RE: [Asterisk-Users] Monitor
Well, for the last part of my email, I now know of AgentCallbackLogin You see.. Asterisk is your friend! :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Lunes, 21 de Febrero de 2005 01:48 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Monitor Guys. How does monitor work? Ive enabled the feature to start monitoring when *! is pressed but I see that my calls are left with some IN and OUT file... how can I merge those into one? Also, when does asterisk records a call? I know I configured it to record queue calls... but what else? Ah! which brings me to another question, when using queues, agents signin and they get MOH until a user calls but... on other call center apps, the agents signin and can actually hangup the phone, which rings when a call comes thru can asterisk behave in this manner or do the agents have to be offhook for this? Thx! __ Anton Krall ___ 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 ___ 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] Asterisk H323 support
Hi, with Openh323 - v1.12.2 and pwlib - v1.5.2 I use asterisk-oh323 v.0.6.3b and it works fine hope it helps cu... --- kolo sos [EMAIL PROTECTED] wrote: Hi, anybody knows what's missing or problem why i cant compile asterisk-oh323 in my machine? i got this compiled successfully ...Openh323 - v1.12.2 ...pwlib - v1.5.2 except ...asterisk-oh323 - v0.6.5 here's the output as i run make... [EMAIL PROTECTED]:~/voip/asterisk-oh323-0.6.5$ make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/home/mkoy/voip/asterisk-oh323-0.6.5/wrapper' ./check_ver /home/mkoy/pwlib pwlib ./check_ver /home/mkoy/openh323 openh323 g++ -DP_LINUX=2.4.26 -ffunction-sections -fdata-sections -D_REENTRANT -Wall -fPIC -DP_USE_PRAGMA -DPHAS_TEMPLATES -I/home/mkoy/pwlib/include/ptlib/unix -I/usr/include/pwlib -I/home/mkoy/pwlib/include -DPTRACING -I/home/mkoy/openh323/include -DHAS_IXJ -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.5.2\ -DOPENH323VERSION=\1.12.2\ -I/home/mkoy/pwlib/include/ptlib/unix -I/home/mkoy/pwlib/include -I/home/mkoy/openh323/include -I/home/mkoy/openh323/include/openh323 -I../asterisk-driver -c asteriskaudio.cxx -o asteriskaudio.o asteriskaudio.cxx: In destructor `virtual PAsteriskSoundChannel::~PAsteriskSoundChannel()': asteriskaudio.cxx:167: error: `baseChannel' undeclared (first use this function) asteriskaudio.cxx:167: error: (Each undeclared identifier is reported only once for each function it appears in.) make[1]: *** [asteriskaudio.o] Error 1 make[1]: Leaving directory `/home/mkoy/voip/asterisk-oh323-0.6.5/wrapper' make: *** [subdirs_build] Error 1 [EMAIL PROTECTED]:~/voip/asterisk-oh323-0.6.5$ Kolosos Philippines __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.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 __ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250 ___ 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] SIP echo on LAN
Check your soundcard controls... maybe it's recording what you hear or PCM, thus sending it again to the other party. Julianjm. On Mon, 21 Feb 2005 09:47:55 +0200, Nic le Roux [EMAIL PROTECTED] wrote: Good Morning, I have a weird situation, I'm testing with Xlite as SIP phone (is it any good ) and dialing an extension (also Xlite on same LAN) and I'm getting a real bad echo on the dialer's side and a not so bad one on the receivers side. Has anyone had something like this ? Aparently one should only get echo when you break out onto a telco network ? ___ 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 ___ 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
[Asterisk-Users] MOH clicks
Guys.. Ive noticed that I have 2 mpg123 processes running, is that ok? also... can you make MOH random? Also, I dont know if there is a problem with my config but when listening to MOH, every 3 or so second I get a click sound which notices because music gets a hickup every 3 or so seconds... is this ok? __ Anton Krall ___ 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] CallerID
You can't change the callerid on an outgoing PSTN call (at least on
analog lines).
To modifiy the callerid on incoming calls, you could do something like
this (not tested):
[incoming-line1]
exten = s,1,setCidName(Line1: . ${CALLERID})
exten = s,2,Goto(Incoming,s,1)
[incoming]
exten = s,1,normal incoming call stuff
then set a different context for each of your zap channels...
Julianjm.
On Mon, 21 Feb 2005 01:37:29 -0600, Anton Krall [EMAIL PROTECTED] wrote:
Guys... I see there is a callerid parameter on zapata.conf... what does that
cid modify? the callerid people see when you call them using any PSTN line?
Is there a way to send the SIP phone the incoming callerid frpm PSTN lines
asrecevied and append some string depending on the line it is coming from?
__
Anton Krall
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[Asterisk-Users] Problems with the FXS module in a TDMxxx card (no sound when receiving a call)
Hi all, I have a brand new TDMxxx card with 3 FXO modules and one FXS. It has replaced my old 3 X100P cards. The FXO part work as before, after some adjustments in the rxgain/txgain part. The problem I have is with the FXS module. I can place calls to SIP/IAX or PSTN destinations without any problems, but the sound received by the other part is much to strong and a little bit distorted. I have tried to modify the txgain up to txgain=-20, but still too strong. ..and this is not all. When I receive a call, from any type of source (IAX,SIP or PSTN), there is no sound (at both ends). No errors in the Asterisk console. I have tried to search through the archive but ... nothing related to this There is any way to enable something like 'iax2 debug' but for Zaptel channel? Any suggestions are welcome. Thank you and best regards, Dan ___ 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
SV: [Asterisk-Users] Snom phone hint exten question
I haven't used it in a while, but I had to put subscribecontext=sip for the phone's (in your case the snom) sip entry. This seems like it has been removed from the wiki. Has it changed or is this incorrect? Hi James, I have just found out that all you need to do is make the hint in the context where the phone registers. That means that all you need to do is put '690,hint,SIP/bt-karen' in your [sip] context, nothing else and it should work. Remember to take the power from the phone for a short while after you have configured this, otherwise it won't work. thorben ___ 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
[Asterisk-Users] ZAP libpri issue crashes PRI?
Hi, I have a problem that is biting at all my customer sites where they have PRIs taking heavy load. This happens both with the stable code stream and with the current CVS. What happens is that after some running, Asterisk starts reporting strange errors on the PRI, eventually calling the PRI down Starts with this sort of thing: Feb 21 09:39:23 DEBUG[18095]: Didn't get a frame from channel: Zap/68-1 Feb 21 09:39:23 DEBUG[18095]: Bridge stops bridging channels Zap/68-1 and Zap/32-1 Feb 21 09:39:23 DEBUG[22374]: Manager received command 'Command' Feb 21 09:39:25 DEBUG[18103]: Launching 'SetVar' Feb 21 09:39:25 DEBUG[18104]: Launching 'SetVar' Feb 21 09:39:33 DEBUG[22374]: Manager received command 'Command' Feb 21 09:39:43 DEBUG[22374]: Manager received command 'Command' Feb 21 09:39:51 DEBUG[18105]: Launching 'SetVar' Feb 21 09:39:53 DEBUG[22374]: Manager received command 'Command' Feb 21 09:39:56 DEBUG[18052]: Write returned -1 (Resource temporarily unavailable) on channel 31 Feb 21 09:40:03 DEBUG[22374]: Manager received command 'Command' Feb 21 09:40:06 DEBUG[18052]: Write returned -1 (Resource temporarily unavailable) on channel 31 Feb 21 09:40:06 DEBUG[17864]: Got RTCP report of 8 bytes Feb 21 09:40:06 DEBUG[17864]: Got RTCP report of 8 bytes Feb 21 09:40:13 DEBUG[22374]: Manager received command 'Command' Feb 21 09:40:23 DEBUG[22374]: Manager received command 'Command' Feb 21 09:40:33 DEBUG[22374]: Manager received command 'Command' Feb 21 09:40:34 DEBUG[18030]: Write returned -1 (Unknown error 500) on channel 45 Feb 21 09:40:34 DEBUG[18030]: Exception on 56, channel 45 Feb 21 09:40:34 DEBUG[18030]: Got event Alarm(4) on channel 45 (index 0) Feb 21 09:40:34 DEBUG[18099]: Exception on 44, channel 33 Feb 21 09:40:34 DEBUG[18099]: Got event Alarm(4) on channel 33 (index 0) Feb 21 09:40:34 DEBUG[18011]: Exception on 55, channel 44 Feb 21 09:40:34 DEBUG[18011]: Got event Alarm(4) on channel 44 (index 0) Feb 21 09:40:34 DEBUG[18063]: Exception on 57, channel 46 Feb 21 09:40:34 DEBUG[18063]: Got event Alarm(4) on channel 46 (index 0) ...etc... It complains about a bunch (though not all) channels. Then, shortly afterwards: Feb 21 09:40:34 DEBUG[22364]: Monitor doohicky got event Alarm on channel 34 Feb 21 09:40:34 WARNING[22364]: Detected alarm on channel 34: Red Alarm Feb 21 09:40:34 WARNING[22364]: Unable to disable echo cancellation on channel 34 Feb 21 09:40:34 DEBUG[22364]: Monitor doohicky got event Alarm on channel 35 Feb 21 09:40:34 WARNING[22364]: Detected alarm on channel 35: Red Alarm Feb 21 09:40:34 WARNING[22364]: Unable to disable echo cancellation on channel 35 Feb 21 09:40:34 DEBUG[22364]: Monitor doohicky got event Alarm on channel 36 Feb 21 09:40:34 WARNING[22364]: Detected alarm on channel 36: Red Alarm Feb 21 09:40:34 WARNING[22364]: Unable to disable echo cancellation on channel 36 ...etc... And the other PRI: Feb 21 09:41:01 DEBUG[22364]: Monitor doohicky got event Alarm on channel 1 Feb 21 09:41:01 WARNING[22364]: Detected alarm on channel 1: Red Alarm Feb 21 09:41:01 WARNING[22364]: Unable to disable echo cancellation on channel 1 Feb 21 09:41:01 DEBUG[22364]: Monitor doohicky got event Alarm on channel 5 Feb 21 09:41:01 WARNING[22364]: Detected alarm on channel 5: Red Alarm Feb 21 09:41:01 WARNING[22364]: Unable to disable echo cancellation on channel 5 Feb 21 09:41:01 DEBUG[22364]: Monitor doohicky got event Alarm on channel 6 Feb 21 09:41:01 WARNING[22364]: Detected alarm on channel 6: Red Alarm Feb 21 09:41:01 WARNING[22364]: Unable to disable echo cancellation on channel 6 Feb 21 09:41:01 DEBUG[22364]: Monitor doohicky got event Alarm on channel 7 Feb 21 09:41:01 WARNING[22364]: Detected alarm on channel 7: Red Alarm Feb 21 09:41:01 WARNING[22364]: Unable to disable echo cancellation on channel 7 Rapidly followed by the PRI coming back up: Feb 21 09:41:07 DEBUG[22364]: Monitor doohicky got event No more alarm on channel 34 Feb 21 09:41:07 NOTICE[22364]: Alarm cleared on channel 34 Feb 21 09:41:07 DEBUG[22364]: Monitor doohicky got event No more alarm on channel 35 Feb 21 09:41:07 NOTICE[22364]: Alarm cleared on channel 35 ..etc.. and: Feb 21 09:41:35 DEBUG[22364]: Monitor doohicky got event No more alarm on channel 1 Feb 21 09:41:35 NOTICE[22364]: Alarm cleared on channel 1 Feb 21 09:41:35 DEBUG[22364]: Monitor doohicky got event No more alarm on channel 5 Feb 21 09:41:35 NOTICE[22364]: Alarm cleared on channel 5 Feb 21 09:41:35 DEBUG[22364]: Monitor doohicky got event No more alarm on channel 6 Feb 21 09:41:35 NOTICE[22364]: Alarm cleared on channel 6 ..etc.. What I find in the one case is that the PRI keeps getting reported RED alarm and recovering, ad infinitum. In another case, I saw millions of lines logged in this vein: Feb 21 09:46:28 DEBUG[22364]: Monitor doohicky got event Event -1 on channel 12 across all the channels that alarmed and recovered. In either case whe whole system is now toast until it is restarted. At a particularly
RE: [Asterisk-Users] Snom phone hint exten question
I haven't used it in a while, but I had to put subscribecontext=sip for the phone's (in your case the snom) sip entry. This seems like it has been removed from the wiki. Has it changed or is this incorrect? Hi James, I have just found out that all you need to do is make the hint in the context where the phone registers. That means that all you need to do is put '690,hint,SIP/bt-karen' in your [sip] context, nothing else and it should work. Remember to take the power from the phone for a short while after you have configured this, otherwise it won't work. thorben Ok your example confused me a little. You put 690,hint,SIP/bt-karen From this section in my extensions from your example I should have exten = 690,hint,SIP/bt-karen exten = 691,hint,SIP/snom-james So set hint on the opposite extensions? [sip] exten = 690,hint,SIP/snom-james exten = 690,1,SetMusicOnHold(random) exten = 690,2,Dial(SIP/snom-james,30,Ttr) exten = 690,3,Voicemail2,u690 exten = 690,103,Voicemail2,b690 exten = 691,hint,SIP/bt-karen exten = 691,1,SetMusicOnHold(random) exten = 691,2,Dial(SIP/bt-karen,30,Ttr) exten = 691,3,Voicemail,u691 exten = 691,103,Voicemail,b691 ___ 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
[Asterisk-Users] CallingCard application AreskiCC RELEASE v1.1
Dear folks, I just made a release of the calling card application AreskiCC Please check it out : http://areski.net/areskicc-doc/ Reported bugs has been fixed. I advice to all users to make the update. Further informations on the release into the CHANGELOG file. Kinds regards, Areski ___ 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
SV: [Asterisk-Users] Snom phone hint exten question
Ok your example confused me a little. You put 690,hint,SIP/bt-karen From this section in my extensions from your example I should have exten = 690,hint,SIP/bt-karen exten = 691,hint,SIP/snom-james So set hint on the opposite extensions? [sip] exten = 690,hint,SIP/snom-james exten = 690,1,SetMusicOnHold(random) exten = 690,2,Dial(SIP/snom-james,30,Ttr) exten = 690,3,Voicemail2,u690 exten = 690,103,Voicemail2,b690 exten = 691,hint,SIP/bt-karen exten = 691,1,SetMusicOnHold(random) exten = 691,2,Dial(SIP/bt-karen,30,Ttr) exten = 691,3,Voicemail,u691 exten = 691,103,Voicemail,b691 Hi James I am sorry I made a typo. You need to set [sip] like this: [sip] exten = 690,hint,SIP/snom-james exten = 691,hint,SIP/bt-karen exten = 690,1,SetMusicOnHold(random) exten = 690,2,Dial(SIP/snom-james,30,Ttr) exten = 690,3,Voicemail2,u690 exten = 690,103,Voicemail2,b690 exten = 691,1,SetMusicOnHold(random) exten = 691,2,Dial(SIP/bt-karen,30,Ttr) exten = 691,3,Voicemail,u691 exten = 691,103,Voicemail,b691 That should work thorben ___ 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
[Asterisk-Users] Re: Re: quadbri and spandsp
-- If I put the parameter 'caller', when I execute the call 'sample.call', the application txfax realizes two calls. One to the fax machine and other to my own asterisk. My Asterisk detects that the incoming call is a fax and begins to save it with 'rxfax'. In another call, 'txfax' says: -- DIS nothing to send [0] DIS nothing to receive [0] -- and the fax machine does not receive anything. Because this happens? (In the 'zapata.conf' I have 'faxdetect=incoming') Thank you. Blas Steve wrote: You need to use the caller parameter. Something like: Channel:Zap/G1/ Application:txfax Data:/root/fax.tif|caller might work better. Regards, Steve ___ 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] Conecting to asterisk server through NAT usingIAX
I did change the port 4569. Also my router forwards those packets. If I start tcpdump port 4569 on my server I receive: 04:25:36.061292 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 24 04:25:39.154871 IP beu164.neoplus.adsl.tpnet.pl.4569 192.168.1.251.4569: UDP, length 24 04:25:39.155919 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 04:25:44.063009 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 04:25:46.063463 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 24 04:25:46.063952 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 04:25:49.119019 IP beu164.neoplus.adsl.tpnet.pl.4569 192.168.1.251.4569: UDP, length 24 04:25:49.120272 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 It means that client is trying to comunicate with asterisk server. But the client says that the server could not be contacted. On asterisk console with iax2 debuging enabled I receive Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 7ms SCall: 1 DCall: 0 [66.234.228.170:4569] USERNAME: nWv96gaD75 REFRESH : 60 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00012ms SCall: 00055 DCall: 1 [66.234.228.170:4569] AUTHMETHODS : 3 CHALLENGE : 164462354 USERNAME: nWv96gaD75 Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 00049ms SCall: 1 DCall: 00055 [66.234.228.170:4569] USERNAME: nWv96gaD75 REFRESH : 60 MD5 RESULT : 478939afef8fa0ec5b480cc939dedf6f Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00047ms SCall: 00055 DCall: 1 [66.234.228.170:4569] USERNAME: nWv96gaD75 DATE TIME : 173363009 REFRESH : 60 APPARENT ADDRES : IPV4 69.208.170.240:4569 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00047ms SCall: 1 DCall: 00055 [66.234.228.170:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00019ms SCall: 2 DCall: 13354 [83.28.32.164:4569] AUTHMETHODS : 1 USERNAME: tester Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00019ms SCall: 2 DCall: 13354 [83.28.32.164:4569] AUTHMETHODS : 1 USERNAME: tester Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: LAGRQ Timestamp: 10022ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Tx-Frame Retry[002] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00019ms SCall: 2 DCall: 13354 [83.28.32.164:4569] AUTHMETHODS : 1 USERNAME: tester Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: LAGRQ Timestamp: 10022ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569]
[Asterisk-Users] Problem with ISDN Dialin via CAPI
Title: Problem with ISDN Dialin via CAPI
Hello List,
because this is my first post to this list, i'd like to introduce myself.
