RE: [Asterisk-Users] SIP extension calls itself intermittently
Thank you very much for the help! I continue to have the mystery calls but not as often. I have attached the debugging info I captured. I also removed every piece of equipment and have a single line coming from the NIU into the X100P Clone card. I do have DSL so a DSL filter is required along is in the middle. What actually causes Asterisk/ZAP to thick there is a call? Thanks, Chip Nov 6 08:31:12 VERBOSE[4851]: -- Starting simple switch on 'Zap/1-1' Nov 6 08:31:20 WARNING[4851]: CallerID returned with error on channel 'Zap/1-1' Nov 6 08:31:20 VERBOSE[4851]: -- Executing Dial(Zap/1-1, SIP/6000|20) in new stack Nov 6 08:31:20 DEBUG[4851]: Setting NAT on RTP to 0 Nov 6 08:31:20 DEBUG[4851]: Outgoing Call for 6000 Nov 6 08:31:20 DEBUG[4851]: Call from user '6000' is 1 out of 0 Nov 6 08:31:20 VERBOSE[4851]: -- Called 6000 Nov 6 08:31:21 VERBOSE[4851]: -- SIP/6000-0821 is ringing Nov 6 08:31:28 DEBUG[4851]: update_user_counter(6000) - decrement outUse counter Nov 6 08:31:28 DEBUG[4851]: Exiting with DIALSTATUS=CANCEL. Nov 6 08:31:28 VERBOSE[4851]: == Spawn extension (from-pstn, s, 1) exited non-zero on 'Zap/1-1' Nov 6 08:31:28 VERBOSE[4851]: -- Executing Hangup(Zap/1-1, ) in new stack Nov 6 08:31:28 VERBOSE[4851]: == Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/1-1' Nov 6 08:31:28 DEBUG[4851]: Hangup: channel: 1 index = 0, normal = 18, callwait = -1, thirdcall = -1 Nov 6 08:31:28 DEBUG[4851]: disabled echo cancellation on channel 1 Nov 6 08:31:28 DEBUG[4851]: Set option TDD MODE, value: OFF(0) on Zap/1-1 Nov 6 08:31:28 DEBUG[4851]: Updated conferencing on 1, with 0 conference users Nov 6 08:31:28 VERBOSE[4851]: -- Hungup 'Zap/1-1' Nov 6 08:31:28 DEBUG[4851]: Acked pending invite 102 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Saturday, November 05, 2005 12:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP extension calls itself intermittently Intermittently Ill get calls from my only SIP extension to itself via the Zap/1. I have no clue and have found nothing online. I have listed my configurations and a sample of the console messages I see why debugging. Right now it only happens to the 6000 extension. Any assistance is appreciated. [from-pstn] exten = s,1,Wait(2) exten = s,2,Answer exten = s,3,Dial(SIP/6000,20) exten = s,4,Voicemail(u6000) exten = s,5,Congestion exten = s,6,Hangup In the above, you don't want to answer and incoming call in your dialplan. When SIP/6000 picks up the phone, an answer is automatically sent back to the pstn. So, change the above to something like this: [from-pstn] exten = s,1,Dial(SIP/6000,20) exten = s,2,Voicemail(u6000) exten = s,102,Voicemail(b6000) exten = s,103,Hangup The following file is /etc/asterisk/zapata.conf, not zaptel.conf zaptel.conf [channels] language=en context=from-pstn switchtype=national busydetect=yes busycount=4 callprogress=yes signalling=fxs_ks rxwink=300 usecallerid=yes cidsignalling=bell cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=400 rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callerid=name (xxx) xxx- busydetect=yes busycount=4 callprogress=yes channel = 1 Far too much junk and duplication in the above. Read the following including the comments. [channels] language=en busydetect=yes busycount=4 cidsignalling=bell cidstart=ring callerid=asreceived ; everything listed above applies to all channels defined below. Therefore ; only have to define them one time. ; the following channel definition is for the pstn line (channel 1) context=from-pstn ; switchtype=national ; this statement is for ISDN, not analog pstn. remove it ; busydetect=yes ; these two statements belong above and apply to all channels. ; busycount=4 callprogress=yes ; this should probably be =no signalling=fxs_ks ; rxwink=300 ; this statement isn't used with fxs_ks, remove it. usecallerid=yes ; cidsignalling=bell ; these two statements belong above and apply to all channels. ; cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes ; not sure about the statement. remove it. callwaitingcallerid=yes threewaycalling=yes transfer=yes; the pstn caller is _not_ going to transfer anything. remove it. cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes ; these two statements are identical. remove one of them. echotraining=400 rxgain=0.0 txgain=0.0 group=1 callgroup=1 ; this statement isn't needed, remove it. pickupgroup=1 ; the pstn line isn't going to pick up a call. remove it. immediate=no callerid=name (xxx) xxx- ; not needed. remove it. callprogress=yes ; used
RE: [Asterisk-Users] SIP extension calls itself intermittently
Thank you very much for the help! I continue to have the mystery calls but not as often. I have attached the debugging info I captured. I also removed every piece of equipment and have a single line coming from the NIU into the X100P Clone card. I do have DSL so a DSL filter is required along is in the middle. What actually causes Asterisk/ZAP to thick there is a call? Thanks, Chip Nov 6 08:31:12 VERBOSE[4851]: -- Starting simple switch on 'Zap/1-1' Nov 6 08:31:20 WARNING[4851]: CallerID returned with error on channel 'Zap/1-1' Don't know. Have you had the line tested by the telco? In the mean time, try busycount=8 to see if that might be a short term solution while waiting on the telco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP extension calls itself intermittently
