Re: [asterisk-users] early media issue from phone co.
Edwin, In your outbound context, you need to have the dialplan evaluate the hangupcause variable and send an appropriate message to your callers. Check out the following URL for some samples that you may adapt for your circumstance. http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause If you need more specific assistance, let me know. Sincerely, Trevor Hammonds -Original Message- From: Edwin Lam Sent: Tuesday, June 08, 2010 4:11 PM Subject: [asterisk-users] early media issue from phone co. hi folks. i have the following puzzle: when i call certain cell phone# using a regular phone POTS. the called cell phone co. usually return a message such as phone travel out of range or phone is busy etc. if the phone is unreachable. now when i have the following setup: sip phone - asterisk - PRI - phone co. i call the same cell# and if it's unavailable. the PRI return cause code 31 and hangup, asterisk will then send a SIP BYE to the sip phone and the channel will simply hangup. how do i get the message on the sip phone? -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- 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] play prompt after hanup
On Monday, August 17, 2009, Rilawich Ango wrote: Thanks. DIALSTATUS works except ANSWER. When the phone hangup, the dialplan doesn't go to s-ANSWER. -- Executing [3...@default:12] Dial(SIP/10.31.0.32-09872150, SIP/3001|50|Tt) in new stack -- Called 3001 -- SIP/3001-0986d1d8 is ringing -- SIP/3001-0986d1d8 answered SIP/10.31.0.32-09872150 == Spawn extension (default, 3001, 12) exited non-zero on 'SIP/10.31.0.32-09872150' You need to ensure you specify the g option when you dial the destination (e.g. Dial(SIP/3001,50,Ttg)). Otherwise the call will jump to the h exten when either party hangs up. Sincerely, Trevor Hammonds ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play prompt after hanup
On Friday, August 14, 2009, Rilawich Ango wrote: Hi, Can I play a prompt after hanging up a call? I have tried below but failed. ... exten = s,n,Dial(SIP/1234) ... exten = h,1,Playback(demo-instruct) -- Executing [...@macro-safedial:2] Playback(SIP/3601-09856bf0, demo-instruct) in new stack [Aug 14 17:24:03] WARNING[2496]: file.c:738 ast_readaudio_callback: Failed to write frame -- SIP/3601-09856bf0 Playing 'demo-instruct' (language 'en') Rilawich, If the channel has been hung up, where do you expect the prompt to be played? Sincerely, Trevor Hammonds ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using asterisk on a shared line
Bill Lovett wrote: Can Asterisk be configured to hang up if another phone picks up? I'm a bit lost as far as terminology goes, but here's my setup. At home, I have asterisk answering calls from the pstn and sending them through to a sip extension or voicemail. All that is working fine. The box running Asterisk isn't on 24/7 so I have a secondary phone connected to the line as well. If Asterisk is not running, I can answer an incoming call from that phone. If asterisk is running, I can answer the call from a sip extension. Can I have it both ways? Can Asterisk back off if the secondary phone answers the call? Currently, if a call comes in and I answer it from the secondary phone Asterisk will continue to ring the sip extension and eventually drop into voicemail. Asterisk is a PBX, not an answering machine, so I would advise against this. It would be best to have Asterisk handle the phone line exclusively, 24/7. However, with that said, it is possible to accomplish what you are asking. Placing a telephone privacy/exclusion adapter on the line cord into Asterisk will cut off the phone line whenever a parallel telephone on the same line is picked up. This means that the instant you pick up any other phone on the line, it would cut off the line to Asterisk. Radio Shack used to sell a couple varieties of these. One was a two-way adapter with one side for phone and the other answering machine. You do not need to plug anything into the phone side for the device to work. The second device was just an inline exclusion device. I was unable to find these at Radio Shack's website. However, I found something similar at the following URLs: (See SER2A, SER2D, and SER3P at Sandman.com) http://www.sandman.com/lineshar.html http://www.trianglecables.com/telanmacorph.html http://www.iec-usa.com/cgi-bin/iec/COM9928 http://www.iec-usa.com/cgi-bin/iec/COM0006 Good luck! Sincerely, Trevor Hammonds ___ -- 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] using asterisk on a shared line
