Re: [asterisk-users] [SPAM] - Re: FW: cdr_addon_mysql.so did notregister itselfduringload - Email found in subject
Did you install the MySQL libraries? Debian's command is - apt-get install libmysqlclient15-dev Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthias Urlichs Sent: 27 November 2008 16:05 To: asterisk-users@lists.digium.com Subject: [SPAM] - Re: [asterisk-users] FW: cdr_addon_mysql.so did notregister itselfduringload - Email found in subject On Thu, 28 Dec 2006 12:34:46 -0600, Savoy, Kevin - Williston, ND wrote: checking for mysql_init in -lmysqlclient... no What do I need to make that say yes? You need to read config.log and check _why_ the link fails. ___ -- 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] [SPAM] - Asterisk and S-Bus - Email found in subject
Have you set port 2 as 'NT' in the mISDN config file (not the Asterisk one)? Also, you will probably need to set it to ptmp. You need to configure them in misdn.conf (the Asterisk one this time). Here's the tail of my misdn.conf (4 x BRI): [trunks] ports = 1,2 ; physical port numbers (as defined in mISDN.conf) context = inbound ; context for incoming calls in extensions.conf msns = * ; accept and process every number that comes in and let my extensions.conf sort it out (easiest way) [extensions] ports = 3,4 ; physical port numbers (as defined in mISDN.conf) context = default ; context for internal users in extensions.conf Ports 1 and 2 are my trunk lines - ports 3 and 4 are my ISDN modems. Hope that helps. Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 28 November 2008 11:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [SPAM] - [asterisk-users] Asterisk and S-Bus - Email found in subject Hi everyone, I've built an Asterisk server (1.4.22) on a Debian Etch 4.0 base system (Kernel 2.6.18). I have so far installed AsteriskGUI and the Zaptel, libPRI and mISDN drivers. The hardware is a dual processor, dual core Xeon 2ghz (per core) server with one Digium Wildcard B410P (4 FXS - 4 FXO) and one Beronet BN2S0 ISDN card (1 TE - 1 NT). The all is working well, however I am unsure of how to configure the BN2S0 card port 2 (NT) to provide an S-Bus that additional ISDN phones can connect to as extensions. The phone is powering up OK and Asterisk is dectecting when the phone is taken off hook however I receive an error about Asterisk not knowing how to handle the call. As of yet, I have not configured any additional users in users.conf for the ISDN phone as Im not quite sure how to configure them to use that channel. Has anyone ever tried such a configuration here that might be able to give me a few pointers? Any and all help is much appreciated. Thanks. Kind Regards, Steven Moughan - LAKE Communications, Beech House, Greenhills Road, Dublin 24, IRELAND int. +353 1 4031112 fax. +353 1 452 0826 www.lakecommunications.com This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify [EMAIL PROTECTED] This footnote also confirms that this email message has been scanned for the presence of computer viruses and other security threats. Registered Office: Lake Communications Ltd, Beech House, Greenhills Road, Dublin 24, Ireland. Registered No. 59890 ___ -- 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] [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject
It looks like you are trying to dial out on your 'NT' instead of your
'TE'.
Try changing Dial(DAHDI/g1/${EXTEN:1}); to Dial(DAHDI/G1/${EXTEN:1});
Oh, and I'd use mISDN for BRI as DAHDI always gave me problems.
HTH
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Re: [asterisk-users] [SPAM] - Re: [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject - Email found in subject
Apart from you were dialling out on your inbound context and vice-versa. The best advice I can give now is to change to mISDN - as this is proven to work with v1.4 and v1.6. Actually - have you tried putting the 100ohm termination on for your NT port? I need to do that with mISDN as it only allows ptmp for ISDN extensions. Cheers Andy ___ -- 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] [SPAM] - Re: CDR Design - Email found in subject
Hi murf, Speaking as someone who designs and builds billing platforms, this is very exciting. One little thing I have most problems with is the good old fax detection. I know that NVFaxDetect et al do actually answer the call and, therefore, get flagged as ANSWERED in the CDR. But, if the call never gets answered after the initial detection - then, to my customer, it is a missed (NO ANSWER) call. Is there anyway of following that scenario? In other words: Call comes in over DAHDI, ZAP or mISDN channel NVFaxDetect samples call It's not fax, so call is passed on... Call never gets answered and caller hangs up. I appreciate that the NV suite are not Digium products - but it would be nice if the new CDR information could at least flag it up as a 'type', so that I can manually (well, via. my code) trace the call through to check for human intervention. Actually, the same would go for v1.6 and it's built in fax detection :). I hope that makes sense. Cheers Andy Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF ___ -- 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] CDR Desgin
Just seconding Freddi's idea - as it makes perfect sense. Otherwise we could quite easily start testing a call that hasn't actually finished yet. ___ -- 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] [SPAM] - Re: CDR Desgin - Email found in subject
...or something along the lines of a setting a variable (like we do for MONITOR_EXEC)... ___ -- 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] [SPAM] - Re: [SPAM] - Re: [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject -Email found in subject - Email found in subject
The DAHDI docs actually include the 'steps' that used to be in Zaptel's 'make b410p'. These steps involve downloading and compiling mISDN. Why re-invent the wheel? Just a thought :-). As for the 100ohm termination bit - it's simply changing a couple of dip switches on the b410p card (as described in the manual). Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: 01 December 2008 17:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [SPAM] - Re: [asterisk-users] [SPAM] - Re: [SPAM] - Dahdi,b410p and looping from 1 port to another - Email found in subject -Email found in subject - Email found in subject 2008/12/1 Andrew Thomas [EMAIL PROTECTED] Apart from you were dialling out on your inbound context and vice-versa. The best advice I can give now is to change to mISDN - as this is proven to work with v1.4 and v1.6. I wanted to try the Digium B410P card with Dahdi as those are now supported in 1.6.X. Hopefully, it would simplify software configuration as both asterisk and dahdi are supported by Asterisk community. As soon as I can get this card running with dahdi, I will give a try with mISDN. Fortunately, you still can use misdn with B410P w/ 1.6.X. Actually - have you tried putting the 100ohm termination on for your NT port? I didn't dare to try as I'm not familiar with concepts behind that. I need to do that with mISDN as it only allows ptmp for ISDN extensions. Cheers Andy ___ -- 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] Dahdi, b410p and looping from 1 port to another
