Re: [asterisk-users] Fax Problem on direct FXO port
Hi Steve, Thanks for the reply, I didn't change anything else, just changed spandsp version to de FFA, I use the spandsp version with more success in other places, but in this particular case, sending faxes works ok with both versions but with spandsp I couldn't receive any fax, with FFA I may get 70% of faxes ok. On May 18, 2012, at 1:35 AM, Steve Underwood wrote: Hi Sebastian, has still some issues that not all faxes pass ok, but does the work == still badly broken Your log doesn't seem to show a spandsp error. It looks more like a bad signal. Did you change anything else when you installed FFA? Usually people move the other way to improve their results. Steve On 05/18/2012 09:38 AM, Sebastian Gutierrez wrote: Rusty, thanks for the reply, the issue seems a spandsp issue, I changed to digium free asterisk fax and works much better, has still some issues that not all faxes pass ok, but does the work. thanks! On May 17, 2012, at 1:06 PM, Rusty Newton wrote: Sebastian, Seeing as this an issue related to faxing using the SpanDSP library; if you do not get an answer leading to a solution here, then you may try asking on the SpanDSP mailing list http://lists.soft-switch.org/mailman/listinfo It's likely that the Asterisk users, specifically using SpanDSP, may be on that list. Thanks, Rusty Newton Open Source Community Support Manager Digium, Inc |www.digium.com |www.asterisk.org On 5/16/2012 12:44 PM, Sebastian Gutierrez wrote: Hi, I´m with asterisk 1.6.2.20 DAHDI Version: 2.5.0.2 Echo Canceller: HWEC, MG2 SpanDSP: spandsp-0.0.6pre20.tgz http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.6pre20.tgz FXO lines. Sending faxes works ok. but receiving gives me error: here is the debug: http://pastebin.com/qfFeXWQW any idea?? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 canreinvite
- Original Message - From: Jonas Kellens jonas.kell...@telenet.be To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 18, 2012 3:56:11 AM Subject: [asterisk-users] Asterisk 1.8 canreinvite Hello, is canreinvite still supported in Asterisk 1.8 ?? Yes, but that name is deprecated. The preferred name to use when configuring this feature is 'directmedia'. I read about directmedia being available in asterisk 1.8, but is it the same ?? Yes. What happens when I use canreinvite in Asterisk 1.8 ? The sip.config.sample delivered with Asterisk provides a good explanation of this feature. To summarize, for two devices on a network that can support the behavior enabled by this feature, this setting allows the media stream to be directed directly between the caller and the callee, such that the media stream no longer passes through Asterisk. Kind regards, Jonas. -- Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 canreinvite
On 05/18/2012 03:56 AM, Jonas Kellens wrote: is canreinvite still supported in Asterisk 1.8 ?? I read about directmedia being available in asterisk 1.8, but is it the same ?? What exactly did you read? 'directmedia' is the new configuration option name for 'canreinvite'; they are the *same* feature. If the document(s) you read didn't make that clear, the authors did you a disservice. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer CDRs
Hello, I'm using attended call transfer in asterisk 1.8.11.0 on a CentOS machine. Each CDR entry of calls that are transferred is repeated once. Every field including uniqueid, calldate, billsec, duration, src, dst, channel, dstchannel is exactly the same. Besides adding a constraint in the database table, isn't there any way I can resolve this call transfer cdr duplication issue in asterisk csv cdrs? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Problem on direct FXO port
On 05/18/2012 04:45 AM, Sebastian Gutierrez wrote: with FFA I may get 70% of faxes ok. Nobody that I work with would consider that acceptable. Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best practices to route calls according holidays
Hi, At the moment, I'm mostly using a Day/Night toggle button to let users deal with week-ends, holidays and opening hours. As Asterisk 1.8 introduces Calendar capabilities, I'm wondering if better alternatives now exist. Is it possible, safe, reliable and easy to refer from Asterisk to a public calendar resource listing holidays, for a given country ? Should you instead refer to a private resource, to avoid depending on an externaly managed resource ? If you go this way, which tools would you recommend to build and update a private calendar ? Suggestions ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] special digits * # on sip dial string
hi guys. sorry if this is a silly question. My recharge application uses * digits if the subscriber wants to send some aditional information to speed up a process, dialing something like *777*123*5000 On my old ss7 network works great, but on my new ngn/sip i think it's not possible because somewhere the call is rejected. -On the NGN/Ericsson side engineer say that the call whas deliverd. -On the asterisk side there is no invite shown on debug. Can sip one or more * signs in a dial? What am i doing wrong. thanks in advance.. rv -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best practices to route calls according holidays
On 05/18/2012 07:57 AM, Olivier wrote: Hi, At the moment, I'm mostly using a Day/Night toggle button to let users deal with week-ends, holidays and opening hours. As Asterisk 1.8 introduces Calendar capabilities, I'm wondering if better alternatives now exist. Is it possible, safe, reliable and easy to refer from Asterisk to a public calendar resource listing holidays, for a given country ? Should you instead refer to a private resource, to avoid depending on an externaly managed resource ? If you go this way, which tools would you recommend to build and update a private calendar ? What we determined to work best for our organization was to have a database of holidays that we observe. This allows several benefits. We define which holidays actually cause the offices to be closed. We can also define what time the office closes so half days for Good Friday and New Years Eve when we close at noon. Some departments may not actually close the same hours as others so there are different calendars for different Queues and Auto Attendants. If a holiday falls on a weekend, we may observe it on a different date so if Christmas falls on Saturday, we are closed on Friday. (We all do not close for Christmas Eve when this happens) We have a special holiday called emergency that can be easily triggered remotely in the case of a major event, typically weather, that would force the offices to be closed. All you need for this is a database, we use MySQL, and a way to query that database, we use func_odbc. Management of the database can be command line, phpmyadmin or a custom front end. Dale -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best practices to route calls according holidays
I have not seen a schedule of holidays by country. We usually do is to enter a MySQL database with the holidays in each country. This will have to work in a particular way because, as happened for example with a client who is an embassy, they celebrate the country holidays and also the country where they are. Remember that the same company also has its holiday (anniversary of the company). Regards, -- Ing CIP. Alejandro Celi Mariátegui a...@linux.org.pe http://cipher.pe/web/asterisk.html El vie, 18-05-2012 a las 14:57 +0200, Olivier escribió: Hi, At the moment, I'm mostly using a Day/Night toggle button to let users deal with week-ends, holidays and opening hours. As Asterisk 1.8 introduces Calendar capabilities, I'm wondering if better alternatives now exist. Is it possible, safe, reliable and easy to refer from Asterisk to a public calendar resource listing holidays, for a given country ? Should you instead refer to a private resource, to avoid depending on an externaly managed resource ? If you go this way, which tools would you recommend to build and update a private calendar ? Suggestions ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 50% of time SendDTMF failed
Did you try putting inband parameter in dtmfmode and dtmf of your sip.conf? Regards, -- Ing CIP. Alejandro Celi Mariátegui a...@linux.org.pe http://cipher.pe/web/asterisk.html El mié, 16-05-2012 a las 16:07 +0100, Shahid H escribió: I am having a problem with SendDTMF() - 50% of time it did not succeed. I suspect it is not sending clear DTMF tones to the IVR. For example: SendDTMF(w3w2ww1w4) Sometime digit 3 and 2 work, and failed to do digit 1. Sometime digit 3 work and failed to do number 2. Sometime all went through fine. dtmfmode=rfc2833 are set in the sip.conf file How do I debug to see what went wrong and how to fix? Asterisk 1.8.12.0 Installed on VPS (XEN, CentOS 5.x, 768 MB Ram, 1000 GB B/W - Located in UK) VOIP Provider in UK. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best practices to route calls according holidays
I think there is Google calendar with public holidays listings for nearly every country. At least I know there is one for Malaysia. And Google calendars are available through number of ways I suppose. Sent from my iPhone On May 19, 2012, at 12:35 AM, Ing CIP. Alejandro Celi Mariáteguia...@linux.org.pe wrote: I have not seen a schedule of holidays by country. We usually do is to enter a MySQL database with the holidays in each country. This will have to work in a particular way because, as happened for example with a client who is an embassy, they celebrate the country holidays and also the country where they are. Remember that the same company also has its holiday (anniversary of the company). Regards, -- Ing CIP. Alejandro Celi Mariátegui a...@linux.org.pe http://cipher.pe/web/asterisk.html El vie, 18-05-2012 a las 14:57 +0200, Olivier escribió: Hi, At the moment, I'm mostly using a Day/Night toggle button to let users deal with week-ends, holidays and opening hours. As Asterisk 1.8 introduces Calendar capabilities, I'm wondering if better alternatives now exist. Is it possible, safe, reliable and easy to refer from Asterisk to a public calendar resource listing holidays, for a given country ? Should you instead refer to a private resource, to avoid depending on an externaly managed resource ? If you go this way, which tools would you recommend to build and update a private calendar ? Suggestions ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hangup not detected?
I have and automated call-in dispatch system where hundreds of people call in
daily for 2-3 minutes each. The extension is set up to get their information,
then text-to-speech the dispatch information (via odbc). It then loops 5 times
then ends the call. These calls are being handled by an 8 port analog digium
card.
Sometimes though, I see calls via 'core show channel dahdi/1-1' that have a
time of 16 hours. I'm not sure if this is a result of dahdi missing the
hangup, ODBC timing out, or TTS failing for some reason. When a channel gets
in this state, the call doesn't seem to progress through the dialplan, they
always display the TTS line. Doing a 'dahdi destroy channel 1-1' doesn't seem
to be effective - the only way I've been able to clear the calls is to do a
'dahdi restart' and/or restart the asterisk service.
For TTS I'm using cepstral with the Swift wrapper.
Here is a snippet of my dialplan:
[AAA_27_EMP]
exten = s,1,Answer
same = n,Set(CDR(accountcode)=27_EMP)
same = n,Set(comp_num=27)
same = n,Set(readprompt=AAA/enter_employee_number)
same = n,Set(truck_text=employee number)
same = n,Set(validate_func=AAA_VALIDATE_EMP_NUM)
same = n,Set(get_param1=27)
same = n,Set(get_param2=E)
same = n,Set(read_length=7)
same = n,Goto(DB_LOOKUP,s,1)
[DB_LOOKUP]
exten = s,1,NoOp()
same = n(getid),Read(account_id,${readprompt},${read_length},,3,5)
same = n,Gotoif($[ ${LEN(${account_id})} = 0]?timeout_hangup)
same = n(validateid),Verbose(validating id ${account_id})
same = n,Set(CDR(userfield)=${account_id})
same = n,GotoIf($[${account_id}==*]?AAACompMenu,s,1)
same =
n,Set(ID_VALIDATED=${validate_func}(${get_param1},${account_id}))
same = n,GotoIf($[${ID_VALIDATED}==0]?badid)
same = n(goodid),Verbose(getting schedule for id ${account_id}
AAA_GET_SCHEDULE(${get_param1},${get_param2},${account_id})))
same =
n,Set(ODBC_ID=${AAA_GET_SCHEDULE(${get_param1},${get_param2},${account_id})})
same = n,GotoIf($[${ODBCROWS} 1]?no_schedule)
same = n,Verbose(odbcrows count: ${ODBCROWS})
same = n,Set(COUNTER=1)
same = n,Set(AAA_OUTPUT=Schedule for ${truck_text} ${account_id}: )
same = n,While($[${COUNTER} = ${ODBCROWS}])
same = n,Set(ARRAY(id,data)=${ODBC_FETCH(${ODBC_ID})})
same = n,Set(AAA_OUTPUT=${AAA_OUTPUT}${data}. )
;same = n,Swift(${data})
same = n,Set(COUNTER=$[${COUNTER} + 1])
same = n,EndWhile()
same = n,ODBCFinish()
same = n,NoOp(${get_param2})
same = n,Set(AAA_CHECKED_IN()=${comp_num}, ${get_param2},
${account_id}, ${AAA_OUTPUT}, S, ${CALLERID(num)}, ${CALLERID(all)},
${UNIQUEID})
same = n,Set(MAX_REPEAT=5)
same = n(readschedule),Swift(${AAA_OUTPUT})
same = n,Set(MAX_REPEAT=$[${MAX_REPEAT}-1])
same = n,Gotoif($[${MAX_REPEAT} = 0]?timeout_hangup)
same = n,Read(return_id,AAA/end_of_schedule,${read_length},,,2)
same = n,Gotoif($[ ${LEN(${return_id})} = 0]?readschedule)
same = n,Set(account_id=${return_id})
same = n,Goto(validateid)
same = n(timeout_hangup),Swift(No ${truck_text} entered. Goodbye)
same = n,Hangup()
same = n(badid),Set(AAA_OUTPUT=Invalid ${truck_text} ${account_id})
same = n,Set(AAA_CHECKED_IN()=${comp_num}, ${get_param2},
${account_id}, ${AAA_OUTPUT}, I, ${CALLERID(num)}, ${CALLERID(all)},
${UNIQUEID})
same = n,Swift(${AAA_OUTPUT})
same = n,Goto(getid)
same = n(no_schedule),Set(AAA_OUTPUT=No schedule found for
${truck_text} ${account_id})
same = n,Set(AAA_CHECKED_IN()=${comp_num}, ${get_param2},
${account_id}, ${AAA_OUTPUT}, N, ${CALLERID(num)}, ${CALLERID(all)},
${UNIQUEID})
same = n,Swift(${AAA_OUTPUT})
same = n,Goto(getid)
Thanks in advance
-Justin
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: RTP stats explaination