Pascal Bruno wrote: Just a little clarification for people refering to Asterisk as a PBX and not an Answering Machine: In fact, Asterisk is neither a PBX nor an Answering Machine. Asterisk is a Telephony Toolkit. You can choose to use it as a PBX or an Answering Machine or both or even in some case as a something different than a PBX or Answering Machine. You should know that already, so this is just a reminder :-) Pascal, I agree with you that Asterisk is a telephony applications toolkit, and not a simple answering machine. However, Asterisk IS a PBX. The term answering machine in the context of this thread implies a device that has only basic answering functionality. Since Asterisk is capable of so much more than this basic functionality, I encouraged the OP to use it full time, rather than as an adjunct device. First line at: http://www.asterisk.org/ Asterisk is the world's leading open source PBX, telephony engine, and telephony applications toolkit. Sincerely, Trevor Hammonds ___ -- 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 to PBX
On Fri, Jul 17, 2009 loganlogan...@gmail.com wrote: Hi, I'm an absolute newbie and wanted to know the following. I want to have a setup where I have a PSTN line connected to my Asterisk box and want to know if it is possible to make more than one simultaneous outbound call through that VoIP gateway? Can Asterisk do this magic of concurrent calls on one PSTN line?? If I put it in other words then can I receive more than one simultaneous call on a PSTN number through Asterisk (the dialplan would forward those calls to different extensions) and the phone line still be able to receive more calls? Do I need some special hardware for the above or a simple SIPURA3000 would be good enough? Please pardon me if this is not the correct list for this question. Thanks. Best Regards, Hitesh Hitesh, The short answer is no. The long answer is that Asterisk (and, indeed, any other PBX) is not be able to make or receive any more analogue calls than the number of available analogue trunks (or lines) to which it has access. You may, however, use an ITSP to send outbound calls over the Internet, leaving your analogue line free for an incoming call. You may also be able to subscribe to Busy Call Forwarding on your POTS line. This way, when your analogue line is in use, additional inbound calls may be forwarded to a DID (telephone number) provided by an ITSP -- essentially increasing the number of lines available to Asterisk. Sincerely, Trevor Hammonds ___ -- 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] PRI hunt group
On Thursday, July 16, 2009, Alex Balashov wrote: C F wrote: If you don't want to port it to the PRI for whatever reason you can convert it to a RCFW (remote call forwarded number) which is around $15.00 plus $8.00 for each additional channel again pricing is for here in Verizon land. Is that true even if the number is out of a rate center that is billed long-distance relative to the destination (but still intra-LATA)? Or do you pay normal LD rates on top of all that in the intra-LATA LD scenario? Alex, Calls forwarded via Remote Call Forwarding are just like calls forwarded from a metered business or residential POTS line. If the destination to which you have selected to forward calls is normally a local call, you will just incur the standard metered call rate. If the call is normally a local toll charge (within the same LATA), you will incur toll charges from the LEC. If the call is long distance, you will need to select an IXC -- who will bill just as if the calls were made from a POTS line. Sincerely, Trevor Hammonds ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Possible WaitUntil Bug
I am having trouble with the WaitUntil application in Asterisk
SVN-branch-1.6.1-r206879. I believe this is a bug, but would like
confirmation.
The relevant dialplan entry is:
exten = 8765,n,WaitUntil(${FutureTime})
The console indicates the following notice:
[Jul 17 17:13:57] NOTICE[4609]: app_waituntil.c:72 waituntil_exec:
WaitUntil called in the past (now 1247876037, arg 1247876040)
After this, the dialplan continues to the next entry without having waited.
My C programming skills are very basic, but looking at
app_waituntil.c, I do not see anything that actually waits for the
appropriate Unix epoch as in the original patch. However, the
function that evaluates if the argument is called in the past is in
error as 1247876037 1247876040.