For a ptmp setup where you have multiple phones. Or even a single phone if the port is set to ptmp. Proof of this point is the way I am using our B410P card. Ports 1 and 2 are TE (ptp) and ports 3 4 are NT (ptmp). I have a single ISDN modem connected to port 3 and the B410P would not even look at it unless the 100ohm termination was switched on. So, to reiterate - ptp needs no 100ohm termination (because the end point provides it - aka TEI 0, but ptmp does - aka TEI 127). Looks like we are going to agree to disagree on this one. ___ -- 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] Dahdi, b410p and looping from 1 port to another - Email found in subject
asterisk-users@lists.digium.com has now been added to the filters white list! Anyway, 100ohm termination isn't required for ptp - but is required for ptmp. I know the DAHDI package(s) no longer include make b410p - hence the reason it is included in the docs. ___ -- 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] [SPAM] - MySQL Error Message - Email found in subject
Give this a go:
exten = s,n,MYSQL(Query resultid ${connid} SELECT `name` FROM `cnam`
WHERE `ani` = '${CALLERID(number)}')
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Re: [asterisk-users] CDR Design
It seems to me that we are confusing billing and logging here. Call billing only really needs the start and finish (like we get now) - but proper call logging requires all steps. Do we leave CDR's as they are (for billing purposes) and have a separate 'event' driven log for call logging? Or do we change the CDR structure to accommodate logging as well? Personally, a separate 'event' log seems preferable to me as this keeps existing billing platforms useable. It just means the logging programs will need to be re-written to look at a new database for events. I know we have the AMI - but that puts out a lot more information than is needed for simple logging (and requires something to prune and store the events in a database of some sort). Any thoughts? Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF ___ -- 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] CDR Design
I think you may have misunderstood me. I didn't say don't have the extra information, I said Let's have the 'extra' information in a different way and leave the existing CDR's as they are. Take the example of a 'real' PBX - the SDX/Lucent/Avaya Index. The Index had 2 options for 'logging' - SMDR (what we know as CDR) and Events. The SMDR gives the call information - after the call has finished (what time, date, number, who answered etc). The Event log gave an Event 'code' every time a handset/trunk changed state (off-hook, dialling, ringing etc.). This Event log helped us provide real time (near as damn it) stats for the system (ring times, hold times etc.) whereas the SMDR just gave us the basic call information. This is what I am suggesting here. Leave the basic CDR's as they are - and focus more on the event driven side (maybe through a TCP/IP port or socket?). Having events put in to a database by Asterisk is putting yet more load on to the server - so why do it if it's not needed? As I joined this 'discussion' late in - I can only assume that murf is doing just this with the CEL bit (if someone can correct me if needed please). In summary: Leave CDR exactly as it is and create a new CEL (Call Event Logging) module (optional in modules.conf) that puts out (and does not accept) call event information (ie. a one-way fire-and-forget output from Asterisk). Hope that makes my positiion a little clearer. Cheers Andrew Thomas ___ -- 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 monitor_filename
You are looking in the wrong place. Have a look at the following: Core show function QUEUE_WAITING_COUNT -= Info about function 'QUEUE_WAITING_COUNT' =- [Syntax] QUEUE_WAITING_COUNT(queuename) [Synopsis] Count number of calls currently waiting in a queue [Description] Returns the number of callers currently waiting in the specified queue. ___ -- 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] CDR Design
Thanks for this Greyman - it's all beginning to make sense now ;). I agree that the 'loss of CDR upon txfr' is a nasty bug which does need to be addressed before anything else (assuming it hasn't been already). But, wouldn't it be better if you could ignore the CDR's completely and use an event based system? This would give you ALL the information you need. All that remains is to filter out the un-required bits. Like I said earlier - the CDR's aren't reliable enough for a billing platform (as you've rightly pointed out) but are OK for very basic call logging (something the customer can look at). Hopefully, the murf'ster will chirp in here :). Cheers Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Grey Man Sent: 05 December 2008 09:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDR Design On Fri, Dec 5, 2008 at 8:26 AM, Andrew Thomas [EMAIL PROTECTED] wrote: In summary: Leave CDR exactly as it is and create a new CEL (Call Event Logging) module (optional in modules.conf) that puts out (and does not accept) call event information (ie. a one-way fire-and-forget output from Asterisk). Hi Andrew and Others, This thread is actually part of a discussion that has been going on for over a year. The links below provide the background to the whole thing. http://www.asterisk.org/node/48358 http://bugs.digium.com/view.php?id=11849 http://lists.digium.com/pipermail/asterisk-users/2008-January/204856.htm l Up until recently the approach was to try and fix the specific bugs with transfer CDRs as a typical bug. There is now a realisation that that is a lot trickier than inially thought so it's been decided to try and come up with a good design for the Asterisk CDR sub-system. Regards, Greyman. ___ -- 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] CDR Design
Quote : I couldn't disagree more. The CDRs is the MOST reliable source for billing purposes ...at the moment. Have you read about Greyman's transfer problem? If you are billing customers purely on the CDR output from Asterisk - then good luck to you :). ___ -- 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] CDR Design
I'd disagree. In some cases a event based system would be the best solution, but in systems with high call volumes, scanning through events looking for the proper billing information and parsing them would be a hard job compared to CDRs. That's just it - you wouldn't be 'scanning' any CDR's - you'd be given Events. Your 3rd party app could then do anything it wanted to with them. Events are real time - not historic (like CDR's). Events are presented as they happen (hold, ring, etc) - CDR's are usually presented AFTER the call has finished so you miss things like hold-times etc. Remember, I am not saying that everyone should stop using the CDR's if they feel comfortable with them - but I, for one, don't trust them for building a stable billing platform or a real time stats package (which more and more customers seem to want these days). If you start to change the CDR's to account for extra bits (using a unique ID) then your 'scanning' actually increases as you will need to tie up all the unique ID's to get one full call progress path. Please note, I am not trying to cause flame wars here - just stating that I'd love an event based stream, that I can parse any way I see fit. I know there's the AMI - but that is a 2-way, give-you-everything solution. All I want is to know when a handset and/or trunk does something (I don't care about SIP registrations etc). I guess we'll just have to wait and see what santa murf gives us all for Christmas :). ___ -- 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] CDR Design
