Re: [asterisk-users] Rate sheet normalization
Internet. Senseless and merciless A E [Gmail] wrote 29.03.2012 10:28: Wow! ...all the poor guy wanted to know was if there was any tool available for normalization of carrier rate sheets! -- _ -- 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] Rate sheet normalization
On Wednesday 28 March 2012, C. Savinovich wrote: The Way to make money is to help folks use the open source items in the most efficient manner Nobody wants to pay me $2,000 to install and configure A2billing, which in my view, is a fairly low price for my time. There are people who do that for less than $500. I do know the tricks and formulas of how to make money in calling cards, but go tell the customer, they don't care. I don't care either, I am very busy in my consulting business, but it is unfortunate that the better income source for developers just doesn't exist anymore. baldrickSounds like a bag of grapefruits to me!/baldrick -- AJS Answers come *after* questions. -- _ -- 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] Official numbering plan - where to get?
DO you know if the doc files from the ITU are available somewhere for download? l. 2012/3/22 Markus unive...@truemetal.org I hope this is not too off-topic. As a kind-of follow up to rate sheet normalization I'm still trying to figure out a solution for: throw 10 ratesheets at a program and get the cheapest codes/providers as output. For this purpose I believe I need a real, detailed, accurate list of all the dialing codes, incl. mobile codes, city codes etc. worldwide as a reference for that particular program. There are thousands of A-Z lists on the web, and there are thousands of codes in them, but nothing is accurate enough or from an official source. So, I spoke with the ITU today and they, funny enough, too don't have such a list. At least they don't have one that is computer parseable, like a .csv or .xls or something like that. What they have is: a single .doc or .pdf file for EACH country (1 file per country), which is not standardized in its content, with lots of text and descriptions, but it has all the codes. They don't even have such a list as a paid service. Feels like 30 years ago. :) Anyway, there is numberingplans.com which provide exactly what I'm looking for, but they don't support one-time purchases, only subscriptions from around 100 to 990 EUR per month, which is above my budget (and I don't need a subscription). Does anyone have an idea where to find such a list for free, or as a one-time purchase? If not, I'll probably go through the effort to compile my own list based on the ITU data. Let me know in case you want a copy then. :) -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- 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] Collaboration Call Center Integrated with Asterisk web and email
A number of call-centers I see use the pause codes in Asterisk to mark different types of activities, like answering to email or IM. It's not much, but easy to implement. l. 2012/3/27 bilal ghayyad bilmar...@yahoo.com Hi All; Is there a collaboration contact center (hope to be open source) Integrated with Asterisk (hope with vicidial), so the agent will be able to receive chat or emails sessions and deal with the customer. If the agent in a call with the customer, then he will not get chat session. Is there like this software? Regards Bilal -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- 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] Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available - why not?
On Mar 28, 2012, at 4:31 PM, Phil Frost wrote: I'm attempting to direct my queue logs at a PostgreSQL table, and seeing this error in the asterisk console: config.c:2256 find_engine: Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available However, everything I know how to check indicates the odbc engine is available. I can't find any more verbose description of the error from Asterisk, so I'm unsure how to proceed. Well, after making a debug build and stepping through the source I solved it. It wasn't apparent to this neophyte that there's res_odbc.so, and then there's res_*config*_odbc.so, which is set to noload in the default modules.conf. -- _ -- 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] Types of bridging
Hello all, What are the different type of bridging used by asterisk in a SIP call? What is the difference between Packet2Packet bridging, Remote bridging and Native bridging? Can someone please explain me the differences or point me to a good documentation of the same. 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
[asterisk-users] IVR functionalities: saying the numbers, saying the date, ... etc
Hi All; I need to use IVR functionalities in Asterisk, I am asking if there is a ready made thing for some IVR functionalities (like saying the numbers, the date the currency ... etc)? Regards Bilal -- _ -- 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] IVR functionalities: saying the numbers, saying the date, ... etc
core show application saydigits core show application SayUnixTime Or better yet core show applications -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Thursday, March 29, 2012 8:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] IVR functionalities: saying the numbers, saying the date, ... etc Hi All; I need to use IVR functionalities in Asterisk, I am asking if there is a ready made thing for some IVR functionalities (like saying the numbers, the date the currency ... etc)? Regards Bilal -- _ -- 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] Official numbering plan - where to get?
