[Asterisk-Users] 4 Port FXO cards
Hi, Do Digium have any plans to release a 4 port fxo card. If yes, when?
Re: [Asterisk-Users] 4 Port FXO cards
anyway, better if Digium can do it quickly, we are suffering a lot with channel banks, we need to replace these channel banks with 4 port cards - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 6:09 PM Subject: Re: [Asterisk-Users] 4 Port FXO cards Surajee Ratnayake wrote: Hi, Do Digium have any plans to release a 4 port fxo card. If yes, when? I think they are in the pipeline.. Initial speculation was that they would be out in September but I guess there have been problems.. I guess the best answer is they will come out when they come out.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Artificially Limiting IAX Calls
- Original Message - From: Brian Schrock [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 23, 2003 3:14 AM Subject: [Asterisk-Users] Artificially Limiting IAX Calls Everyone, Can I artifically limit the amout of IAX calls going out of an asterisk server... I am worried that if every call requires x amount of bandwidth and their internet link is only as big as 2x what happens when the third call comes up. So far I have seen it just starts up and will kill the other two calls. How have other folks on here dealt with this issue? Brian J. Schrock Anistone Technologies, LLC 6926 Avery Rd. Dublin, OH 43017 Phone: 614-798-9106 Use a global variable to keep a count of IAX calls going through that IP link Surajee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unexpected Call Termination!
hi, I hav a softPBX setup. Our set up has 2 servers, one is connected to an ISDN PRI E1 coming from PSTN central office and the other server is connected to another E1 which is coming from a Nortel PBX. and 2 servers are connected to a LAN. So when a Nortel PBX users want to get an out side call they go though our servers. But there are some complains coming to us saying that most of the calls do get cut after several time. that is when some body is engaged in a call with an outside number, suddenly call terminates unexpectedly. This is very disturbing for us. Can anybody pls help us with this situation. Surajee
Re: [Asterisk-Users] Call Time out Problem-Very Urgent!
hi, yes we tried without |t, but it didn't work.. still keeps on ringing forever... :-( Surajee - Original Message - From: Surajee Ratnayake To: [EMAIL PROTECTED] Sent: Monday, September 08, 2003 12:49 PM Subject: [Asterisk-Users] Call Time out Problem-Very Urgent! hi, I have a problem in call time out,An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a Nortel PBX is conneted to my server.But when i do a Dialout(from both E1s)the calls do not timeout.For ex. Dial(Zap/g4/123456|20|t) suppose other side is ringing and is not answering.even after 20 seconds, call doesn't get timeout pls gv me a solutions..its really urgent.. Surajee
Re: [Asterisk-Users] Call Time out Problem-Very Urgent!
Is it a problem with E1, bcos, when we dial a SIP extension from the same asterisk box it timeouts but not the Zap ones.. We tried without |t, but it didn't work.. still keeps on ringing forever... :-( Surajee - Original Message - From: [EMAIL PROTECTED] To: Surajee Ratnayake [EMAIL PROTECTED] Sent: Sunday, September 07, 2003 12:29 PM Subject: Re: [Asterisk-Users] Call Time out Problem-Very Urgent! surajee: what happens if you remove the |t ? still no timeout ? -wasim On Mon, 8 Sep 2003, Surajee Ratnayake wrote: hi, I have a problem in call time out, An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a Nortel PBX is conneted to my server. But when i do a Dialout(from both E1s)the calls do not timeout. For ex. Dial(Zap/g4/123456|20|t) suppose other side is ringing and is not answering. even after 20 seconds, call doesn't get timeout pls gv me a solutions.. its really urgent.. Surajee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Time out Problem-Very Urgent!