Original patchl:
https://issues.asterisk.org/view.php?id=11487
I would appreciate if anyone else is able to confirm this error.
Sincerely,
Trevor Hammonds
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[asterisk-users] AGI to announce temperature from weather.com XML file
I would like to have the ability to have Asterisk announce the temperature -- not using TTS -- within the dialplan. For a non-Asterisk project, I have a cron job that periodically pulls down an XML file from weather.com containing local weather data (TWC's user agreement requires that data be cached locally). Using sed, I also create a text file that contains only the numeric value of the current temperature, created from that XML file (e.g. tmp65/tmp in the XML file becomes a text file with 65 as its only contents). I am hoping someone on the list has an example of a lightweight AGI script that I may modify to either read the simple text file and set a dialplan variable to the current temperature, or hopefully a more-sophisticated one which will parse the XML file to set the dialplan variable. The end goal is to have Asterisk play the speech files temperature sixty five degrees or the equivalent non-English files per the channel's current language setting. Thank you. Any assistance will be greatly appreciated. Sincerely, Trevor Hammonds ___ -- 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] Using the PBX Directory from a Blackberry
From: Tzafrir Cohen Sent: Thursday, July 02, 2009 11:47 PM On Thu, Jul 02, 2009 at 09:02:05PM -0700, Trevor Hammonds wrote: Anyhow, on the Blackberry, when you hold down the Alt key and press the alpha character, the device sends out the correct digit tone associated with that character, like on a regular phone keypad. Any chance of answering the OR? There is no DTMF tone associated with digits (over D). So they must be using something else. Any idea what? The BlackBerry sends the DTMF of the number normally associated with the letter as on a standard touch tone keypad. (e.g. 2 for A, B, and C; 3 for D, E, and F; etc.). For the special case of Q and Z (not originally assigned by Bell), 7 is sent for Q and 9 for Z. Sincerely, Trevor Hammonds ___ -- 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] Using the PBX Directory from a Blackberry
From: Tzafrir Cohen Sent: Friday, July 03, 2009 12:41 AM On Fri, Jul 03, 2009 at 12:30:07AM -0700, Trevor Hammonds wrote: From: Tzafrir Cohen Sent: Thursday, July 02, 2009 11:47 PM On Thu, Jul 02, 2009 at 09:02:05PM -0700, Trevor Hammonds wrote: Anyhow, on the Blackberry, when you hold down the Alt key and press the alpha character, the device sends out the correct digit tone associated with that character, like on a regular phone keypad. Any chance of answering the OR? There is no DTMF tone associated with digits (over D). So they must be using something else. Any idea what? The BlackBerry sends the DTMF of the number normally associated with the letter as on a standard touch tone keypad. (e.g. 2 for A, B, and C; 3 for D, E, and F; etc.). For the special case of Q and Z (not originally assigned by Bell), 7 is sent for Q and 9 for Z. Errr.. missed that. Isn't that exactly what app_directory exdxpects, anyway? Yup! I believe the original poster was just pointing out that you CAN use a BlackBerry if you know how to send the digits (Alt+letter). -- Trevor ___ -- 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] Using the PBX Directory from a Blackberry
From: Darrick Hartman Sent: Thursday, July 02, 2009 8:06 PM On 07/02/2009 10:14 AM, Tzafrir Cohen wrote: On Thu, Jul 02, 2009 at 09:53:18AM -0500, JR Richardson wrote: Hi All, A couple of customers called complaining that folks were dialing into their PBX trying to use the Directory to locate users, from a Blackberry, and getting frustrated due to the incompatibility of dialing alpha characters on the the qwerty keyboard and not getting through. The issue of course is the Directory application only recognizes numeric digit tones, not alpha characters (not sure is there is actually tones generated when the alpha characters are pressed, it just doesn't work). Anyhow, on the Blackberry, when you hold down the Alt key and press the alpha character, the device sends out the correct digit tone associated with that character, like on a regular phone keypad. Is it a tone? Or the letter itself in SIP / RTP signalling? This is a 'bug' or 'feature' of blackberry phones. The phones switch the keypad to numeric when in a phone call. You need to memorize abc=2, def=3... Sure would be nice if there was an option to send the DTMF for 5 when pressing the alpha key j k or l, but I don't believe this is possible. You do not need to memorize the alpha-to-numeric conversion on a BlackBerry. The OP mentioned the solution: Anyhow, on the Blackberry, when you hold down the Alt key and press the alpha character, the device sends out the correct digit tone associated with that character, like on a regular phone keypad. Sincerely, Trevor Hammonds ___ -- 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] music on hold file formats