Pardon me, Granted ;). I have created realtime stats package that's based on CDR, you see new info immediately after call leg/event is over I see what you are saying but can you show hold-times etc? For example, call comes in to A, A puts call on hold, A dials B, B answers A, A transfers call to B, B speaks to caller. Basic PBX functionality - but how long did it take B to answer A? What if B is an external number (trunk to trunk)? To illustrate - dial an external number and, while on that call, check your CDR's - there isn't any. Now put that call on hold, still none, now call another internal extension - still none. Now hang up and transfer the call. Now there is one CDR for your call. That isn't real-time - that's historic (ie. it happens AFTER the call is finished). The CDR that's produced here will show your call to the outside world - and its duration etc. So far, so good (for historic reporting). Now get the person you transferred the call to to hang up. Another CDR record - but this show as you talking to the internal extension - not the external extension talking to the outside world. Therefore, if the 2nd extension stays on that call for a long time - who's picking up the bill? Current CDR's are lacking in this respect - and I think this is what murf is trying to sort out (please jump in here murf). ___ -- 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] top posting again [was: Re: CDR Design]
Address added to spam filter. Please do NOT e-mail me again. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 05 December 2008 13:27 To: asterisk-users@lists.digium.com Subject: [asterisk-users] top posting again [was: Re: CDR Design] Top posting strikes again: On Fri, Dec 05, 2008 at 01:39:59PM +0200, [EMAIL PROTECTED] wrote: Quote : Like I said earlier - the CDR's aren't reliable enough for a billing platform (as you've rightly pointed out) but are OK for very basic call logging (something the customer can look at). Who wrote that? [snip the rest of the reply] Andrew Thomas wrote: [snip] Like I said earlier - the CDR's aren't reliable enough for a billing platform (as you've rightly pointed out) but are OK for very basic call logging (something the customer can look at). Why didn't you place your reply here? We have archives of the list. We can spot the original message. [snip more useless quoting resulted from top-posting] -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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 ___ -- 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] CDR Design
Amen! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Apostolos Pantsiopoulos Sent: 05 December 2008 13:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDR Design Quote : That's just it - you wouldn't be 'scanning' any CDR's - you'd be given Events. Your 3rd party app could then do anything it wanted to with them. A 3rd party live application introduces one more point of failure in the whole process. A 3rd party CDRs aggregator can run at its own pace (and multiple times if any issue arises). A 3rd party live application could get choked on heavy loads and introduce inconsistency. I think what Vlasis suggests is that there are times that you need an event-based system (PBX, predictive dialing etc). And there are times that you need bulk non-realtime processing of the CDRs (sometimes the billing can be done days or weeks after the actual call). In the first situation you need a real time system, but in the second situation you don't. Being a programmer that dealt with both situations I can say that we need both approaches in asterisk :). In fact the LEGO paradigm would be the ideal solution. I think that asterisk should cope with both situations instead of just choosing one. I think we all agree on that. -- --- Apostolos Pantsiopoulos Kinetix Tele.com R D email: [EMAIL PROTECTED] --- Andrew Thomas wrote: I'd disagree. In some cases a event based system would be the best solution, but in systems with high call volumes, scanning through events looking for the proper billing information and parsing them would be a hard job compared to CDRs. That's just it - you wouldn't be 'scanning' any CDR's - you'd be given Events. Your 3rd party app could then do anything it wanted to with them. Events are real time - not historic (like CDR's). Events are presented as they happen (hold, ring, etc) - CDR's are usually presented AFTER the call has finished so you miss things like hold-times etc. Remember, I am not saying that everyone should stop using the CDR's if they feel comfortable with them - but I, for one, don't trust them for building a stable billing platform or a real time stats package (which more and more customers seem to want these days). If you start to change the CDR's to account for extra bits (using a unique ID) then your 'scanning' actually increases as you will need to tie up all the unique ID's to get one full call progress path. Please note, I am not trying to cause flame wars here - just stating that I'd love an event based stream, that I can parse any way I see fit. I know there's the AMI - but that is a 2-way, give-you-everything solution. All I want is to know when a handset and/or trunk does something (I don't care about SIP registrations etc). I guess we'll just have to wait and see what santa murf gives us all for Christmas :). ___ -- 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 ___ -- 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] top posting again [was: Re: CDR Design] - Or was it top posting?
Thanks for that, it IS appreciated - but, everyone, can we please not argue this matter any more. Some see it as top posting - some don't. I really don't care either way. No if we could just get back to the subject in hand and not clog up this list with flames. Thanks Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mr Gabriel Sent: 05 December 2008 14:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] - Or was it top posting? - Original Message - From: Mr Gabriel [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, 5 December, 2008 13:58:09 GMT +00:00 GMT Britain, Ireland, Portugal Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] - Original Message - From: Andrew Thomas [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, 5 December, 2008 13:49:59 GMT +00:00 GMT Britain, Ireland, Portugal Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] Address added to spam filter. Please do NOT e-mail me again. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 05 December 2008 13:27 To: asterisk-users@lists.digium.com Subject: [asterisk-users] top posting again [was: Re: CDR Design] Top posting strikes again: On Fri, Dec 05, 2008 at 01:39:59PM +0200, [EMAIL PROTECTED] wrote: Quote : Like I said earlier - the CDR's aren't reliable enough for a billing platform (as you've rightly pointed out) but are OK for very basic call logging (something the customer can look at). Who wrote that? [snip the rest of the reply] Andrew Thomas wrote: [snip] Like I said earlier - the CDR's aren't reliable enough for a billing platform (as you've rightly pointed out) but are OK for very basic call logging (something the customer can look at). Why didn't you place your reply here? We have archives of the list. We can spot the original message. [snip more useless quoting resulted from top-posting] ** SNIP ** Which address did you add to the spam filter? Was in the asterisk address? If you don't want to recieve any more mails from the list, then you should unregister, not add it to the spam list :) ** SNIP ** I don't think this one was a TOP POST read the whole message! ___ -- 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 DECT phones as SIP phones?
