[asterisk-users] PRI Random Disconnected
Dear Group, I am having some problem with PRI, my calls randomly get disconnected and after I am running Debug, I got the out from CLi screen... Cli messages, -- Executing Dial(Zap/31-1, zap/g1/100||rTt) in new stack -- Making new call for cr 32809 -- Requested transfer capability: 0x10 - 3K1AUDIO Protocol Discriminator: Q.931 (8) len=30 Call Ref: len= 2 (reference 41/0x29) (Originator) Message type: SETUP (5) [04 03 90 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a1 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 03 21 81 20] Calling Number (len= 5) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) ' ' ] [70 04 c1 31 30 30] Called Number (len= 6) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '100' ] -- Called g1/100 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 41/0x29) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) -- Zap/1-1 is proceeding passing it to Zap/31-1 Protocol Discriminator: Q.931 (8) len=12 Call Ref: len= 2 (reference 41/0x29) (Terminator) Message type: STATUS (125) [08 02 81 e4] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unknown (100), class = Protocol Error (6) ] [14 01 09] Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Incoming Call Proceeding (9) -- Processing IE 8 (cs0, Cause) -- Processing IE 20 (cs0, Call State) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 41/0x29) (Terminator) Message type: ALERTING (1) -- Zap/1-1 is ringing Protocol Discriminator: Q.931 (8) len=12 Call Ref: len= 2 (reference 41/0x29) (Terminator) Message type: CONNECT (7) [4c 05 01 80 31 30 30] Connected Number (len= 7) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Ext: 1 Presentation: Presentation permitted, user number not screened (0) '100' ] -- Processing IE 76 (cs0, Connected Number) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 41/0x29) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/1-1 answered Zap/31-1 -- Attempting native bridge of Zap/31-1 and Zap/1-1 ---Extension 100 pickup the call, and get disconnected Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 41/0x29) (Terminator) Message type: DISCONNECT (69) [08 02 85 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the remote user (5) Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 1 got hangup request /. Zapata.conf file [channels] context=from-alcatel switchtype=national overlapdial=yes signalling=pri_net pridialplan=local resetinterval=never facilityenable = yes priindication=outofband group=1 usecallerid=yes hidecallerid=no threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=-3.5 txgain=-3.5 busydetect=yes ;Busydetect I did set to no, and calls does not hangup so I turn to yes. relaxdtmf=yes immediate=no channel=1-15 channel=17-31 Equipment Topologies Connection: Analog Trunk line PSTN (Alcatel Ompcx Office) -- Asterisk (E1 as PRI_NET) Incoming goes into Alcatel and Alcatel Divert it to Asterisk via E1 Asterisk as a Auto-Attendance System. Asterisk R1.2.5 / Zaptel 1.2.5 / Libpri 1.2.2 .CentOS 4.3 final (Kernel 2.6.9-34.01EL) Appreciate if some one can help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma A200 hangup detection
Hi, Does some one experience the Sangoma A20X-ec series card that cant detect the hangup tone? I got * server 1.2.5 and running on Centos 4.3, I hock up to two PSTN lines and each time some one call in and my phone delay 1-2 sec (this is Asterisk delay nothing to do with Sangoma) and it rings on my phone, however, end on the day I got not less that 10 empty messages. I found out that Sangoma FXO port's does not hangup the lines after the external caller hangup the trunk line's. It take about 30sec later.so bad. I did feedback to Sangoma about this and never one of the tries successescan some one help me on this? Or this is the nature of Sangoma A2XX card? I did tried with TDM4XX no hangup issues on FXO port. My zaptel.conf file fxsks=3-4 ...I did tried out ls b4 I ask Sangoma loadzone=sg Zapata.conf [channels] context = from-pstn3 switchtype = national usecallerid=yes hidecallerid=no transfer=yes echocancel = yes echocancelwhenbridged = yes echotrainning = yes busydetect=yes busycount=1 callprogress=yes relaxdtmf=yes rxgain =-2.5 txgain =-2.5 signalling=fxs_ks group=1 channel=3-4 Any advice? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Centos cause Asterisk crash
Hi, Can some one who experience that does those file necessary for the CentOS and Asterisk installation /etc/cron.daily/00-makewhatis.cron /etc/cron.daily/slocate.cron /etc/cron.daily/prelink /etc/cron.daily/rpm /etc/cron.weekly/00-makewhatis.cron I experience that those file cause my Asterisk Server crash. Can I just disable them and run the Asterisk stable? Any reply will be appreciated. Thank you in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP Adapter
Hi all, I am seeking for the SIP Adapter which is providing the dual FXs ports. I can get some in the market, did some one experience that using Zyxel P-2002 ATA compatible with Asterisk? Further more, does Auto-Provisioning ATA useful to work with Asterisk? Please advice, Good experience ATA is needed. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help -- voicemail