http://www.itu.int/oth/T0202.aspx?parent=T0202 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri Sent: Thursday, March 29, 2012 5:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Official numbering plan - where to get? DO you know if the doc files from the ITU are available somewhere for download? l. 2012/3/22 Markus unive...@truemetal.org I hope this is not too off-topic. As a kind-of follow up to rate sheet normalization I'm still trying to figure out a solution for: throw 10 ratesheets at a program and get the cheapest codes/providers as output. For this purpose I believe I need a real, detailed, accurate list of all the dialing codes, incl. mobile codes, city codes etc. worldwide as a reference for that particular program. There are thousands of A-Z lists on the web, and there are thousands of codes in them, but nothing is accurate enough or from an official source. So, I spoke with the ITU today and they, funny enough, too don't have such a list. At least they don't have one that is computer parseable, like a .csv or .xls or something like that. What they have is: a single .doc or .pdf file for EACH country (1 file per country), which is not standardized in its content, with lots of text and descriptions, but it has all the codes. They don't even have such a list as a paid service. Feels like 30 years ago. :) Anyway, there is numberingplans.com which provide exactly what I'm looking for, but they don't support one-time purchases, only subscriptions from around 100 to 990 EUR per month, which is above my budget (and I don't need a subscription). Does anyone have an idea where to find such a list for free, or as a one-time purchase? If not, I'll probably go through the effort to compile my own list based on the ITU data. Let me know in case you want a copy then. :) -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- 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] Official numbering plan - where to get?
http://www.itu.int/oth/T0202.aspx?parent=T0202 But don't do it. Because I'm doing it right now. So let's not waste energy and do the same task twice. A complete list will soon be available, for free. And then we on this list here will start a web project to keep it updated. I'll let you know once the list is ready. I've already registered opennumberingplan.org :) Am 29.03.2012 11:12, schrieb Lenz Emilitri: DO you know if the doc files from the ITU are available somewhere for download? l. 2012/3/22 Markus unive...@truemetal.org mailto:unive...@truemetal.org I hope this is not too off-topic. As a kind-of follow up to rate sheet normalization I'm still trying to figure out a solution for: throw 10 ratesheets at a program and get the cheapest codes/providers as output. For this purpose I believe I need a real, detailed, accurate list of all the dialing codes, incl. mobile codes, city codes etc. worldwide as a reference for that particular program. There are thousands of A-Z lists on the web, and there are thousands of codes in them, but nothing is accurate enough or from an official source. So, I spoke with the ITU today and they, funny enough, too don't have such a list. At least they don't have one that is computer parseable, like a .csv or .xls or something like that. What they have is: a single .doc or .pdf file for EACH country (1 file per country), which is not standardized in its content, with lots of text and descriptions, but it has all the codes. They don't even have such a list as a paid service. Feels like 30 years ago. :) Anyway, there is numberingplans.com http://numberingplans.com which provide exactly what I'm looking for, but they don't support one-time purchases, only subscriptions from around 100 to 990 EUR per month, which is above my budget (and I don't need a subscription). Does anyone have an idea where to find such a list for free, or as a one-time purchase? If not, I'll probably go through the effort to compile my own list based on the ITU data. Let me know in case you want a copy then. :) -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- 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] IVR functionalities: saying the numbers, saying the date, ... etc
You can take a look at phpivr project - https://sites.google.com/site/grygoriim/devel/phpivr Sent from my iPhone On Mar 29, 2012, at 8:49 PM, Eric Wieling ewiel...@nyigc.com wrote: core show application saydigits core show application SayUnixTime Or better yet core show applications -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Thursday, March 29, 2012 8:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] IVR functionalities: saying the numbers, saying the date, ... etc Hi All; I need to use IVR functionalities in Asterisk, I am asking if there is a ready made thing for some IVR functionalities (like saying the numbers, the date the currency ... etc)? Regards Bilal -- _ -- 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 -- _ -- 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] SIP jitter and packlost channel variables
Hi,
A client of ours get lots of problem with there voice quality when the do a lot
SIP calls.