My name is Thorsten Müller, 26 years old and live near Frankfurt/Main in germany.
Okay, now to the reason for this posting:
I just installed my first asterisk server (Debian 2.2 Kernel 2.4.18-686) with AVM ISDN Fritz PCI card (passive).
I followed the configuration on http://voip-info.org and the Voice-Over-IP telephony between two PC's with SJPhone works perfectly.
But connecting this machine to the ISDN line drives me crazy.
I installed the capi driver from the AVM website and installed chan_capi without bigger problems.
Followed these instructions: http://voip-info.org/tiki-index.php?page=Asterisk+How+to+connect+with+CAPI
After entering the asterisk console i tried
*CLI capi info
Contr1: 2 B channels total, 2 B channels free.
(looks good)
*CLI capi debug
CAPI Debugging Enabled
When i call my msn with a normal phone i see the following content:
=
-- CONNECT_IND ID=001 #0x0020 LEN=0047
Controller/PLCI/NCCI = 0x101
CIPValue = 0x10
CalledPartyNumber = c1523065
CallingPartyNumber = 21 816101806124
CalledPartySubaddress = default
CallingPartySubaddress = default
BC = 80 90 a3
LLC = default
HLC = 91 81
AdditionalInfo = default
Feb 21 11:32:30 NOTICE[384]: chan_capi.c:1932 capi_handle_msg: CONNECT_IND ID=001 #0x0020 LEN=0047
Controller/PLCI/NCCI = 0x101
CIPValue = 0x10
CalledPartyNumber = c1523065
CallingPartyNumber = 21 816101806124
CalledPartySubaddress = default
CallingPartySubaddress = default
BC = 80 90 a3
LLC = default
HLC = 91 81
AdditionalInfo = default
== CONNECT_IND (PLCI=0x101,DID=523065,CID=6101806124,CIP=0x10,CONTROLLER=0x1)
Feb 21 11:32:30 ERROR[384]: chan_capi.c:2051 capi_handle_msg: did not find device for msn = 523065
-- INFO_IND ID=001 #0x0021 LEN=0022
Controller/PLCI/NCCI = 0x101
InfoNumber = 0x70
InfoElement = c1523065
Feb 21 11:32:30 ERROR[384]: chan_capi.c:1198 find_pipe: unable to find a pipe for PLCI = 0x101 MN = 0x21
Feb 21 11:32:30 NOTICE[384]: chan_capi.c:1302 pipe_msg: INFO_IND ID=001 #0x0021
LEN=0022
Controller/PLCI/NCCI = 0x101
InfoNumber = 0x70
InfoElement = c1523065
-- INFO_IND ID=001 #0x0022 LEN=0016
Controller/PLCI/NCCI = 0x101
InfoNumber = 0x18
InfoElement = 89
Feb 21 11:32:30 ERROR[384]: chan_capi.c:1198 find_pipe: unable to find a pipe for PLCI = 0x101 MN = 0x22
Feb 21 11:32:30 NOTICE[384]: chan_capi.c:1302 pipe_msg: INFO_IND ID=001 #0x0022
LEN=0016
Controller/PLCI/NCCI = 0x101
InfoNumber = 0x18
InfoElement = 89
-- DISCONNECT_IND ID=001 #0x0023 LEN=0014
Controller/PLCI/NCCI = 0x101
Reason = 0x0
== DISCONNECT_IND PLCI=0x101 REASON=0
Feb 21 11:32:30 ERROR[384]: chan_capi.c:1198 find_pipe: unable to find a pipe for PLCI = 0x101 MN = 0x23
Afterwards i did a search with google (and google groups) about: did not find device for msn but i wasn't really succesfull.
Here are my entries of /etc/asterisk/capi.conf:
=
; CAPI config
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=523065
incomingmsn=523065
controller=1
softdtmf=1
accountcode=
context=demo
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2
Here are the entries of /etc/asterisk/extensions.conf
==
s,1,Dial,CAPI/523065:b${EXTEN}|30
s,1,Dial,CAPI/523065:${EXTEN}|30|r
Here's my /etc/isdn/capi.conf
=
# card file proto io irq mem cardnr options
#b1isa b1.t4 DSS1 0x150 7 - - P2P
b1pci b1.t4 DSS1 - - - -
c4 c4.bin DSS1 - - - -
c4 - DSS1 - - - -
c4 - DSS1 - - - - P2P
c4 - DSS1 - - - - P2P
#c2 c2.bin DSS1 - - - -
#c2 - DSS1 - - - -
#t1isa t1.t4 DSS1 0x340 9 - 0
#t1pci t1.t4 DSS1 - - - -
#fcpci - - - - - -
#fcclassic - - 0x150 10 - -
Sorry, for the long posting, but i want to add a little debug output.
Can someone please point me to the right direction.
Thanks a lot
Thorsten
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Re: [Asterisk-Users] Conecting to asterisk server through NAT usingIAX
This was set on linksys wrt54. I turned on the forwarding to asterisk server on port 4569. I believe that by default all outgoing packets pass through. Bart, Hallo Did you allow udp outgoing on 4569 as well.. i found udp bit different than tcp when comming to firewalls liaan - Original Message - From: Bartosz Wegrzyn - asterisk [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 21, 2005 12:29 PM Subject: Re: [Asterisk-Users] Conecting to asterisk server through NAT usingIAX I did change the port 4569. Also my router forwards those packets. If I start tcpdump port 4569 on my server I receive: 04:25:36.061292 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 24 04:25:39.154871 IP beu164.neoplus.adsl.tpnet.pl.4569 192.168.1.251.4569: UDP, length 24 04:25:39.155919 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 04:25:44.063009 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 04:25:46.063463 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 24 04:25:46.063952 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 04:25:49.119019 IP beu164.neoplus.adsl.tpnet.pl.4569 192.168.1.251.4569: UDP, length 24 04:25:49.120272 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 It means that client is trying to comunicate with asterisk server. But the client says that the server could not be contacted. On asterisk console with iax2 debuging enabled I receive Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 7ms SCall: 1 DCall: 0 [66.234.228.170:4569] USERNAME: nWv96gaD75 REFRESH : 60 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00012ms SCall: 00055 DCall: 1 [66.234.228.170:4569] AUTHMETHODS : 3 CHALLENGE : 164462354 USERNAME: nWv96gaD75 Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 00049ms SCall: 1 DCall: 00055 [66.234.228.170:4569] USERNAME: nWv96gaD75 REFRESH : 60 MD5 RESULT : 478939afef8fa0ec5b480cc939dedf6f Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00047ms SCall: 00055 DCall: 1 [66.234.228.170:4569] USERNAME: nWv96gaD75 DATE TIME : 173363009 REFRESH : 60 APPARENT ADDRES : IPV4 69.208.170.240:4569 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00047ms SCall: 1 DCall: 00055 [66.234.228.170:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00019ms SCall: 2 DCall: 13354 [83.28.32.164:4569] AUTHMETHODS : 1 USERNAME: tester Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00019ms SCall: 2 DCall: 13354 [83.28.32.164:4569] AUTHMETHODS : 1 USERNAME: tester Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: LAGRQ Timestamp: 10022ms SCall: 2 DCall: 13354 [83.28.32.164:4569]
Re: [Asterisk-Users] How many line appearance can Snom 200 handle?
You can only set up the 220 with an extended key pad. both phones, the 220 as well as the 200, support up to 7 SIP lines/registrations. Regards Nils Ohlmeier On Monday 21 February 2005 05:53, dkwok wrote: Snom 200 has be set up with extended key pad. The product literature also mention multiple sip registration. How many registration can it handle? It does not seem to appear in the user manual. David Kwok -- snom technology AGPascalstrasse 10bD-10581 Berlin Nils Ohlmeier mailto:[EMAIL PROTECTED] http://www.snom.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] Conecting to asterisk server through NAT usingIAX
well, it seems like the 2 are communicating correctley.. just went through all the logs what is the error that you recieve? - Original Message - From: Bartosz Wegrzyn - asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 21, 2005 12:52 PM Subject: Re: [Asterisk-Users] Conecting to asterisk server through NAT usingIAX This was set on linksys wrt54. I turned on the forwarding to asterisk server on port 4569. I believe that by default all outgoing packets pass through. Bart, Hallo Did you allow udp outgoing on 4569 as well.. i found udp bit different than tcp when comming to firewalls liaan - Original Message - From: Bartosz Wegrzyn - asterisk [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 21, 2005 12:29 PM Subject: Re: [Asterisk-Users] Conecting to asterisk server through NAT usingIAX I did change the port 4569. Also my router forwards those packets. If I start tcpdump port 4569 on my server I receive: 04:25:36.061292 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 24 04:25:39.154871 IP beu164.neoplus.adsl.tpnet.pl.4569 192.168.1.251.4569: UDP, length 24 04:25:39.155919 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 04:25:44.063009 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 04:25:46.063463 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 24 04:25:46.063952 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 04:25:49.119019 IP beu164.neoplus.adsl.tpnet.pl.4569 192.168.1.251.4569: UDP, length 24 04:25:49.120272 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 It means that client is trying to comunicate with asterisk server. But the client says that the server could not be contacted. On asterisk console with iax2 debuging enabled I receive Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 7ms SCall: 1 DCall: 0 [66.234.228.170:4569] USERNAME: nWv96gaD75 REFRESH : 60 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00012ms SCall: 00055 DCall: 1 [66.234.228.170:4569] AUTHMETHODS : 3 CHALLENGE : 164462354 USERNAME: nWv96gaD75 Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 00049ms SCall: 1 DCall: 00055 [66.234.228.170:4569] USERNAME: nWv96gaD75 REFRESH : 60 MD5 RESULT : 478939afef8fa0ec5b480cc939dedf6f Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00047ms SCall: 00055 DCall: 1 [66.234.228.170:4569] USERNAME: nWv96gaD75 DATE TIME : 173363009 REFRESH : 60 APPARENT ADDRES : IPV4 69.208.170.240:4569 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00047ms SCall: 1 DCall: 00055 [66.234.228.170:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00019ms SCall: 2 DCall: 13354 [83.28.32.164:4569] AUTHMETHODS : 1 USERNAME: tester Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00019ms SCall: 2 DCall: 13354 [83.28.32.164:4569] AUTHMETHODS : 1 USERNAME: tester Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
[Asterisk-Users] How to ECT (explicit call transfer) ?
