[asterisk-users] Query regarding Pulse Dialing at 20 pps
Hi, I have a query regarding pulse dialing at 20 pps. An Analog Phone is directly connected to the FXS port of Asterisk PBX. When the analog phone pulse-dials at 20 pps, the pulse digits were not decoded correctly by Asterisk. For e.g. when the user dials a 2, Asterisk decodes the pulse digit as 1. Does anyone know how this problem could be solved ? Thank you. regards, Kwang Mien ___ --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] Make/Break ratio for Pulse Dialing
Hi, Does anyone know how I could configure the make/break ratio of pulse dialing in Asterisk ? regards, Kwang Mien ___ --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
Re: [asterisk-users] Make/Break ratio for Pulse Dialing
Thanks for your suggestion. I have compiled according to http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse +dialing dialing at 10 pps works fine with Asterisk with the newly compiled wctdm. but when I dial at 20 pps, the pulses cannot be decoded correctly. I tried changing the make/break ratio but dialing at 20 pps still has the decoding problem. Does anyone have any suggestion ? @0 PPS may not work Check the Wiki first to solve the debounce problem, then recompile. there are also references in the Wiki to make break ratios Some of us who use old rotary phones have difficulty with 10 pps. Seems the Zaptel authors didn't completely do their homework with pulse dial Someone smart with C needs to rework wctdm John Novack Chan Kwang Mien wrote: Hi, Does anyone know how I could configure the make/break ratio of pulse dialing in Asterisk ? regards, Kwang Mien ___ --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 ___ --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] Query on Call Parking
Hi, I understand that 700 is the default extension to initiate a Call Park. Does anyone know of a way to configure Asterisk such that it has more than one park extension for e.g. parkexten = 700,800,900 regards, Kwang Mien ___ --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] Query about Call Detail Record in Asterisk
Hi, My testbed is as follows: sipphone -- Asterisk PBX 1 -- Asterisk PBX 2 -- PSTN -- Analog Phone I understand that one of the fields in the CDR (Call Detail Record) is the Answer field which is the time when call is answered. Is it right that : a) the Answer field of the CDR at Asterisk PBX 1 shows the time when Asterisk PBX 2 answers the call from Asterisk PBX 1 ? b) the answer field of the CDR at Asterisk PBX 2 shows the time when the Analog Phone answers the call from Asterisk PBX 2 ? regards, Kwang Mien ___ --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] hint for Hold
Hi, hint is used to monitor the status channels by using extensions in the dialplan. When an IP phone holds a call, there aren't any extensions sent to Asterisk. Does anyone know how I could monitor Hold ? For example, when an IP phone holds an existing call, the button on the phone that monitors the Hold blinks. regards, Kwang Mien ___ --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] Extension for Incoming Call through Zap Channel
Hi, In my zapata.conf, I have the following lines signalling = fxs_ks context = fromfxs channel = 1 When there is an incoming Zap call at Zap channel 1, the context fromfxs is entered and the entry s extension in the context is executed. Would it be possible to jump to a particular extension in context fromfxs instead of the s extension ? for e.g. when there is an incoming Zap call at Zap channel 1, the 123 extension in the context fromfxs is executed ? Thank you. regards, Kwang Mien ___ --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] Hangup Problem with PSTN and ISDN
Hi, sipphone -- Asterisk PBX -- PSTN -- Cell Phone sipphone sets up a call to Cell Phone. When Cell Phone hangs up, it takes about 1 minute for Asterisk to show the Hangup Zap 1/1 message, after which sipphone hangs up. During the time before Asterisk shows the Hangup message, Busy Tone can be heard at sipphone. Does anyone know why Asterisk took 1 minute to hangup ? Am I right to say that Disconnect Supervision is enabled in PSTN ? Is the Busy Tone generated by Asterisk ? If that is so, then Asterisk must have known that the line is hung up. I conducted another experiment. sipphone -- Asterisk PBX -- ISDN -- PSTN -- Cell Phone sipphone sets up a call to Cell Phone. When Cell Phone hangs up, Asterisk does not hangup at all. From the ISDN messages, it shows that Asterisk receives the Disconnect message and seem to be disconnected from ISDN. However, there isn't any Hangup message shown. There isn't any tone at sipphone when Cell Phone hangs up. Is the ISDN Hangup problem related to Disconnect Supervision ? Thank you. Regards, Kwang Mien ___ --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] Problems with Hangup