In a application I log the rtpqos audio jitter an lost packets. (see Below)
Does anybody know what the numbers mean?
If I look at a sample of the channel variables, I see the following number.
local_lostpackets = 7706
local_jitter = 2
local_maxjitter = 11
local_minjitter = 0
..
..
remote_lostpackets = 0
remote_jitter = 0
remote_maxjitter = 7
remote_minjitter = 14000
..
..
The only thing I see is this:
http://www.voip-info.org/wiki/view/Asterisk+func+channel
Regards,
Arjan Kroon
Mobillion BV
exten = s,n,Set(A_SIP_DATA=${CHANNEL(rtpqos,audio,local_lostpackets)})
exten = s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_jitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_maxjitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_minjitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_normdevjitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_stdevjitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_lostpackets)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_jitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_maxjitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_minjitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_normdevjitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_stdevjitter)})
--
_
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Re: [asterisk-users] Rate sheet normalization
Am 28.03.2012 20:17, schrieb C. Savinovich: The Way to make money is to help folks use the open source items in the most efficient manner Nobody wants to pay me $2,000 to install and configure A2billing, which in my view, is a fairly low price for my time. There are people who do that for less than $500. I do know the tricks and formulas of how to make money in calling cards, but go tell the customer, they don't care. I don't care either, I am very busy in my consulting business, but it is unfortunate that the better income source for developers just doesn't exist anymore. I would make much more money if A2billing weren't open source and I would selling copies of my own version of A2billing, probably even developing a better product, definitely with the request that initiated this thread included ;) Why don't you go ahead then and start to code such a program? If you don't have the time personally then you could hire someone and at the same time create a new job in the USA. :) I'm pretty sure there are a bunch of people who would be happy to pay money for a better a2billing. Including myself :) -- _ -- 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] Types of bridging
On Mar 29, 2012, at 08:43 , Deepesh D wrote: What are the different type of bridging used by asterisk in a SIP call? What is the difference between Packet2Packet bridging, Remote bridging and Native bridging? Packet2Packet bridging is when RTP datagrams are forwarded by Asterisk without modification. This imposes little load on the CPU. Obviously this can only happen if both ends are using the same codec, and likely there are likely other less obvious conditions that must be met. Remote bridging happens when Asterisk can direct both ends to send media (RTP probably) to each other directly, by a SIP reINVITE, for example. Only works if both ends have a route to each other, Asterisk is configured to do it, each end shares a codec, and probably a dozen other more subtle conditions are true. In this case there is no load on Asterisk as it's not even in the media path. It also means it can't do things like intercept and act on DTMF or monitor the call. Native bridging is when media is forwarded with Asterisk, but for whatever reason (different codecs, maybe) Asterisk must inspect or modify the stream. Could mean a significant CPU load. -- Phil Frost Macprofessionals office 248-893-0738 direct 248-662-0809 -- _ -- 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] AGI variables being wrong