Hey Guys
Im trying to find out how to transfer a call with ECT (explicit call
transfer) ?
Im currently transferring a call as following:
exten=2,1,Dial(capi/720:07,18)
exten = 2,2,Goto(2-${DIALSTATUS},1)
exten = 2-NOANSWER,1,Dial(capi/720:07979)
exten = 2-CHANUNAVAIL,1,Goto(1,1)
exten = 2-BUSY,1,Dial(capi/720:07979)
If I wanna transfer a call with ECT (call deflection), do I'll do that in
the extensions.conf file ?
Thx for the help
Matt
P.S: I've already looked on google, but could not find any help..
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[Asterisk-Users] LineJACK dial problem
hallo all,
i have a quicknet LineJACK card and it seems to work ok, the only problem
is, that when i use this in extentions.conf,
exten = _[1-9]., 1, Dial(IAX2/krath:[EMAIL PROTECTED]/${EXTEN},50,Ttr)
exten = _[1-9]., 2, Congestion
it dials only 2 digits, e.g when i dial 1234 it dials only 12, if i change
the exten to:
exten = _[1-9]XX.
it dials only 4 digits but not more? if i use 2 exten statemants, on for 4
digits and one for 3, it uses only the shortest?
could sombody give me an hint what the problem could be?
thanks for help,
alex
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[Asterisk-Users] Calls from IAX2 trunk start again when hung
Hi all, I'm having a weird problem. The setup is Asterisk A with a TDM400P/4xFXO, DSL line, and Asterisk B with TDM400P/4xFXO on another DSL line. Both boxes are behind their own NAT. Asterisk B forwards calls from it's four PSTN ports to Asterisk A over an IAX2 trunk, which works fine using the GSM codec. Asterisk A dials the SIP phones on it's local segment. The problem is that when the inbound PSTN call ends, the hangups are detected, but for some reason, Asterisk B starts a new call all over again, Asteriks A receives it, the SIP phones ring, but when one of them picks up there is a dialtone, busy tone, or silence. Is there anything I may be missing here? I can post .conf files, but I don't think it has anything to do with those. Calls on the local PSTN ports of Asterisk A work fine. This setup is in Spain, FYI. Regards, Mike ___ 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] Calls from IAX2 trunk start again when hung
I'm having a weird problem. The setup is Asterisk A with a TDM400P/4xFXO, DSL line, and Asterisk B with TDM400P/4xFXO on another DSL line. Both boxes are behind their own NAT. Asterisk B forwards calls from it's four PSTN ports to Asterisk A over an IAX2 trunk, which works fine using the GSM codec. Asterisk A dials the SIP phones on it's local segment. The problem is that when the inbound PSTN call ends, the hangups are detected, but for some reason, Asterisk B starts a new call all over again, Asteriks A receives it, the SIP phones ring, but when one of them picks up there is a dialtone, busy tone, or silence. Is there anything I may be missing here? I can post .conf files, but I don't think it has anything to do with those. Calls on the local PSTN ports of Asterisk A work fine. This setup is in Spain, FYI. Kind of sounds like an issue with detecting pstn line supervision events, but almost impossible to guess at root cause unless you provide something to look at. Might try some of the cli debug commands; 'zap debug', 'iax2 debug', etc. Look those over very closely and you're likely to spot the problem. If not, post the results. Include * version data as well. ___ 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] Calls from IAX2 trunk start again when hung
Rich Adamson wrote: I'm having a weird problem. The setup is Asterisk A with a TDM400P/4xFXO, DSL line, and Asterisk B with TDM400P/4xFXO on another DSL line. Both boxes are behind their own NAT. Asterisk B forwards calls from it's four PSTN ports to Asterisk A over an IAX2 trunk, which works fine using the GSM codec. Asterisk A dials the SIP phones on it's local segment. The problem is that when the inbound PSTN call ends, the hangups are detected, but for some reason, Asterisk B starts a new call all over again, Asteriks A receives it, the SIP phones ring, but when one of them picks up there is a dialtone, busy tone, or silence. Is there anything I may be missing here? I can post .conf files, but I don't think it has anything to do with those. Calls on the local PSTN ports of Asterisk A work fine. This setup is in Spain, FYI. Kind of sounds like an issue with detecting pstn line supervision events, but almost impossible to guess at root cause unless you provide something to look at. Might try some of the cli debug commands; 'zap debug', 'iax2 debug', etc. Look those over very closely and you're likely to spot the problem. If not, post the results. Include * version data as well. Hi Richard, Thanks for the pointers, I will try those debugs and will post the results. Best regards, Mike ___ 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] Snom phone hint exten question
I am going to now sit in a corner and go quietly insane while playing the banyo with no strings. Still doesn't work, I dialed in an outside line and picked up the receive on extension 691, yet the light on the snom phone did not come on. I dialed out of extension 691 to an outside line, yet still the light did not come on. Snom190 has firmware 3.56m the button is set to Destination 691 Hi James, I am using the latest CSV-HEAD of *, I do not think it works with * stable. Thorben ___ 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] Problem with ISDN Dialin via CAPI
On Mon, Feb 21, 2005 at 11:36:45AM +0100, Müller, Thorsten wrote: Hello List, because this is my first post to this list, i'd like to introduce myself. My name is Thorsten Müller, 26 years old and live near Frankfurt/Main in germany. Okay, now to the reason for this posting: I just installed my first asterisk server (Debian 2.2 Kernel 2.4.18-686) with AVM ISDN Fritz PCI card (passive). I followed the configuration on http://voip-info.org and the Voice-Over-IP telephony between two PC's with SJPhone works perfectly. But connecting this machine to the ISDN line drives me crazy. I installed the capi driver from the AVM website and installed chan_capi without bigger problems. Followed these instructions: http://voip-info.org/tiki-index.php?page=Asterisk+How+to+connect+with+CAPI After entering the asterisk console i tried *CLI capi info Contr1: 2 B channels total, 2 B channels free. (looks good) *CLI capi debug CAPI Debugging Enabled When i call my msn with a normal phone i see the following content: = -- CONNECT_IND ID=001 #0x0020 LEN=0047 Controller/PLCI/NCCI= 0x101 CIPValue= 0x10 CalledPartyNumber = c1523065 CallingPartyNumber = 21 816101806124 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo = default Feb 21 11:32:30 NOTICE[384]: chan_capi.c:1932 capi_handle_msg: CONNECT_IND ID=001 #0x0020 LEN=0047 Controller/PLCI/NCCI= 0x101 CIPValue= 0x10 CalledPartyNumber = c1523065 CallingPartyNumber = 21 816101806124 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo = default == CONNECT_IND (PLCI=0x101,DID=523065,CID=6101806124,CIP=0x10,CONTROLLER=0x1) Feb 21 11:32:30 ERROR[384]: chan_capi.c:2051 capi_handle_msg: did not find device for msn = 523065 -- INFO_IND ID=001 #0x0021 LEN=0022 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x70 InfoElement = c1523065 I think the outgoingmsn is missing / the one you are looking for Here are my entries of /etc/asterisk/capi.conf: = ; CAPI config ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=523065 incomingmsn=523065 outgoingmsn=HERE_WE_NEED_A_VALID_MSN controller=1 softdtmf=1 accountcode= context=demo ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 -- Tho/\/\as ___ 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
[Asterisk-Users] compiling cvs-head today?
Anyone having problems compiling the current cvs head this morning? New cvs checkout on RH9, followed by appropriate make clean and make install. System was running cvs head from Nov 23 with TDM card, PRI, SIP phones on local wire, and IAX. Appears zaptel and libpri compiled correctly, however the first attempt in the asterisk src directory yielded: gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-HEAD-02/21/05-06:22:04\ -DASTERISK_VERSION_NUM=99 -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN-c -o say.o say.c say.c: In function `ast_say_number_full_tw': say.c:2128: internal error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See URL:http://bugzilla.redhat.com/bugzilla/ for instructions. The bug is not reproduceable, so it is likely a hardware or OS problem make: *** [say.o] Error 1 Then, with another simple make clean and make install, yielded: gcc -c -Wall -pipe -g3 -O '-D__RCSID(x)=' '-D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. readline.c -o readline.o_a /usr/bin/ar cru libedit.a editline.o_a np/strlcat.o_a np/strlcpy.o_a np/fgetln.o_a np/vis.o_a np/unvis.o_a history.o_a tokenizer.o_a readline.o_a ranlib libedit.a make[1]: Leaving directory `/usr/src/asterisk/editline' make[1]: Entering directory `/usr/src/asterisk/db1-ast' gcc -Wall -c -D__DBINTERFACE_PRIVATE -O2 -I. -Iinclude -Ihash -o hash.o hash/hash.c hash/hash.c: In function `__hash_open': hash/hash.c:243: internal error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See URL:http://bugzilla.redhat.com/bugzilla/ for instructions. make[1]: *** [hash.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/db1-ast' make: *** [db1-ast/libdb1.a] Error 2 Not sure as yet what might have changed other then the asterisk box has been stable since Nov 23 (other then the occasional TDM card lockup that requires a restart of the drivers). This might not be an asterisk problem, just not sure as yet. Thoughts? ___ 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
[Asterisk-Users] bug? Unterminated comment detected beginning on line 0
Hi, Using latest cvs. A comment-line begins with semicolon ; However - if the line contains ;-- or like this ; -- blabla bla -- You get this error and * stops reading that file: Feb 21 13:47:12 WARNING[17393]: config.c:664 config_text_file_load: Unterminated comment detected beginning on line 0 Shouldn't Asterisk skip any line beginning with a semicolon? Or should a comment now be terminated too? /Stig - N Y H E T E R! - IP-telefoni, spara tusenlappar om året! - Rikstäckande ADSL 0,25-24Mbit - Internetaccess (Modem/ISDN64+128 via Ymex - utan abonnemangskostnad! - Eposttjänster, även UUCP, Uppringd SMTP, MX fallback, DomänPOP - Surf24 - en billig bredbandstjänst från Ymex för kunder i Härnösand/Älandsbro. - Get your emailed Web-forms into a database of your choice!!! Checkout DBFORM V1.0, see details at http://www.ymex.se - Ymex AB| Alvägen 7 | 871 52 Härnösand | Sweden | http://www.ymex.se/ ___ 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
[Asterisk-Users] Monitoring calls through a transfer
We have the following scenario: Incoming call to a queue, Agent A answers. Agent A determines after about 20 seconds that agent B needs to deal with this call. A puts call on hold, calls and speaks to B, and then transfers the call to B. B speaks to the incomming caller for 5 minutes. That's all fine. However, the CDR records the call as incomming to agent A for 5 minutes, and the agent monitoring recording is also determined as belonging to A. Trouble is that we need to find all calls that B received (both directly and through a transfer) and look at them. How can we do this ? CVS Head 17/02/2005. Julian. ___ 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
[Asterisk-Users] Any luck with attended transfer and ATA186?
