[asterisk-users] iax stresstest client
Hello Everybody, does anyone knows an opensource stresstest client for the IAX protocol, like sipp? Thanx for your answer. Daniel -- _ -- 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] channel variables in AGI
On Wed, 18 Aug 2010, Anthony Messina wrote: For example, see: http://messinet.com/trac/asterisk-fax-gw/browser/fax-gw.agi#L622 Wow. I thought I knew a bit about bash. I made notes on 19* different lines I have no clue what they do. It's going to take me hours to figure these out so I can add them to my repertoire. *) I'm sure there's more nuggets in there but my eyes are glazing over. -- 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
[asterisk-users] Opensource Speech recognition for Asterisk
Hi Everyone, Has anyone got any opensource speech recognition software to work with Asterisk? Please only list WORKING ones. Not the theoretically should work ones! 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
Re: [asterisk-users] Opensource Speech recognition for Asterisk
I yet have to see ANY working speech recognition software, free or not. This technology is nothing more than a joke so far, not practical at any level. As for free, there is nothing decent. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-21 4:31 PM, bruce bruce bruceb...@gmail.com wrote: Hi Everyone, Has anyone got any opensource speech recognition software to work with Asterisk? Please only list WORKING ones. Not the theoretically should work ones! 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
[asterisk-users] Mobile answer machine cut off
We are having some strange issue where a call from asterisk dials a mobile number. If the number answers, we put the call through to an agent SIP phone. All works fine. If, however, the call goes straight through to the mobiles voicemail service *and* the agent phone is a Cisco 79xx, then the call is dropped (from the mobile end) about 1 second into the call. If the SIP phone is an Aastra9133i, then there is no problem. Has anyone seen anything like this ? Thanks Julian -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] Mobile answer machine cut off
Crap, sorry, meant to add that we are running 1.4 svn head Julian On 21 August 2010 23:38, Julian Lyndon-Smith aster...@dotr.com wrote: We are having some strange issue where a call from asterisk dials a mobile number. If the number answers, we put the call through to an agent SIP phone. All works fine. If, however, the call goes straight through to the mobiles voicemail service *and* the agent phone is a Cisco 79xx, then the call is dropped (from the mobile end) about 1 second into the call. If the SIP phone is an Aastra9133i, then there is no problem. Has anyone seen anything like this ? Thanks Julian -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] Opensource Speech recognition for Asterisk
On Sat, Aug 21, 2010 at 6:21 PM, Zeeshan Zakaria zisha...@gmail.com wrote: I yet have to see ANY working speech recognition software, free or not. This technology is nothing more than a joke so far, not practical at any level. As for free, there is nothing decent. I disagree, while not Open Source like the OP requested, both Nuance and Microsoft Speech Server are nothing to laugh at. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] Opensource Speech recognition for Asterisk
Then may be these big multi-billion dollar corporations should use one of them, with whom we all deal regarding various services, and who put us through these voice recognition time-wasting activity in a hope that the poor caller will eventually give up, or will wait painfully long until one of their agent will get time to attend call in person. Your experience could be different and better then most, and you have certainly complete right of your own opinion. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-21 6:57 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sat, Aug 21, 2010 at 6:21 PM, Zeeshan Zakaria zisha...@gmail.com wrote: I yet have to see ANY... I disagree, while not Open Source like the OP requested, both Nuance and Microsoft Speech Server are nothing to laugh at. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Pr... -- _ -- 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] NVidia component out
I realize this is getting a bit outside myth...but hopefully someone can offer some ideas... I'm using the latest NVIDIA drivers on Fedora 12, with Nvidia 8600GT. Although the dual DVI outputs work great, the driver just won't detect anything connected to the component video connector. Is anyone out there successfully using the component video out on their Nvidia card with a recent driver? -- _ -- 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] setting variable for a DID number