Have a look at ATA devices. Any good VoIP equipment reseller should have them available. http://www.voip-info.org/wiki-ATA is worth a look. Cheers Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: 05 December 2008 14:17 To: 'Asterisk Users List' Subject: [asterisk-users] Using DECT phones as SIP phones? I see a variety of DECT 6 phones available CHEAP at costco. Is there a way to convert these to SIP? I recall someone talking about a Siemens devices that works with all DECT phones, making them SIP (I think) ___ -- 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] top posting again [was: Re: CDR Design]
Q: What is the most annoying thing in e-mail? Spam and useless replies when I've already asked for this topic to be closed *sigh*. -- -Original Message- -- From: [EMAIL PROTECTED] [mailto:asterisk-users- -- [EMAIL PROTECTED] On Behalf Of Gergo Csibra -- Sent: 05 December 2008 14:41 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] -- -- Friday, December 5, 2008, 2:49:59 PM, Andrew wrote: -- -- Address added to spam filter. Please do NOT e-mail me again. -- -- A: Because it messes up the order in which people normally read text. -- Q: Why is top-posting such a bad thing? -- A: Top-posting. -- Q: What is the most annoying thing in e-mail? -- -- -- -- Best regards, -- Gergomailto:[EMAIL PROTECTED] -- -- -- ___ -- -- 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] top posting again [was: Re: CDR Design]
Well, it seems this opened one large can of worms. Anyway, just to repeat my previous plea - and to echo David's request - can we please stop all this 'top post' rubbish and move on with our lives? Thanks and Merry Christmas Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David fire Sent: 06 December 2008 03:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] is enougth of this pointless topic... you are spammers now... 2008/12/6 Bob Gustafson [EMAIL PROTECTED] If I notice that someone has started a bottom post, I will follow. But, if I am the first, I will top post. When I look at a new email, I don't like to scroll to the bottom to find out what is new. If you know of a mail reader which will automatically scroll to the top of the latest info, let me know. If there is a technological fix, perhaps these threads will die down. Bob G On Sat, 2008-12-06 at 14:47 +1300, Duncan Turnbull wrote: I like the discussion, I doubt it will end. I prefer top posting because I reply to all my customers that way, my mail client isn't that smart and I think technology should meet the needs rather than force you to adopt work arounds. I can fully understand though others preferring it, but I don't. All the presented evidence so far suggest bottom posting is a work around to a list archive function that is less than ideal or a politeness to get around a way of doing things that doesn't really apply so much anymore. I would have thought someone could make a better list archive model, I don't believe bottom posting is intuitive and therefore being picked up by many newcomers to the game. An alternate is to get a filter that sorts the whole thing out depending on preferences ;-), but who can be bothered. I haven't seen a signup requirement to this list requiring bottom posting, and neither have I on the many other lists I am on. In fact if I look at most of my lists the majority of posters over time have tended to top posting. Doesn't mean its right but it appears to be happening. Cheers Duncan ___ -- 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 is enougth of this pointless topic... you are spammers now... -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] CDR Design
I've just spotted another interesting CDR 'feature'. Data calls don't get flagged as such. In other words - if I make an ISDN modem call to another ISDN modem via. the PSTN, the source and destination channels are set correctly (as is everything else in the current CDR) but there is no record if it being a data call. Can the 'new style' (whatever it turns out to be) differentiate between data and voice calls (eg. B and D channel ones on ISDN)? Just one more thing to keep Papa Murf busy over the holidays :). Cheers Andy -- -Original Message- -- From: [EMAIL PROTECTED] [mailto:asterisk-users- -- [EMAIL PROTECTED] On Behalf Of Anthony Francis -- Sent: 10 December 2008 18:19 -- To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com -- Subject: Re: [asterisk-users] CDR Design -- -- -- -- Steve Murphy wrote: -- Just to be pedantic, how would src_cid be different from the clid -- field -- that cdr's have now? -- -- and the same with src_exten vs. src; -- -- A simple example might help to let this sink into my brain. -- -- murf -- -- -- The main thing is that the originating number shouldn't be linked to -- the -- callerid. This way you can do things like allow no callerid while -- maintaining billing integrity. -- Here is the CDR columns for one one of my providers that exhibits -- this: -- -- -- -- -- -- *Field Number* -- -- -- -- *Field Name* -- -- -- -- *Description* -- -- -- -- *Type* -- -- -- -- *Length* -- -- -- -- *Example* -- -- -- -- -- -- 1 -- -- -- -- SwitchBatchNbr -- -- -- -- Sequential, positive integer assigned to each CDR file imported into -- the -- system -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 5594 -- -- -- -- -- -- 2 -- -- -- -- RecNbr -- -- -- -- Sequential, positive integer assigned to each CDR within a CDR file. -- Together with the SwitchBatchNbr, a unique combination. -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 2354 -- -- -- -- -- -- 3 -- -- -- -- SwitchNbr -- -- -- -- Unique number identifying the switch from which the CDR was processed -- or -- assigned -- -- -- -- Numeric -- -- -- -- Integer -- -- -- -- 13 -- -- -- -- -- -- 4 -- -- -- -- CustNbr -- -- -- -- The unique, numeric number assigned to a customer -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 1025 -- -- -- -- -- -- 5 -- -- -- -- AuthCode -- -- -- -- The authorization code used in the call. Can be the Switch/Trunk -- combination (dedicated), ANI, Travel Card, 