Hi, Using latest cvs. I (as many otheres it seems) can't get Attended transfer to work with Cisco ATA186 (using SIP) Has anyone else had any luck? Same with 3-part calling, if one drops off, all are disconnected... /Stig ___ 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
[Asterisk-Users] * Call Monitoring
I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? -Daniel ___ 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
[Asterisk-Users] Re: * Mobile Phone Mobile Network
I've used a Nokia 32 unattended (remote) for the past year or so. David Uzzell [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] | Ok I have a question. Seen it come and go around the mailling list for a | while but never really seen an answer that seems to sort it out. | | What is needed is some interface from * Mobile Phone Mobile Network | Service. | | At this point all the providers in AUS that I have found are charging a | Premium Rate for Land Line Mobile Network services. | | What I would like to do is be able to purchase a low rate Mobile SIM | that I can chuck into a Mobile Phone and have it setup so that I route | the Mobile calls through it. | | Rembering that most if not all mobile phones can be accessed via RS232 | interface. | | Anyone done this or seen it done or know how to do it using * and whatever? | | Cheers | David | ___ | 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 | ___ 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] Snom phone hint exten question
I am going to now sit in a corner and go quietly insane while playing the banyo with no strings. Still doesn't work, I dialed in an outside line and picked up the receive on extension 691, yet the light on the snom phone did not come on. I dialed out of extension 691 to an outside line, yet still the light did not come on. Snom190 has firmware 3.56m the button is set to Destination 691 Hi James, I am using the latest CSV-HEAD of *, I do not think it works with * stable. Thorben Just downloaded the latest cvs 21/2/05 and compiled and installed it. Still nothing, the led's work on the snom but naybe its just buggered, *sigh* James ___ 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] Conference between 2 lines
Is there a way to make a join conference between 2 lines? like when you have 2 incoming calls and you merge them together with you? how can you do this on * if its possible? Transfert them both to a conference room, then join that conference. At least, that's how I would do it. ___ 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] Conecting to asterisk server through NAT usingIAX
Send us your DIAX configuration. Denis. Em Seg 21 Fev 2005 07:29, Bartosz Wegrzyn - asterisk escreveu: I did change the port 4569. Also my router forwards those packets. If I start tcpdump port 4569 on my server I receive: 04:25:36.061292 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 24 04:25:39.154871 IP beu164.neoplus.adsl.tpnet.pl.4569 192.168.1.251.4569: UDP, length 24 04:25:39.155919 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 04:25:44.063009 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 04:25:46.063463 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 24 04:25:46.063952 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 04:25:49.119019 IP beu164.neoplus.adsl.tpnet.pl.4569 192.168.1.251.4569: UDP, length 24 04:25:49.120272 IP 192.168.1.253.4569 beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 It means that client is trying to comunicate with asterisk server. But the client says that the server could not be contacted. On asterisk console with iax2 debuging enabled I receive Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 7ms SCall: 1 DCall: 0 [66.234.228.170:4569] USERNAME: nWv96gaD75 REFRESH : 60 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00012ms SCall: 00055 DCall: 1 [66.234.228.170:4569] AUTHMETHODS : 3 CHALLENGE : 164462354 USERNAME: nWv96gaD75 Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 00049ms SCall: 1 DCall: 00055 [66.234.228.170:4569] USERNAME: nWv96gaD75 REFRESH : 60 MD5 RESULT : 478939afef8fa0ec5b480cc939dedf6f Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00047ms SCall: 00055 DCall: 1 [66.234.228.170:4569] USERNAME: nWv96gaD75 DATE TIME : 173363009 REFRESH : 60 APPARENT ADDRES : IPV4 69.208.170.240:4569 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00047ms SCall: 1 DCall: 00055 [66.234.228.170:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00019ms SCall: 2 DCall: 13354 [83.28.32.164:4569] AUTHMETHODS : 1 USERNAME: tester Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00019ms SCall: 2 DCall: 13354 [83.28.32.164:4569] AUTHMETHODS : 1 USERNAME: tester Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: LAGRQ Timestamp: 10022ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Tx-Frame Retry[002] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00019ms SCall: 2 DCall: 13354 [83.28.32.164:4569] AUTHMETHODS : 1 USERNAME: tester Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: LAGRQ Timestamp: 10022ms SCall: 2 DCall: 13354 [83.28.32.164:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] USERNAME: tester REFRESH : 300 Tx-Frame Retry[-01] --
Re: [Asterisk-Users] MOH clicks
On Mon, 21 Feb 2005 02:42:41 -0600, Anton Krall [EMAIL PROTECTED] wrote: Guys.. Ive noticed that I have 2 mpg123 processes running, is that ok? also... can you make MOH random? Yes, this if fine. Please read the archives. Use google. Use the wiki. Again, on the random, read the samples, use the docs, check the wiki. Come on man, this info is readily available. Also, I dont know if there is a problem with my config but when listening to MOH, every 3 or so second I get a click sound which notices because music gets a hickup every 3 or so seconds... is this ok? Check the version of mpg123 you are using. There are some specifics in the wiki on which version works. Most of us would recommend version Version 0.59r (1999/Jun/15) Otherwise, watch your CLI when in it hiccups. Could be something else going on. We don't mind helping, but it does get old answering very well documented configs/problems. If you are going to do much with Asterisk, you will have to spend a lot of time on the wiki. You won't always get spoon fed. -Chuji ___ 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] * Call Monitoring
Daniel Corbe wrote: I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? -Daniel ___ 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 http://www.voip-info.org/wiki-Example+Argus+Config ___ 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] A bit of a survey: What do do if you need more than 4 C.O. lines
I corresponded with Voicetronix around Christmas last year. Jim, there is a dealer in Ottawa although I got better answers from emails to Aus. There are two things that they don't do that the Zap cards do: Distinctive Ring Detection and fax detection. They went out of their way to say they were customer driven and features get in because customers ask. The gentleman made a claim of effort to get fax detection to work which sounded like it was a no-brainer in their code. If it is easy as claimed, I would expect to see it appear just because I enquired. I am particularly interested in the Dist Ring Detection however for they make cheap DID's for low volume like home offices, dedicated voicemail numbers, etc. David Cook From: Jim Van Meggelen [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan4C.O. lines To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=windows-1250 I haven't followed this thread closely but have you looked into the Voicetronix OpenSwitch cards? http://www.voicetronix.com.au/hda.htm I've looked at them, but never heard much about them. Is anyone using them? Can anyone give a comparison vs. the TDM400? I'm using a Voicetronix OpenLine4, and it works well under asterisk. Initially I had some echo problems, but Voicetronix support is excellent and solved them (I've just updated the wiki with the bal# values they gave me). I can't compare it to the TDM400, not having used one, but you can use multiple Voicetronix OpenSwitch 6 and 12 cards in one system without the interrupt problem of the TDM400. That sounds like the ticket, then. ___ 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] * Call Monitoring
On Mon, 21 Feb 2005 08:00:40 -0500, Daniel Corbe [EMAIL PROTECTED] wrote: I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? There aren't any specific tools that do exactly what you want afaik. It wouldn't take much to taylor a few things yourself though. As for the PRI processing calls. You could always drop a call file in from the cron every 10 minutes that makes a call out and back in. Then you you can run a script that looks over your CDR to verify that the call was received. Have it call a specific context or application to look for. As for calls failing this could be a challange. What do you consider failing? You could use something like my-swatch to tail the log file looking for certain patterns. PRI alarms would be an obvious. Might take you a day or so to get these things going, but it would be well worth your time and piece of mind. -Chuji ___ 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] IAX2: Connection rejected
carrier via a PRI, they will dictate what the DID looks like. Some will be the last 4 digits, others will be all 10. (assuming US). They do this, because it would be to difficult to maintain your extension mapping on their side. You purchase a DID. When a call comes in it says, This is the number they were calling, you do your own matching to whatever extension you want. Now, what about the folks who are trying to call other countries, and potentially be called by other DIDs themselves? I'm assuming this sort of thing is very likely. Did you set a username? On some weired reason that is needed in 1.0.5 for IAX to work. Wessel ___ 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
[Asterisk-Users] [SOLVED] Problem with ISDN Dialin via CAPI
Title: [SOLVED] Problem with ISDN Dialin via CAPI Hi, i was able to solve my problem. During my playing around with * and capi i changed several options in config files. I did this while my * was running. To test if my changes where successful i entered reload on * console. This didn't help. But after i stopped asterisk and startet it again, everything worked perfect. So it seems that doing a reload while asterisk is running doesn't reload all settings. Thanks for your help Thorsten -Ursprngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Gesendet: Montag, 21. Februar 2005 13:43 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] Problem with ISDN Dialin via CAPI On Mon, Feb 21, 2005 at 11:36:45AM +0100, Mller, Thorsten wrote: Hello List, because this is my first post to this list, i'd like to introduce myself. My name is Thorsten Mller, 26 years old and live near Frankfurt/Main in germany. Okay, now to the reason for this posting: I just installed my first asterisk server (Debian 2.2 Kernel 2.4.18-686) with AVM ISDN Fritz PCI card (passive). I followed the configuration on http://voip-info.org and the Voice-Over-IP telephony between two PC's with SJPhone works perfectly. But connecting this machine to the ISDN line drives me crazy. I installed the capi driver from the AVM website and installed chan_capi without bigger problems. Followed these instructions: http://voip-info.org/tiki-index.php?page=Asterisk+How+to+connect+with+CAPI After entering the asterisk console i tried *CLI capi info Contr1: 2 B channels total, 2 B channels free. (looks good) *CLI capi debug CAPI Debugging Enabled When i call my msn with a normal phone i see the following content: = -- CONNECT_IND ID=001 #0x0020 LEN=0047 Controller/PLCI/NCCI = 0x101 CIPValue = 0x10 CalledPartyNumber = c1523065 CallingPartyNumber = 21 816101806124 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo = default Feb 21 11:32:30 NOTICE[384]: chan_capi.c:1932 capi_handle_msg: CONNECT_IND ID=001 #0x0020 LEN=0047 Controller/PLCI/NCCI = 0x101 CIPValue = 0x10 CalledPartyNumber = c1523065 CallingPartyNumber = 21 816101806124 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo = default == CONNECT_IND (PLCI=0x101,DID=523065,CID=6101806124,CIP=0x10,CONTROLLER=0x1) Feb 21 11:32:30 ERROR[384]: chan_capi.c:2051 capi_handle_msg: did not find device for msn = 523065 -- INFO_IND ID=001 #0x0021 LEN=0022 Controller/PLCI/NCCI = 0x101 InfoNumber = 0x70 InfoElement = c1523065 I think the outgoingmsn is missing / the one you are looking for Here are my entries of /etc/asterisk/capi.conf: = ; CAPI config ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=523065 incomingmsn=523065 outgoingmsn=HERE_WE_NEED_A_VALID_MSN controller=1 softdtmf=1 accountcode= context=demo ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 -- Tho/\/\as ___ 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 ___ 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
[Asterisk-Users] Problems with the FXS module in a TDMxxx card (no sound when receiving a call)
Hi All, As my previous mail was not posted on the list for more than 10 hours now, I'll try to resend it. Thank you, Dan - Original Message - From: Dan [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 21, 2005 11:12 AM Subject: Problems with the FXS module in a TDMxxx card (no sound when receiving a call) Hi all, I have a brand new TDMxxx card with 3 FXO modules and one FXS. It has replaced my old 3 X100P cards. The FXO part work as before, after some adjustments in the rxgain/txgain part. The problem I have is with the FXS module. I can place calls to SIP/IAX or PSTN destinations without any problems, but the sound received by the other part is much to strong and a little bit distorted. I have tried to modify the txgain up to txgain=-20, but still too strong. ..and this is not all. When I receive a call, from any type of source (IAX,SIP or PSTN), there is no sound (at both ends). No errors in the Asterisk console. I have tried to search through the archive but ... nothing related to this There is any way to enable something like 'iax2 debug' but for Zaptel channel? Any suggestions are welcome. Thank you and best regards, Dan ___ 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
[Asterisk-Users] Disable musiconhold
I'm trying to disable musiconhold, simply because I don't need it. But then chan_iax2 is complaining : undefined symbol: ast_moh_stop. Is there a way to completely disable moh (maybe a compilation option) ? Thank you. Yves ___ 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
[Asterisk-Users] Illegal instruction on startup
Hello, I have done some browsing through the wiki and on Google and havent been able to find anything that looks like what is happening to me. When I start Asterisk by typing asterisk vvvc, I get Illegal instruction and nothing else. Nothing before and nothing after. This is a Via Cyrix III 667MHz CPU with 192MB RAM running on Slackware 10.1 (Kernel 2.4.29) as a fresh install. I downloaded Asterisk, compiled mpeg123 and installed it, then compiled and installed Asterisk, then installed the sample data. I tried to start it up, and got the above error. Any pointers on where to look would be great. Thanks. -- Tommy Vielkanowitz Shared Resources of NC, LLC ___ 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] Extra sounds (Weather)
Hi
This is my script for my local forecast for SE England. I have had
problems getting festival to work integrated so I have cron run this
script every 3 hours and use Playback to play it in Asterisk:
Script
--
#!/bin/sh
cd /var/lib/asterisk/sounds
curl
http://www.bbc.co.uk/weather/ukweather/printables/print_regional_outloo
k.shtml?pmslondon 2/dev/null \
| (sed -n '/print area open/,/print area close/ { s/.*//;s/deg
C/Celsius/;s/deg F/Fahrenheit/;s/ deg$/ /;s/^C /Celsius
/;s/^F)/Fahrenheit)/;p }' date +'B B C forecast, %A %e %B at %l %p')
\
| /usr/local/bin/text2wave -f 8000 - -o wx.tmp.wav
sox wx.tmp.wav -r 8000 -c 1 wx.tmp.gsm
mv wx.tmp.gsm wx.gsm;rm -rf wx.tmp.wav
Extensions.conf
---
;Weather forecast for SE England 0_0 WX (199)
exten = 199,1,Answer
exten = 199,2,Playback(wx)
exten = 199,3,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Liaan vd
Merwe
Sent: 16 February 2005 11:57
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Extra sounds (Weather)
Hi Trevor
This i know
I just send you a other script doing the same task this will give you a
guideline to make you own
- Original Message -
From: Trevor G. Hammonds [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Wednesday, February 16, 2005 1:50 PM
Subject: RE: [Asterisk-Users] Extra sounds (Weather)
Liaan vd Merwe wrote on Wednesday, 16 February 2005
2:53 AM:
This is the example script (extracted from that
link) you will need
to find a weather page for your region an then
change the urls and
grep statements chow L
Once again, this is NOT the script mentioned at Eric
Wieling's former
site,
http://www.fnords.org/~eric/asterisk/, referenced it
the message in the
archives at
http://lists.digium.com/pipermail/asterisk-users/2003-November/025983.ht
ml.
Sincerely,
Trevor Hammonds
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RE: [Asterisk-Users] can't enable trunking :(
Sorry if this subject has been covered, but, my boss claims to have a working IAX2 trunking set-up without a timing source on one side of the connection. In all the posts and documentation regarding this subject, this appears to be impossible. My questions are: 1) could this be true? 2) if it is - is there a load limit where it breaks down? In summary, will IAX2 trunking work without some kind of timing source on both sides? Thanks in advance and sorry if this is a rehash. --Doug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Eric Wieling Sent: Thursday, February 17, 2005 10:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] can't enable trunking :( Muhammad Muzzamil Luqman wrote: I have successfully installed and configured the asterisk, the incoming and the outgoing calls are working fine, its a tremendous solution :) Now i want to enable trunking between two asterisk boxes, in the iax.conf i have put: [karachi] ... ... ... trunk=yes ... ... ... everything seems to work fine but when i load asterisk it says: -- Feb 17 10:59:14 WARNING[18726]: chan_iax2.c:7536 build_user: Unable to support trunking on user 'karachi' without zaptel timing Feb 17 10:59:14 WARNING[18726]: chan_iax2.c:7345 build_peer: Unable to support trunking on peer 'karachi' without zaptel timing -- I tried to install the ztdummy and i succeeded on one of the box but for the other i am having problems :( If you can't install Zaptel (a real driver, ztdummy, zaprtc, etc) then you can't use trunking. Remember trunking is only really useful when you have 3 or more calls at the same time between the same two Asterisk systems. Trunking with only one call actually uses MORE bandwidth. ___ 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 ___ 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] * Call Monitoring
Okay
here's a quick and dirty little perl script to monitor the PRI Status
and mimic nagios plugin output.