Hi, Did someone experience that Asterisk OS 1.2.5 voicemail issues? Problem description: Some one call to the extensions 200, After 10 sec ring then go to voicemail [EMAIL PROTECTED] Announcement Please leave me a messages.blar blar.. When I completed to leave a message... IF : I press the pound #key ... Then it says Transfer IF : I Press the zero 0key Then it say Please confirm your recording IF : I hangup after leaving a message...then things get normal. What is this Funny. Pls some one reply. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 21, Issue 70
Ok, I did check it before and nothings related to this # key, if it's then system will announce that Please key in the extensions, but not in this case. By default, the blind transfer is #1. Some one can help? Check your features.conf file for conflicting key set. # is the default key for blind transfer feature. []'s MM chan (Alpha Trilogies Networks) wrote: Hi, Did someone experience that Asterisk OS 1.2.5 voicemail issues? Problem description: Some one call to the extensions 200, After 10 sec ring then go to voicemail [EMAIL PROTECTED] Announcement Please leave me a messages.blar blar.. When I completed to leave a message... IF : I press the pound #key ... Then it says Transfer IF : I Press the zero 0key Then it say Please confirm your recording IF : I hangup after leaving a message...then things get normal. What is this Funny. Pls some one reply. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI Setup
Hi, Need help from you guys. I had my Asterisk Set-up using PRI card TE110p, and everything working ok. However, I had bad experience with Asterisk answering call The problem was, when outsider calling into Asterisk... Asterisk answered call... CLI Accepting Overlap call from (CALLERIDNUM) to (unspecified) channel 0/31 CLI Starting Simple Switch on Zap31/1 Asterisk wait for 3sec Then jump to the context, and my phone rings. Can some one advice that, reducing the number of sec before passes it to the next context or task??? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GUI Web interface
Hi, I need some advice from your. I need to develop the GUI which is similar to [EMAIL PROTECTED], and the Web programming is PHP, and I wish to let the end user to do the programming via Web Interface and input data are directly put into the PostgreSQL, what step should I do? For example interfacing between the Web Interface to the PostgreSQL and to the Asterisk without reloading. Any advice or reference pls. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Problems with installing a TE110P on a Dell Poweredge 850
Please check your zaptel, TE110P is the T1/E1, you should state the parameter accordingly. /etc/zaptel.conf -- span=1,0,0,esf,b8zs fxsgs=1 ..where is your d-, b- channels? loadzone=us defaultzone=us Your mistake! [channels] context=default switchtype=national signalling=fxo_ls .it must be the same as zaptel.conf possible start thhe asterisk # asterisk # asterisk -c Debug the output of the status This is my advice. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G729A
Hi all, Will G729A codec exhaust the CPU power? If yes, how many concurrent sessions that P4 server board that can stand? Pls advise. Btw, if G729A has been purchased and installed, what will happen to the Asterisk Server crash say hard-disk when down or faulty, any where to do back up first such as tar commands? Any advice will be appreciated tq ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE:Toshiba DK424 / Asterisk / DTMF problems
Anthony, I will suggest you to use E1, you got 30 channels to communicate. I did the integration with Toshiba CTX using E1, and no problem at all. Asterisk as Pri_net Toshiba as PRI_cpe /etc/zaptel.conf span=1,1,0,ccs,hdb3 ;this is depanding on your pbx settingyou got to test it out, you could place crc4 or omit... bchan=1-15 dchan=16 bchan=17-31 loadzone=sg defaultzone=sg context=toshiba-intercom group=2 usecallerid=yes hidecallerid=no transfer=yes cancallforward=yes echocancel=yes echotraining=yes busydetect=yes busycount=2 immediate=no ;context=from-zap-trunk-1 switchtype=euroisdn overlapdial=yes signalling=pri_net pridialplan=local priindication=outofband channel=1-15 channel=17-31 Message: 3 Date: Thu, 2 Mar 2006 11:58:22 -0500 From: Anthony Cennami [EMAIL PROTECTED] Subject: [Asterisk-Users] Toshiba DK424 / Asterisk / DTMF problems To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I have a Toshiba DK424 connected via T1 EM to a TE110P card. Intermittently when a user dials a number I am getting 'getdtmf' errors on the Ast server and the calls do not go through. If they dial the number once or twice more, it works fine and I receive no DTMF problems. On another note, end users are complaining about intermittent disconnects. T1 is ESF/B8ZS - 24 chan. Other than those two problems the voice quality appears OK and I haven't really seen too many other problems. If there's anybody here running a similar config can you let me know if you've encountered this and what solutions you've devised. zapata.conf [trunkgroups] [channels] language=en group=1 context=from-pbx signalling=em_w relaxdtmf=yes usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=no callwaitingcallerid=no threewaycalling=no transfer=no canpark=no cancallforward=no callreturn=no echocancel=no echocancelwhenbridged=no busydetect=yes rxgain=5.0 txgain=5.0 callgroup=1 pickupgroup=1 immediate=no channel = 1-24 [zaptel.conf] span=1,1,0,esf,b8zs em=1-24 -- Anthony D Cennami -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060302/16e364 38/attachment-0001.htm -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