In our app we do not forward packet immediately. After enough packet received to increase rtp packetization time (ptime) the we forward the message over raw socket and set dscp to be 10 so that this time packets can escape iptable rules. From client side the RTP stream analysis shows nearly every stream as problematic. summery for some streams are given below : Stream 1: Max delta = 1758.72 ms at packet no. 40506 Max jitter = 231.07 ms. Mean jitter = 9.27 ms. Max skew = -2066.18 ms. Total RTP packets = 468 ? (expected 468) ? Lost RTP packets = 0 (0.00%) ? Sequence errors = 0 Duration 23.45 s (-22628 ms clock drift, corresponding to 281 Hz (-96.49%) Stream 2: Max delta = 1750.96 ms at packet no. 45453 Max jitter = 230.90 ms. Mean jitter = 7.50 ms. Max skew = -2076.96 ms. Total RTP packets = 468 ? (expected 468) ? Lost RTP packets = 0 (0.00%) ? Sequence errors = 0 Duration 23.46 s (-22715 ms clock drift, corresponding to 253 Hz (-96.84%) Stream 3: Max delta = 71.47 ms at packet no. 25009 Max jitter = 6.05 ms. Mean jitter = 2.33 ms. Max skew = -29.09 ms. Total RTP packets = 258 ? (expected 258) ? Lost RTP packets = 0 (0.00%) ? Sequence errors = 0 Duration 10.28 s (-10181 ms clock drift, corresponding to 76 Hz (-99.05%) Any idea where should we look for the problem? A maximum jitter of 230 milliseconds looks pretty horrendous to me. This is going to cause really serious audio stuttering on the receiving side, and/or will force the use of such a long jitter buffer by the receiver that the audio will suffer from an infuriating amount of delay. Even a local call would sound as if it's coming from overseas via a satellite-radio link. I suspect it's likely due to a combination of two things: (1) The fact that you are deliberately delaying the forwarding of the packets. This adds latency, and if you're forwarding packets in batches it will also add jitter. (2) Scheduling delays. If your forwarding app fails to run its code on a very regular schedule - if, for example, it's delayed or preempted by a higher-priority task, or if some of its code is paged/swapped out due to memory pressure and has to be paged back in - this will also add latency and jitter. Pushing real-time IP traffic up through the application layer like this is going to be tricky. You may be able to deal with issue (2) by locking your app into memory with mlock() and setting it to run at a real-time scheduling priority. Issue (1) - well, I really think you need to avoid doing this. Push the packets down into the kernel for retransmission as quickly as you can. If you need to rate-limit or rate-pace their sending, use something like the Linux kernel's traffic-shaping features. Is there other network traffic flowing to/from this particular machine? It's possible that other outbound traffic is saturating network-transmit buffers somewhere - either in the kernel, or in an upstream communication node such as a router or DSL modem. If this happens, there's no guarantee that high priority or expedited delivery packets would be given priority over (e.g.) FTP uploads... many routers/switches/modems don't pay attention to the class-of-service on IP packets. To prevent this, you'd need to use traffic shaping features on your system, to pace the transmission of *all* packets so that the total transmission rate is slightly below the lowest-bandwidth segment of your uplink. You'd also want to use multiple queues to give expedited-deliver packets priority over bulk-data packets. The Ultimate Linux traffic-shaper page would show how to accomplish this on a Linux system; the same principles with different details would apply on other operating systems. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: RTP stats explaination
On Fri, 18 May 2012, Dave Platt wrote: A maximum jitter of 230 milliseconds looks pretty horrendous to me. This is going to cause really serious audio stuttering on the receiving side, and/or will force the use of such a long jitter buffer by the receiver that the audio will suffer from an infuriating amount of delay. Even a local call would sound as if it's coming from overseas via a satellite-radio link. Won't a cell-to-cell call experience delays in the 300ms range? Many moons ago I remember listening with a cell while tapping on the table with another cell and being stunned with the magnitude of the delay and that most people manage to carry on conversations without noticing. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] British Telecom ISDN BRI line issues