Hi, my test-bed is : sipphone -- Asterisk PBX -- PSTN -- Cell Phone sipphone was able to setup a connection to Cell Phone. When sipphone hangs up, Cell Phone also hangs up. However, when Cell Phone hangs up, sipphone was not able to hang up. Could it be that Asterisk was not able to recognise the hangup tone when the Cell Phone hangs up ? Does anyone know what the reason is ? Thank you. regards, Kwang Mien ___ --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] Problems with Codecs in Asterisk
Hi, My test-setup is as follows : sip1 -- Asterisk -- sip2 ^ |--- sip3 In sip.conf, [sip1] type=friend host=dynamic secret=pass disallow=all allow=g729 allow=ulaw [sip2] type=friend host=dynamic secret=pass disallow=all allow=g729 [sip3] type=friend host=dynamic secret=pass disallow=all allow=ulaw sip1 supports g.729 and g.711u only sip2 supports g.729 only sip3 supports g.711u only sip1 is able to establish a call to sip2. However, I have problem establishing a call from sip1 to sip3. sip3 rings but when I answered it, it hanged up. The Logs are : -- Executing Dial(SIP/2006-389a, SIP/2003) in new stack -- Called 2003 Aug 8 09:55:15 WARNING[6937]: channel.c:2725 ast_channel_make_compatible: No path to translate from SIP/2003-b5f8(4) to SIP/2006-389a(256) -- SIP/2003-b5f8 is ringing -- SIP/2003-b5f8 answered SIP/2006-389a Aug 8 09:55:16 WARNING[6937]: channel.c:2725 ast_channel_make_compatible: No path to translate from SIP/2006-389a(256) to SIP/2003-b5f8(4) Aug 8 09:55:16 WARNING[6937]: app_dial.c:1608 dial_exec_full: Had to drop call because I couldn't make SIP/2006-389a compatible with SIP/2003-b5f8 == Spawn extension (phones, 2003, 1) exited non-zero on 'SIP/2006-389a' I think the codecs used by sip3 and sip1 are incompatible. Does anyone know how I could make them compatible ? Thank you. Regards, Kwang Mien ___ --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
RE: [asterisk-users] Problems with Codecs in Asterisk
From the SIP messages exchange, sip1 informs Asterisk in the INVITE message that it supports g.729 and g.711u. Asterisk then compares its first allowed codec which is g.729 with the supported codec by sip1. Since sip1 supports g.729 and it is an allowed codec, Asterisk chooses g.729 as the codec between itself and sip1. Asterisk then forwards the INVITE message but the codec in the INVITE is changed to g.711u. sip3 replied that it supports g.711u in the OK message. Asterisk then realised that the codec between itself and sip3 is different from the codec between itself and sip1. There is a need for transcoding. And since there isn't any g.729 Licence, the connection breaks. In short, once Asterisk is sure that the first codec of the allowed list is supported by sip1, it will use that codec and will ignore the remaining codec, in this case, g.711u. Intuitively, I thought that since sip1 supports both g.729 and g.711u, it should be able to connect to a g.729 phone or a g.711u phone via Asterisk using the same sip.conf. I have the same problem here, why does asterisk not use ulaw with Sip1 - Sip3 ? As it has allow=g729 and allow=ulaw in Sip1, should it not fallback onto ulaw when the g729 fails? Thanks, Dean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Rosli Sukri Sent: 08 August 2006 13:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems with Codecs in Asterisk either 1)pay digium for g.729 license or 2)allow g.729 for sip3 - sip 1 - sip2 work cause it will pass thru, - sip 2 - sip3 fails because since asterisk wants to do transcoding to 729-711 and no license if bandwidth is a concern just use GSM (if available as a codec on the phone) On 8/8/06, Chan Kwang Mien [EMAIL PROTECTED] wrote: Hi, My test-setup is as follows : sip1 -- Asterisk -- sip2 ^ |--- sip3 In sip.conf, [sip1] type=friend host=dynamic secret=pass disallow=all allow=g729 allow=ulaw [sip2] type=friend host=dynamic secret=pass disallow=all allow=g729 [sip3] type=friend host=dynamic secret=pass disallow=all allow=ulaw sip1 supports g.729 and g.711u only sip2 supports g.729 only sip3 supports g.711u only sip1 is able to establish a call to sip2. However, I have problem establishing a call from sip1 to sip3. sip3 rings but when I answered it, it hanged up. The Logs are : -- Executing Dial(SIP/2006-389a, SIP/2003) in new stack -- Called 2003 Aug 8 09:55:15 WARNING[6937]: channel.c:2725 ast_channel_make_compatible: No path to translate from SIP/2003-b5f8(4) to SIP/2006-389a(256) -- SIP/2003-b5f8 is ringing -- SIP/2003-b5f8 answered SIP/2006-389a Aug 8 09:55:16 WARNING[6937]: channel.c:2725 ast_channel_make_compatible: No path to translate from SIP/2006-389a(256) to SIP/2003-b5f8(4) Aug 8 09:55:16 WARNING[6937]: app_dial.c:1608 dial_exec_full: Had to drop call because I couldn't make SIP/2006-389a compatible with SIP/2003-b5f8 == Spawn extension (phones, 2003, 1) exited non-zero on 'SIP/2006-389a' I think the codecs used by sip3 and sip1 are incompatible. Does anyone know how I could make them compatible ? Thank you. Regards, Kwang Mien ___ --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 ___ --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 ___ --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