800 number, DID. -- -- -- -- Numeric -- -- -- -- Float -- -- -- -- 2145551212 -- -- -- -- -- -- 6 -- -- -- -- AcctCd -- -- -- -- The Account Code dialed with the CDR -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 2331 -- -- -- -- -- -- 7 -- -- -- -- CallMMDD -- -- -- -- Call date at time of answer (MMDD format) -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 20020131 -- -- -- -- -- -- 8 -- -- -- -- CallHHMMSS -- -- -- -- Call time at time of answer (HHMMSS format) -- -- -- -- Numeric -- -- -- -- Long Integer -- -- -- -- 205618 -- -- 9 -- -- -- -- DestNbr -- -- -- -- -- -- Destination Phone Number -- -- -- -- Char -- -- -- -- 18 -- -- -- -- 2145551212 -- -- -- -- -- -- -- -- -- -- 10 -- -- -- -- DialedNumber -- -- -- -- -- -- Dialed Number -- -- -- -- Char -- -- -- -- 18 -- -- -- -- 12145551212 -- -- -- -- -- -- -- -- -- -- 11 -- -- -- -- ThirdPartyNbr -- -- -- -- -- -- Third Party Number -- -- -- -- Char -- -- -- -- 18 -- -- -- -- 2145551212 -- -- -- -- -- -- 12 -- -- -- -- DestCity -- -- -- -- -- -- Destination city name -- -- -- -- Char -- -- -- -- 15 -- -- -- -- Dallas -- -- 13 -- -- -- -- DestState -- -- -- -- -- -- Destination state name -- -- -- -- Char -- -- -- -- 2 -- -- -- -- TX -- -- 14 -- -- -- -- DestOCN -- -- -- -- -- -- Destination OCN -- -- -- -- Char -- -- -- -- 4 -- -- -- -- 9100 -- -- 15 -- -- -- -- DestLata -- -- -- -- -- -- Destination LATA -- -- -- -- Numeric -- -- -- -- integer -- -- -- -- 552 -- -- 16 -- -- -- -- IntraInter -- -- -- -- Flag indicating jurisdiction: 1=Intralata, 2=Intrastate, 3=Interstate, -- 4=Canada, 5=Intl, Mexico -- -- -- -- Numeric -- -- -- -- Integer -- -- -- -- 1 -- -- 17 -- -- -- -- CallType --
Re: [asterisk-users] CDR Design
-- -Original Message- -- From: [EMAIL PROTECTED] [mailto:asterisk-users- -- [EMAIL PROTECTED] On Behalf Of Steve Murphy -- Sent: 11 December 2008 16:26 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] CDR Design -- -- On Thu, 2008-12-11 at 11:37 +, Andrew Thomas wrote: -- I've just spotted another interesting CDR 'feature'. Data calls -- don't -- get flagged as such. In other words - if I make an ISDN modem call -- to -- another ISDN modem via. the PSTN, the source and destination -- channels -- are set correctly (as is everything else in the current CDR) but -- there -- is no record if it being a data call. -- -- Can the 'new style' (whatever it turns out to be) differentiate -- between -- data and voice calls (eg. B and D channel ones on ISDN)? -- -- -- How do you picture this information appearing in a CDR? -- Via another field, or some indication in an existing field? -- -- murf -- Either/or is fine by me :). As long as there is some sort of indication I can parse - then I'm a happy bunny. Cheers Andy ___ -- 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] Variables for dial plan
Use setvar=variablename=value
Eg: under [client1]
setvar=dialplan=NZ
Then just reference ${dialplan} in your extensions.conf
Cheers
Andy
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Michael
-- Sent: 15 December 2008 04:36
-- To: asterisk-users@lists.digium.com
-- Subject: [asterisk-users] Variables for dial plan
--
-- I want to have a arbitary named variable within the client's user
-- details in
-- sip.conf
--
-- [client1]
-- dialplan=NZ
-- ..
--
-- In extensions.conf (Logic expressed using PHP style)
--
-- if ($dialplan == NZ) {
-- $NAT = 0;
-- $INT = 00;
-- };
--
-- and in the [outgoing] section
--
-- ; Australia
-- exten = _${INT}61[278]NXX.,1,Set(CDR(UserField)=AUSTRALIA)
-- exten =
_${INT}61[278]NXX.,n,Dial(SIP/SIP_PROVIDER/0${EXTEN:4:9})
--
-- How can I implement this in Asterisk style?
--
-- Thanks,
--
-- Michael
--
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Re: [asterisk-users] Dedicated Fax Line
Since when can you segment PRI channels off at the telco end? I know you could do with DASS - but I'm not aware you can do it with PRI. Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Tim Nelson -- Sent: 15 December 2008 21:04 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] Dedicated Fax Line -- -- Tell your PRI provider that you want one of those channels exclusively -- bound to your fax DID. Also, it should be removed from the normal hunt -- group where the rest of your calls come in. Then, the only way that -- PRI channel will be used is when someone calls your fax number/DID. -- -- Tim Nelson -- Systems/Network Support -- Rockbochs Inc. -- (218)727-4332 x105 -- -- - Johnny Edge je...@visafirst.com wrote: -- -- Sorry I didn't clarify more. -- -- I have one number for fax and 9 more for regulars calls, all of them -- terminated on the same 20 chan PRI. When there 20 active calls I -- can't -- send/recv faxes. Inbound faxes are sent to e-mail. I wish to make -- sure -- the fax line is separate and is not used for anything else but -- faxes. -- -- -- -- From: asterisk-users-boun...@lists.digium.com -- [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of -- Olivier -- Sent: 15 December 2008 22:30 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] Dedicated Fax Line -- -- -- -- -- 2008/12/15 Johnny Edge je...@visafirst.com -- -- -- Hello folks, -- -- I have a 20 channel fractional PRI and I would like to dedicate -- one -- of the lines for a Fax service (in and outbound). -- -- -- -- Do you imply casual incoming calls not to be answered, to be replied -- a -- busy tone or to deflected elsewhere ? -- How do you expect inbound faxes to be treated ? Switched to an -- analog -- fax machine ? E-mailed ? -- -- -- -- -- Is this possible with Asterisk and what conf would I need for -- that? -- -- Thanks, -- -- -JE -- ___ -- -- 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 -- -- ___ -- -- 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] Dedicated Fax Line