On Tue, Jun 23, 2009 at 5:40 PM, Ron nha...@gmail.com wrote: I have a portal where a user can upload their own MP3, but when a user is using a g729 codec, the music on hold has a crackly sound. using g711 it's very clear. That could be for any number of reasons, including a overly lossy mp3 to begin with. the mp3 moh is very clear when i use g711, only on g729 i'm having the issue, could that be an issue of lossy MP3 still? Ron, Keep in mind that G.729a is a low bit-rate CODEC optimized for speech compression. It does not do well with music, or even DTMF tones. G.711, on the other hand, has greater dynamic range, so music sounds better. It would be nice if Asterisk had an option like CCM that forces MoH to use G.711, while the voice calls still use G.729. Sincerely, Trevor Hammonds ___ -- 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] Understanding Call Handling In Asterisk
On May 29, 2009, Varun Rapelly varun.rape...@spectross.com wrote: Hi, I am a newbie to Asterisk; need help understanding three-way conferencing call-transfer features implemented over standard extensions i.e. on a TDM800P card (4 FXO + 4FXS) In Asterisk I have observed that if an extension is already participating in an active call (e.g. Ext A Ext B communicating): 1. An incoming call to one of these active extensions would be presented with call-wait beeps (e.g. Ext A receives call-wait beeps as Ext C is attempting to call Ext A). 2. The call waiting may be answered by pressing Hook-Flash, placing the previously active call on hold (e.g. C answered; A C communicate; B placed on hold). 3. The calls could be toggled by subsequent Hook-Flash's (e.g. A B communicate; C placed on hold). Yes, this is normal behaviour on pretty much every analogue PBX or telco switch. Queries: 1. If the extension which received call-wait beeps hangs-up then the call waiting/the call placed on hold returns as a new call. I was expecting the call to be transferred (A hangs-up, B C communicate), how could the call be transferred? I expected this feature to be available in Asterisk as this is a very normal feature available on any PBX and used extensively in Call Transfer. When you transfer a call, the person initiating the transfer has to be MAKING a call. Example: Ext A receives a call from Ext B. Ext A wants to transfer the call to Ext C. Ext A puts the first call on hold with a hook flash, dials Ext C, then either waits for the Ext C to answer and announces the transfer (e.g. an attended transfer) OR simply hangs up as soon as the call to Ext C starts ringing (e.g. an un-attended or blind transfer). The behaviour you explain is not something available on any switch that I am aware of, and would be highly problematic if it were. If this feature were available, you could get a circumstance where two people who are calling you end up being bridged together on a call, unknown to you. As a bad example, your wife and your girlfriend end up talking to each other because you hung up while one of them call-waited you while you were talking to the other. 2. How could the extension that received call-wait beeps initiate a three-way conference with the other extensions (A, B C in three-way conference)? I expected this feature to be available in Asterisk as this is a very normal feature available on any PBX and used extensively in 3-way Call Conference. Again, this is NOT a feature available on any analogue PBX that I am aware of. If it were, this would, again, mean that you may get unwanted parties connected together. With the above example, you answer your girlfriend's call while talking to your wife, and all three of you end up in the same conference. Unfortunately, POTS lines do not handle transfering multiple inbound calls very well (with call waiting). This is not an Asterisk issue, POTS lines were not designed to do anything other than handle a single call at a time. You may be able to handle transfering a call-waited call with DTMF signalling. I am certain someone else on the list will be able to give you a definitive answer on that. Sincerely, Trevor Hammonds ___ -- 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] Problem with Free Fax For Asterisk
