Re: [asterisk-users] Call in progress tones
On Thu, May 28, 2009 at 12:10 AM, David Backeberg dbackeb...@gmail.com wrote: On Tue, May 26, 2009 at 8:46 PM, Mikel Lindsaar raasd...@gmail.com wrote: Does anyone know of a way to have tones played during the call progress stage of the call? You could detect what was dialed and route accordingly, and have a caveat play before the dial to those particular calls instead. Well.. no, what I want is to be able to play a specific tone before the call is established. For example, Dial app allows you to specify r to force ring tone, even though the other end is not ringing yet, how do I override that and only provide a specific tone (not a ring tone) until the other end is busy, or congested, or rings.? Mikel ___ -- 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] Call in progress tones
Hello all, I've played with background and play sounds apps and googled around and asked the list before to no avail. Does anyone know of a way to have tones played during the call progress stage of the call? We (especially on some international circuits) get up to 5 seconds of silence before the phone starts ringing or is busy. I don't want to force R on the Dial app as then you can get ring ring, ring ring, rin, beep beep beep when the phone is busy. What I want is something to play during the call setup or making progress stage of the call. just a series of beeps about 800ms apart until the phone call is actually set up... so then you would get something like bip, bip, bip, bip, bip, ring ring, ring ring... for ringing or bip, bip, bip, bip, bip, beep, beep, beep... for busy. Any ideas? Mikel ___ -- 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] Making use of SIP making progress messages
Hi all, Is there any way to make use of the SIP making progress messages? I find that about the time the SIP peer says making progress is the time the other end actually starts to ring, or is busy etc. Before that time, I want to generate an in progress tone using playtones to let the user know that their call is doing something. I need to be able to call stop tones once the in progress call is received. Any ideas? Mikel ___ -- 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] Recognizing the making progress notification
Hello all,
I have an extension like this in my test box:
exten = _,1,Answer()
exten = _,n,Playtones(350/100,0/750)
exten = _,n,Dial(SIP/${EXTEN})
The idea is that occasionally the SIP peer might take a few seconds to
connect the call, and this provides an in progress sound instead of
silence.
I don't want to provide a ring tone followed by a busy tone etc. I
feel that this confuses users.
What currently happens though is that Playtones continues until Dial
actually connects the call... if I implemented this, the user would
never hear the ring tone, which is disconcerting.
What I am not sure how to do is how to capture that the sip peer is up
to making progress and then switching to playing a dial tone, busy
tone, congested, etc.
I thought about using a goto after Dial and jump to ANSWERED, or BUSY,
etc... but this doesn't allow the end caller to hear the ring tone and
only handles it for BUSY or CONGESTED.
How do I go about doing this?
Thanks in advance
Mikel
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[asterisk-users] Question on phone line pass through
Hi all, I have a TDM2400 full of FXS cards and a TDM800 full of FXO cards. If I have a fax machine on the FXS port dialing out through asterisk on the TDM800 FXO, should I be expecting any problems? Or should this just work as expected? (ie, flawlessly with the asterisk box essentially transparent to the whole operation). I am doing it this way to allow many faxes and modems to share a dial out pool. Mikel -- http://lindsaar.net/ Rails, RSpec and Life blog ___ -- 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] CISCO 7940 United_States/7960-tones.xml
I have a smartnet contract for this phone, and have searched high and low for this file on the Cisco website. I need: United_States/7960-tones.xml English_United_States/7960-font.xml Every road seems to lead to the Call manager express downloads... I don't have a CME, so that's basically useles. Can anyone point me in the right direction? Mikel ___ -- 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] CISCO 7940 United_States/7960-tones.xml
Thanks Mark, The phone starts, I can get to settings and such... just it keeps looking for this file. I just opened a TAC ticket and getting it handled. Mikel On Thu, Jan 8, 2009 at 3:26 AM, Mark G. Thomas m...@misty.com wrote: Mikel, On Thu, Jan 08, 2009 at 12:52:02AM +1100, Mikel Lindsaar wrote: I have a smartnet contract for this phone, and have searched high and low for this file on the Cisco website. I need: United_States/7960-tones.xml English_United_States/7960-font.xml Every road seems to lead to the Call manager express downloads... I don't have a CME, so that's basically useles. Can anyone point me in the right direction? Those files aren't directly included in the CME downloads. I think their contents must be included in the binary phone load or internal to CME. Using CME, if one sets cnf-file location flash:, then does a create cnf-files, they are then written out to the CCME flash, however they default to being on system:, not the tftp server flash. Have you tried resetting your phone to factory defaults -- **# to unlock the settings menu? You might not actually need these files. Mark -- Mark G. Thomas (m...@misty.com) http://mail-cleaner.com/ ___ -- 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 -- http://lindsaar.net/ Rails, RSpec and Life blog ___ -- 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] chan_sccp and CISCO CP-7914 Module
Hello list, Has anyone on list had experience with getting the 7914 module working with Asterisk, probably using the chan-sccp drivers? I have 7912 and 7940 phones working OK, I need something for reception, and I was thinking a 7960 with two of these modules would handle it, but need to know that someone has got it working somewhere (seeing lines that are free, busy etc) Mikel ___ -- 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] Good comparisons on cheaper VOIP phones
Thanks all for your replies. I have an aastra 9133i here for testing and am getting a polycom 320 to try out. But today, I got my hands on an older Cisco 7912G with SIP software installed. It connected fine to the Asterisk box, works with the PoE stuff I have, sounds good and doesn't seem to have any problems. Best all, I can buy near new for about $60 each in Australian dollars (thats about 45USD with the Aussie dollar being what it is :) The handsets look OK, they are nice and solid feeling and very easy to use / not complex. Any reason not to use the 7912G ? Seems with the SIP image they work just dandy... Mikel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound fax issues
On Tue, Dec 23, 2008 at 1:28 AM, Danny Nicholas da...@debsinc.com wrote:
What does your extensions.conf look like for this call? If you can
insert a ww into your Dial command (ie, change 18005551212 to ww18005551212)
this may improve your dialing behavior.
In an attempt to isolate the problem, I reduced the extensions.conf to:
; Fax Lines
exten = _.,1,Dial(${AAPT}${EXTEN},,R)
What does putting ww at the front do?
Regards
Mikel
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Re: [asterisk-users] Good comparisons on cheaper VOIP phones
On Tue, Dec 23, 2008 at 11:01 PM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: We have one 7912 which we bought for evaluation. The main drawback is that it has hands free speaker but no microphone. That's true. But we will be getting higher models for the speaker function. Did you find or know of a way to do paging with the Cisco 7912G ? Looking around on Google didn't come up with much. Mikel ___ -- 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] Outbound fax issues
On Wed, Dec 24, 2008 at 12:02 AM, Barry L. Kline blkl...@attglobal.netwrote: What does putting ww at the front do? Each w makes Asterisk wait a 1/2 second before sending the DTMF to dial. (It may be a 1/4 second each 'w') I thought so, in that case, it is not the problem here. My problem is that the fax dials out, connects, but can't handle some perceived line noise Need to work out or find out what the best settings are for a fax machine connected to a VOIP device per my diagram previously... Mikel ___ -- 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] Outbound fax issues
Hello all. I have the following setup: Fax machine | Sipura SPA-3120 | SIP 100BaseT | Asterisk 1.4 | IAX2 100BaseT | Asterisk 1.6 | ISDN PRI TE210P | Traditional Telco The fax lands on the Internal Asterisk 1.4 box, the sip config for this extension looks like: [35081] type=friend secret= qualify=yes port=5060 nat=no host=dynamic dtmfmode=rfc2833 dial=SIP/35081 context=fax-line canreinvite=no callerid=device 35081 disallow=all allow=ulaw allow=alaw Now, inbound faxing (from the other side of the Telco to me) is working and from what I can tell, never fails to receive. Sending though is a bit touch and go. Sometimes works, sometimes doesn't. It's about a 40% success rate, and does not seem to depend on what number dialed (ie, the problem has been basically isolated to my internal network). The symptoms are long handshake times with the fax trying to get carrier, then failing. Redialing 2-6 times eventually gets the fax through. The extensions.conf simply answers and dials out through the ISDN line. No special config here. I believe from what I have read via our friends at Google and voip forums around the place, that it is probably an echo or jitter problem, but what I have found so far has been a bit vague. Does anyone have any pointers on what I should be looking for to improve the outbound call? Thanks! Mikel ___ -- 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] Good comparisons on cheaper VOIP phones
