Re: [Asterisk-Dev] INFO and Duration=250
Doubling the value to 500 did not seem to effect the length of the tone played at allhm. Back to the drawing board for me. Anyone know what this value is supposed to effect? James Sizemore wrote: I did a bit of searching around and found this class in chan_sip.c: I am going to test the Duration at 500, and see how this effect things. If anyone has already played with these values, and had any bad gotchas please let me know. == static int add_digit(struct sip_request *req, char digit) { char tmp[256]; int len; char clen[256]; snprintf(tmp, sizeof(tmp), "Signal=%c\r\nDuration=250\r\n", digit); len = strlen(tmp); snprintf(clen, sizeof(clen), "%d", len); add_header(req, "Content-Type", "application/dtmf-relay"); add_header(req, "Content-Length", clen); add_line(req, tmp); return 0; } == James Sizemore wrote: I have a gateway using a Digium card to convert a PRI call to a sip call then I transport the sip call to a Cisco IAD where it is converted back to a PRI. This all works well except DTMF is sent with a duration of .25sec. PRI specs says this should be .25sec to .5sec so this is with in spec, however the PBX on the other side of the IAD does not reliable work with the DTMF tones the minimum allowable length. I found in the INFO packets where the DTMF is set to a duration or 250, I would like to change this to 500. Which file and class would be the correct place to change this value at? == INFO packet options I would like to change == Content-Type: application/dtmf-relay Content-Length: 24 Signal=5 Duration=250 == ___ ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Bug 4301 - ztdummy accuracy problem
Since the bug has been closed, I sent the question here. I saw patch for bug 4301 has been included in zaptel 1.2-beta1, but with a limitation kernel version >= 2.6.13. Does it mean USE_RTC will not work for kernel version < 2.6.13? I tested meetme with OH323 driver and encountered the increasing delay problem. My kernel is 2.6.9. I removed the kernel limitation by hand and recompile ztdummy, but the delay problem is not solved. Should I upgrade kernel and test again? ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] INFO and Duration=250
On 10/17/2005, "James Sizemore" <[EMAIL PROTECTED]> wrote: > Doubling the value to 500 did not seem to effect the length of the > tone played at allhm. Back to the drawing board for me. > Anyone know what this value is supposed to effect? > >>> I have a gateway using a Digium card to convert a PRI >>> call to a sip call then I transport the sip call to a Cisco >>> IAD where it is converted back to a PRI. This all works >>> well except DTMF is sent with a duration of .25sec. >>> PRI specs says this should be .25sec to .5sec so this >>> is with in spec, however the PBX on the other side of >>> the IAD does not reliable work with the DTMF tones >>> the minimum allowable length. I found in the INFO packets >>> where the DTMF is set to a duration or 250, I would like >>> to change this to 500. >>> >>> Which file and class would be the correct place to change >>> this value at? Since it is the duration of the DTMF that goes INTO the PRI, I would be looking at where that happens - the Cisco config or the Digium config. Don't know a thing about the Cisco - but there is a toneduration in the zapata.conf. Since you say you get a call from the Digium card -> SIP -> Cisco -> PRI I really think you need to look at the Cisco. Brett ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Change codec on running SIP channel
Anyone knows what is the proper way to change the codec of an established SIP channel? Michael. ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] timezone on Cisco phones
Hello, How it is possible to configure cisco's ip phone in order to specifying the correct time zone. For example: the correct hour of Paris. Now the Ipphone use the Universal time given by the NTP Server :-( Following the docs online # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: 192.168.2.100 ; SNTP Server IP Address sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast (default) time_zone: GTM ; Time Zone Phone is in thank you in advance Piero Martino Voiptech - Ideas for Voice Business Unit di Raser srl P.zza vescovio, 7 00199 roma (rm) vat 06997891004 tel. +39 (0) 6.40.40.90.04 mobile +39 -335 -6661098 fax. +39(0) 6 23325757 www.voiptech.it ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Bug 4301 - ztdummy accuracy problem
Are you using the name/record playback option? On 10/18/05, Chih-Wei Huang <[EMAIL PROTECTED]> wrote: BJ Weschke wrote:>> The bug was closed because the ztdummy behavior is not the specific cause > for the delay problem. That patch with USE_RTC was intended to make use of> the real time resource available within the kernel >= 2.6.13 instead of> relying on a OHCI USB resource which was the case previously. But USE_RTC also compile for kernel < 2.6.13..(at least, 2.6.9 in my system)Is kernel >= 2.6.13 necessary?> If you're looking for a "fix" in MeetMe, some have reported success with > applying the patches available in bug 5374.Thank you for the info.I tried that patch.Unfortunately, it seems not improve the increasing delay problemof meetme.Any suggestion or experience for the delay problem? especially using with OH323 channel driver.___Asterisk-Dev mailing listAsterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-devTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Bug 4301 - ztdummy accuracy problem
BJ Weschke wrote: Are you using the name/record playback option? Sorry... I don't understand... Do you mean the meetme options? I only use p option. Here is my dialplan: [macro-conf] exten => s,1,Set(LANGUAGE()=${LOCAL_LANGUAGE}) exten => s,2,MeetMe(${MACRO_EXTEN}|p) and I call the macro by exten => 111,1,Macro(conf) Do you mean the options will affect the delay problem? I'll try other options later... On 10/18/05, Chih-Wei Huang <[EMAIL PROTECTED]> wrote: BJ Weschke wrote: The bug was closed because the ztdummy behavior is not the specific cause for the delay problem. That patch with USE_RTC was intended to make use of the real time resource available within the kernel >= 2.6.13 instead of relying on a OHCI USB resource which was the case previously. But USE_RTC also compile for kernel < 2.6.13.. (at least, 2.6.9 in my system) Is kernel >= 2.6.13 necessary? If you're looking for a "fix" in MeetMe, some have reported success with applying the patches available in bug 5374. Thank you for the info. I tried that patch. Unfortunately, it seems not improve the increasing delay problem of meetme. Any suggestion or experience for the delay problem? especially using with OH323 channel driver. ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev