Re: [asterisk-users] Use of 603 Declined
29 jan 2010 kl. 17.20 skrev Kristian Kielhofner: On Fri, Jan 29, 2010 at 10:31 AM, Kevin P. Fleming kpflem...@digium.com wrote: Well, that's the problem, and it's the reason why 603 is so commonly used. This is a situation where the current request has failed, but there is no indication that repeating the request will also fail. 403 means that the request should not be repeated without either changing it or authenticating as a different entity, which is a different scenario. This is true... Authenticating as a different entity would/could potentially match other peers (causing a 407) and probably isn't technically correct. However, if they didn't match an existing peer (to be challenged or not) using Asterisk's standard peer matching, how did they end up in the nocrackers context anyway? Either way I wasn't considering 5xx responses because of Olle's request. It is very likely that there is no standard-defined 4xx code for 'cannot process this call right now', only the 5xx and 6xx variants. Asterisk has certainly bent standards (which real world implementation hasn't) before. It seems to me that the best reply is the one that's most likely to encourage correct behavior from the far end... 603 almost certainly doesn't do that. In this scenario any forking proxy faced with a 603 coming from Asterisk has to break RFC compliance just to successfully complete the request on another host. Nasty. Are we back to the next-most-generic SIP error, 503 (as originally suggested by Alex)? 503 is definitely more wrong than 603. The 603 is often used when a user presses the red button on a phone and denies the call. After reading through the RFC and the complete list of defined responsed codes on IANA, I haven't found a good alternative. There's no code for call terminated, which is what we want to say. We might have played early media, then terminate the call setup. There are two actions here: - Find the code path and make sure we set a resonable hangup cause. We should have proper hangup causes set on hangups. - Agree on a reasonable error code that doesn't hurt proxy forking and doesn't claim that our server has a problem - which can cause the server to be taken out of load balancing clusters. IANA lists these for our menu: Request Failure 4xx 400 Bad Request 401 Unauthorized 402 Payment Required 403 Forbidden 404 Not Found 405 Method Not Allowed 406 Not Acceptable 407 Proxy Authentication Required 408 Request Timeout 410 Gone 412 Conditional Request Failed[RFC3903] 413 Request Entity Too Large 414 Request-URI Too Long 415 Unsupported Media Type 416 Unsupported URI Scheme 417 Unknown Resource-Priority [RFC4412] 420 Bad Extension 421 Extension Required 422 Session Interval Too Small[RFC4028] 423 Interval Too Brief 428 Use Identity Header [RFC4474] 429 Provide Referrer Identity [RFC3892] 430 Flow Failed [RFC5626] 433 Anonymity Disallowed [RFC5079] 436 Bad Identity-Info [RFC4474] 437 Unsupported Certificate [RFC4474] 438 Invalid Identity Header [RFC4474] 439 First Hop Lacks Outbound Support [RFC5626] 440 Max-Breadth Exceeded [RFC5393] 470 Consent Needed[RFC5360] 480 Temporarily Unavailable 481 Call/Transaction Does Not Exist 482 Loop Detected 483 Too Many Hops 484 Address Incomplete 485 Ambiguous 486 Busy Here 487 Request Terminated 488 Not Acceptable Here 489 Bad Event [RFC3265] 491 Request Pending 493 Undecipherable 494 Security Agreement Required [RFC3329] Now, you have to read the deinitions in the RFCs to understand these, one can't just pick one where the english text sounds reasonable close to what we want to do, since there are logic in various servers that we will affect. 403 is too strong, the server will propably not contact us ever again. Here's something interesting: 21.4.25 487 Request Terminated The request was terminated by a BYE or CANCEL request. This response is never returned for a CANCEL request itself. This is only used in combination with Cancel's, but in this case the call was cancelled by the dialplan, not by the caller. It's a misuse, but a bit clever one. Now, I realize that we also need a setting to indicate whether Asterisk is authorative for a domain or not. If we're the only owners of a domain, we should generate 6xx class errors, if not, 4xx error. So this also applies to 486/600 busy 488/606 and 404/604. If we start separating 4xx and 6xx replies, we might as well do it right. So the domain configuration needs an option per domain whether we're just part of a cluster handling a domain or if we're THE
[asterisk-users] forward call back up same trunk to external cell phone problem