Hello list, I am doing some work for a non profit group. As part of this, I am going to be putting in a 30 handset Asterisk solution. We are trying to keep the costs down as much as possible, as this job includes cabling, I am looking at POE solutions. On the switch side, I am considering something like some Netgear ProSafe FS726TP 24 port switches, or maybe the equivalent Linksys SRW224MP 24 port switch. About 4 of these will run the phones and computers on the network connecting back to a gigabit switch handling the phone and other servers. On the phone side VOIP phones The price range sort of limits me to: * Aastra 9112i * Snom 300 * Polycom 320 * Cisco CP-7906G (But I believe this won't handle SIP out of the box?) Any good bad stories of the above? One thing I like about the Aastra is being able to go POE from a switch, to the Aastra, then out of the second port on the Aastra and into the PC. Regards Mikel ___ -- 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 not reading fast DTMFs, was: PBX - PRI - * - Telco not working
I plug the NEC back straight to the Telco and all works well again. I just got on the phone to Digium and we've raised a ticket with some pri intense debugging going on. I'll update the list on findings. On Wed, Nov 19, 2008 at 10:32 AM, Brent Davidson [EMAIL PROTECTED] wrote: I have a weird thought... Is the PBX possibly passing the digits both inband and via PRI signaling so Asterisk is getting two digit streams at the same time and totally freaking out? You know.. that is probably it What the NEC system is doing I think is when you pick up the POTS phone to dial, you go to the NEC's LCR program (least cost routing). It then reads the first digits of your call. When it determines how to route your call (in our case, we have made it route everything out to the PRI) it then must send the digits out via PRI signaling. Maybe it captures three digits before deciding what to do, so it sends them out via PRI signaling. It would also capture the remaining digits and send them too via PRI signaling, but then the analog phone is ALSO sending the remaining digits via inband audio and then asterisk gets the first three via pri signaling, and the last 5 via inband, and instead of putting the pri signaling first and the inband second, is interleaving it. This must be how the Telco actually managed to router the call. Because it must go 'pri signaled digits first, inband second'. Because if you take the pri signal digits (which we assume are the first three) and put them at the start, you can see the number, all in the correct sequence. Thanks for this idea, I'm going to send it off to Digium and get it added to the ticket. Mikel -- http://lindsaar.net/ Rails, RSpec and Life blog ___ -- 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 not reading fast DTMFs, was: PBX - PRI - * - Telco not working
On Wed, Nov 19, 2008 at 9:08 PM, Hakan C [EMAIL PROTECTED] wrote: Did you try relaxdtmf = yes in your Zaptel/DAHDI conf? Yup, no difference. Mikel ___ -- 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 not reading fast DTMFs, was: PBX - PRI - * - Telco not working
On Mon, Nov 17, 2008 at 10:20 AM, Tony Mountifield [EMAIL PROTECTED] wrote: If I do this from an NEC digital extension I get 14149692, but if I do it from an NEC POTS extension I get 1942124000 That looks like when you pick up the analogue phone and dial 9, it immediately opens the outgoing line and sends the 141 acces code, but is doing so at the same time you carry on dialling 692. So the digits clash with each other. Notice you have 1414 interleaved with 922000. It appears like the digits generated by the NEC (1414) are overriding the digits coming in from the phone, and either obliterating the latter, or splitting them up (in the case of the 2, which gets chopped in half by a short burst of 1). OK, I removed the 1414 prefix from the NEC system. And now I have found a basic problem. If I connect a POTS phone to the analogue extensions and dial fast (like an autodial) asterisk doesn't read the digits properly. If I connect manually and dial slowly, asterisk reads all the digits correctly and can handle the call. Is there any way that i can get asterisk to read the faster DTMF digits? Mikel -- http://lindsaar.net/ Rails, RSpec and Life blog ___ -- 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 not reading fast DTMFs, was: PBX - PRI - * - Telco not working