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lists Pleasants Sent: 05 November 2005 01:59 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP extension calls itself intermittently Intermittently Ill get calls from my only SIP extension to itself via the Zap/1. I have no clue and have found nothing online. I have listed my configurations and a sample of the console messages I see why debugging. Right now it only happens to the 6000 extension. Any assistance is appreciated. Thanks, Chip -- Starting simple switch on 'Zap/1-1' Nov 4 14:00:54 WARNING[4156]: chan_zap.c:5476 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Wait(Zap/1-1, 2) in new stack -- Executing Answer(Zap/1-1, ) in new stack -- Executing Dial(Zap/1-1, SIP/6000|20) in new stack -- Called 6000 -- SIP/6000-3d34 is ringing == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' bart*CLI = Chip, The output above looks to me as an incoming call. I think you would have seen the SIP extension calling on ZAP/1-1 if initiated from your end, and would not call in as the line was busy. I get this now and again in the UK, usually in an evening time when the Telco do an auto check of line status. Regards Dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP extension calls itself intermittently
Intermittently Ill get calls from my only SIP extension to itself via the Zap/1. I have no clue and have found nothing online. I have listed my configurations and a sample of the console messages I see why debugging. Right now it only happens to the 6000 extension. Any assistance is appreciated. [from-pstn] exten = s,1,Wait(2) exten = s,2,Answer exten = s,3,Dial(SIP/6000,20) exten = s,4,Voicemail(u6000) exten = s,5,Congestion exten = s,6,Hangup In the above, you don't want to answer and incoming call in your dialplan. When SIP/6000 picks up the phone, an answer is automatically sent back to the pstn. So, change the above to something like this: [from-pstn] exten = s,1,Dial(SIP/6000,20) exten = s,2,Voicemail(u6000) exten = s,102,Voicemail(b6000) exten = s,103,Hangup The following file is /etc/asterisk/zapata.conf, not zaptel.conf zaptel.conf [channels] language=en context=from-pstn switchtype=national busydetect=yes busycount=4 callprogress=yes signalling=fxs_ks rxwink=300 usecallerid=yes cidsignalling=bell cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=400 rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callerid=name (xxx) xxx- busydetect=yes busycount=4 callprogress=yes channel = 1 Far too much junk and duplication in the above. Read the following including the comments. [channels] language=en busydetect=yes busycount=4 cidsignalling=bell cidstart=ring callerid=asreceived ; everything listed above applies to all channels defined below. Therefore ; only have to define them one time. ; the following channel definition is for the pstn line (channel 1) context=from-pstn ; switchtype=national ; this statement is for ISDN, not analog pstn. remove it ; busydetect=yes ; these two statements belong above and apply to all channels. ; busycount=4 callprogress=yes ; this should probably be =no signalling=fxs_ks ; rxwink=300 ; this statement isn't used with fxs_ks, remove it. usecallerid=yes ; cidsignalling=bell ; these two statements belong above and apply to all channels. ; cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes ; not sure about the statement. remove it. callwaitingcallerid=yes threewaycalling=yes transfer=yes; the pstn caller is _not_ going to transfer anything. remove it. cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes ; these two statements are identical. remove one of them. echotraining=400 rxgain=0.0 txgain=0.0 group=1 callgroup=1 ; this statement isn't needed, remove it. pickupgroup=1 ; the pstn line isn't going to pick up a call. remove it. immediate=no callerid=name (xxx) xxx- ; not needed. remove it. callprogress=yes ; used this above. remove it. channel = 1 Cleaned up, your zapata.conf should look about like this: [channels] language=en busydetect=yes busycount=6 ; changed this from 4 to 6. cidsignalling=bell cidstart=ring callerid=asreceived ; the following channel definition is for the pstn line (channel 1) context=from-pstn callprogress=no ; changed this from yes to no. signalling=fxs_ks usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 group=1 immediate=no channel = 1 Now, to answer your original question after cleaning up the statements, try busycount=6 and callprogress=no. What is likely happening is that either a bridged analog phone is connected to your pstn line and when someone is talking on that phone, asterisk interprets the audio as ringing. If there is no bridged analog phone, then * is likely interpreting noise of some sort as ringing. The callprogress=yes statement that you had doesn't really do anything for you, so either remove it or change it to callprogress=no. Several of the above statements that are left are simply repeating default values. If those statements were not there, it would function the same. But, while you're learning what these things do, I found it useful to leave them as is so as to remind you what those default values are actually in use. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users