Hi If I have an incoming call coming down a SIP trunk to a particular internal SIP extension- I can answer the extension fine, all works well However, if I change extension.conf from dialling the internal extension to forward the call to an external cell phone (up the same trunk as the incoming leg of the call) I cannot get any audio and get the following error message on the console: [Jan 30 08:38:42] WARNING[27575]: rtp.c:1145 ast_rtp_read: RTP Read too short i.e. change from [voipfone_incoming] exten = s,1,Dial(SIP/203,20,t) to [voipfone_incoming] exten = s,1,Dial(SIP/07123123...@voipfone,20,t) What's wrong?! John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1 Asterisk server, multiple registrations to ITSP
How can I seperate outgoing calls to the same ITSP but with a different DID ?? I have defined a peer in sip.conf for the incoming calls that are routed to a certain [context]. Then in the dialplan this [context] send the incoming calls to the correct context based on the callerID that was send with the incoming call... But how can I send an outgoing call to the ITSP and seperate the outgoing DID's ??? If I define the outgoing DID's [peer] in sip.conf as followed : ; outgoing calls [did1] type=peer host=sip.ITSP username=user1 secret=passwd1 fromuser=user1 [did2] type=peer host=sip.ITSP username=user2 secret=passwd2 fromuser=user2 then incoming calls are matched against these peer-definitions and I get the NOTICE [Jan 29 18:49:07] NOTICE[6314]: chan_sip.c:14703 handle_request_invite: Call from 'did2' to extension 329990102 rejected because extension not found. So how to create multiple outgoing peer definitions without having an incoming call be matched against these peers Thank you ! Jonas. On Fri, 2010-01-29 at 16:51 +, Robert Lister wrote: On Fri, 2010-01-29 at 15:09 +0100, jonas kellens wrote: Hello list ! Having troubles with multiple registrations to one and the same ITSP. This sip.conf : register = user1:pass...@sip.itsp register = user2:pass...@sip.itsp ; outgoing conversations [user1-out] type=peer host=sip.ITSP Try setting type=friend instead of peer for these and see what happens. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Aastra RFP-32 and CLID
Gentlemen, I did borrow an Aastra RFP 32 for some tests that i wanted to do. Everything seems to be working except CLID. Setup as below: DECT handset - GAP - Aastra RFP-32 - SIP - Asterisk - SIP Phone When SIP Phone calls DECT handset, the display on the DECT handset only shows the number of SIP phone, not the name. Did unplug the Aastra RFP-32 and register X-Lite to the same extension. Ofc X-Lite showed both number and name, so it must be me that have missded something in the Aastra RFP-32. :( Any suggestions? Med vänliga hälsningar MAGNUS BENNGRD Direktnr 031-799 89 75 Fältspatsgatan 2 421 30 Västra Frölunda Tel. 031-799 89 00 Fax 031-799 89 01 www.inputinterior.se [1] Links: -- [1] http://www.inputinterior.se inline: 7-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk IPv6 update - we need an update
Friends, Before the Christmas holidays, I did send this letter and did not get a lot of response, but some. Since then, I've been able to get interest from a few parties that are willing to fund parts of this work, including Digium, the main sponsor of Asterisk. I will also apply for additional funding from a foundation here in Sweden and hope to get some more responses so that we can fund this project together. If anyone out there has interest or feedback regarding IPv6, Asterisk and VoIP, I'll be happy to get in contact. I've documented some of my thoughts on how to proceed, based on the work already done by Marc Blanchet (and of course work together with him) on my blog, http://www.voip-forum.com/asterisk/2010-01/voip-users-care-ipv6/ My hope is that we can get this done and integrated in Asterisk 1.8, but that requires some immediate attention from the community, as well as help with testing and feedback when we start rolling. Marcs code is already out there, so you can start testing NOW in your IPv6-enabled network. http://www.asteriskv6.org/ IPv6 is a boring topic, and if you do it right, no one will thank you for it. It just needs to be done. My work with IPv6 started the summer of 1995 and since then people have been shouting We need to migrate now!. We've done that so long so that no one listens any more and now it's getting really critical. The IP numbering authorities, like ARIN and RIPE, have already outlined how they will have to change procedures for IPv4 assignments every six months from now, making it harder and harder to get addresses. For VoIP - sip trunks, calling each other across the Internet, it's critical to have public IP addresses unless you want to stay with your lovely Telco on the other end of the copper cables. Personally, I'm not sure how to design software for this migration properly. In order to educate myself and collegues that develop and build SIP solutions, I'm going to organize an event this spring which combines testing and training. I do hope that the Asterisk community will join me and support the developer team in our efforts to make Asterisk - the leading Open Source PBX - running perfectly well on both IPv4 and IPv6 networks. It needs to be done, we will get it done. And no one will thank us for it, since everyone just expects Asterisk to work as we have done for the last 10 years... With IPv6 greetings! /Olle Vidarebefordrat brev: Från: Olle E. Johansson o...@edvina.net Datum: 17 december 2009 09.39.40 CET Till: Asterisk Non-Commercial Discussion Users Mailing List - asterisk-users@lists.digium.com Ämne: [asterisk-users] Asterisk IPv6 update - we need an update Svara till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Friends, At the first Astricon I was very happy to see Marc Blanchet as one of the attendees. I knew he was one of the IPv6 gurus and I wanted someone to show some interest in Asterisk and IPv6. Well, he did not only get interested in it, but started coding on it. The results have been available for quite some time at http://www.asteriskv6.org/ and Marc has tested it at several SIPits for interoperability. This patch is very large and affects large areas of Asterisk. In order to support IPv6, we need to update the way we interact with sockets, with DNS, with URI's. The SIP channel needs to handle multiple UDP as well as TCP sockets in both protocols. The ACL's we use for all VoIP protocols and manager needs support for IPv6. And much more. Marc hasn't been able to spend time to keep it up to date with the everchanging trunk. I feel we need to move this forward and try to divide the large patch into smaller pieces that can be reviewed separately by the developer team and be merged gradually. First, Marcs branch needs a serious overhaul to get up to date with trunk. In order to work on this, Marc and I needs funding. I have a few interested parties, but need more interested parties that can commit to funding during the first half of 2010 for this project. It's not a small task, the current estimate is at least one month's work for each of us for updating, cutting it up, merging, going through the review process, testing and finalizing with new tests at SIPit or a similar event. If your organization is interested, please let me know off list and we'll discuss from there. My e-mail is as always o...@edvina.net. Please don't hesitate to mail me with any questions you might have about this project. Thank you for your support. Best regards, /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set CDR userfield for Queues
Thereis an option with log queue stored into database. You can do an
cross join from cdr and queue log using callid. With this solution you
can track your call over your asterisk system.
I wrote about this in old post and submit an complete solution.
Regards,
On Sun, Jan 24, 2010 at 1:14 PM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
Yeah, after hours of trying Friday, I got working by a macro.. I didn't like
the outcome using a context, the macro cdr records looked cleaner.
[macro-queue]
exten = s,1,Answer()
exten = s,n,Queue(${ARG1}|rn)
exten = s,n,Set(MEMBERINTERFACE='NOANSWER')
exten = s,n,VoiceMail(${ARG1},u)
exten = s,n,Hangup()
exten = h,1,Set(CDR(userfield)=${MEMBERINTERFACE})
Which is called from:
exten = _?,1,Macro(queue,${EXTEN})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deep D
Sent: Friday, January 22, 2010 11:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set CDR userfield for Queues
I just added a line with 'h'extension.
My dialplan is like this
[mycontext]
exten = s,1,Queue(6000)
exten = h,1,Set(CDR(userfield)=${MEMBERINTERFACE})
On Sat, Jan 23, 2010 at 12:14 AM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
setinterfacevar=yes
Needs to be under each queue
What does your dialplan end up looking like?
I would like to add to mine, and stop running a cron job..
exten = 5000,1,Answer
exten = 5000,n,Queue(5000|rn)
exten = 5000,n,VoiceMail(5000,u)
exten = 5000,n,Hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deep D
Sent: Friday, January 22, 2010 1:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set CDR userfield for Queues
The 'h' extension worked. Thanks.
The other option of 'memebermacro' did not work. On the asterisk
console I could see that the macro is executed and cdr userfield is
set when agent answers the call, but the userfield doesn't show up in
the generated cdr.
Also I had one more question. Doesn't setinterfacevar=yes work when
it is declared in the general section? I had to declare it for each
queues.