If I connect a POTS phone to the analogue extensions and dial fast (like an autodial) asterisk doesn't read the digits properly. If I connect manually and dial slowly, asterisk reads all the digits correctly and can handle the call. Is there any way that i can get asterisk to read the faster DTMF digits For example. On the POTS phone I dial: 95523025 And the following comes up in the caller log: == CDR updated on DAHDI/21-1 -- Executing [EMAIL PROTECTED]:1] Dial(DAHDI/21-1, DAHDI/g2/29350525,,Tr) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g2/29350525 -- DAHDI/38-1 is proceeding passing it to DAHDI/21-1 -- Channel 0/7, span 2 got hangup request, cause 1 -- Hungup 'DAHDI/38-1' So it gets all the right digits... just interleaved. 2 9 3 5 0 5 25 955 2 3 025 Any ideas? As I said before, if i manually dial the digits with 1 second lags between each button press, it calls out fine. Mikel -- http://lindsaar.net/ Rails, RSpec and Life blog ___ -- 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 not reading fast DTMFs, was: PBX - PRI - * - Telco not working
On Wed, Nov 19, 2008 at 2:13 AM, Tony Mountifield [EMAIL PROTECTED]wrote: In article [EMAIL PROTECTED], Mikel Lindsaar [EMAIL PROTECTED] wrote: For example. On the POTS phone I dial: 95523025 And the following comes up in the caller log: == CDR updated on DAHDI/21-1 -- Executing [EMAIL PROTECTED]:1] Dial(DAHDI/21-1, DAHDI/g2/29350525,,Tr) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g2/29350525 -- DAHDI/38-1 is proceeding passing it to DAHDI/21-1 -- Channel 0/7, span 2 got hangup request, cause 1 -- Hungup 'DAHDI/38-1' So it gets all the right digits... just interleaved. As I said before, if i manually dial the digits with 1 second lags between each button press, it calls out fine. Well that IS weird! It looks to me like the NEC is collecting up some digits itself (e.g. that it receives before it gets Answer status from Asterisk), and then sending on the collected digits once it has connected, but these are then overlapping with the rest of the digits that are being passed through from the phone in-band. I think the source of your problems now is the behaviour of the NEC unit. So you need to understand exactly what it does with DTMF and how it wants to interact with the Asterisk unit behind it. I don't think Asterisk is the problem any more... Which I would agree with 100% if it were not for the fact that this same NEC system was working without ANY modification on and E1 the day before. The setup was: NEC == E1 == Telco To which I changed it to: NEC == CAT5 == TE210P:1 = * = TE210P:2 == E1 == Telco ie... just inserted the Asterisk box in between. I plug the NEC back straight to the Telco and all works well again. Unless Asterisk is expecting inband DTMF and the NEC was doing out of band with the Telco :/ That would make sense.. but how to force it to out of band? Mikel -- http://lindsaar.net/ Rails, RSpec and Life blog ___ -- 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 not reading fast DTMFs, was: PBX - PRI - * - Telco not working
I plug the NEC back straight to the Telco and all works well again. I just got on the phone to Digium and we've raised a ticket with some pri intense debugging going on. I'll update the list on findings. Mikel -- http://lindsaar.net/ Rails, RSpec and Life blog ___ -- 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] Debugging Asterisk
Hello all, Two questions: 1) What do people on the list do to debug phone quality issues. Phone quality seems to be a very subjective thing. But are there metrics that you can work against? Like maybe generating a tone and measuring the return quality etc? It looks like all trial and error right now. If that is the way it is, then fine. But anything more accurate / scientific? 2) Also wondering what people do when parsing asterisk verbose output in the log. Specifically, following a certain call. Asterisk's verbose output logs in sequence of action, which is good, but if you have 40-50 workstations going at once, tracking the progress of one call you are trying to make can be difficult. Obviously you can follow the channel as it goes through. But I am wondering if there is a smarter way, like telling asterisk to only log on certain numbers etc. Any hints or tricks on this would be appreciated. Mikel ___ -- 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] PBX - PRI - * - Telco not working
On Sat, Nov 15, 2008 at 11:05 PM, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Mikel Lindsaar [EMAIL PROTECTED] wrote: I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. NEC - E1 - TE210P:1 - * - TE210P:2 - E1 - Telco Incomming calls from the telco to the asterisk box to the NEC work fine with indials and everything. Works sweet. Outbound from the NEC to the Asterisk box fail. Giving an long dial tone that then times out. You need to understand how the NEC interacted with the Telco before the Asterisk box was inserted. You might want to try changing immediate=no to immediate=yes on span 1. If that's the case, you might need a different dialplan too: If that doesn't work, you could try doing this at the Asterisk CLI prompt: Yup, didn't work. Ended up timing out on the WaitExten command and looking for the t exten, which means it didn't receive a number to dial... Here is the verbose output: -- Accepting call from '' to 's' on channel 0/31, span 1 -- Executing [EMAIL PROTECTED]:1] WaitExten(DAHDI/31-1, ) in new stack [Nov 16 14:10:01] WARNING[7729]: pbx.c:7787 pbx_builtin_waitexten: Timeout but no rule 't' in context 'from-nec' == Spawn extension (from-nec, s, 1) exited non-zero on 'DAHDI/31-1' -- Hungup 'DAHDI/31-1' Because this didn't work, I changed it back to immediate = no. And removed the WaitExten from the from-nec context. Now I don't get the constant dial tone any more, I get an immediate busy with asterisk reporting: -- Extension 's' in context 'from-nec' from '' does not exist. Rejecting call on channel 0/31, span 1 Which means it's not getting the number to dial somehow. Then post the contents of /tmp/pri.txt if it's not too huge, or else put it up on a file server or web site and post the URL. Isn't too long. There are no other calls happening at the moment, so is nice and short: Protocol Discriminator: Q.931 (8) len=19 Call Ref: len= 2 (reference 1/0x1) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) User information layer 1: A-Law (35) [18 03 a1 83 9f] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Preferred Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 31 ] [6c 02 21 81] Calling Number (len= 4) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '' ] -- Making new call for cr 1 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 108 (cs0, Calling Party Number) q931.c:3509 q931_receive: call 1 on channel 31 enters state 6 (Call Present) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, peerstate Call Initiated q931.c:3104 q931_release_complete: call 1 on channel 31 enters state 0 (Null) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Any ideas? Mikel ___ -- 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] PBX - PRI - * - Telco not working
On Sun, Nov 16, 2008 at 3:13 AM, Mikel Lindsaar [EMAIL PROTECTED] wrote: On Sat, Nov 15, 2008 at 11:05 PM, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Mikel Lindsaar [EMAIL PROTECTED] wrote: I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. NEC - E1 - TE210P:1 - * - TE210P:2 - E1 - Telco Incomming calls from the telco to the asterisk box to the NEC work fine with indials and everything. Works sweet. Outbound from the NEC to the Asterisk box fail. Giving an long dial tone that then times out. Isn't too long. There are no other calls happening at the moment, so is nice and short: Here is a pastie of: dahdi/system.conf http://www.pastie.org/315597 asterisk/chan_dahdi.conf http://www.pastie.org/315608 asterisk/dahdi_channels.confhttp://www.pastie.org/315612 Mikel ___ -- 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] PBX - PRI - * - Telco not working
OK, made some progress.
With some help from friends on #Asterisk, found that the NEC was doing post
connect dialing.
So then, added the following to extensions.conf:
[from-nec]
exten = s,1,Answer()
exten = s,n,Set(TIMEOUT(digit)=2)
exten = s,n,Set(TIMEOUT(response)=5)
exten = s,n,Read(DialedNumber)
exten = s,n,Dial(DAHDI/g2/${DialedNumber},,T)
Which now captures the number and tries to dial, but then I get:
-- Accepting call from '' to 's' on channel 0/28, span 1
-- Executing [EMAIL PROTECTED]:1] Answer(DAHDI/28-1, ) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(DAHDI/28-1, TIMEOUT(digit)=2) in
new
stack
-- Digit timeout set to 2
-- Executing [EMAIL PROTECTED]:3] Set(DAHDI/28-1, TIMEOUT(response)=5)
in
new stack
-- Response timeout set to 5
-- Executing [EMAIL PROTECTED]:4] Read(DAHDI/28-1, DialedNumber) in new
stack
-- User entered '1414040040'
-- Executing [EMAIL PROTECTED]:5] Dial(DAHDI/28-1,
DAHDI/g2/1414040040,,T) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g2/1414040040
-- DAHDI/32-1 is proceeding passing it to DAHDI/28-1
-- Channel 0/1, span 2 got hangup request, cause 31
[Nov 16 16:09:30] WARNING[5828]: app_dial.c:827 wait_for_answer: Unable to
forward voice or dtmf
-- Hungup 'DAHDI/32-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'DAHDI/28-1' status is 'CHANUNAVAIL'
-- Channel 0/28, span 1 got hangup request, cause 16
-- Hungup 'DAHDI/28-1'
Ideas?
Mikel
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Re: [asterisk-users] PBX - PRI - * - Telco not working
Good. Got the darn thing working.
Problem was the NEC Xen Master does post-connect DTMF to dial. So I had to
read the digits after connect.
Then, I had to configure the PRI to be pridialplan = unknown
Thanks for your help Tony.
Here is the end result for Google's sake
#
### /etc/dahdi/system.conf
#
#
# Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER)
span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
# Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2
span=2,2,0,ccs,hdb3,crc4
# termtype: te
bchan=32-46,48-62
dchan=47
echocanceller=mg2,32-46,48-62
# Global data
loadzone = au
defaultzone = au
#
### /etc/asterisk/chan_dahdi.conf
#
[trunkgroups]
[channels]
#include dahdi-channels.conf
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
callgroup=1
pickupgroup=1
language=en
hidecallerid=no
callerid=asreceived
restrictcid=no
#
### /etc/asterisk/dahdi-channels.conf
#
; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER)
group=1
context=from-nec
switchtype = euroisdn
signalling = pri_net
channel = 1-15,17-31
; Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2
group = 2
switchtype = euroisdn
signalling = pri_cpe
pridialplan=unknown
context = from-pstn
channel=32-46,48-62
#
### extensions.conf (relevant bits)
#
[from-pstn]
exten = _55XX,1,Dial(DAHDI/g1/${EXTEN},,T)
[from-nec]
exten = s,1,Answer()
exten = s,n,Set(TIMEOUT(digit)=2)
exten = s,n,Set(TIMEOUT(response)=5)
exten = s,n,Read(DialedNumber)
exten = s,n,Dial(DAHDI/g2/${DialedNumber},,Tr)
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Re: [asterisk-users] PBX - PRI - * - Telco not working
On Sun, Nov 16, 2008 at 9:37 AM, Tony Mountifield [EMAIL PROTECTED]wrote: Actually, if Read() works, then WaitExten should have worked too. I expect what was missing was the Answer(). So this ought to work as an alternative: It doesn't for some reason... I end up getting the timeout call [Nov 16 23:22:51] WARNING[8213]: pbx.c:7787 pbx_builtin_waitexten: Timeout but no rule 't' in context 'from-nec' == Spawn extension (from-nec, s, 3) exited non-zero on 'DAHDI/28-1' -- Hungup 'DAHDI/28-1' Also, there is one improvement I think might be made to either, and that is to replace the 'r' flag (always generate ringing) with a call to Progress() before the Dial: I'll give that a shot And ideas why the waitexten is not getting the digits? I am getting another weird error as well that might shed some light. Calling from the NEC phone digital handsets works 100%. Calling from the telco and into the NEC system also works fine. But when I try to call out using a POTS phone connected to an analog line on the NEC, it seems like the numbers get mangled on either their way to Asterisk (unlikely as the NEC was making calls out the E1 from POTS lines yesterday) or from the translation that the asterisk box is doing to talk to the NEC. For example, I dial: 9692 the NEC system prepends it with 1414 (which is our telco access prefix) and then dials. If I do this from an NEC digital extension I get 14149692, but if I do it from an NEC POTS extension I get 1942124000 Strange :) Incomming from Telco = * = NEC = POTS line works fine. Wrap your brain cells around that one :) Mikel ___ -- 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] Preserving DID numbers on PRI pass through
Dear Tony, Thanks. Found that problem and that now works :) Now I have a different problem, but different thread for that. Mikel ___ -- 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] PBX - PRI - * - Telco not working
Hello all.
I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box.
NEC - E1 - TE210P:1 - * - TE210P:2 - E1 - Telco
Incomming calls from the telco to the asterisk box to the NEC work fine with
indials and everything. Works sweet.
