ... Apr 11 22:49:52 DEBUG[16385] chan_iax2.c: Received VNAK: resending outstanding frames Apr 11 22:49:52 DEBUG[16385] chan_iax2.c: Received VNAK: resending outstanding frames Apr 11 22:49:52 DEBUG[16385] chan_iax2.c: Received VNAK: resending outstanding frames Apr 11 22:49:52 DEBUG[16385] chan_iax2.c: Received VNAK: resending outstanding frames Apr 11 22:49:52 DEBUG[16385] chan_iax2.c: Received VNAK: resending outstanding frames Apr 11 22:49:52 DEBUG[16385] chan_iax2.c: Received VNAK: resending outstanding frames Apr 11 22:49:52 DEBUG[16385] chan_iax2.c: Received VNAK: resending outstanding frames Apr 11 22:49:52 DEBUG[16385] chan_iax2.c: Received VNAK: resending outstanding frames Apr 11 22:49:52 DEBUG[16385] chan_iax2.c: Received VNAK: resending outstanding frames Apr 11 22:49:53 DEBUG[16385] chan_iax2.c: Raw Hangup 59.167.XXX.XXX:8435, src=3, dst=9660 ___ --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] Re: Received VNAK: resending outstanding frames?
On Tue, 2006-04-11 at 23:00 +1000, Carey O'Shea wrote: I only receive 4 google results on my error. So some help would be appreciated. I could not even determine what VNAK was. Let me describe my problem. I have an IAX hardware phone here that connects and operates fine within my internal network. However, outside my internal network, the hardware phone fails to register. The plot thickens: outside my internal network I've tested numerous IAX softphones and strangely enough they function fine, where the hardware phone does not. Of course (seeing as how the softphones work externally) I have both TCP and UDP 4569 port forwarded to Asterisk server and there is no firewall on the Asterisk server. So I am guessing there is some NAT issue or some configuration issue with this hardware IAX phone I have. Below are the messages I receieve in my full log (about 20 or 30 of them each second for many seconds, then after the flood of messages it reports Raw Hangup, and then soon enough it starts again). I'm using a PA1686 IAX hardware phone. I have a few PA1686 phones doing IAX through firewalls and NAT, so it is possible :-) Try running : iax2 debug Can you send us the few lines in the log before the VNAK's start? Tim Panton [EMAIL PROTECTED] Thanks for the reply Tim. I would love to send you those lines... but unfortunately I accidentally put my PA168S phone into PPPoE mode and now it doesn't try to pick up a local DHCP address, and just sits there endlessly trying to connect to a bogus PPPoE account. I need to know how to reset this phone to defaults. Can't see any pinhole or anything, so perhaps it needs to be opened up and there is a way to do this inside of the phone? Or any other ideas so that I can access it :-) Carey O'Shea. ___ --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