On Fri, Jan 22, 2010 at 10:37 PM, Carlos Chavez cur...@telecomabmex.com
wrote:
On Fri, 2010-01-22 at 20:25 +0530, Deep D wrote:
I want to do something like this
exten = 1234,n,Queue(6000,c)
exten = 1234,n,Set(CDR(userfield)=${Agent}) ;; where Agent is the
agent who answered the call
exten = 1234,n,Hangup
Actually because the user will hangup within the Queue application
you
cannot do that. You will have to use the h extension to make the change
to the userfield. Something like this:
h,1,Set(CDR(userfield)=${MEMBERINTERFACE})
Make sure you have setinterfacevar=yes in your queue.conf so that
variable is created when the user is connected to the agent. Another
possibility is to run a macro by using membermacro=somemacro and set
the userfield within that macro. I think that option is only available
on Asterisk 1.6.X and not for older ones though. You can also run an
AGI script (you can set it as an option in the Queue commando) that will
set the userfield as this AGI is run just before the call is bridged to
the agent but the ${MEMBERINTERFACE} is already set.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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[asterisk-users] MATH
I want to create a script for IVR that compiles responses, aggregates
them to a total number.
Then, run an equation based on the result.
Press 1 for X (X is a positive number 500)
Press 2 for Y (Y is a positive number 200)
Press 3 for Z (Z is a positive number 300)
Press 20 to calculate the results
= 500+200+300 =1000
then,
exten = s,n,Read(NUMBER,,1000)
exten = s,n,SayDigits(${NUMBER})
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Re: [asterisk-users] MATH
For all calls combined, or for the current call?
On Sat, Jan 30, 2010 at 2:48 PM, Thomas Perron thomas.per...@gmail.comwrote:
I want to create a script for IVR that compiles responses, aggregates
them to a total number.
Then, run an equation based on the result.
Press 1 for X (X is a positive number 500)
Press 2 for Y (Y is a positive number 200)
Press 3 for Z (Z is a positive number 300)
Press 20 to calculate the results
= 500+200+300 =1000
then,
exten = s,n,Read(NUMBER,,1000)
exten = s,n,SayDigits(${NUMBER})
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Re: [asterisk-users] Use of 603 Declined
Olle E. Johansson wrote: Here's something interesting: 21.4.25 487 Request Terminated The request was terminated by a BYE or CANCEL request. This response is never returned for a CANCEL request itself. This is only used in combination with Cancel's, but in this case the call was cancelled by the dialplan, not by the caller. It's a misuse, but a bit clever one. Yes, I think 487 seems to be a logical choice here; it's very close to what 487 is normatively used for. Now, I realize that we also need a setting to indicate whether Asterisk is authorative for a domain or not. If we're the only owners of a domain, we should generate 6xx class errors, if not, 4xx error. So this also applies to 486/600 busy 488/606 and 404/604. If we start separating 4xx and 6xx replies, we might as well do it right. So the domain configuration needs an option per domain whether we're just part of a cluster handling a domain or if we're THE domain handler. That would be a good idea, yes. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MATH
total up for current call.
then read back the number
2010/1/30 Håkon Nessjøen haa...@avelia.no:
For all calls combined, or for the current call?
On Sat, Jan 30, 2010 at 2:48 PM, Thomas Perron thomas.per...@gmail.com
wrote:
I want to create a script for IVR that compiles responses, aggregates
them to a total number.
Then, run an equation based on the result.
Press 1 for X (X is a positive number 500)
Press 2 for Y (Y is a positive number 200)
Press 3 for Z (Z is a positive number 300)
Press 20 to calculate the results
= 500+200+300 =1000
then,
exten = s,n,Read(NUMBER,,1000)
exten = s,n,SayDigits(${NUMBER})
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Re: [asterisk-users] MATH
Try something like:
exten = 1,1,WaitExten(3)
exten = 1,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten = 1,n,WaitExten(3)
exten = 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
exten = 2,n,WaitExten(3)
exten = 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
exten = 3,n,WaitExten(3)
exten = 9,1,SayNumber(${TOTAL})
Or something. Never used either math or saynumber before, but according to
the documentation, something like this should work..
On Sat, Jan 30, 2010 at 3:06 PM, Thomas Perron thomas.per...@gmail.comwrote:
total up for current call.
then read back the number
2010/1/30 Håkon Nessjøen haa...@avelia.no:
For all calls combined, or for the current call?
