[asterisk-users] Strange behaviour Panasonic KX-TD1232
CF, Adding www after Dial doesnt solve the trouble. I think we are talking the same but I dont express correctly. Did you saw my dialplan? I dont think I would have to add r. Yes, I have installed a 4 FXO Card, with fxsks signalling. What I mean is I understand FXO doesnt give the tone, but Panasonic. The cadence of ringing on Panasonic is a little different to the PSTNs cadence, FXO detects properly PSTN cadence when a call goes to or come from PSTN, and when a call goes from Panasonic to Asterisk, but doesnt make same job with a call going from Asterisk to Panasonic. The Sip phone behind Asterisk make the call, keeps ringing until Panasonic extension answer. Its normal, but even Panasonic user pick up the phone the Sip phone keeps ringing to user on Sip pone, nobody answer his call, however user on Panasonic pick up and doesnt hear anything. Im going crazy ___ --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] Strange behaviour Panasonic KX-TD1232
Ok Here goes dialplan [general] static=yes writeprotect=yes [incoming] exten = s,1,Answer exten = s,2,Background(pbx) exten = s,3,Set(TIMEOUT(response)=5) exten = 1001,1,Dial,SIP/1001|20 exten = 1001,2,Hangup exten = 1001,102,Congestion,3 exten = 1002,1,Dial,SIP/1002|20 exten = 1002,2,Hangup exten = 1002,102,Congestion,3 [sip] include = outgoing exten = 1001,1,Dial(SIP/1001,20) exten = 1001,2,Hangup exten = 1001,102,Congestion,3 exten = 1002,1,Dial(SIP/1002,20) exten = 1002,2,Hangup exten = 1002,102,Congestion,3 [outgoing] exten = 0,1,Dial,Zap/g1 exten = 0,2,Hangup exten = 0,102,Congestion exten = 9,1,Dial,Zap/g1/9 exten = 9,2,Hangup exten = 9,102,Congestion Here goes zapata [trunkgroups] [channels] context=default switchtype=national signalling=fxs_ks rxwink=300 usecallerid=no hidecallerid=no callwaiting=no callprogress=yes ;progzone=us usecallingpres=yes threewaycalling=no transfer=no cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes busydetect=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no signalling=fxs_ks group=1 callerid=asreceived context=incoming channel =1 channel =2 channel =3 channel =4 ___ --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] Strange behaviour Panasonic KX-TD1232
I think still didnt explain me clearly The problem is when I dial 0, in this case the asterisk take Zap (connected directly to ext 200 from Panasonic), Panasonic gives tone, dial another extension (ie 100), the extension rings but when answer the phone asterisk keeps ringing it doesnt detect when you pick up the phone. ___ --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] Strange behaviour Panasonic KX-TD1232
Ok Ok, the figure doesnt help.Here we go again - -- --- --| SIP | - | ASTERISK | -- | PANASONIC | --- | PSTN | - -- --- -- | | Ext1 Ext2Here is my dialplan[incoming]exten = s,1,Answerexten = s,2,Background(prueba-pbx)exten = s,3,Set(TIMEOUT(response)=5)exten = 1001,1,Dial,SIP/1001|20exten = 1001,2,Hangupexten = 1001,102,Congestion,3exten = 1002,1,Dial,SIP/1002|20exten = 1002,2,Hangupexten = 1002,102,Congestion,3[sip]include = outgoingexten = 1001,1,Dial(SIP/1001,20)exten = 1001,2,Hangupexten = 1001,102,Congestion,3exten = 1002,1,Dial(SIP/1002,20)exten = 1002,2,Hangupexten = 1002,102,Congestion,3[outgoing]exten = 0,1,Dial,Zap/g1exten = 0,2,Congestionexten = 0,102,Congestionexten = 9,1,Dial,Zap/g1/9exten = 9,2,Congestionexten = 9,102,CongestionWhen I make a call from PSTN to SIP, first Answer the Panasonic, after this I digit an Extension and the call goes to asterisk, then I dial to sip and the call goes on successfully. When I make a call from Ext1 or Ext2 to SIP, I dial an extension and the call goes to asterisk, then I dial to sip and the call goes on.When I make a call from SIP to PSTN, first dial 9 and asterisk gets a Zap sending 9 to get PSTN line, the dial the PSTN number and the call goes on.When I make a call from SIP to Ext1 (Ext2 ExtN), the Sip phone keeps ringing and user behind Ext1 doesn't hear anything.Your help will be appreciated. ___ --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] Strange behaviour Panasonic KX-TD1232
Ok, Im going to stop pictures I have a Digium 4 FXO Card in my asterisk, and connect to Panasonic through 2 extensions (configured in a pool) This means when you dial 200 (example) in Panasonic, the call goes to asterisk and it answers. In this sense, the answer is yes replacing asterisk by a conventional phone, I can dial and the phone rings. The only way in wich call doesnt work is from Sip to Panasonic Ext. I really dont think the problem is asterisk, but ringing cadence and ringback tones from Panasonic. ___ --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] Strange behaviour Panasonic KX-TD1232