hi, yes, i had 'callprogress=yes', and i commented it. now the time out is working. Thank you very much by disabling callprogress in an analog environment, does it affet the call disconnection? Surajee - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 08, 2003 10:32 PM Subject: Re: [Asterisk-Users] Call Time out Problem-Very Urgent! Do you have callprogress=yes in zapata.conf ? If yes, then comment it out. Also you could send some trace from the console including pri debug span span-no Martin On Mon, 8 Sep 2003, Surajee Ratnayake wrote: Is it a problem with E1, bcos, when we dial a SIP extension from the same asterisk box it timeouts but not the Zap ones.. We tried without |t, but it didn't work.. still keeps on ringing forever... :-( Surajee - Original Message - From: [EMAIL PROTECTED] To: Surajee Ratnayake [EMAIL PROTECTED] Sent: Sunday, September 07, 2003 12:29 PM Subject: Re: [Asterisk-Users] Call Time out Problem-Very Urgent! surajee: what happens if you remove the |t ? still no timeout ? -wasim On Mon, 8 Sep 2003, Surajee Ratnayake wrote: hi, I have a problem in call time out, An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a Nortel PBX is conneted to my server. But when i do a Dialout(from both E1s)the calls do not timeout. For ex. Dial(Zap/g4/123456|20|t) suppose other side is ringing and is not answering. even after 20 seconds, call doesn't get timeout pls gv me a solutions.. its really urgent.. Surajee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Time out Problem-Very Urgent!
hi, I have a problem in call time out,An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a Nortel PBX is conneted to my server.But when i do a Dialout(from both E1s)the calls do not timeout.For ex. Dial(Zap/g4/123456|20|t) suppose other side is ringing and is not answering.even after 20 seconds, call doesn't get timeout pls gv me a solutions..its really urgent.. Surajee
Re: [Asterisk-Users] What is the best IP phone?
can the pingtel phonebe used smoothly with asterisk? - Original Message - From: Steve Totaro To: [EMAIL PROTECTED] Sent: Saturday, September 06, 2003 11:15 PM Subject: Re: [Asterisk-Users] What is the best IP phone? i like the pingtel phones. www.pingtel.com - Original Message - From: Surajee Ratnayake To: [EMAIL PROTECTED] Sent: Saturday, September 06, 2003 4:40 AM Subject: [Asterisk-Users] What is the best IP phone? hi, Can anybody suggest me a good, reliable, robust, SIP supported hardware IP phone? Surajee
Re: [Asterisk-Users] What is the best IP phone?
how about the call transfer feature? is it working fine? and can u pls let me know the price too.. - Original Message - From: Steve Totaro To: [EMAIL PROTECTED] Sent: Sunday, September 07, 2003 12:16 AM Subject: Re: [Asterisk-Users] What is the best IP phone? i setup asterisk for the first time yesterday morning and had two pingtel phones working by noon (with message waiting indicators) the phones have a nice web interface for config and speed dials. its a java phone too. they are a little pricey and maybe too funky looking for some people though - Original Message - From: Surajee Ratnayake To: [EMAIL PROTECTED] Sent: Saturday, September 06, 2003 6:08 AM Subject: Re: [Asterisk-Users] What is the best IP phone? can the pingtel phonebe used smoothly with asterisk? - Original Message - From: Steve Totaro To: [EMAIL PROTECTED] Sent: Saturday, September 06, 2003 11:15 PM Subject: Re: [Asterisk-Users] What is the best IP phone? i like the pingtel phones. www.pingtel.com - Original Message - From: Surajee Ratnayake To: [EMAIL PROTECTED] Sent: Saturday, September 06, 2003 4:40 AM Subject: [Asterisk-Users] What is the best IP phone? hi, Can anybody suggest me a good, reliable, robust, SIP supported hardware IP phone? Surajee
[Asterisk-Users] ISDN PRI E1-CLI and DNIS
hi everybody, my question is specific to ISDN signalling, in my set up, i want to get cli and dnis into my asterisk box, and i am going to use ISDN PRI E1s coming from telco. To get cli and dnis, do i need to apply for QSIG from the telecom, or is there some other type? and i got to know that still asterisk does not support QSIG??? sorry for asking this kind of question in the asterisk mailing list, Thank you inadvance, Surajee
Re: [Asterisk-Users] Call Transfer Problem
yes, u are quite right, you can find this feature in almost every pbx now. We are also wondering whether, presently some one is implementing this feature or not, if no body is doing that, we can start on that Surajee - Original Message - From: George Lin To: [EMAIL PROTECTED] Sent: Wednesday, June 04, 2003 3:36 AM Subject: RE: [Asterisk-Users] Call Transfer Problem so, What should the call initiator do if s/he wants to transfer the call initiated by himself/herself, by using flash keypad or what else ? I can see such application can be used in some big office, where the BOSS always asks the secretary to make the call, once the call is connected, then the secretary can trasfer the call to the BOSS. in order to let the BOSS talk on the phone. am I right ?? Please let me know once the feature is implemented. George Lin -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Surajee RatnayakeSent: Monday, June 02, 2003 1:05 AMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Call Transfer Problem U get the following output when u execute the "show application Dial" command in the Asterisk prompt, -= Info about application 'Dial' =- [Synopsis]: Place an call and connect to the current channel [Description]: Dial(Technology/resource[Technology2/resource2...][|timeout][|options][|URL]):Requests one or more channels and places specified outgoing calls on them.As soon as a channel answers, the Dial app will answer the originatingchannel (if it needs to be answered) and will bridge a call with the channelwhich first answered. All other calls placed by the Dial app will be hunp upf a timeout is not specified, the Dial application will wait indefinitelyuntil either one of the called channels answers, the user hangs up, or allchannels return busy or error. In general, the dialler will return 0 if itwas unable to place the call, or the timeout expired. However, if allchannels were busy, and there exists an extension with priority n+101 (wheren is the priority of the dialler instance), then it will be the nextexecuted extension (this allows you to setup different behavior on busy fromno-answer). This application returns -1 if the originating channel hangs up, or if thecall is bridged and either of the parties in the bridge terminate the call.The option string may contain zero or more of the following characters: 't' -- allow the called user transfer the calling user 'T' -- to allow the calling user to transfer the call. 'r' -- indicate ringing to the calling party, pass no audio until answered. 'm' -- provide hold music to the calling party until answered. 'd' -- data-quality (modem) call (minimum delay). 'c' -- clear-channel data call (PRI-PRI only). 'H' -- allow caller to hang up by hitting *. 'C' -- reset call detail record for this call. 'P[(x)]' -- privacy mode, using 'x' as database if provided. In addition to transferring the call, a call may be parked and then pickedup by another user. The optionnal URL will be sent to the called party if the channel supportsit. Surajee - Original Message - From: George Lin To: [EMAIL PROTECTED] Sent: Monday, June 02, 2003 1:11 PM Subject: FW: [Asterisk-Users] Call Transfer Problem Hi, Which document describes the Dial with T option ? Could you let me know or email it to me. Thanks, George Lin -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Surajee RatnayakeSent: Sunday, June 01, 2003 9:10 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Call Transfer Problem hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial applicationintructs to use "T" to achieve that. and we learnt that it has not been implemented yet in Asterisk. Is this true? Is some one workin on this issue? if the answer is NO, we can give a try to implement it, with a help of u all , ofcourse :-) (cos, we are quite new to asterisk-only 1 month of experience, but amazed of its great
[Asterisk-Users] Limit Concurrent IAX Channels
hi All, Since the quality drops with the increased usageof IAX channels, between 2 Asterisk servers, I want to limit the number of simultaneously used IAX channels. ie basically to limit the calls between the 2 Asterisk servers, can anybody pls tell me a method/hint to achieve this, i am helpless :-( Thanx in advance, Surajee
Re: [Asterisk-Users] Limit Concurrent IAX Channels
I did search in the list and i googled the web, but i couldn't find anything related to my problem, if u hav that post can u pls send it to me, or at least gv me some key words to search for.. Surajee - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 02, 2003 8:12 PM Subject: Re: [Asterisk-Users] Limit Concurrent IAX Channels hi surajee, some time ago martin pycko posted an example to the list how to make use of global variables and gotoif to limit the number of channels. the ML archive is your friend :) regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon: +49 30 79705392 fax: +49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Mon, 2003-06-02 um 17.01 schrieb Surajee Ratnayake: hi All, Since the quality drops with the increased usage of IAX channels, between 2 Asterisk servers, I want to limit the number of simultaneously used IAX channels. ie basically to limit the calls between the 2 Asterisk servers, can anybody pls tell me a method/hint to achieve this, i am helpless :-( Thanx in advance, Surajee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit Concurrent IAX Channels