On Sat, Jan 30, 2010 at 2:48 PM, Thomas Perron thomas.per...@gmail.com
wrote:
I want to create a script for IVR that compiles responses, aggregates
them to a total number.
Then, run an equation based on the result.
Press 1 for X (X is a positive number 500)
Press 2 for Y (Y is a positive number 200)
Press 3 for Z (Z is a positive number 300)
Press 20 to calculate the results
= 500+200+300 =1000
then,
exten = s,n,Read(NUMBER,,1000)
exten = s,n,SayDigits(${NUMBER})
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Re: [asterisk-users] Help for MOH - sounding scratchy/static on hold
I am also having this issue with the MOH. Would be nice to find a solution! From: Steve Edwards asterisk@sedwards.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Fri, January 29, 2010 3:43:12 PM Subject: Re: [asterisk-users] Help for MOH - sounding scratchy/static on hold On Fri, 29 Jan 2010, Danny Nicholas wrote: Mpg123 works well for us. You have to get your files into mp3 format, but LAME does this simply. Why would you want to compress files when you will have to decompress them again every single time the are used? I'd rather use the CPU cycles to process more calls. Are you in a severely storage challenged environment? You should store all of your audio encoded to match the codec used by the channel. -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] microphone on Polycom 550/650
Yes, the external calls are going over DAHDI. The problem is on the Polycom phones b/c if I pick up the handset, the other end can hear me fine. The problem is when using the hands-free (speakerphone) instead the handset. Here are some similar posting of the same issue. http://www.trixbox.org/forums/vendor-forums-certified/polycom/increasing-speakerphone-tx-gain http://www.trixbox.org/forums/vendor-moderated-forums/polycom/430-sound-volume-gain Most of our phones are IP 550. Where and what do I need to adjust the setting to fix this issue? Any Polycom experts in this mailing list? From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Fri, January 29, 2010 10:18:29 AM Subject: Re: [asterisk-users] microphone on Polycom 550/650 You don’t state this, but the assumption would be that your external calls are DAHDI based, so you might need to tweak txgain in dahdi.conf. From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hin lee Sent: Friday, January 29, 2010 12:08 PM To: Asterisk Users Subject: [asterisk-users] microphone on Polycom 550/650 I have quite a number of users complaining that when they are using handsfree to talk over a landline, the other end can't hear them. It's like the person is speaking 5 feet away and can barely hear their voice. However between internal SIP calls, it's fine. What could be the problem? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] microphone on Polycom 550/650
There's no issue here. There are parameters in the Polycom config files for the various gain settings. The headset, handset and speakerphone volume and mic gains are all separate. They can be tweaked as you like via the config files. Michael --Original Message Text--- From: hin lee Date: Sat, 30 Jan 2010 08:37:24 -0800 (PST) Yes, the external calls are going over DAHDI. The problem is on the Polycom phones b/c if I pick up the handset, the other end can hear me fine. The problem is when using the hands-free (speakerphone) instead the handset. Here are some similar posting of the sa issue. http://www.trixbox.org/forums/vendor-forums-certified/polycom/increasing -speakerphone-tx-gain http://www.trixbox.org/forums/vendor-moderated-forums/polycom/430-sound- volume-gain Most of our phones are IP 550. Where and what do I need to adjust the setting to fix this issue? Any Polycom experts in this mailing list? From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Fri, January 29, 2010 10:18:29 AM Subject: Re: [asterisk-users] microphone on Polycom 550/650 You donât state this, but the assumption would be that your external calls are DAHDI based, so you might need to tweak txgain in dahdi.conf. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hin lee Sent: Friday, January 29, 2010 12:08 PM To: Asterisk Users Subject: [asterisk-users] microphone on Polycom 550/650 I have quite a number of users complaining that when they are using handsfree to talk over a landline, the other end can't hear them. It's like the person is speaking 5 feet away and can barely hear their voice. However between internal SIP calls, it's fine. What could be the problem? -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FAX over ISDN PRI
Hi,
All I just want to be able to detect the fax signail while doing an outbout
call taking advance of the out_dialout feature of asterisk. So for to have a
clear image on how i am doing it.