Outbound from the NEC to the Asterisk box fail. Giving an long dial tone
that then times out.
Ie, pick up NEC handset, dial to get outside line, are given a dial tone,
and then press numbers on keypad but dialtone continues and then eventually
get an fast busy signal.
Looking at the console, it doesn't seem like the * system is seeing the
dialed number data from the NEC box. the reason I say this is that the
asterisk box complains that it is accepting call from '' to ''
Which just seems like no data is passing across.
It is an NEC Xen Master that is configured to connect to an E1 and works
fine. This E1 line has been replaced by a direct link to the * box to make
the NEC box think the * box is the Telco – so I know this at least works.
And I can also send a call into the NEC system fine.
Does anyone know how to debug this? How can I see what data the NEC box is
sending over the signaling channel in * ?
dahdi/system.conf looks like:
# Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1
# To NEC System
span=1,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
# Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2
# To E1
span=2,1,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47
echocanceller=mg2,32-46,48-62
# Global data
loadzone = au
defaultzone = au
asterisk/dahdi-channels.conf looks like:
; SPAN 1 connects to the NEC PBX system
; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) B8ZS/ESF
group=1
context=from-nec
switchtype = euroisdn
signalling = pri_net
channel = 1-15,17-31
immediate = no
overlapdial = yes
; SPAN 2 connects to Telstra – 30 channel E1
; Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 B8ZS/ESF
group=2
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel = 32-46,48-62
immediate = no
overlapdial = yes
prindication = outofband
asterisk/extensions.conf (relative bit) looks like:
[from-pstn]
exten = _73XX,1,Dial(DAHDI/g1/${EXTEN},,T)
[from-nec]
exten = _X.,1,Dial(DAHDI/g2/${EXTEN},,T)
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[asterisk-users] Preserving DID numbers on PRI pass through
Hello all,
I have the following working (somewhat) setup:
TELCO
|
|
E1 (30 Chan -- TE210 SPAN 2)
|
|
Asterisk box 1.6 with
DAHDI drivers loaded
Digium TE210p
|
|
E1 (30 Chan -- TE210 SPAN 1)
|
|
NEC PBX
From the NEC system I can make calls out. From a line on the other side of
the Telco, I can make calls in.
What I am trying to get though is how to pass through the DID range.
The E1 that I am connecting to the Telco with, used to connect direct to the
NEC system and already has hunt group calling enabled for all 30 channels.
Also, I was given a 100 number indial range (from 00 - 99).
If the E1 is connected to the NEC directly, I can call 7320 and the NEC
will route that directly to a certain handset.
But when I put the asterisk system in between, I am getting the call
through, but it does not then call the correct number it seems..
The telco side has the following in extensions.conf:
[from-pstn]
exten = _73XX,1,Dial(DAHDI/g1/${ARG1},,T)
[from-nec]
exten = s,1,Dial(DAHDI/g1/${ARG1},,T)
The dahdi_channels.conf has:
; SPAN 1 connects to the NEC PBX system
; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER)
group=0,11
context=from-nec
switchtype = euroisdn
signalling = pri_net
channel = 1-15,17-31
group = 63
; SPAN 2 connects to Telstra on 02 7300 - 30 channel E1
; Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel = 32-46,48-62
group = 63
Which is then included in chan_dahdi.conf with an #include...
What am I missing? Is there something more I should read on this?
I can't seem to find my answers in the Asterisk book nor online
Thanks
Mikel
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[asterisk-users] Connecting TE212p to NEC XenMaster
Hello list. Looking around I can't seem to find answers to what I am after, so here goes: I have an NEC Xen Master system (3 unit) basically maxed out. I want to connect a spare E1 card to the back of an existing Asterisk system terminating on a TE212P so I can divert out VOIP calls and eventually migrate over to Asterisk using the NEC system for handling the existing digital handsets only. So two questions: 1) Has anyone got any experience in connecting up this type of NEC system to a TE212 to route calls from the NEC system TO the Asterisk box? Any feedback / gotchas? 2) In this scenario, is the echo canceller needed? Or do I only really need it if I plan to route the call traffic out into another E1 circuit and to a telco? Thanks. Mikel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