I can use a global variable to keep a count of number of channels used on a particular channel, and i can increase this count when i am calling the 'Dial' application, but the problem is, how can i reduce the counter(when a channel is free), i can not find a place in dail plan to do 'counter--' operation i am desperate :-( - Original Message - From: Surajee Ratnayake [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 02, 2003 9:28 PM Subject: Re: [Asterisk-Users] Limit Concurrent IAX Channels I did search in the list and i googled the web, but i couldn't find anything related to my problem, if u hav that post can u pls send it to me, or at least gv me some key words to search for.. Surajee - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 02, 2003 8:12 PM Subject: Re: [Asterisk-Users] Limit Concurrent IAX Channels hi surajee, some time ago martin pycko posted an example to the list how to make use of global variables and gotoif to limit the number of channels. the ML archive is your friend :) regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon: +49 30 79705392 fax: +49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Mon, 2003-06-02 um 17.01 schrieb Surajee Ratnayake: hi All, Since the quality drops with the increased usage of IAX channels, between 2 Asterisk servers, I want to limit the number of simultaneously used IAX channels. ie basically to limit the calls between the 2 Asterisk servers, can anybody pls tell me a method/hint to achieve this, i am helpless :-( Thanx in advance, Surajee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit Concurrent IAX Channels
Thank you very much! i was exactly looking for that - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 03, 2003 9:16 AM Subject: Re: [Asterisk-Users] Limit Concurrent IAX Channels On Mon, 2003-06-02 at 21:58, Surajee Ratnayake wrote: I can use a global variable to keep a count of number of channels used on a particular channel, and i can increase this count when i am calling the 'Dial' application, but the problem is, how can i reduce the counter(when a channel is free), i can not find a place in dail plan to do 'counter--' operation i am desperate :-( exten = h,1,setvar -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Transfer Problem
hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial applicationintructs to use "T" to achieve that. and we learnt that it has not been implemented yet in Asterisk. Is this true? Is some one workin on this issue? if the answer is NO, we can give a try to implement it, with a help of u all , ofcourse :-) (cos, we are quite new to asterisk-only 1 month of experience, but amazed of its great performance) Thank you very much, Surajee
Re: [Asterisk-Users] Call Transfer Problem
U get the following output when u execute the "show application Dial" command in the Asterisk prompt, -= Info about application 'Dial' =- [Synopsis]: Place an call and connect to the current channel [Description]: Dial(Technology/resource[Technology2/resource2...][|timeout][|options][|URL]):Requests one or more channels and places specified outgoing calls on them.As soon as a channel answers, the Dial app will answer the originatingchannel (if it needs to be answered) and will bridge a call with the channelwhich first answered. All other calls placed by the Dial app will be hunp upf a timeout is not specified, the Dial application will wait indefinitelyuntil either one of the called channels answers, the user hangs up, or allchannels return busy or error. In general, the dialler will return 0 if itwas unable to place the call, or the timeout expired. However, if allchannels were busy, and there exists an extension with priority n+101 (wheren is the priority of the dialler instance), then it will be the nextexecuted extension (this allows you to setup different behavior on busy fromno-answer). This application returns -1 if the originating channel hangs up, or if thecall is bridged and either of the parties in the bridge terminate the call.The option string may contain zero or more of the following characters: 't' -- allow the called user transfer the calling user 'T' -- to allow the calling user to transfer the call. 'r' -- indicate ringing to the calling party, pass no audio until answered. 'm' -- provide hold music to the calling party until answered. 'd' -- data-quality (modem) call (minimum delay). 'c' -- clear-channel data call (PRI-PRI only). 'H' -- allow caller to hang up by hitting *. 'C' -- reset call detail record for this call. 'P[(x)]' -- privacy mode, using 'x' as database if provided. In addition to transferring the call, a call may be parked and then pickedup by another user. The optionnal URL will be sent to the called party if the channel supportsit. Surajee - Original Message - From: George Lin To: [EMAIL PROTECTED] Sent: Monday, June 02, 2003 1:11 PM Subject: FW: [Asterisk-Users] Call Transfer Problem Hi, Which document describes the Dial with T option ? Could you let me know or email it to me. Thanks, George Lin -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Surajee RatnayakeSent: Sunday, June 01, 2003 9:10 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Call Transfer Problem hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial applicationintructs to use "T" to achieve that. and we learnt that it has not been implemented yet in Asterisk. Is this true? Is some one workin on this issue? if the answer is NO, we can give a try to implement it, with a help of u all , ofcourse :-) (cos, we are quite new to asterisk-only 1 month of experience, but amazed of its great performance) Thank you very much, Surajee
Re: [Asterisk-Users] A Major Problem!