Hello, Ive got asterisk running and almost working with Panasonic KX-TD1232 I said almost, because theres a strange behaviour when I make calls. --- - - --- | SIP | -- | ASTERISK | -- | PANASONIC | | PSTN | --- - - -- | | --- --- | Ext1| | Ext2| --- --- When I make a call from PSTN to SIP, the call goes on successfully. When I make a call from SIP to PSTN, the call goes on successfully. When I make a call from Ext1 or Ext2 to SIP, the call goes on successfully. When I make a calla from SIP to Ext1 (Ext2 ExtN), the Sip phone keeps ringing and user behind Ext1 doesnt hear anything. It seams appear like Asterisk doesnt detect the answer on Ext1 Is there any way to figure it out?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Don't Hit # after 9 to get PSTN line
Hi all, Iv got a problem taking lines to call from SIP to PSTN. I have to press # after 9 to get ringtone, otherwise I would have to wait above 15 seconds. [out] exten = 9,1,Dial,Zap/g1/9 exten = 9,2,Hangup exten = 9,102,Congestion The problem occurs when the user doesnt complete the call, and hangup after pressing only 9. If these events occur twice consecutively, Asterisk attempts to native bridge between 2 channels. I think the problem is that # is being used like a transfer trigger. But when I deactivate these feature, I have to wait 15 second after press 9 no get line. What can I do?? What should I do to get line without spend this time? Pablo ___ --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: Don't Hit # after 9 to get PSTN line
Really dont. Dialplan is very simple, please take a look [incoming] exten = s,1,Answer exten = s,2,Background(prueba-pbx) exten = s,3,Set(TIMEOUT(response)=5) exten = 1001,1,Dial,SIP/1001|20 exten = 1001,2,Hangup exten = 1001,102,Congestion,3 exten = 1002,1,Dial,SIP/1002|20 exten = 1002,2,Hangup exten = 1002,102,Congestion,3 exten = 1003,1,Dial,SIP/1003|20 exten = 1003,2,Hangup [sip] include = out exten = 1001,1,Dial(SIP/1001,20) exten = 1001,2,Hangup exten = 1001,102,Congestion,3 exten = 1002,1,Dial(SIP/1002,20) exten = 1002,2,Hangup exten = 1002,102,Congestion,3 exten = 1003,1,Dial(SIP/1003,20) exten = 1003,2,Hangup [out] exten = 9,1,Dial,Zap/g1/9 exten = 9,2,Hangup exten = 9,102,Congestion And yes, Im trying asterisk behind and Ericsson MD110 PBX, and when I hit 9 I ask for an internal line and re-send 9 to get an external line. Thanks Pablo ___ --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] Don't Hit # after 9 to get PSTN line
I really dont understand what you say. Ive been searching in my SIP device (Innomedia 3308), and there isnt any option to disable 3-way calling. Do you refer to sip.conf??? Pablo ___ --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: Don't Hit # after 9 to get PSTN line
Steven, Ive been searching that you say, but certainly I dont know where to search or those lines isnt there. I found these: Configuring VoIP DigitMap dialing pattern - empty - Configure FXS Setting Parameters Ringing Timeout = 180 second Ringing Cadence = 0 Ringing Repetition = 0 Dial Tone Timeout = 16 seconds Echo Cancellation: Yes Prefix Digit = NULL Configuring SIP Settings Current SIP Proxy Servers = 192.168.42.3 Use Outbound Proxy = No Current Local SIP Port = 5060 Response Code for Retry Registration = Retry Registration Interval = 0 seconds Current SIP Domain = Current Exponential Backoff = 500 ms Current Exponential Cap = 2000 ms Current Non-INVITE retry = 4 times Current INVITE msg retry = 4 times Current REGISTER expiration = 3600 seconds Current Session Timer = 0 seconds Current Bullet Interval = 0 seconds Current Number of Codecs = 1 Current Codec List = G729A Digitmap Partial Match Timeout = 16 Digitmap Critical Timeout = 4 Cancel Call Waiting Invoke String = *72 Call Transfer Invoke String = *90 CID Block Invoke String = *67 CID Display Invoke String = *82 Call Park Invoke String = *98 Call Retrieve Invoke String = *99 Outside Line Access Number = 9 Use User-Agent Header = Yes Set Jitter Buffer Adaptive = Yes Use SIP INFO for DTMF = No Re-registration Credential Enable = No Current SIP PING Interval = 0 seconds Current SIP PING Proxy Require Header = Current SIP External IP address = Use SIP INFO for Flash Event = No So, what do you think?? Pablo ___ --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] asterisk + centos 4.3
Not sure what you want, but I have asterisk running on Centos 4.3 and theres no problems. ___ --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] Delay on ring after dial Out
Hello, Ive asterisk installed, but It has a particularity. When I dial an extension in context internal (exten = 1001,1,Dial,SIP/1001), the call finishes successfully. But, when I dial to get a trunk line (trunk like an analog line from PSTN) it takes above 15 sec to give me tone (exten = 0,1,Dial,Zap/g1). What its wrong? What can I do to remove this delay. After de 15 secs, I can call normally, but you really dont wanna wait 15 secs every time you will call. Thanks Pablo ___ --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] how to decrease answer time !
Pablo Mora, Ing. GERENTE DE OPERACIONES ESPOLTEL S.A. Malecón 100 y Loja Telf.:2514477 ___ --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