I can only assume it's a T1 thing - as E1's tend not to have that facility. Oh well, you live and learn :) Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Tim Nelson -- Sent: 16 December 2008 15:08 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] Dedicated Fax Line -- -- I've worked with many providers who are able to do this. In fact, -- we're using such a setup on our office PRI. I'm not sure how they're -- achieving this on their end however... -- -- Tim Nelson -- Systems/Network Support -- Rockbochs Inc. -- (218)727-4332 x105 -- -- - Andrew Thomas a...@datavox.co.uk wrote: -- -- Since when can you segment PRI channels off at the telco end? I -- know -- you could do with DASS - but I'm not aware you can do it with PRI. -- -- -- Andrew Thomas -- Technical Services Manager -- DataVox Ltd -- Saddleworth Business Centre -- Huddersfield Road -- Delph, Oldham -- OL3 5DF -- -- ___ -- -- 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] libpri and NT-Point to multi-point
If you are connecting to BRI lines then you should be TE - not NT. You can run as TE ptp or ptmp with mISDN (not sure about DAHDI yet - not tried the new release). HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 17 December 2008 08:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] libpri and NT-Point to multi-point Hi, At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point to multi-point mode. Here (France), most small PBXes are connected to ISDN through BRI trunks in PtmP (don't know why but it seems the general case). So this NT-PtmP function would be very helpful to easily slide an Asterisk box between an existing PBX and the network. Does the same case apply elsewhere (UK, Germany, Italy, ...) ? Do you think this is needed ? Regards ___ -- 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] libpri and NT-Point to multi-point
I have piggy backed a few PBX's off the back of a B410P (4 x BRI) card with no problems. The ones I used for testing were the Avaya IP Office, Siemens Hi-Path/Hi-Com and various old Panasonics. All I had to do was to turn on the 100ohm termination on my S0 ports (set as NT on the B410P of course). I actually have a similar set-up at the moment on our main asterisk system. 2 x BRI trunks (ports 1 2 set as TE ptp - for the DDI's etc) and 2 x BRI S0 bus (ports 3 4 set as NT ptmp) running and ISDN modem on each (good old Fritz! ones). So it is possible to run ptmp on NT ports using mISDN - just remember to turn on 100ohm termination on the ISDN card if you only have one device per port. HTH Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 17 December 2008 09:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] libpri and NT-Point to multi-point Hello Andrew, 2008/12/17 Andrew Thomas a...@datavox.co.uk If you are connecting to BRI lines then you should be TE - not NT. Yes of course, you're right. I was mostly referring to this : ISDN --BRI asterisk -BRI- legacy PBX Then, in this case, as legacy PBX has a set of TE-PtmP or TE-PtP interfaces, asterisk box should also include such NT-PtP or NT-PtmP interfaces. For instance, would you say that in the UK, most PBXes are using TE-PtP ? You can run as TE ptp or ptmp with mISDN (not sure about DAHDI yet - not tried the new release). HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 17 December 2008 08:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] libpri and NT-Point to multi-point Hi, At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point to multi-point mode. Here (France), most small PBXes are connected to ISDN through BRI trunks in PtmP (don't know why but it seems the general case). So this NT-PtmP function would be very helpful to easily slide an Asterisk box between an existing PBX and the network. Does the same case apply elsewhere (UK, Germany, Italy, ...) ? Do you think this is needed ? Regards ___ -- 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] RDNIS and asterisk
Where are you actually doing the diverting? In Asterisk at the telco
exchange?
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith
-- Sent: 17 December 2008 11:07
-- To: Asterisk Users Mailing List - Non-Commercial Discussion
-- Subject: Re: [asterisk-users] RDNIS and asterisk
--
-- No, ${exten} is the final destination number
--
-- myphone calls 123456, which is diverted to 22334455 would givc an
-- ${exten} of 22334455, but I wanted to know the 123456.
--
-- Julian
-- Andrew Thomas wrote:
-- Isn't that the ${exten} number? In other words, the number
called.
--
--
--
-- -- -Original Message-
-- -- From: asterisk-users-boun...@lists.digium.com
-- [mailto:asterisk-users-
-- -- boun...@lists.digium.com] On Behalf Of Tony Mountifield
-- -- Sent: 17 December 2008 10:17
-- -- To: asterisk-users@lists.digium.com
-- -- Subject: Re: [asterisk-users] RDNIS and asterisk
-- --
-- -- In article 49483005.8030...@dotr.com,
-- -- Julian Lyndon-Smith aster...@dotr.com wrote:
-- -- I have a couple of numbers that are diverted to a number
-- that is
-- -- conected to an isdn30 card, running asterisk 1.4.
-- --
-- -- eg.
-- --
-- -- 123456 = 22334455
-- -- 654321 = 22334455
-- --
-- -- What I would like to know is the number of the orginal
-- number
-- -- dialled
-- -- (123456 or 654321). I thought that RDNIS was the answer,
but
-- it
-- is
-- -- always coming up blank.
-- --
-- -- When I did a debug on the pri span, I saw the following
-- message
-- --
-- -- Unable to handle ROSE operation 15
-- --
-- -- is this the cause of my problem ?
-- --
-- -- Don't know about that error, but in the pri debug output,
did
-- you
-- see
-- -- any mention of the originally dialled number, or only the
-- translated
-- -- number?
-- --
-- -- If the originally dialled number is not presented in an
-- information
-- -- element somewhere, then it would be a bit of a challenge
for
-- Asterisk
-- -- to infer it! :-)
-- --
-- -- Just found this message, which seems to refer to the same
-- issue:
-- --
-- http://lists.digium.com/pipermail/asterisk-users/2007-
-- July/191858.html
-- --
-- -- If your original number does appear in a ROSE IE simlar to
-- that
-- shown
-- -- in
-- -- the above message, then it may be that libpri needs
updating
-- to
-- handle
-- -- it.
-- --
-- -- Can you get a second destination number on the same ISDN30
and
-- then
-- -- divert one of the original numbers to that instead?