Greetings! I have the following line in features.conf: parse = *9,peer/both,AGI,/etc/asterisk/agi/map.pl What that script does is parsing AGI variables and doing some things based on them, nothing special. During outgoing call, those variables get messed up. Let's look at an example: number 404 calls 201, it is being routed over PRI line. When 'agi debug' is active, one can see what parameters are being fed to script: AGI Tx agi_request: /etc/asterisk/agi/map.pl AGI Tx agi_channel: Zap/63-1 AGI Tx agi_language: en AGI Tx agi_type: Zap AGI Tx agi_uniqueid: 1322049810.4307 AGI Tx agi_callerid: 044201 AGI Tx agi_calleridname: unknown AGI Tx agi_callingpres: 3 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 0 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: 481 AGI Tx agi_rdnis: unknown AGI Tx agi_context: from_pstn AGI Tx agi_extension: AGI Tx agi_priority: 1 AGI Tx agi_enhanced: 0.0 AGI Tx agi_accountcode: Looking at the callerid and dnid being swapped, one can say that for some reason Asterisk sees this call as incoming from PRI (context kind of approves this). I gave it lots of thinking, and the only conclusion I could come to was - it's because I run my application on 'peer'. But it's not a problem, as I could just swap them back in my script. The problem is, as you can see, our dnid is 481, however we are calling from 404. And moreover - each time I try to call, I get different dnid, like 401, 408 and so on. I thought that it could be last called number on PRI - but it is not. If the call is really incoming (comes from PSTN) - all variables get passed correctly, and my script is happy. When I issue 'show channel' command during active call, I see that variables are incorrect in Asterisk on A-leg (i.e. SIP/404-someid), and variables are correct on B-leg (i.e. ZAP/63-1 in our example). Is that some bug, or misconfiguration, or maybe wrong programming? -- With Best Regards Mikhail Lischuk -- _ -- 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] Types of bridging
Earlier I was using asterisk 1.4 and 1.6. In these version it used to do native bridging and the CPU load was not very high. Now after switching to asterisk 1.8 it has started to do remote bridging and the CPU load has often started to peak. Could this be a configuration issue. I have done the same SIP settings that was earlier there in 1.4 and 1.6. I have 'directmedia=yes' and 'directrtpsetup=yes' in sip.conf and both the peers use the same codecs and there are no nat issues as well Please help On Thu, Mar 29, 2012 at 7:29 PM, Phil Frost p...@macprofessionals.com wrote: On Mar 29, 2012, at 08:43 , Deepesh D wrote: What are the different type of bridging used by asterisk in a SIP call? What is the difference between Packet2Packet bridging, Remote bridging and Native bridging? Packet2Packet bridging is when RTP datagrams are forwarded by Asterisk without modification. This imposes little load on the CPU. Obviously this can only happen if both ends are using the same codec, and likely there are likely other less obvious conditions that must be met. Remote bridging happens when Asterisk can direct both ends to send media (RTP probably) to each other directly, by a SIP reINVITE, for example. Only works if both ends have a route to each other, Asterisk is configured to do it, each end shares a codec, and probably a dozen other more subtle conditions are true. In this case there is no load on Asterisk as it's not even in the media path. It also means it can't do things like intercept and act on DTMF or monitor the call. Native bridging is when media is forwarded with Asterisk, but for whatever reason (different codecs, maybe) Asterisk must inspect or modify the stream. Could mean a significant CPU load. -- Phil Frost Macprofessionals office 248-893-0738 direct 248-662-0809 -- _ -- 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] Rate sheet normalization
On 03/29/2012 03:47 PM, Markus wrote: I'm pretty sure there are a bunch of people who would be happy to pay money for a better a2billing. Including myself :) Have you looked at jbilling.com ? It's F/OSS with commercial support. Regards, Patrick -- _ -- 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] Mute DTMF
On Thu, Mar 29, 2012 at 12:09 PM, Todd Routhier fonema...@gmail.com wrote: I have been breaking my head on this, can't find a solution. Anyone know a way to mute DTMF on SIP? I have already tried changing the dtmfmode option and messing with different codec/dtmfmode settings but so far, not having any luck. It's not an asterisk based solution but you could use RFC2833 signalling and then drop the RTP DTMF packets at your firewall. -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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