jzhehd Ksjfyejzhehd Sent from an unlocked GSM Palm Pre Plus on T-Mobile On Aug 19, 2010 12:50 PM, Tino lt;t...@sparksupport.comgt; wrote: But when i call my DID number following dialplans are being executed.nbsp; What i need is to set a variable with one value for one DID number and set the same variable with another value for another DID number. Also any contexts should be able to use this variable. - nbsp;NoOp(SIP/5070-5407, Received incoming SIP connection from unknown peer to lt;DID NUMBERgt;) in new stack nbsp;nbsp;nbsp; -- Executing [lt;DID NUMBERgt;@from-sip-external:2] Set(SIP/5070-5407, DID=lt;DID NUMBERgt;) in new stack nbsp;nbsp;nbsp; -- Executing [lt;DID NUMBERgt;@from-sip-external:3] Goto(SIP/5070-5407, s|1) in new stack nbsp;nbsp;nbsp; -- Goto (from-sip-external,s,1) nbsp;nbsp;nbsp; -- Executing [...@from-sip-external:1] GotoIf(SIP/5070-5407, 1?checklang:noanonymous) in new stack nbsp;nbsp;nbsp; -- Goto (from-sip-external,s,2) nbsp;nbsp;nbsp; -- Executing [...@from-sip-external:2] GotoIf(SIP/5070-5407, 0?setlanguage:from-trunk|lt;DID NUMBERgt;|1) in new stack nbsp;nbsp;nbsp; -- Goto (from-trunk,lt;DID NUMBERgt;,1) nbsp;nbsp;nbsp; -- Executing [lt;DID NUMBERgt;@from-trunk:1] Set(SIP/5070-5407, __FROM_DID=lt;DID NUMBERgt;) in new stack nbsp;nbsp;nbsp; -- Executing [lt;DID NUMBERgt;@from-trunk:2] Gosub(SIP/5070-5407, app-blacklist-check|s|1) in new stack nbsp;nbsp;nbsp; -- Executing [...@app-blacklist-check:1] LookupBlacklist(SIP/5070-5407, ) in new stack nbsp;nbsp;nbsp; -- Executing [...@app-blacklist-check:2] GotoIf(SIP/5070-5407, 0?blacklisted) in new stack nbsp;nbsp;nbsp; -- Executing [...@app-blacklist-check:3] Set(SIP/5070-5407, CALLED_BLACKLIST=1) in new stack nbsp;nbsp;nbsp; -- Executing [...@app-blacklist-check:4] Return(SIP/5070-5407, ) in new stack nbsp;nbsp;nbsp; -- Executing [lt;DID NUMBERgt;@from-trunk:3] ExecIf(SIP/5070-5407, 0 |Set|CALLERID(name)=Anonymous) in new stack nbsp;nbsp;nbsp; -- Executing [lt;DID NUMBERgt;@from-trunk:4] Set(SIP/5070-5407, __CALLINGPRES_SV=allowed_not_screened) in new stack nbsp;nbsp;nbsp; -- Executing [lt;DID NUMBERgt;@from-trunk:5] SetCallerPres(SIP/5070-5407, allowed_not_screened) P.S : used lt;DID NUMBERgt; in place of actual DID number -- _ -- 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] Opensource Speech recognition for Asterisk
The Lumenvox works fine in my limited use, easy to setup, good dictionary options but it always depends on your circumstance. http://www.lumenvox.com/partners/digium/Asterisk.aspx Most of it is being really careful in planning the customer experience. The technology is secondary to the business analysis focussing on why and what the caller wants and making the most easy and efficient method of getting them there. Voice recognition is a pain for people with accents and poor lines and when people have written bad call flows but by making sure you get someone to an operator really quickly if you can't work out what they said then you can alleviate a few issues. The primary advantage of voice recognition is to give more choice to the caller and route them through more quickly. If you can't do that or don't need that complexity then don't use it Cheers Duncan On 22/08/2010, at 11:09 AM, Zeeshan Zakaria wrote: Then may be these big multi-billion dollar corporations should use one of them, with whom we all deal regarding various services, and who put us through these voice recognition time-wasting activity in a hope that the poor caller will eventually give up, or will wait painfully long until one of their agent will get time to attend call in person. Your experience could be different and better then most, and you have certainly complete right of your own opinion. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-21 6:57 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sat, Aug 21, 2010 at 6:21 PM, Zeeshan Zakaria zisha...@gmail.com wrote: I yet have to see ANY... I disagree, while not Open Source like the OP requested, both Nuance and Microsoft Speech Server are nothing to laugh at. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Pr... -- _ -- 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] Calling Line Identity - any ideas
In simple words , Paddy should go with my trick, that is what i got from this reply Regards On Sat, Aug 21, 2010 at 5:14 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Nasir Iqbal na...@ictinnovations.com wrote: With all honor and respect you deserve, Do I need your permission to express my point of view on community forum ? also it would be quiet helpful for us if you understand well the requirement of post *snip* Nasir, You don't need my permission to post on a public forum...However, neither do I, and I took issue with what you said, and found that your comment about those who are dealing with high load traffic offensive, since it made the assumption that I was just some new guy who deals with hobby/small Asterisk systems and doesn't know what he's talking aboutTherefore, I made it abundantly clear that I wasn't, and that I definitely took issue with that statement. However, I will say that yes, I did mis-take something the OP said... Paddy: Now, here's idea I came up with (haven't tested yet, too busy writing a system for an international interpretation company's telecom needs) First of all, you should have a separate context for outbound calls made by internal extensions... so, in THAT context have code to set the CID to what you wish (you can do logic control and if you're feeling spiffy you can even lookup what CLID to use based on the extension making the call). Second, calls that are being passed from the outside world onto should pass through a different context, performing pretty much the same function... Third, both of THOSE contexts should then pass to a third context that performs the dialout using the multiple targets... Let me know if that works...I know I can make this do what you want, but I'm not trying to do all the work, just point you in a direction, since I get paid to actually do the work ;-) Cheers all, and remember, some of us have been doing this a while, and get grumpy... ;-) there's still no conceivable reason What can be? except performance! (as asterisk has to create one additional leg and bridge it) Which is very conceivable to those who are dealing with high load traffic. And what will be the option, if other outgoing call requires different custom CLI while using the same trunk? 