-- --
-- -- Cheers
-- -- Tony
-- -- --
-- -- Tony Mountifield
-- -- Work: t...@softins.co.uk - http://www.softins.co.uk
-- -- Play: t...@mountifield.org - http://tony.mountifield.org
-- --
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Re: [asterisk-users] RDNIS and asterisk
-- Where are you actually doing the diverting? In Asterisk at the telco -- exchange? ...or at... ___ -- 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] libpri and NT-Point to multi-point
I would say the 'norm' in the UK is TE-ptp and NT-ptp or NT-ptmp (depends what is on the end of the port(s)). If using NT-ptmp, then a 100ohm resistor is usually needed in the circuit somewhere - aka ISDN balun - (unless the card has this facility - like the B410P has). HTH Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 17 December 2008 10:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] libpri and NT-Point to multi-point 2008/12/17 Andrew Thomas a...@datavox.co.uk I have piggy backed a few PBX's off the back of a B410P (4 x BRI) card with no problems. The ones I used for testing were the Avaya IP Office, Siemens Hi-Path/Hi-Com and various old Panasonics. All I had to do was to turn on the 100ohm termination on my S0 ports (set as NT on the B410P of course). I actually have a similar set-up at the moment on our main asterisk system. 2 x BRI trunks (ports 1 2 set as TE ptp - for the DDI's etc) and 2 x BRI S0 bus (ports 3 4 set as NT ptmp) running and ISDN modem on each (good old Fritz! ones). Fine, so, using this setup as an example, would say the norm in the UK, is to connect to ISDN-BRI in ptp (reading from ports 1 and 2 configuration), or to connect using ptmp (reading from ports 3 and 4 configuration, dedicated to isdn modems) ? So it is possible to run ptmp on NT ports using mISDN - just remember to turn on 100ohm termination on the ISDN card if you only have one device per port. HTH Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 17 December 2008 09:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] libpri and NT-Point to multi-point Hello Andrew, 2008/12/17 Andrew Thomas a...@datavox.co.uk If you are connecting to BRI lines then you should be TE - not NT. Yes of course, you're right. I was mostly referring to this : ISDN --BRI asterisk -BRI- legacy PBX Then, in this case, as legacy PBX has a set of TE-PtmP or TE-PtP interfaces, asterisk box should also include such NT-PtP or NT-PtmP interfaces. For instance, would you say that in the UK, most PBXes are using TE-PtP ? You can run as TE ptp or ptmp with mISDN (not sure about DAHDI yet - not tried the new release). HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 17 December 2008 08:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] libpri and NT-Point to multi-point Hi, At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point to multi-point mode. Here (France), most small PBXes are connected to ISDN through BRI trunks in PtmP (don't know why but it seems the general case). So this NT-PtmP function would be very helpful to easily slide an Asterisk box between an existing PBX and the network. Does the same case apply elsewhere (UK, Germany, Italy, ...) ? Do you think this is needed ? Regards ___ -- 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 ___ -- 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] RDNIS and asterisk
Isn't that the ${exten} number? In other words, the number called.
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Tony Mountifield
-- Sent: 17 December 2008 10:17
-- To: asterisk-users@lists.digium.com
-- Subject: Re: [asterisk-users] RDNIS and asterisk
--
-- In article 49483005.8030...@dotr.com,
-- Julian Lyndon-Smith aster...@dotr.com wrote:
-- I have a couple of numbers that are diverted to a number that is
-- conected to an isdn30 card, running asterisk 1.4.
--
-- eg.
--
-- 123456 = 22334455
-- 654321 = 22334455
--
-- What I would like to know is the number of the orginal number
-- dialled
-- (123456 or 654321). I thought that RDNIS was the answer, but it
is
-- always coming up blank.
--
-- When I did a debug on the pri span, I saw the following message
--
-- Unable to handle ROSE operation 15
--
-- is this the cause of my problem ?
--
-- Don't know about that error, but in the pri debug output, did you
see
-- any mention of the originally dialled number, or only the
translated
-- number?
--
-- If the originally dialled number is not presented in an
information
-- element somewhere, then it would be a bit of a challenge for
Asterisk
-- to infer it! :-)
--
-- Just found this message, which seems to refer to the same issue:
--
http://lists.digium.com/pipermail/asterisk-users/2007-July/191858.html
--
-- If your original number does appear in a ROSE IE simlar to that
shown
-- in
-- the above message, then it may be that libpri needs updating to
handle
-- it.
--
-- Can you get a second destination number on the same ISDN30 and
then
-- divert one of the original numbers to that instead?
--
-- Cheers
-- Tony
-- --
-- Tony Mountifield
-- Work: t...@softins.co.uk - http://www.softins.co.uk
-- Play: t...@mountifield.org - http://tony.mountifield.org
--
-- ___
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Re: [asterisk-users] Setup ReceiveFax(), fax2mail, mime-construct - but now Sendmail :(
You don't really need to use any local MTA if you use the sendEmail
script.
I got it from - http://www.caspian.dotconf.net/menu/Software/SendEmail/
This actually works by 'talking' directly to any SMTP server - even
remote ones (I use our Exchange server for our e-mails).
HTH
Andy
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Anthony Messina
-- Sent: 20 December 2008 02:47
-- To: Asterisk Users Mailing List - Non-Commercial Discussion
-- Subject: Re: [asterisk-users] Setup ReceiveFax(), fax2mail,mime-
-- construct - but now Sendmail :(
--
-- On Friday 19 December 2008 20:24:11 sean darcy wrote:
-- Using 1.6 on Fedora Core 9 I'm trying to receive faxes. I've got
-- this far:
--
-- [incoming-fax]
-- exten =
--
--
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%
-- M)}-0
-- ${CALLERIDNUM}) exten = s,2,ReceiveFAX(${FAXFILE}.tif)
-- exten = s,3,Hangup()
-- exten=h,1,System(/usr/local/bin/fax2mail --cid-number
-- 0${CALLERIDNUM}
-- --cid-name home fax --dest-name admin --dest-email
-- ${admin_email}
-- -f ${FAXFILE})
--
-- which all seems work well on the CLI. No errors.
--
-- fax2mail uses mime-contruct to send the fax by sendmail. That
didn't
-- work.
--
-- No email. /var/log/maillog:
--
-- Dec 19 21:04:53 asterisk sendmail[2628]: mBH2mWvQ006043:
-- to=ad...@myco.com, ctladdr=r...@localhost.localdomain (0/0),
-- delay=2+23:16:09, xdelay=00:00:00, mailer=esmtp, pri=6640305,
-- relay=mx01.1and1.com., dsn=4.0.0, stat=Deferred: Connection
timed
-- out
-- with mx01.1and1.com.