-Daniel
On Mon, 21 Feb 2005 07:50:45 -0600, Brian Roy [EMAIL PROTECTED] wrote:
On Mon, 21 Feb 2005 08:00:40 -0500, Daniel Corbe
[EMAIL PROTECTED] wrote:
I need to make sure the PRIs connected to my box stay up and I need to
make sure calls are not failing for any reason. Are there any *
monitoring packages like this?
There aren't any specific tools that do exactly what you want afaik.
It wouldn't take much to taylor a few things yourself though.
As for the PRI processing calls. You could always drop a call file in
from the cron every 10 minutes that makes a call out and back in. Then
you you can run a script that looks over your CDR to verify that the
call was received. Have it call a specific context or application to
look for.
As for calls failing this could be a challange. What do you consider
failing? You could use something like my-swatch to tail the log file
looking for certain patterns. PRI alarms would be an obvious.
Might take you a day or so to get these things going, but it would be
well worth your time and piece of mind.
-Chuji
#!/usr/bin/perl
###
# Michael Jastremski
# Monitor Asterisk PBX via Manager Interface
# http://megaglobal.net/docs/
###
# Based upon:
#
# TACI - Trivial Asterisk Call Interface v.02
# Last update 3/30/2004
# Tony Wasson [EMAIL PROTECTED]
#
#
# Modified by Daniel Corbe to monitor PRI spans
# [EMAIL PROTECTED]
#
# -Daniel
#
$ENV{'PATH'}='';
$ENV{'BASH_ENV'}='';
$ENV{'ENV'}='';
$| = 1;
use Net::Telnet ();
use File::Basename;
use lib /usr/local/nagios/libexec;
use utils qw(%ERRORS);
my $mgr_user = nagios;
my $mgr_secret = XyXyXyXyXy;
my $failed = 0;
my $reason = undef;
my $server_ip = 127.0.0.1;
my $prispan = $ARGV[0];
$tn = new Net::Telnet (Port = 5038,
Prompt = '/.*[\$%#] $/',
Output_record_separator = '',
Errmode= 'return'
);
$tn-open($server_ip);
$tn-waitfor('/0\n$/');
$tn-print(Action: Login\nUsername: $mgr_user\nSecret: $mgr_secret\n\n);
unless($tn-waitfor('/Authentication accept*/'))
{
$failed = 1;
$reason = Failed Connect;
}
else
{
$tn-print(Action: Command\n);
$tn-print(Command: pri show span $prispan\n\n);
#Response: Follows
#Primary D-channel: 24
#Status: Provisioned, Up, Active
unless($tn-waitfor('/Response: Follows\nPrimary D-channel: (.*)?\nStatus:
Provisioned, Up, Active/'))
{
$failed = 1;
$reason = PRI Span # . $prispan . is down;
}
else
{
$tn-print(Action: Logoff\n\n);
}
}
print PRI Span #$prispan is up\n unless $failed;
print $reason\n if $failed;
exit $ERRORS{'CRITICAL'} if $failed;
exit $ERRORS{'OK'};
exit 0;
__END__
TODO:
-- Maybe check other variables?
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[Asterisk-Users] setting caller id number and using sip type=peer for incomming calles.
Just to bug you all (feel free to rant at me), a client wants to set his caller*ID number for outbound calls though us to PSTN. the client is using SIP to us, he can set the caller*ID name fine. if he sets his caller*ID number to anything other than his account number (8440101), the call is dropped into the default context (and then hung up by our dial plan). To get around this i updated CVS HEAD and changed the sip entity from type=user to type=peer (yes peer!) (type=friend works too but im making a point) the client now must register to set his outbound caller*ID Number. it works because when a call comes in asterisk checks its list of registered users connection info and matches against a peer entity. this seems to be working but it hardly seems correct, i mean using a peer for inbound calls when the docs all say it is for outbound calls. im not up on the sip protocol but wouldnt it be better if, when receiving an unknown connection (ie when caller*ID number is set to a pstn number) it first sends an authentication request to the client, on return it checks that username/secret against its list of users. if it still doesnt find it then drop it into the guest account. iv posted a bug with a bit more detail but it was closed as a configuration issue (which i suppose it is...) http://bugs.digium.com/bug_view_page.php?bug_id=0003621 Morgan Gilroy, Telappliant Support ___ 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] Monitoring calls through a transfer
Asterisk wrote: We have the following scenario: Incoming call to a queue, Agent A answers. Agent A determines after about 20 seconds that agent B needs to deal with this call. A puts call on hold, calls and speaks to B, and then transfers the call to B. B speaks to the incomming caller for 5 minutes. That's all fine. However, the CDR records the call as incomming to agent A for 5 minutes, and the agent monitoring recording is also determined as belonging to A. Trouble is that we need to find all calls that B received (both directly and through a transfer) and look at them. How can we do this ? How are you performing the transfer? Have you tried the following? show application ResetCDR -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Amateur - Problema when installing
On 21 Feb 2005, at 01:21, Lyle Giese wrote: Hmmm, maybe you need to re-read the instructions? You missed a major step. make clean; make install *is* the recommended way to compile and install. See http://www.asterisk.org/index.php?menu=download. As for your error - I don't think there really is one. It looks like libpri successfully installed. I get the same .depend: file missing errors and they haven't ever caused me problems. Hope this helps, Phil. Try doing a make before make install. make;make install Lyle - Original Message - From: Paulo - Ibest [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, February 18, 2005 6:28 AM Subject: [Asterisk-Users] Amateur - Problema when installing Friends, I'm in trouble, I tried to install de Asterisk, based on the site manual, into a RedHat 9.0, I followed every step, and it doesn't work. When I does the libpri make install, the message is: quote: [EMAIL PROTECTED] zaptel]# cd .. [EMAIL PROTECTED] src]# cd libpri/ [EMAIL PROTECTED] libpri]# make clean; make install ___ 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] Monitoring calls through a transfer
Andrew Thompson wrote: Asterisk wrote: We have the following scenario: Incoming call to a queue, Agent A answers. Agent A determines after about 20 seconds that agent B needs to deal with this call. A puts call on hold, calls and speaks to B, and then transfers the call to B. B speaks to the incomming caller for 5 minutes. That's all fine. However, the CDR records the call as incomming to agent A for 5 minutes, and the agent monitoring recording is also determined as belonging to A. Trouble is that we need to find all calls that B received (both directly and through a transfer) and look at them. How can we do this ? How are you performing the transfer? The agents are using the transfer button (SIP, using Cisco 7940) Have you tried the following? show application ResetCDR No, but I was looking at ForkCDR :) However, I am at a loss at how to intercept the transfer and forkcdr ... Many thanks for any suggestions Julian ___ 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] A bit of a survey: What do do if you need morethan 4 C.O. lines
On Mon, February 21, 2005 8:47 am, David Cook said: There are two things that they don't do that the Zap cards do: I've received nothing but positive and rapid support to any issues I've had with my Voicetronix card. We should make the distinction between the low-level VPB device driver (analgous to the Zaptel drivers) and the VPB channel in asterisk. The former seems to be 100% Voicetronix's responsibility while the later seems to be somehow shared between them and Digium/Asterisk. That said... I bought one of the Voicetronix cards to see if their FXO interface could do a better job handling echo that the TDM400 which I was having trouble with at one site. The Voicetronix FXO interface didn't have any problem with echo. No tweaking was required either; it just worked out of the box. Yay! I have/had a couple issues: - I had to patch their VPB driver to get it to work with udev on my Fedora Core3 machine and get the /dev devices to appear. I've sent the patch to Voicetronix. Ben indicated he'd look at it when he got to the next rewrite of the driver. - I had to adjust the hard-coded timer for the CallerID so it would catch it here on my US phone service. - I'm having no CallerID detected on one line but I've not tracked that down yet. The FXO interface works as far as I can tell. I also wanted to see about their FXS implementation and was surpried to find it very lacking compared to the Zaptel feature set. I found the VPB channel in Asterisk didn't support CallerID delivery to stations, *XX service codes, and ambiguous dial plans with timeouts (this had been fixed). I've also had problems with the ringback not being sent to calling stations (this too appears to be fixed) and crosstalk between interfaces on the card (still looking into this one). As a station interface, I don't see much reason to use Voicetronix. I'd like to reiterate that these guys have been absolutely nothing but helpful, responsive, curteous, etc. Ben has handled (or is handling) all my issues and he's been great. He's taught me much along the way. These guys deserve kudos for their effort to support *. Attenention needs to be given, however, to the VPB station channel in * if this card is to be a viable alternative to the feature-rich Zaptel station channels. Paul -- Paul A. DugasDugas Enterprises, LLC [EMAIL PROTECTED]1711 Indian Ridge Drive p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA ___ 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] SIP echo on LAN
Hi There, Thanks for your reply. Where can I read up on doing this or maybe you could point me in the right direction. I don't believe that I have recording enabled. | Julian Wrote: | | Check your soundcard controls... maybe it's recording what you hear | or PCM, thus sending it again to the other party. | | Julianjm. | | Nic le Roux wrote: Good Morning, I have a weird situation, I'm testing with Xlite as SIP phone (is it any good ) and dialing an extension (also Xlite on same LAN) and I'm getting a real bad echo on the dialer's side and a not so bad one on the receivers side. Has anyone had something like this ? Aparently one should only get echo when you break out onto a telco network ___ 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
[Asterisk-Users] Polycom Phone Calling Party ID
I am trying to get the name and number to show up for an incoming calls on my Polycom IP 500. Right now only the name shows up, but in the call list both name and number show up. Any help on what to change in the config file would be greatly appreciated. Thanks Mark ___ 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
[Asterisk-Users] Anyone using SuperMicro SuperServer 6014P-8R?
Hi, Is anyone here using the SuperMicro SuperServer 6014P-8R with Asterisk? I'm especially interested if you've used it with a TE405P or TE410P. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Polycom Phone Calling Party ID
Mark Floyd wrote: I am trying to get the name and number to show up for an incoming calls on my Polycom IP 500. Right now only the name shows up, but in the call list both name and number show up. Any help on what to change in the config file would be greatly appreciated. Watch the display. Once you answer the phone, the number should show up right below the name. Nick ___ 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
[Asterisk-Users] Adit 600 MGCP configuration
I've finally got my Adit 600 and are configuring it right now. But I have to say, there aren't much documentation for it. I've setup MGCP and Asterisk seems to find it. But all channels (40 FXS channels) are Down! But the MGCP itself is Up according to the statistics. I can't find any documents how to set each channel to Up in the CLI. Any suggestions? Thanks! ___ 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
[Asterisk-Users] Polycom Phone Calling Party ID
I am trying to get the name and number to show up for an incoming calls onmy Polycom IP 500. Right now only the name shows up, but in the call listboth name and number show up. Any help on what to change in the config filewould be greatly appreciated. Watch the display. Once you answer the phone, the number should show up right below the name.Nick Yes that works, but it would be nice to see who is calling before I pick up. Is there a way to make that happen, have both name and number, or just number show up for incoming calls before I answer. Thanks ___ 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] * Call Monitoring
On Mon, 2005-02-21 at 08:00 -0500, Daniel Corbe wrote: I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? Interesting you should ask this today... I got to work this morning and was wondering why some of my calls were still diverting to my mobile. Eventually I realised that they were diverting on no answer. A restart of asterisk, reload of modules etc made no differences, I couldn't do anything with the line. Eventually I worked out it was a telco problem (no dialtone/etc) so I logged the fault. I looked at zttool and it showed a red alarm... In around 10-20 minutes I hacked zttool.c and converted it into a very basic cli version (which doesn't need newt) and would just dump the current status of all the spans. Similar to what you see on screen when you first start zttool. Then, I threw together some simple shell scripting to analyse/send the report to BigBrother (www.bb4.org). So far it is working nicely, by tomorrow night (yes, 27 hours after reporting it) hopefully my line should come back, and the alarm should change to OK... I'll put the package etc onto www.deadcat.net (BB addons website) and drop a post here when it is done. Will also put it onto www.websitemanagers.com.au/asterisk/ BTW, I did need to suid the zttool-cli command to root, as the normal BB user doesn't have the needed permissions. I haven't looked into this, but if anyone has a suggestion on a better way to do this, feel free to let me know. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] setting caller id number and using sip type=peer for incomming calles.
Morgan Gilroy wrote: To get around this i updated CVS HEAD and changed the sip entity from type=user to type=peer (yes peer!) (type=friend works too but im making a point) the client now must register to set his outbound caller*ID Number. Yes, that is normal. SIP has difficulty separating the remote party identification from the authentication identification (although it can be done). this seems to be working but it hardly seems correct, i mean using a peer for inbound calls when the docs all say it is for outbound calls. In CVS HEAD, soon _all_ SIP entries will be type=peer, because it's more logical this way. im not up on the sip protocol but wouldnt it be better if, when receiving an unknown connection (ie when caller*ID number is set to a pstn number) it first sends an authentication request to the client, on return it checks that username/secret against its list of users. if it still doesnt find it then drop it into the guest account. I believe it can already be configured to work that way, if you disable access to guest connections (I've not tried it, though). Remember also that it works this way because there are number of providers out there (Broadvoice being one) that will _not_ authenticate when they send you a call, only when they register. ___ 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] Snom phone hint exten question
El 21/02/2005, a las 12:30, James Bean escribió: Still doesn't work, I dialed in an outside line and picked up the receive on extension 691, yet the light on the snom phone did not come on. I dialed out of extension 691 to an outside line, yet still the light did not come on. Snom190 has firmware 3.56m the button is set to Destination 691 Be sure to reboot the snom after every change, fooled with it a little bit too. But get it woorking now. Atentament. ·· Adrià Vidal ___ 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] Polycom Phone Calling Party ID
Mark Floyd wrote: Yes that works, but it would be nice to see who is calling before I pick up. Is there a way to make that happen, have both name and number, or just number show up for incoming calls before I answer. You identify incoming callers by number, and not by name? Odd. In any case, no, unless you can get Polycom to change their firmware. There are no configuration options that control this behavior. ___ 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] Anyone using SuperMicro SuperServer 6014P-8R?