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 First, the reason is, why use a BAD IDEA when there's perfectly good solutions in front of the user There was no mention on this ONE call going outbound over the trunk needing a different CID...the request was as follows: Client needs to call an INTERNAL extension, where the INTERNAL CallerID will be used, and at the SAME TIME, a call to an EXTERNAL number (which would necessitate USING THEIR PROVIDER TRUNK), using the EXTERNAL CallerID Now, p-lease tell me how just configuring the damned trunk's outbound CID is NOT more sensible, efficient, and just friggin' COMMON SENSE TO START WITH...over using a Local channel call, which would require slightly more typing, and using something that I've almost NEVER found a good reason to use, and if you'd care to search the damn archives, you'll see that I was pushing upwards of 5k CONCURRENT CALLS back in 2005, WITH 1.4 Trunk and the RealTime addiiton (which was experimental)... For the love of whatever you find holy and good and true...don't come at me like that...I'm really not in the mood anymore...I put 3-4 solid years of helpjng newbies figure out why shit didn't work, reporting REAL bugs and issues to thew developers and even assisting with some of the fixesI feel entitled (yes, I know that's an asshole thing to say) to a little common respect Now...anyone for a pint? I'm off to vent some frustration with people who jump on the WRONG bandwagon and try to take over Sherwood Mother-F'in' McGowanb... Telecommunications and Tattooing You konw anyone else who combines those two professions? I'd like to buy that guy a drink! -- _ -- 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
Re: [asterisk-users] channel variables in AGI
On Saturday, August 21, 2010 02:19:00 pm Steve Edwards wrote: Wow. I thought I knew a bit about bash. I made notes on 19* different lines I have no clue what they do. It's going to take me hours to figure these out so I can add them to my repertoire. *) I'm sure there's more nuggets in there but my eyes are glazing ove Believe me, I've glazed over the Bash man page for quite some time to get that interface going ;) If you're interested in mail to fax (and back), give it a shot. I could use some testers. Have a good night. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Opensource Speech recognition for Asterisk
On 8/21/2010 6:09 PM, Zeeshan Zakaria wrote: Then may be these big multi-billion dollar corporations should use one of them, with whom we all deal regarding various services, and who put us through these voice recognition time-wasting activity in a hope that the poor caller will eventually give up, or will wait painfully long until one of their agent will get time to attend call in person. Your experience could be different and better then most, and you have certainly complete right of your own opinion. Zeeshan A Zakaria -- www.ilovetovoip.com http://www.ilovetovoip.com On 2010-08-21 6:57 PM, Paul Belanger paul.belan...@polybeacon.com mailto:paul.belan...@polybeacon.com wrote: On Sat, Aug 21, 2010 at 6:21 PM, Zeeshan Zakaria zisha...@gmail.com mailto:zisha...@gmail.com wrote: I yet have to see ANY... I disagree, while not Open Source like the OP requested, both Nuance and Microsoft Speech Server are nothing to laugh at. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com mailto:paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com http://blog.polybeacon.com -- _ -- Bandwidth and Colocation Pr... Zeeshan, You have to figure, Speech is a complex thing. I work at a company that sells ASR system outsourcing, and from using the products, with my run of the mill accent-less American language use, I haven't seen much of a problem, compared to other systems. It is very hard to make a computer understand long and short vowel and consonant sounds as being the same work as the ones said within the parameters of their dictionaries. It is very difficult to develop these especially in languages that the developers are not fluent in. As a side note, most of the BIG multimillion dollar companies outsource their call center functionality. As for our poster, it depends on how much time you want to dedicate to a dictionary set for recognition. If you are willing to spend a bit though, Nuance, and Holly Connect are good products, as well as the mentioned (in another post) Lumenvox. ~Seann smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] NVidia component out
It may be a bit outside Myth, but it's even further outside from Asterisk :) Sorry, can't help. Julian On 22 August 2010 01:33, Michelle Dupuis mdup...@ocg.ca wrote: I realize this is getting a bit outside myth...but hopefully someone can offer some ideas... I'm using the latest NVIDIA drivers on Fedora 12, with Nvidia 8600GT. Although the dual DVI outputs work great, the driver just won't detect anything connected to the component video connector. Is anyone out there successfully using the component video out on their Nvidia card with a recent driver? -- _ -- 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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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