I have my .call that I move the /var/spool/asterisk/ouotgoing like:
(numbers were changed to preserve privacy)
Channel: Local/9...@dial_out/n
CallerID: Fax Test
MaxRetries: 0
RetryTime: 300
WaitTime: 60
Archive: yes
Context: dial_go
Extension: s
Priority: 1
Set: Q_NAME=511
Set: ANI=
extension.conf
[dial_out]
;
exten = _X.,1,Macro(recordcall,${Q_NAME},${CALLERID(number)}) ;; this is a
custom macro that I created for personal recording puporse
exten = _X.,n,Dial(DAHDI/g1/${EXTEN:0}) ;; this is en sime truck dial
macro that dial into the SPANS - I know it is OK for sure
exten = _X.,n,Hangup()
[dial_go]
;
exten = s,1,Answer
exten = s,n,Wait(2)
exten = s,n,Goto(${Q_NAME},1)
; If none of above happen, send to queue
exten = _s-.,1,Goto(${Q_NAME},1)
;
exten = fax,1,QueueLog(${Q_NAME}|${UNIQUEID}|NONE|FAXDETECTED|${ANI})
exten = fax,n,Hangup()
;
exten = _512,1,Goto(voicemenu-custom-1,s,1)
;
exten = _5[01]X,1,Queue(${EXTEN}${Q_TIMEOUT})
exten = _5[01]X,n,Hangup()
;
chan_dahdi.conf
faxdetect = incoming
I have successfully detected and Answer Machine with AMD application, but as
I experience some dificulties to detect fax I have removed on AMD structure
form the dial plan, only to focus on fax detection. Here is the asterisk
console output when placing a call to fax destination.
[Jan 30 17:49:14] -- Attempting call on Local/9990...@dial_out/n for
s...@dial_go:1 (Retry 1)
[Jan 30 17:49:14] -- Executing [9990...@dial_out:1]
Macro(Local/9990...@dial_out-dda8,2, recordcall|511|9990909) in new
stack
[Jan 30 17:49:14] -- Executing [...@macro-recordcall:1]
GotoIf(Local/9990...@dial_out-dda8,2, 1?5:2) in new stack
[Jan 30 17:49:14] -- Goto (macro-recordcall,s,5)
[Jan 30 17:49:14] -- Executing [...@macro-recordcall:5]
Set(Local/9990...@dial_out-dda8,2,
FILEREC=2010-01-30-17-49-14-SRC-511-DST-9990909) in new stack
[Jan 30 17:49:14] -- Executing [...@macro-recordcall:6]
Set(Local/9990...@dial_out-dda8,2,
FILE_PATH=2010/01/30/2010-01-30-17-49-14-SRC-511-DST-9990909) in new stack
[Jan 30 17:49:14] -- Executing [...@macro-recordcall:7]
Set(Local/9990...@dial_out-dda8,2,
CDR(userfield)=2010-01-30-17-49-14-SRC-511-DST-9990909.wav) in new stack
[Jan 30 17:49:14] -- Executing [...@macro-recordcall:8]
MixMonitor(Local/9990...@dial_out-dda8,2,
/opt/rec/2010/01/30/2010-01-30-17-49-14-SRC-511-DST-9990909.wav|b) in new
stack
[Jan 30 17:49:14] -- Executing [...@macro-recordcall:9]
MacroExit(Local/9990...@dial_out-dda8,2, ) in new stack
[Jan 30 17:49:14] -- Executing [9990...@dial_out:2]
Macro(Local/9990...@dial_out-dda8,2, dialerdial|DAHDI/g1/9990909|511) in
new stack
[Jan 30 17:49:14] -- Executing [...@macro-dialerdial:1]
Dial(Local/9990...@dial_out-dda8,2, DAHDI/g1/9990909) in new stack
[Jan 30 17:49:14] -- Requested transfer capability: 0x00 - SPEECH
[Jan 30 17:49:14] -- Called g2/9990909
[Jan 30 17:49:14] == Begin MixMonitor Recording
Local/9990...@dial_out-dda8,2
[Jan 30 17:49:14] -- DAHDI/32-1 is proceeding passing it to
Local/9990...@dial_out-dda8,2
[Jan 30 17:49:15] -- DAHDI/32-1 is ringing
[Jan 30 17:49:21] -- DAHDI/32-1 answered Local/9990...@dial_out-dda8,2
[Jan 30 17:49:21] -- Executing [...@dial_go:1]
Answer(Local/9990...@dial_out-dda8,1, ) in new stack
[Jan 30 17:49:21] -- Executing [...@dial_go:2]
Wait(Local/9990...@dial_out-dda8,1, 2) in new stack
[Jan 30 17:49:23] -- Executing [...@dial_go:3]
Goto(Local/9990...@dial_out-dda8,1, 511|1) in new stack
[Jan 30 17:49:23] -- Goto (dial_go,511,1)
[Jan 30 17:49:23] -- Executing [...@dial_go:1]
Queue(Local/9990...@dial_out-dda8,1, 511180) in new stack
[Jan 30 17:49:23] -- Started music on hold, class 'default', on channel
'Local/9990...@dial_out-dda8,1'
[Jan 30 17:49:23] -- outgoing agentcall, to agent '10017', on
'Local/3...@default-d45d,1'
[Jan 30 17:49:23] -- Executing [3...@default:1]
Dial(Local/3...@default-d45d,2, SIP/3601) in new stack
[Jan 30 17:49:23] -- Called 3601
[Jan 30 17:49:23] -- SIP/3601-096664f0 is ringing
[Jan 30 17:49:23] -- Agent/10017 is ringing
Thanks to all
Mariano
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Re: [asterisk-users] FAX over ISDN PRI
Mariano Lecuona wrote: All I just want to be able to detect the fax signail while doing an outbout call taking advance of the out_dialout feature of asterisk. So for to have a clear image on how i am doing it. The faxdetect functionality in Asterisk is not intended to detect answering FAX machines; it is for detection of calling FAX machines. The open source NVFaxDetect application may be able to do what you want. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX over ISDN PRI