On February 21, 2005 10:25 am, Tony Mountifield wrote: Is anyone here using the SuperMicro SuperServer 6014P-8R with Asterisk? I'm especially interested if you've used it with a TE405P or TE410P. I'm actually using a 7043P-8R with a TE405P (hacked it so it ran in a 3.3V slot) -- it works but I am starting to suspect that it's teh system board that's been the cause of all my frustrations. I am changing it out shortly to verify this. -A. ___ 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
[Asterisk-Users] X-IMail-SPAM-Connection DNS Sudo ANI vs True ANI
Hello All, I'm having problems with international calling via Global Crossing. I'm told I need to send a true ani versus a sudo ani. What is the difference and how can I configure asterisk to do this. Global Crossing is denying calls with sudo anis. Thanks, Keith ___ 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] Illegal instruction on startup
On Mon, 2005-02-21 at 09:26 -0500, Tommy Vielkanowitz wrote: Hello, I have done some browsing through the wiki and on Google and havent been able to find anything that looks like what is happening to me. When I start Asterisk by typing asterisk vvvc, I get Illegal instruction and nothing else. Nothing before and nothing after. This is a Via Cyrix III 667MHz CPU with 192MB RAM running on Slackware 10.1 (Kernel 2.4.29) as a fresh install. I downloaded Asterisk, compiled mpeg123 and installed it, then compiled and installed Asterisk, then installed the sample data. I tried to start it up, and got the above error. Any pointers on where to look would be great. Thanks. Strike 1, you sent HTML email. Strike 2, you obviously didn't google. strike 3, well not yet.. http://www.google.com/search?q=Illegal+instruction+site%3Alists.digium.com -- Steven Critchfield [EMAIL PROTECTED] ___ 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] setting caller id number and using sip type=peerfor incomming calles.
To get around this i updated CVS HEAD and changed the sip entity from type=user to type=peer (yes peer!) (type=friend works too but im making a point) the client now must register to set his outbound caller*ID Number. Yes, that is normal. SIP has difficulty separating the remote party identification from the authentication identification (although it can be done). I see.. this seems to be working but it hardly seems correct, i mean using a peer for inbound calls when the docs all say it is for outbound calls. In CVS HEAD, soon _all_ SIP entries will be type=peer, because it's more logical this way. you mean amalgamating user and peer so there will eventually only be one type for both incoming and outgoing calls, (hopefully have an option to disable enable in/out bound calls). As long as it is the way it is supposed to be working I will quit complaining :) im not up on the sip protocol but wouldn't it be better if, when receiving an unknown connection (ie when caller*ID number is set to a pstn number) it first sends an authentication request to the client, on return it checks that username/secret against its list of users. if it still doesn't find it then drop it into the guest account. I believe it can already be configured to work that way, if you disable access to guest connections (I've not tried it, though). Remember also that it works this way because there are number of providers out there (Broadvoice being one) that will _not_ authenticate when they send you a call, only when they register. Now that's a bit of a bitch :/ but at the moment registration will have to do. But it would still be nice to have the ability to have a client that doesn't have to register, ie they have multiple servers that can dial though us and set caller id number. If I get time I might tinker with this myself for some fun :). Thanks for your reply. ___ 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] Adit 600 MGCP configuration
On Mon, 2005-02-21 at 16:33 +0100, Daniel Nyström wrote: I've finally got my Adit 600 and are configuring it right now. But I have to say, there aren't much documentation for it. I've setup MGCP and Asterisk seems to find it. But all channels (40 FXS channels) are Down! But the MGCP itself is Up according to the statistics. I can't find any documents how to set each channel to Up in the CLI. Go get the documentation from CAC. It shouldn't be that difficult to get the big PDF file and have docs for all the cards. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] compiling cvs-head today?
On Mon, 2005-02-21 at 06:36 -0600, Rich Adamson wrote: Anyone having problems compiling the current cvs head this morning? New cvs checkout on RH9, followed by appropriate make clean and make install. System was running cvs head from Nov 23 with TDM card, PRI, SIP phones on local wire, and IAX. See URL:http://bugzilla.redhat.com/bugzilla/ for instructions. The bug is not reproduceable, so it is likely a hardware or OS problem make: *** [say.o] Error 1 hash/hash.c:243: internal error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See URL:http://bugzilla.redhat.com/bugzilla/ for instructions. Looks like a hardware problem as you had failures in different locations but both where a gcc seg fault. This means either your CPU is hot and starting to spit out randomness or your memory is failing and producing randomness. Could be something else like low power supply and therefor faulty writing/reading of data to/from memory. Any way around it looks like you are in for either a while of debugging hardware or a hardware replacement regiment. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Adit 600 MGCP configuration
I don't believe the adit 600 has an up/down for channels. Are the channels connected to something. You might look at the 'connect' command and see if that helps. To bring the FXS channels up on my box I needed to connect them to the T1 (in your case it would be the MGCP) The t1 syntax is I believe 'connect a:01:1-8 1:1-8' The MGCP should be similiar but would probably be something other than a as it is presumably in a different slot. As far as more documentation, the adit 600 user manual seems to be plenty adequate (the pdf version is over 12Meg) I believe it is available on their website, if you have problems finding it, contact me offlist and I can send you a copy. Hope this helps, Jon. On Monday 21 February 2005 09:33 am, Daniel Nyström wrote: I've finally got my Adit 600 and are configuring it right now. But I have to say, there aren't much documentation for it. I've setup MGCP and Asterisk seems to find it. But all channels (40 FXS channels) are Down! But the MGCP itself is Up according to the statistics. I can't find any documents how to set each channel to Up in the CLI. Any suggestions? Thanks! ___ 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 ___ 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
[Asterisk-Users] Terminating problem
Hello list, I have been working with asterisk for a coupleof months and nowI have run into aproblem, I have the following setup PSTN ==Asterisk(remote behind nat)===IAX==Asterisk(local public ip)OH323Gateway I want to terminate incoming calls from the gateway in the remote asterisk. My problem is that when I started testing this setup the calls coming to the pstn lines from sip and h323 clientsworked, now that I want to terminate my calls in the pstn lines coming from the gatewaythe calls just hangupwith no error message and just a message saying no one is available to answer. I have been using my own phone lines in the office for testing, and this happens every time that thecall is passedand the phonewhere the call is supposed to land receives the call, in that moment asterisk hangs for no apparent reason, the asterisk with the zap channels is stable v.1.0.3 and the one with OH323 is cvs with openh323 and pwlib Janus patch and OpenH323 v.0.7.0. If anyone has done something similar I would appreciatethe help, any clues or extra information you need I would gladly send. Dan Flores ___ 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
[Asterisk-Users] VoIP Test Phone
Hi folks, Does anyone know if there is a small test board that has a mike and speaker? Board should run an OS that supports asterisk. I want to load asterisk on to it and test out. Thanks in advance, Kiran ___ 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] CallerID
Worked Great! Thx Julian..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian J. M.
Sent: Lunes, 21 de Febrero de 2005 02:46 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID
You can't change the callerid on an outgoing PSTN call (at least on analog
lines).
To modifiy the callerid on incoming calls, you could do something like this
(not tested):
[incoming-line1]
exten = s,1,setCidName(Line1: . ${CALLERID}) exten =
s,2,Goto(Incoming,s,1)
[incoming]
exten = s,1,normal incoming call stuff
then set a different context for each of your zap channels...
Julianjm.
On Mon, 21 Feb 2005 01:37:29 -0600, Anton Krall [EMAIL PROTECTED]
wrote:
Guys... I see there is a callerid parameter on zapata.conf... what
does that cid modify? the callerid people see when you call them using any
PSTN line?
Is there a way to send the SIP phone the incoming callerid frpm PSTN
lines asrecevied and append some string depending on the line it is coming
from?
__
Anton Krall
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[Asterisk-Users] Problems with the FXS module in a TDMxxx card (no sound when receiving a call
Hi all, I have a brand new TDMxxx card with 3 FXO modules and one FXS. It has replaced my old 3 X100P cards. The FXO part work as before, after some adjustments in the rxgain/txgain part. The problem I have is with the FXS module. I can place calls to SIP/IAX or PSTN destinations without any problems, but the sound received by the other part is much to strong and a little bit distorted. I have tried to modify the txgain up to txgain=-20, but still too strong. ..and this is not all. When I receive a call, from any type of source (IAX,SIP or PSTN), there is no sound (at both ends). No errors in the Asterisk console. I have tried to search through the archive but ... nothing related to this There is any way to enable something like 'iax2 debug' but for Zaptel channel? Any suggestions are welcome. Thank you and best regards, Dan ___ 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] setting caller id number and using sip type=peerfor incomming calles.
Morgan Gilroy wrote: you mean amalgamating user and peer so there will eventually only be one type for both incoming and outgoing calls, (hopefully have an option to disable enable in/out bound calls). Yes, exactly (and there will be other settings as well, to identify the type of peer (network, trunk, endpoint) for other reasons). Now that's a bit of a bitch :/ but at the moment registration will have to do. But it would still be nice to have the ability to have a client that doesn't have to register, ie they have multiple servers that can dial though us and set caller id number. If I get time I might tinker with this myself for some fun :). That's coming too, but in a different way. Actually if your remote peer can send you Remote-Party-ID headers now, you can set trustrpid=yes in your peer definition and the CLID/CNAM will come from that instead of the From header, so the From header can contain only authentication information. If the remote peer is Asterisk, it cannot currently send Remote-Party-ID, but watch Mantis for a patch in a few days to enable it :-) ___ 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] CallerID
I'm doing something like that on my system -- http://muware.com/asterisk -Original Message- From: Anton Krall [mailto:[EMAIL PROTECTED] Sent: Monday, February 21, 2005 1:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] CallerID Guys... I see there is a callerid parameter on zapata.conf... what does that cid modify? the callerid people see when you call them using any PSTN line? Is there a way to send the SIP phone the incoming callerid frpm PSTN lines asrecevied and append some string depending on the line it is coming from? __ Anton Krall ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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
[Asterisk-Users] Call Announce
Has anyone implemented callannounce? Here is an example. 1. Caller dials into asterisk 2. Call chooses Sales Extension 3. SalesPErson picks up the line and asterisk says you have a call from Sales Press 1 to accept the call Press 2 to send to voicemail. press 3 to hear caller ID Is something like this hard to implement? Thanks! -Randy ___ 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] compiling cvs-head today?