Good to hear you got it working with Digium's help. One thing: if bri_presistentlayer means that the drivers will force the D-channel to always be up then do not be surprised if BT disables the ISDN port. They don't like it if a customer forces them to power the D-channel all the time at their expense. And with their ISDN team not contactable it may be a bit of a challenge to get them to enable the port again (after you promised not to mess with heir D-channel again...). Hi Patrick, it seems like you have the magic ball, I think what you described is exactly what happened: After we tested the server+ link and we were able to have simultaneous calls (as expected), and knowing that this server was not touched (not even rebooted), it is back not dialing through PTP link.The server remained working from Friday to at least Monday then boom, when I called BT...of course no explanation I am just wondering: their mechanism to miss up things, is it automatic or manual ( I think automatic). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: RTP stats explaination
On 05/18/2012 12:51 PM, Steve Edwards wrote: On Fri, 18 May 2012, Dave Platt wrote: A maximum jitter of 230 milliseconds looks pretty horrendous to me. This is going to cause really serious audio stuttering on the receiving side, and/or will force the use of such a long jitter buffer by the receiver that the audio will suffer from an infuriating amount of delay. Even a local call would sound as if it's coming from overseas via a satellite-radio link. Won't a cell-to-cell call experience delays in the 300ms range? Many moons ago I remember listening with a cell while tapping on the table with another cell and being stunned with the magnitude of the delay and that most people manage to carry on conversations without noticing. Yes, cellular networks have largish latencies, but no jitter. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 50% of time SendDTMF failed
I suggest you put in dtmfmode=auto (so rfc2833 / inband ) can be selected dynamically. Wanted to check with the community if this feature holds true on latest versions of Asterisk ? Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Fri, May 18, 2012 at 10:11 PM, Ing CIP. Alejandro Celi a...@linux.org.pe wrote: ** Did you try putting inband parameter in dtmfmode and dtmf of your sip.conf? Regards, -- Ing CIP. Alejandro Celi Mariátegui a...@linux.org.pe http://cipher.pe/web/asterisk.html El mié, 16-05-2012 a las 16:07 +0100, Shahid H escribió: I am having a problem with SendDTMF() - 50% of time it did not succeed. I suspect it is not sending clear DTMF tones to the IVR. For example: SendDTMF(w3w2ww1w4) Sometime digit 3 and 2 work, and failed to do digit 1. Sometime digit 3 work and failed to do number 2. Sometime all went through fine. dtmfmode=rfc2833 are set in the sip.conf file How do I debug to see what went wrong and how to fix? Asterisk 1.8.12.0 Installed on VPS (XEN, CentOS 5.x, 768 MB Ram, 1000 GB B/W - Located in UK) VOIP Provider in UK. Thanks --_-- Bandwidth and Colocation Provided by http://www.api-digital.com --New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-use http://lists.digium.com/mailman/listinfo/asterisk-usersrs -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BroadVoice Unlimited World PLUS - Dialplan Update (18/May/2012)
We are Broadvoice users some years ago, using Unlimited World PLUS product that seems reasonably acceptable. The problem is that we review Broadvoice config updates a lot of times to make our changes, but they are adding a lot of new countries to the flat rate (which is what alone is all that you want to use to avoid overruns), but the instructions installation are not updated on their website. http://www.broadvoice.com/support_install_asterisk.html It seems that they do in order to use the destinations to which there is no flat rate billing for the additional traffic minutes. That is why we have proceeded to update the Dialplan so they can make use of the new destinations as a flat fee. We will verify the destinations which are not allowed cell phones so they can improve and optimize their Asterisk Dialplan. http://cipher.pe/web/nuestra-experiencia/26-broadvoice-unlimited-world-plus-dialplan-actualizado-18-may-2012.html Hope that you help you Regards, -- Ing CIP. Alejandro Celi Mariátegui a...@linux.org.pe http://cipher.pe -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