thanks.. 2010/1/30 Kevin P. Fleming kpflem...@digium.com Mariano Lecuona wrote: All I just want to be able to detect the fax signail while doing an outbout call taking advance of the out_dialout feature of asterisk. So for to have a clear image on how i am doing it. The faxdetect functionality in Asterisk is not intended to detect answering FAX machines; it is for detection of calling FAX machines. The open source NVFaxDetect application may be able to do what you want. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] microphone on Polycom 550/650
I have several models running; IP335, IP430, IP450 IP60. I'm running v3.2.1. I don't have convenient access to the configs at the moment. But if you download the newer firmware from Polycom their stock configs will give you all that you need. http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html Michael --Original Message Text--- From: hin lee Date: Sat, 30 Jan 2010 16:01:19 -0800 (PST) Which model do you have? what version of sip are you running on the Polycom? I am on sip 3.0.2. Can you email me your sip.cfg so I can compare the differences? From: Michael Graves mgra...@mstvp.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sat, January 30, 2010 9:02:44 AM Subject: Re: [asterisk-users] microphone on Polycom 550/650 There's no issue here. There are parameters in the Polycom config files for the various gain settings. The headset, handset and speakerphone volume and mic gains are all separate. They can be tweaked as you like via the config files. Michael --Original Message Text--- From: hin lee Date: Sat, 30 Jan 2010 08:37:24 -0800 (PST) Yes, the external calls are going over DAHDI. The problem is on the Polycom phones b/c if I pick up the handset, the other end can hear me fine. The problem is when using the hands-free (speakerphone) instead the handset. Here are some similar posting of the sa issue. http://www.trixbox.org/forums/vendor-forums-certified/polycom/increasing -speakerphone-tx-gain http://www.trixbox.org/forums/vendor-moderated-forums/polycom/430-sound- volume-gain Most of our phones are IP 550. Where and what do I need to adjust the setting to fix this issue? Any Polycom experts in this mailing list? From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Fri, January 29, 2010 10:18:29 AM Subject: Re: [asterisk-users] microphone on Polycom 550/650 You donââ¬â¢t state this, but the assumption would be that your external calls are DAHDI based, so you might need to tweak txgain in dahdi.conf. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hin lee Sent: Friday, January 29, 2010 12:08 PM To: Asterisk Users Subject: [asterisk-users] microphone on Polycom 550/650 I have quite a number of users complaining that when they are using handsfree to talk over a landline, the other end can't hear them. It's like the person is speaking 5 feet away and can barely hear their voice. However between internal SIP calls, it's fine. What could be the problem? -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium fax - sending fax call file vs manager originate
On Fri, Jan 29, 2010 at 3:09 PM, Hristo Benev hris...@smartbox.ca wrote: If I use call file with spool Fax is send but if I use manager I get Any suggestions? Well, one obvious solution is to just use call file. Problem solved. Try changing your call manager setup to use a Local channel instead, and set up a context that does the dial within that context. That should give you better introspection into where things are failing / what you're doing wrong. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911, location
Thanks very much everybody who contributed their thoughts. I would try to get some DID's so that each physical location can be identified by 911 call Center. Regards Shahnawaz On 2010-01-29, at 2:41 PM, Kevin P. Fleming wrote: Leif Neland wrote: 2: Often callers are answered with an automated message This is 911, please hold, just to give pranksters/misdiallers a chance to hang up before disturbing the operator. Unless 911 records the incoming call right from the start, they will never hear the im-at message. And even if they do, they have to know the message is there to seek on the recording. In the US at least, calls to PSAPs are recorded from the instant the last digit is dialed, before the call is even routed and ringing (on wireline networks where this is possible, anyway). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MATH
thanks for the response.