On Mon, 2005-02-21 at 06:36 -0600, Rich Adamson wrote: Anyone having problems compiling the current cvs head this morning? New cvs checkout on RH9, followed by appropriate make clean and make install. System was running cvs head from Nov 23 with TDM card, PRI, SIP phones on local wire, and IAX. See URL:http://bugzilla.redhat.com/bugzilla/ for instructions. The bug is not reproduceable, so it is likely a hardware or OS problem make: *** [say.o] Error 1 hash/hash.c:243: internal error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See URL:http://bugzilla.redhat.com/bugzilla/ for instructions. Looks like a hardware problem as you had failures in different locations but both where a gcc seg fault. This means either your CPU is hot and starting to spit out randomness or your memory is failing and producing randomness. Could be something else like low power supply and therefor faulty writing/reading of data to/from memory. Any way around it looks like you are in for either a while of debugging hardware or a hardware replacement regiment. Okay... this one is at a site 50 miles away where they are off on holiday today. Guess I'll wait for someone to show up. ;) Thanks ___ 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
[Asterisk-Users] Hitachi Wireless SIP handset
Anyone tried this yet? Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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
[Asterisk-Users] Queue Question
Is there a way to prioritize calls in multiple queues based on hold time? I have three queues set up on my Asterisk PBX with agents logged into all three queues. I've noticed that sometimes calls in one queue will make it through in a couple minutes while another queue will be backed up with people having been on hold for 30+ minutes. Is it possibly the fact that I am set for the rrmemory ring strategy? Thanks, Shaun ___ 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
[Asterisk-Users] IAX channel unable to create
I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by [EMAIL PROTECTED] on a i686 running Linux Box B is running: Asterisk 1.0.3 built by [EMAIL PROTECTED] on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12 WARNING[-1233155152]: channel.c:1860 ast_request: No channel type registered for 'IAX' Feb 21 12:48:12 NOTICE[-1233155152]: app_dial.c:696 dial_exec: Unable to create channel of type 'IAX' Feb 21 12:48:14 WARNING[-1116300368]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Non-critical Response) My box A iax.conf: [general] port=5036 bindport=5036 bandwidth=low allow=ulaw disallow=lpc10 jitterbuffer=no tos=lowdelay [slave] type=friend secret=4435 context=voice-mail defaultip=192.168.2.232 qualify=yes My Box A extension.conf [voice-mail] exten = _2000,1,Dial(IAX/master:[EMAIL PROTECTED]/[EMAIL PROTECTED]) My box B iax.conf [general] port=5036 bindport=5036 bandwidth=low allow=ulaw disallow=lpc10 tos=lowdelay [master] type=friend secret=4435 context=home defaultip=192.168.1.2 qualify=yes My Box B extension.conf [home] exten = _24xx,1,Dial(IAX2/slave:[EMAIL PROTECTED]/[EMAIL PROTECTED]) Thanks in advance Kurt ___ 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
[Asterisk-Users] Asterisk to Asterisk via IAX2 Help
Hi, I have two asterisk machines, chomper and otao. otao is otao.ieworks.net, has a public IP address (66.101.11.61), but no PSTN connections. chomper is at my house, behind NAT, but has a single X100P PSTN connection. I would like to establish two way calling between otao and chomper. Right now, I can call my extension on otao (2101) from my x-lite softphone on chomper, but I cannot call my extension (7101) on chomper from my sipura/hardphone on 2101 (Connected to otao). This is the error from otao: otao*CLI -- Executing Dial(SIP/2101-02b8, IAX2/chomper:[EMAIL PROTECTED]/101|20|Tt) in new stack -- Called chomper:[EMAIL PROTECTED]/101 Feb 21 11:25:25 WARNING[6894]: chan_iax2.c:5562 socket_read: Call rejected by 69.173.140.135: No authority found -- Hungup 'IAX2/chomper/16384' == No one is available to answer at this time -- Executing Playback(SIP/2101-02b8, goodbye) in new stack -- Playing 'goodbye' (language 'en') -- Executing Hangup(SIP/2101-02b8, ) in new stack == Spawn extension (from-sip, 7101, 3) exited non-zero on 'SIP/2101-02b8' This is the error from chomper: Feb 21 11:25:25 NOTICE[23368]: chan_iax2.c:5449 socket_read: Rejected connect attempt from 66.101.11.61 What am I missing? I've read through the wiki on this exact topic and must be having a dense day ;) Links I have looked at: http://www.voip-info.org/wiki-Asterisk+-+dual+servers http://www.voip-info.org/wiki-Asterisk+config+iax.conf http://www.voip-info.org/wiki-Asterisk+iax+channels http://www.voip-info.org/tiki-index.php?page=IAX2 I've also googled for answers, but what I've found seems to be related to IAX clients, rather than asterisk to asterisk via IAX2. Anyway, thanks for the clues in advance. Darren Relevant configs below: otao:iax.conf = [chomper] type=friend username=chomper host=dynamic ;secret=aragorn context=from-sip qualify=200 trunk=yes permit=0.0.0.0/0.0.0.0 otao:extensions.conf == ; 7101: [EMAIL PROTECTED] exten = 7101,1,Dial(IAX2/chomper:[EMAIL PROTECTED]/101,20,Tt) exten = 7101,2,Playback(goodbye) exten = 7101,3,Hangup chomper:iax.conf === [general] register = chomper:[EMAIL PROTECTED] ; ; VOIP * server on static IP [otao] type=friend host=otao.ieworks.net ;secret=aragorn context=from-sip qualify=200 trunk=yes permit=0.0.0.0/0.0.0.0 chomper:extensions.conf [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here. ; ; Set my info up. IAXINFO=chomper:SECRET DIALOUTANALOG=Zap/1 ; ; Extension 2101 is found on otao: exten = 2101,1,Dial(IAX2/[EMAIL PROTECTED]/2101) ___ 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] bridging iaxtel calls to PSTN
On Sun, 20 Feb 2005 02:43:46 -0700, [EMAIL PROTECTED] wrote: Hello, I just started using asterisk, and have a question. I have setup two asterisk servers, A and B. A has a Digium TDM400 11B card (1 FXO and 1 FSX modules) and is connected to the PSTN. B has same, but is NOT connected to PSTN. I want to configure B to call A via iaxtel, and connect to the PSTN using A's line. How can I configure iaxtel dial plan for B in extensions.conf? I want to be able to make a call to local US number (where A is located) from B, using iaxtel. Can anyone please help me? All I have seen so far is just making calls from A to B and vice versa using the iaxtel 1700 number, but I haven't seen any examples of how to bridge the iaxtel calls to PSTN. Help please. chuks. NB: I don't mean toll free number, I mean just local dialing. Don't bother with IAXTel. It's very frequently down. Just have each server register with the other and trunk between them. That way you just use dialplan logic to make the A place a call on B's resource. The wiki has a good section on trunking between servers ova IAX. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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
[Asterisk-Users] Polycom Phone Calling Party ID
Mark Floyd wrote: Yes that works, but it would be nice to see who is calling before I pick up. Is there a way to make that happen, have both name and number, or just number show up for incoming calls before I answer. You identify incoming callers by number, and not by name? Odd. In any case, no, unless you can get Polycom to change their firmware. There are no configuration options that control this behavior. The reason I want to do this is I get a lot of calls from cell phones and the caller ID name shows up unavailable. Do you know if a new firmware release is on the way? ___ 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
[Asterisk-Users] Why can't I make inter IAX calls between 2 Asterisk servers
Hello, two questions: 1: How can I open/enable network connection to B? scenerio: I have 2 Asterisk servers, A and B, running Fedora Core1 on my local network.B refuses any network connection attempts from A, i.e. I can't even telnet or FTPto B from A, but I canto A from B. This makes B refuse any IAX connection attempt from A. 2: what's wrong with my configurations, why can't I dial A from B, and vice versa? scenerio: A and B each has an analog device connected to their Zap/1channels, on extensions 2000 and 3000 respectively. I am trying to make IAX calls to each extension from the other, i.e call 3000 (on B)from A, and call 2000 (on A) from B. I get two different errors. While calling ext 2000 (on B) from A, connection was refused because of problem 1 above. While calling ext 3000 (on A) from B, it says context/extension does not exist on A. Here are my config files: A's extension.config [internal] exten = 3000,1,DIAL(IAX2/chuks:[EMAIL PROTECTED]/3000) exten = 3000,2,congestion include - from-iax [from-iax] exten = s,1,Wait(2) exten = s,2,Answer exten = 2000,3,Dial(Zap/1,20) NB: A's zapata.conf points to the internal context A's iax.conf [general] port=5036 bandwidth=high disallow=lpc10 tos=lowdelay [michael] type=friend secret=password auth=plaintext host=192.168.1.107 context=from-iax allow=all trunk=yes B's extension.config [internal] exten = 2000,1,DIAL(IAX2/michael:[EMAIL PROTECTED]/2000) ;A is on 192.168.1.103 exten = 2000,2,congestion include - from-iax [from-iax] exten = s,1,Wait(2) exten = s,2,Answer exten = 3000,3,Dial(Zap/1,20) NB: B's zapata.conf points to the internal context B's iax.conf [general] port=5036 bandwidth=high disallow=lpc10 tos=lowdelay [chuks] type=friend secret=password auth=plaintext host=192.168.1.103 context=from-iax allow=all trunk=yes At least I thought I'd hear A ring when I dial 2000 from B, instead, I get the congestion (busy) tone. Can anyone tell me what I'm doing wrong? If I can open up B's network connectios, I know I'll get the same problem each way. thx, chuks [EMAIL PROTECTED] ___ 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] Mandrake CAPI
I was looking at the exercise as a bit of Linux lerning for myself, so I guess Mandrake 10.1 and mISDN? Does anyone have working examples? Ray -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy Sent: 20 February 2005 23:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Mandrake CAPI Or you could go to a 2.6 kernel and use the mISDN drivers. Craig - Original Message - From: Razza [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, February 20, 2005 8:00 PM Subject: [Asterisk-Users] Mandrake CAPI All, I have been trying to get CAPI4Linux working on my machine and being frank am failing miserably! I am looking for any help available, I am real newbie (so please be gentle) and choose to run Mandrake 9.2 as it appears quite friendly (or so I thought!). I have been following the guidance found at http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI for the AVM card (actually I have a BT Speedway - apparently the same thing). I guess the best approach is to detail what I have done in tandem with the guidance? So here we go - Type - # modprobe capi Great! I get no response (which is expected!), so move to step 2 (http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install) Guidance states 'Download and install your kernel sources' - I installed these as part of the original installation, so I'll ignore. I download and install the CAPI driver - # cd /usr/src # wget ftp://ftp.avm.de/cardware/fritzcrd.pci/linux/suse.82/fcpci-suse8.2-03. 11 .02.tar.gz # tar -xzvf fcpci-suse8.2-03.11.02.tar.gz # cd fritz Great! Looking good! Guidance states modify the makefile in /usr/src/src.drv as follows - Replace - CARD_PATH = /lib/modules/`uname -r`/misc with - CARD_PATH = /lib/modules/$(uname -r)/kernel/drivers/isdn/avmb1 I am aware this chap is running Debian and I am running Mandrake, so after searching decided to modify the line as such - CARD_PATH = /lib/modules/2.4.22-10mdk/kernel/drivers/isdn/avmb1 Guidance states modify the KRNLINCL lines for the correct include path - KRNLINCL= /usr/src/kernel-headers-`uname -r`/include #KRNLINCL= /lib/modules/`uname -r`/build/include #KRNLINCL= /usr/src/linux/include And modify the lines as thus - DEFINES = -DMODULE -D__KERNEL__ -DNDEBUG \ -D__$(CARD)__ -DTARGET=\$(CARD)\ CCFLAGS = -c $(DEFINES) -O2 -Wall -I $(KRNLINCL) With - DEFINES = -DMODULE -DMODVERSIONS -D__KERNEL__ -DNDEBUG \ -D__$(CARD)__ -DTARGET=\$(CARD)\ CCFLAGS = -c $(DEFINES) -march=i686 -O2 -Wall -I $(KRNLINCL) \ -include $(KRNLINCL)/linux/modversions.h Again aware of the Debian V's Mandrake configuration, I searched the web and found the following guidance for Mandrake (using the google translation feature - http://translate.google.com/translate?hl=en http://translate.google.com/translate?hl=ensl=deu=http://ixi.thepen gu in.de/prev=/search%3Fq%3Dcapi%2Bmandrake%26hl%3Den%26lr%3D%26rls%3DRNWE ,RNWE:2004-35,RNWE:en sl=deu=http://ixi.thepenguin.de/prev=/search%3Fq%3Dcapi%2Bmandrake%26 hl%3Den%26lr%3D%26rls%3DRNWE,RNWE:2004-35,RNWE:en ) And made the following changes to the makefile in /usr/src/src.drv as that seemed more appropriate and saved the file - KRNLINCL =/usr/src/linux/include DEFINES = Dmodule Dmodversions D__kernel __ Dndebug \ D__$(card) __ Dtarget=\$(card) \ CCFLAGS = C $(defines) -march=i586 -O2 barrier i $(krnlincl) \ include/usr/src/linux/include/linux/modversions.h Going back to the original Guidance (http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install) I am instructed to modify the defs.h file in /usr/src/fritz/src.drv as follows - #if LINUX_VERSION_CODE KERNEL_VERSION(2, 5, 0) with #if LINUX_VERSION_CODE KERNEL_VERSION(2, 4, 23) Great, I'm now ready to run the make command! Unfortunately the first couple of responses are as follows which to me looks very bad? And not sure what to do next? [EMAIL PROTECTED] src.drv]# make cc C Dmodule Dmodversions D__kernel__ DNDEBUG D Dtarget=\\ -march=i586 -O2 barrier i /usr/src/linux/include include/usr/src/linux/include/linux/modversions.h main.c -o main.o cc: C: No such file or directory cc: Dmodule: No such file or directory cc: Dmodversions: No such file or directory cc: D__kernel__: No such file or directory cc: DNDEBUG: No such file or directory cc: D: No such file or directory cc: Dtarget=: No such file or directory cc: barrier: No such file or directory cc: i: No such file or directory cc: include/usr/src/linux/include/linux/modversions.h: No such file or directory For completeness I Have included the makefile and defs.h files Makefile SOURCES = main.c driver.c tables.c queue.c lib.c tools.c OBJECTS = $(patsubst %.c,%.o,$(SOURCES)) LIBRARY =
Re: [ nocadm ] Re: [Asterisk-Users] HDLC Bad FCS / HDLC Abort
Eric Wieling wrote: Yes. There are lots of messages in the mailing list archives regarding this problem, some of them even include things to try. You didn't see these messages when you searched the mailing list archives? Yes, I've read then. They say it can be caused by interruptions. I tried asterisk on kernels 2.4.29 and on 2.6.10 and I got the same errors. At this moment, I am running the latest version available on cvs on kernel 2.6.10 and I took off all other cards (4 FXO - Intel Ambient 3200) and still get the errors. The TE405P card is not loosing interruptions any more... Have a see... # cat /proc/interrupts CPU0 0:4454368 XT-PIC timer 1:589 XT-PIC i8042 2: 0 XT-PIC cascade 8: 2 XT-PIC rtc 10:4421380 XT-PIC t4xxp 11: 10671 XT-PIC eth0 14: 3163 XT-PIC ide0 15: 3125 XT-PIC ide1 NMI: 0 ERR: 0 # cat /proc/zaptel/1 Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 B8ZS/ESF 1 TE4/0/1/1 FXOKS (In use) 2 TE4/0/1/2 FXOKS (In use) 3 TE4/0/1/3 FXOKS (In use) 4 TE4/0/1/4 FXOKS (In use) [...] # cat /proc/zaptel/4 Span 4: TE4/0/4 TE410P (PCI) Card 0 Span 4 HDB3/CCS 73 TE4/0/4/1 Clear (In use) 74 TE4/0/4/2 Clear (In use) [...] I contacted Digium Instalation Support and I am waiting for their response. []s Alex Robertson Alex G Robertson wrote: Some news. It is not caused by transmission lines, conectors or anything like that. The telco tecnician just came here and analyzed the circuit and he got no erros! He sugested me to loop my PRI port in the balum attached in my asterisk box. And Surprise... I got the same errors! The error is on my hardware/software. []s Alex Robertson Alex G Robertson wrote: Hi everybody, I just installed asterisk, but this NOTICE dont stop appearing on my log file;; Feb 17 18:30:11 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:29:42 NOTICE[1336]: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 Feb 17 18:29:41 NOTICE[1336]: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 Feb 17 18:29:41 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:27:11 NOTICE[1336]: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 Feb 17 18:26:51 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:25:11 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:24:41 NOTICE[1336]: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 Feb 17 18:22:21 NOTICE[1336]: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 Feb 17 18:21:16 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:15 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:15 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:15 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:15 NOTICE[1336]: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 Feb 17 18:21:15 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:14 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:14 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:14 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:13 NOTICE[1336]: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 Feb 17 18:21:12 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:11 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 Feb 17 18:21:01 NOTICE[1336]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4 And from time to time this is happening -- B-channel 0/1 successfully restarted on span 4 -- B-channel 0/2 successfully restarted on span 4 -- B-channel 0/3 successfully restarted on span 4 -- B-channel 0/4 successfully restarted on span 4 [...] -- B-channel 0/29 successfully restarted on span 4 -- B-channel 0/30 successfully restarted on span 4 -- B-channel 0/31 successfully restarted on span 4 And the conversation stops. Telco, with a traffic analyzer, says that the clock is sliding. Does anybody knows what can it be? Hardware, software, transmission (conectors) etc ? Thanks in advance. -- Alex G Robertson NOC - Microlink ___ 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] How many line appearance can Snom 200 handle?