no, we dont have a busydetect=yes line in the zapata.conf, we will put it and giv it a try, btw, what will be the case with an E1 line, will the same problem occur? Surajee - Original Message - From: Michael Bielicki [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 30, 2003 6:51 PM Subject: Re: [Asterisk-Users] A Major Problem! Do you have busydetect set to yes in zapata,.comf ? uou need that for analog lines and you cannot have that for E1 lines :) regards Michael Bielicki On Friday 30 May 2003 1:38 pm, Surajee Ratnayake wrote: hi, we are experiecing the following probem, if anybody have come across such a problem or a solution to this please let us know. our set up is, an Asterisk server equipped with, 4 port station interface card ,single port fxo card and several soft sip phones we have found problems with the following scenarios, outside caller (calling through fxo interface) -- sip phone/ station interface phone calls to a conference outside caller (calling through fxo interface)--- confernce the problem is, once the outside caller(calling through fxo interface) disconnects the line, Asterisk does not detects the disconnection, other party can hear the 'engage like tone' coming from the other side.This continues till the other party(probalby the sip phone or the station interface phone) hangs up. If the fxo user was in a conference if he disconnets the line, other confencees can here the 'engage like tone' , this is very disturbing. The biggest problem is, the fxo line remains busy, till the sip/station phone user disconnects the line. Can anybody give us a solution for this. In the near future, we are going to add some E1 lines too(with E400P cards), once this is done, will the above call disconnection problem occur in that configuration too..or is this a common problem only with analog ? Thank you very much, Surajee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A Major Problem!
yes, that solves the problem, thank you very much, but my other problem remains, will this be a problem when it comes to E1 lines? i am very sorry for keep on asking this - Original Message - From: Surajee Ratnayake [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 30, 2003 7:04 PM Subject: Re: [Asterisk-Users] A Major Problem! no, we dont have a busydetect=yes line in the zapata.conf, we will put it and giv it a try, btw, what will be the case with an E1 line, will the same problem occur? Surajee - Original Message - From: Michael Bielicki [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 30, 2003 6:51 PM Subject: Re: [Asterisk-Users] A Major Problem! Do you have busydetect set to yes in zapata,.comf ? uou need that for analog lines and you cannot have that for E1 lines :) regards Michael Bielicki On Friday 30 May 2003 1:38 pm, Surajee Ratnayake wrote: hi, we are experiecing the following probem, if anybody have come across such a problem or a solution to this please let us know. our set up is, an Asterisk server equipped with, 4 port station interface card ,single port fxo card and several soft sip phones we have found problems with the following scenarios, outside caller (calling through fxo interface) -- sip phone/ station interface phone calls to a conference outside caller (calling through fxo interface)--- confernce the problem is, once the outside caller(calling through fxo interface) disconnects the line, Asterisk does not detects the disconnection, other party can hear the 'engage like tone' coming from the other side.This continues till the other party(probalby the sip phone or the station interface phone) hangs up. If the fxo user was in a conference if he disconnets the line, other confencees can here the 'engage like tone' , this is very disturbing. The biggest problem is, the fxo line remains busy, till the sip/station phone user disconnects the line. Can anybody give us a solution for this. In the near future, we are going to add some E1 lines too(with E400P cards), once this is done, will the above call disconnection problem occur in that configuration too..or is this a common problem only with analog ? Thank you very much, Surajee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A Major Problem!