Kevin P. Fleming kpflem...@digium.com wrote: Trevor Hammonds wrote: New development: I've assigned an external DID to the fax extension, and fax calls come in fine, with a strange burst of noise about one second into the preamble. However, I am still unable to transfer an inbound call from another DID to the fax extension. OK, so transfers are involved; that could have some effect on the audio path configuration, especially if transcoding was in use for the voice call before it was transferred. It was all G.711u (ulaw) from the SIP provider to the phone. It looks as though Asterisk is having a problem converting the fax (SLIN) to ulaw, for some reason. ___ -- 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] Problem with Free Fax For Asterisk
I have installed Free Fax For Asterisk 1.6.1_1.0.8 on Asterisk SVN-branch-1.6.1-r194876M and am getting a peculiar error message. Current scenario, while testing, is to receive an inbound fax call via a regular DID and then transfer the call to the internal fax extension. When I do so, I receive the following error message on the console, several times: WARNING[2895]: chan_sip.c:5369 sip_write: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x40 (slin)(64)/0x4 (ulaw)(4) The incoming caller does not hear the fax tones. When I call the fax extension directly, internally, there appears to be no trouble (I hear the fax preamble tones). Any ideas? Also, the documentation is a little unclear on one issue. I would like to know if this version of FFA (1.6.1_1.0.8) will work with Asterisk 1.6.2. Thank you. Sincerely, Trevor Hammonds ___ -- 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] Open source SIP client
Perhaps DHAVAL INDRODIYA should just click here: http://tinyurl.com/rbddau Sincerely, Trevor Hammonds On Mon, May 18, 2009 at 3:36 AM, Alex Balashov abalas...@evaristesys.com wrote: What on earth are you talking about? What open source client? What is your requirement? What exactly is presumed under the operative verb give? Have you considered investigating more deeply the basic mechanics of the written English language? Or do you simply have not the foggiest clue what these collections of syllables intended to convey meaning -- words -- mean? DHAVAL INDRODIYA wrote: hi all, can anybody help me to give Opensource SIP client information which can be modified as per our requirment ___ -- 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] Problem with Free Fax For Asterisk
Kevin P. Fleming kpflem...@digium.com wrote: Trevor Hammonds wrote: I have installed Free Fax For Asterisk 1.6.1_1.0.8 on Asterisk SVN-branch-1.6.1-r194876M and am getting a peculiar error message. Current scenario, while testing, is to receive an inbound fax call via a regular DID and then transfer the call to the internal fax extension. When I do so, I receive the following error message on the console, several times: WARNING[2895]: chan_sip.c:5369 sip_write: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x40 (slin)(64)/0x4 (ulaw)(4) WARNING != ERROR :-) I get that. Sorry for the misstatement. :-) In spite of that, are there *no* other messages before that which would indicate a problem setting up a translation path or similar problems? The only console messages are related to the FAX setup and the call transfer event. New development: I've assigned an external DID to the fax extension, and fax calls come in fine, with a strange burst of noise about one second into the preamble. However, I am still unable to transfer an inbound call from another DID to the fax extension. Also, the documentation is a little unclear on one issue. I would like to know if this version of FFA (1.6.1_1.0.8) will work with Asterisk 1.6.2. Asterisk 1.6.2 is still in beta, so its APIs are still subject to change. FFA for Asterisk 1.6.1 *may* work fine with it, or may not. Once Asterisk 1.6.2 reaches the release candidate state, if new FFA modules are required then they will be produced and the FFA download selector page will be updated to reflect that. Thanks for the clarification and your assistance. Sincerely, Trevor Hammonds ___ -- 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