Yes 7 lines on the SNOM 200 SIP phone. Use a web browser to connect to your phone's IP address. There is a world of things it can do via its built-in web server. Just don't change the setting that says where to get the photos from, leave it as from the phone. Each line can be configured to register with a different server and with different accounts. Great little phone, even though the ringers sounds are goofy. Race The Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Sunday, February 20, 2005 11:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How many line appearance can Snom 200 handle? Snom 200 has be set up with extended key pad. The product literature also mention multiple sip registration. How many registration can it handle? It does not seem to appear in the user manual. David Kwok ___ 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 ___ 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] Snom phone hint exten question
On 13:27, Mon 21 Feb 05, Thorben Jensen wrote: I am going to now sit in a corner and go quietly insane while playing the banyo with no strings. Still doesn't work, I dialed in an outside line and picked up the receive on extension 691, yet the light on the snom phone did not come on. I dialed out of extension 691 to an outside line, yet still the light did not come on. Snom190 has firmware 3.56m the button is set to Destination 691 Hi James, I am using the latest CSV-HEAD of *, I do not think it works with * stable. It works on the Debian 1.0.5 version. ___ 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
[Asterisk-Users] voice recognition xml
Anyone here technical enough to design a voice recognition voice xml interchange for asterisk please email me; Ive been speaking with a contact of mine that is in the voice recognition space and he is interested in donating some technical support to the Asterisk community to assist with this project. This can only help benefit the Asterisk Community if this comes off. If this got up and running it would mean that Asterisk users would be able to offer voice recognition capabilities to their clients (or on their own installations) in an on-net ASP capability. Email me and Ill send you the details of the working group. Cheers, Dean ___ 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
[Asterisk-Users] Why can't I make inter IAX calls between 2 Asterisk servers
Hello, two questions: 1: How can I open/enable network connection to B? scenerio: I have 2 Asterisk servers, A and B, running Fedora Core1 on my local network.B refuses any network connection attempts from A, i.e. I can't even telnet or FTPto B from A, but I canto A from B. This makes B refuse any IAX connection attempt from A. 2: what's wrong with my configurations, why can't I dial A from B, and vice versa? scenerio: A and B each has an analog device connected to their Zap/1channels, on extensions 2000 and 3000 respectively. I am trying to make IAX calls to each extension from the other, i.e call 3000 (on B)from A, and call 2000 (on A) from B. I get two different errors. While calling ext 2000 (on B) from A, connection was refused because of problem 1 above. While calling ext 3000 (on A) from B, it says context/extension does not exist on A. Here are my config files: A's extension.config [internal] exten = 3000,1,DIAL(IAX2/chuks:[EMAIL PROTECTED]/3000) exten = 3000,2,congestion include - from-iax [from-iax] exten = s,1,Wait(2) exten = s,2,Answer exten = 2000,3,Dial(Zap/1,20) NB: A's zapata.conf points to the internal context A's iax.conf [general] port=5036 bandwidth=high disallow=lpc10 tos=lowdelay [michael] type=friend secret=password auth=plaintext host=192.168.1.107 context=from-iax allow=all trunk=yes B's extension.config [internal] exten = 2000,1,DIAL(IAX2/michael:[EMAIL PROTECTED]/2000) ;A is on 192.168.1.103 exten = 2000,2,congestion include - from-iax [from-iax] exten = s,1,Wait(2) exten = s,2,Answer exten = 3000,3,Dial(Zap/1,20) NB: B's zapata.conf points to the internal context B's iax.conf [general] port=5036 bandwidth=high disallow=lpc10 tos=lowdelay [chuks] type=friend secret=password auth=plaintext host=192.168.1.103 context=from-iax allow=all trunk=yes At least I thought I'd hear A ring when I dial 2000 from B, instead, I get the congestion (busy) tone. Can anyone tell me what I'm doing wrong? If I can open up B's network connectios, I know I'll get the same problem each way. thx, chuks [EMAIL PROTECTED] ___ 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] compiling cvs-head today?
On Mon, 2005-02-21 at 12:11 -0600, Rich Adamson wrote: On Mon, 2005-02-21 at 06:36 -0600, Rich Adamson wrote: Anyone having problems compiling the current cvs head this morning? New cvs checkout on RH9, followed by appropriate make clean and make install. System was running cvs head from Nov 23 with TDM card, PRI, SIP phones on local wire, and IAX. See URL:http://bugzilla.redhat.com/bugzilla/ for instructions. The bug is not reproduceable, so it is likely a hardware or OS problem make: *** [say.o] Error 1 hash/hash.c:243: internal error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See URL:http://bugzilla.redhat.com/bugzilla/ for instructions. Looks like a hardware problem as you had failures in different locations but both where a gcc seg fault. This means either your CPU is hot and starting to spit out randomness or your memory is failing and producing randomness. Could be something else like low power supply and therefor faulty writing/reading of data to/from memory. Any way around it looks like you are in for either a while of debugging hardware or a hardware replacement regiment. Okay... this one is at a site 50 miles away where they are off on holiday today. Guess I'll wait for someone to show up. ;) If they are gone for holiday, it very well could be heat related. Try your compiles a few hours after they get into the office and see if the heat levels have changed. -- Steven Critchfield [EMAIL PROTECTED] ___ 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
[Asterisk-Users] Asterisk@home Linux has no KDE
Hi Folks, I installed [EMAIL PROTECTED] on my PC. It went through the installation and all. But now i get a command line login window. Doesn't it has a KDE or some other type of OS GUI (i am not talking about [EMAIL PROTECTED] web GUI)? After i login, just the command line interface comes out. Any command to type here to get Linux OS GUI? Thanks, Kiran ___ 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
[Asterisk-Users] why can't I make toll free calls via IAXTEL
Hello, can someone tell me what's wrong with this? I can't make toll free calls via iaxtel. Here's the definition in my extensions.conf [iaxtel-trunks] ; ;outbound 1-700 and toll free calls go via iaxtel ;be sure to include the iaxtel-trunks context in dialing context ;add function here to continue ring tone when 9 is dialed ; ignorepat=9 exten = _91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1) exten = _91888NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1) exten = _91877NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1) exten = _91866NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1) exten = _91800NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1) Note: IAXTEL_INFO is already definedas username:password and here's my iax.conf [general] port=5036 bandwidth=high disallow=lpc10 tos=lowdelay ;to register with iaxtel register = username:[EMAIL PROTECTED] ; ; Trust Caller*ID Coming from iaxtel.com ; [iaxtel] type=friend context=from-iaxtel auth=cleartext ;inkeys=iaxtel when i make an 800 number call for instance, registration goes through and iaxtel can find me. But there is an endless silence, sort of like an endles loop, and the only output I see is a "timeout on Zap/1-1" and it tries the whole thing again...and goes on forever, and the call never goes through. Is there anything wrong with my configuration above? thx, Chuks. ___ 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
[Asterisk-Users] ZAP FXS vs ethernet FXS
What are the advantages/disadvantages of using a ZAP FXS port versus using one of the many small ethernet FXS devices on the market. The ZAP FXS talks directly to asterisk over PCI. Is this an advantage? The ethernet devices I assume speak either iax2 or sip, does this cripple the functionality of the attached FXS device for things like callwaiting,callerid,distinctive ring, etc... Does anyone have experience with both types of devices and would recommend one over the other? Thanks, Jon. ___ 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
[Asterisk-Users] South Korea DID wanted
Sorry for the cross post, but I'm still trying to find a Seoul DID. I received an email from LiveVoip.com that said they have service in South Korea, but when I called them they said they didn't offer such service. If you have the capability to offer a DID please let me know what your pricing structure is. This would be fairly low personal usage, probably around 400 minutes a month. Thank you! Justin ___ 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] compiling cvs-head today?
-Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Looks like a hardware problem as you had failures in different locations but both where a gcc seg fault. This means either your CPU is hot and starting to spit out randomness or your memory is failing and producing randomness. Could be something else like low power supply and therefor faulty writing/reading of data to/from memory. Any way around it looks like you are in for either a while of debugging hardware or a hardware replacement regiment. The first thing I usually do in these situations (after making sure the machine's fans are all running and dust-free) is run MEMTEST-86. http://www.memtest86.com/ It's not foolproof, but in my experience it catches more memory problems than any other utility. ___ 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: [ nocadm ] Re: [Asterisk-Users] HDLC Bad FCS / HDLC Abort
Sergey Kuznetsov wrote: This is happens because of imperfect HDLC code. Do you mean the software? The source code? []s -- Alex G Robertson NOC - Microlink ___ 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
[Asterisk-Users] why can't I make toll free calls via IAXTEL
Hello, can someone tell me what's wrong with this? I can't make toll free calls via iaxtel. Here's the definition in my extensions.conf [iaxtel-trunks] ; ;outbound 1-700 and toll free calls go via iaxtel ;be sure to include the iaxtel-trunks context in dialing context ;add function here to continue ring tone when 9 is dialed ; ignorepat=9 exten = _91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1) exten = _91888NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1) exten = _91877NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1) exten = _91866NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1) exten = _91800NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1) Note: IAXTEL_INFO is already definedas username:password and here's my iax.conf [general] port=5036 bandwidth=high disallow=lpc10 tos=lowdelay ;to register with iaxtel register = username:[EMAIL PROTECTED] ; ; Trust Caller*ID Coming from iaxtel.com ; [iaxtel] type=friend context=from-iaxtel auth=cleartext ;inkeys=iaxtel when i make an 800 number call for instance, registration goes through and iaxtel can find me. But there is an endless silence, sort of like an endles loop, and the only output I see is a "timeout on Zap/1-1" and it tries the whole thing again...and goes on forever, and the call never goes through. Is there anything wrong with my configuration above? thx, Chuks. ___ 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: [ nocadm ] Re: [Asterisk-Users] HDLC Bad FCS / HDLC Abort
Alex G Robertson wrote: Eric Wieling wrote: Yes. There are lots of messages in the mailing list archives regarding this problem, some of them even include things to try. You didn't see these messages when you searched the mailing list archives? Yes, I've read then. They say it can be caused by interruptions. Did you confirm you are not running graphics? (X, frame buffer, etc). Did you confirm you have unmasked IDE interrupts (-u1 to haparm)? --Eric ___ 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] compiling cvs-head today?
On Mon, 2005-02-21 at 12:11 -0600, Rich Adamson wrote: On Mon, 2005-02-21 at 06:36 -0600, Rich Adamson wrote: Anyone having problems compiling the current cvs head this morning? New cvs checkout on RH9, followed by appropriate make clean and make install. System was running cvs head from Nov 23 with TDM card, PRI, SIP phones on local wire, and IAX. See URL:http://bugzilla.redhat.com/bugzilla/ for instructions. The bug is not reproduceable, so it is likely a hardware or OS problem make: *** [say.o] Error 1 hash/hash.c:243: internal error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See URL:http://bugzilla.redhat.com/bugzilla/ for instructions. Looks like a hardware problem as you had failures in different locations but both where a gcc seg fault. This means either your CPU is hot and starting to spit out randomness or your memory is failing and producing randomness. Could be something else like low power supply and therefor faulty writing/reading of data to/from memory. Any way around it looks like you are in for either a while of debugging hardware or a hardware replacement regiment. Okay... this one is at a site 50 miles away where they are off on holiday today. Guess I'll wait for someone to show up. ;) If they are gone for holiday, it very well could be heat related. Try your compiles a few hours after they get into the office and see if the heat levels have changed. This one is located in a data center with a fair air handler in place, so more likely its a mem or power supply issue. Rich ___ 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
[Asterisk-Users] FWD using IAX2
Guys Ive setup FWD using IAX according to all the docs and I tried the give me a call url on FWD webpage and I do get the call but when asked to say my name, I hear a voice saying it didnt get it.. seems my voice is not getting thru to FWD... anybody had this problem while setting up FWD with IAX2? __ Anton Krall ___ 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] * Call Monitoring
Yeah, I'd be interested in porting your work so it runs under nagios. Please post your results when you're finished. -Daniel On Tue, 22 Feb 2005 02:54:22 +1100, Adam Goryachev [EMAIL PROTECTED] wrote: On Mon, 2005-02-21 at 08:00 -0500, Daniel Corbe wrote: I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? Interesting you should ask this today... I got to work this morning and was wondering why some of my calls were still diverting to my mobile. Eventually I realised that they were diverting on no answer. A restart of asterisk, reload of modules etc made no differences, I couldn't do anything with the line. Eventually I worked out it was a telco problem (no dialtone/etc) so I logged the fault. I looked at zttool and it showed a red alarm... In around 10-20 minutes I hacked zttool.c and converted it into a very basic cli version (which doesn't need newt) and would just dump the current status of all the spans. Similar to what you see on screen when you first start zttool. Then, I threw together some simple shell scripting to analyse/send the report to BigBrother (www.bb4.org). So far it is working nicely, by tomorrow night (yes, 27 hours after reporting it) hopefully my line should come back, and the alarm should change to OK... I'll put the package etc onto www.deadcat.net (BB addons website) and drop a post here when it is done. Will also put it onto www.websitemanagers.com.au/asterisk/ BTW, I did need to suid the zttool-cli command to root, as the normal BB user doesn't have the needed permissions. I haven't looked into this, but if anyone has a suggestion on a better way to do this, feel free to let me know. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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