yes, i will be using PRI E1s, so i hope i won't get any bad luck regarding call disconnection, thank you very much guys for ur quick responses, Surajee - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 30, 2003 7:30 PM Subject: Re: [Asterisk-Users] A Major Problem! On Fri, 2003-05-30 at 08:23, Surajee Ratnayake wrote: yes, that solves the problem, thank you very much, but my other problem remains, will this be a problem when it comes to E1 lines? i am very sorry for keep on asking this Depends on the signalling of your E1 line. If you are doing a PRI(PRA?) you will have absolute signaling that will bring up and down phone lines with no problems. If you are using a RBS type line, it will depend on the type of signalling they are providing to you. - Original Message - From: Surajee Ratnayake [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 30, 2003 7:04 PM Subject: Re: [Asterisk-Users] A Major Problem! no, we dont have a busydetect=yes line in the zapata.conf, we will put it and giv it a try, btw, what will be the case with an E1 line, will the same problem occur? Surajee - Original Message - From: Michael Bielicki [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 30, 2003 6:51 PM Subject: Re: [Asterisk-Users] A Major Problem! Do you have busydetect set to yes in zapata,.comf ? uou need that for analog lines and you cannot have that for E1 lines :) regards Michael Bielicki On Friday 30 May 2003 1:38 pm, Surajee Ratnayake wrote: hi, we are experiecing the following probem, if anybody have come across such a problem or a solution to this please let us know. our set up is, an Asterisk server equipped with, 4 port station interface card ,single port fxo card and several soft sip phones we have found problems with the following scenarios, outside caller (calling through fxo interface) -- sip phone/ station interface phone calls to a conference outside caller (calling through fxo interface)--- confernce the problem is, once the outside caller(calling through fxo interface) disconnects the line, Asterisk does not detects the disconnection, other party can hear the 'engage like tone' coming from the other side.This continues till the other party(probalby the sip phone or the station interface phone) hangs up. If the fxo user was in a conference if he disconnets the line, other confencees can here the 'engage like tone' , this is very disturbing. The biggest problem is, the fxo line remains busy, till the sip/station phone user disconnects the line. Can anybody give us a solution for this. In the near future, we are going to add some E1 lines too(with E400P cards), once this is done, will the above call disconnection problem occur in that configuration too..or is this a common problem only with analog ? Thank you very much, Surajee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another Problem!
hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial applicationintructs to use "T" to achieve that. and we learnt that it has not been implemented yet in Asterisk. Is this true? Is some one workin on this issue? if the answer is NO, we can give a try to implement it, with a help of u all , ofcourse :-) (cos, we are quite new to asterisk-only 1 month of experience, but amazed of its great performance) Thank you very much, Surajee
[Asterisk-Users] A Major Problem!
hi, we are experiecing the following probem, if anybody have come across such a problem or a solution to this please let us know. our set up is, an Asterisk server equipped with,4 port station interface card,single port fxo card and several soft sip phones we have found problems with the following scenarios, outside caller (calling through fxo interface) -- sip phone/ station interface phone calls to a conference outside caller (calling through fxo interface)--- confernce the problem is, once the outside caller(calling through fxo interface) disconnects the line, Asterisk does not detects the disconnection, other party can hear the 'engage like tone' coming from the other side.This continues till the other party(probalby the sip phone or the station interface phone) hangs up. If the fxo user was in a conference if he disconnets the line, other confencees can here the 'engage like tone' , this is very disturbing. The biggest problem is, the fxo line remains busy, till the sip/station phone user disconnects the line. Can anybody give us a solution for this. In thenear future, we are going to add some E1 lines too(with E400P cards), once this is done, will the above call disconnection problem occur in that configuration too..or is this a common problem only with analog ? Thank you very much, Surajee
Re: [Asterisk-Users] Asterisk for call logging.......?
asterisk logs cdrs in a database as well as in a log file, look for /etc/asterisk/cdr_mysql.conf for db details, cdr log file is located in /var/log/asterisk/cdr-csv/ directory Surajee - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, May 28, 2003 11:25 AM Subject: [Asterisk-Users] Asterisk for call logging...? Hi All, I wonder if Asterisk can be setup as a Call Centre and do Call Logging? I want to log all incoming and outgoing calls. What sort of logging mechanisms does it supports? Any help is greatly appreciated. Thanks in advance! Regards, Denzel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