I tried to simplify and am now tuning the following, but it is not
responding with anything.
something wrong with timing?
here is what I have:
exten = 1625,1,Answer()
exten = 1625,n,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten = 1625,n,WaitExten(3)
exten = 9625,1,Answer()
exten = 9625,n,SayNumber(${TOTAL})
output from the console
[Jan 30 22:25:16] WARNING[22987]: func_math.c:194 math: '' is not a valid number
-- Executing [1...@default:2] Set(SIP/64.85.162.137-c00d10e0,
TOTAL=) in new stack
-- Executing [1...@default:3]
WaitExten(SIP/64.85.162.137-c00d10e0, 3) in new stack
[Jan 30 22:25:19] WARNING[22987]: pbx.c:7855 pbx_builtin_waitexten:
Timeout but no rule 't' in context 'default'
== Spawn extension (default, 1625, 3) exited non-zero on
'SIP/64.85.162.137-c00d10e0'
2010/1/30 Håkon Nessjøen haa...@avelia.no:
Try something like:
exten = 1,1,WaitExten(3)
exten = 1,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten = 1,n,WaitExten(3)
exten = 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
exten = 2,n,WaitExten(3)
exten = 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
exten = 3,n,WaitExten(3)
exten = 9,1,SayNumber(${TOTAL})
Or something. Never used either math or saynumber before, but according to
the documentation, something like this should work..
On Sat, Jan 30, 2010 at 3:06 PM, Thomas Perron thomas.per...@gmail.com
wrote:
total up for current call.
then read back the number
2010/1/30 Håkon Nessjøen haa...@avelia.no:
For all calls combined, or for the current call?
On Sat, Jan 30, 2010 at 2:48 PM, Thomas Perron thomas.per...@gmail.com
wrote:
I want to create a script for IVR that compiles responses, aggregates
them to a total number.
Then, run an equation based on the result.
Press 1 for X (X is a positive number 500)
Press 2 for Y (Y is a positive number 200)
Press 3 for Z (Z is a positive number 300)
Press 20 to calculate the results
= 500+200+300 =1000
then,
exten = s,n,Read(NUMBER,,1000)
exten = s,n,SayDigits(${NUMBER})
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Re: [asterisk-users] beroNet BN4S0e PCI Express ISDN Card with chan_dahdi
On Fri, Jan 29, 2010 at 12:08:19AM +0100, Laurent CARON wrote: Hi, I'm currently trying to get a BN4S0e (which is basically a BN4S0 with a PCIe connector) working with dahdi. The module is loading properly but the card is not detected by the module. Is support on dahdi planned for this card ? Yes, please see https://issues.asterisk.org/view.php?id=16493 Basically the driver needs minimal fixing. Probably just to add the PCI ID to the list. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MATH
what is wrong with this please:
;exten = 4,1,WaitExten(3)
exten = 4,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten = 4,n,WaitExten(3)
exten = 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
exten = 2,n,Waitexten(3)
exten = 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
exten = 3,n,WaitExten(3)
exten = 9,1,SayNumber(${TOTAL})
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