-- Dec 19 21:04:53 asterisk sendmail[2628]: mBH2mWvS006043:
-- to=ad...@myco.com, ctladdr=r...@localhost.localdomain (0/0),
-- delay=2+23:16:09, xdelay=00:00:00, mailer=esmtp, pri=6640312,
-- relay=mx00.1and1.com., dsn=4.0.0, stat=Deferred: Connection
timed
-- out
-- with mx00.1and1.com.
--
-- I've avoided MTA's like sendmail for a _long_ time. So I need
help.
--
-- 1. Is this the right list to try to resolve this? If not, which
-- list?
--
-- 2. postfix seems to considered much easier to configure than
-- sendmail.
-- Do I install postfix? If so, will this work out of the box?
--
-- 3. If sendmail, what's the magic configuration?
--
--
-- i'm still working on this, but take a look at
-- http://messinet.com/viewvc/asterisk-fax-gw/trunk/
--
-- currently, i use postfix, which seems easier to me to configure
than
-- sendmail
--
-- --
-- Anthony - http://messinet.com -
http://messinet.com/~amessina/gallery
-- 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
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Re: [asterisk-users] Install app_rxfax and app_txfax in 1.4with Lenny
JFYI - I run (successfully) agx-addons with 1.4.22 and Etch. Make sure you have the right version of SpanDSP installed (as well as the tiff libraries). -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Loic Didelot -- Sent: 19 December 2008 20:33 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] Install app_rxfax and app_txfax in -- 1.4with Lenny -- -- Hi, -- I tried agx-addons with different version. I got it working till -- asterisk version 1.4.21 included on ubuntu with libtiff4. -- -- Starting from asterisk 1.4.22 it did not longer work. -- -- Loic -- -- On Wed, 2008-12-17 at 17:12 +0100, Olivier wrote: -- Hi, -- -- I've read README file in agx-ast-addons-1.4.17.5.tar.bz2 -- It says Install libTiff =3.8 and 4.0 -- -- Should you really use this agx-ast-addons to get app_rxfax and -- app-_txfax running with latest 1.4.22 ? -- If positive, should you take this libtiff warning into account ? -- If positive, where can you find such libtiff version as Debian -- repository (I didn't check alternate distrib) includes libtiff4 but -- no -- libtiff3 not libtiff. -- -- Cheers -- ___ -- -- 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 -- -- -- Loïc DIDELOT -- MIXvoip S.a. -- Tel: +352 20 20 -- Fax: +352 20 90 -- ldide...@mixvoip.com -- http://www.mixvoip.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 ___ -- 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] Install app_rxfax and app_txfax in 1.4withLenny
...described in the README file ;). SpanDSP - 0.0.4 pre 16 LibTiff - = 3.8 but 4.0 I had to trawl around for the right SpanDSP - but I can e-mail a copy to whomever wants one (drop me a personal e-mail and I'll attach it by return) HTH Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 22 December 2008 09:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Install app_rxfax and app_txfax in 1.4withLenny Hi Andrew, 2008/12/22 Andrew Thomas a...@datavox.co.uk JFYI - I run (successfully) agx-addons with 1.4.22 and Etch. Make sure you have the right version of SpanDSP installed (as well as the tiff libraries). which are (thinking of both SpanDSP and libiff) ? -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Loic Didelot -- Sent: 19 December 2008 20:33 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] Install app_rxfax and app_txfax in -- 1.4with Lenny -- -- Hi, -- I tried agx-addons with different version. I got it working till -- asterisk version 1.4.21 included on ubuntu with libtiff4. -- -- Starting from asterisk 1.4.22 it did not longer work. -- -- Loic -- -- On Wed, 2008-12-17 at 17:12 +0100, Olivier wrote: -- Hi, -- -- I've read README file in agx-ast-addons-1.4.17.5.tar.bz2 -- It says Install libTiff =3.8 and 4.0 -- -- Should you really use this agx-ast-addons to get app_rxfax and -- app-_txfax running with latest 1.4.22 ? -- If positive, should you take this libtiff warning into account ? -- If positive, where can you find such libtiff version as Debian -- repository (I didn't check alternate distrib) includes libtiff4 but -- no -- libtiff3 not libtiff. -- -- Cheers -- ___ -- -- 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 -- -- -- Loïc DIDELOT -- MIXvoip S.a. -- Tel: +352 20 20 -- Fax: +352 20 90 -- ldide...@mixvoip.com -- http://www.mixvoip.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 ___ -- 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] Install app_rxfax and app_txfax in 1.4withLenny
I'm not sure mate - as I don't use HylaFax on that particular server. Hopefully, someone else can help. Sorry. -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Michael -- Sent: 22 December 2008 10:58 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] Install app_rxfax and app_txfax in -- 1.4withLenny -- -- On Mon, 22 Dec 2008 23:46:28 Andrew Thomas wrote: -- ...described in the README file ;). -- -- SpanDSP - 0.0.4 pre 16 -- LibTiff - = 3.8 but 4.0 -- -- -- I had to trawl around for the right SpanDSP - but I can e-mail a -- copy to -- whomever wants one (drop me a personal e-mail and I'll attach it by -- return) -- -- HTH -- Andy -- -- How can I compile this without first installing Libtiff 3.8+ ? -- -- I have a patched version of 3.7.2 that I need to keep to maintain my -- Hylafax -- Jbig/Jpeg support. (Unless someone knows of a more recent Hylafax -- libtiff -- patch) -- -- Michael -- -- ___ -- -- 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] integration with Microsoft CRM?
Try http://forums.vtiger.com/viewtopic.php?t=14314 Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Louis-David Mitterrand -- Sent: 21 January 2009 11:15 -- To: asterisk-users@lists.digium.com -- Subject: Re: [asterisk-users] integration with Microsoft CRM? -- -- On Wed, Jan 21, 2009 at 09:02:51AM -0200, David fire wrote: -- how hard is to integrate whit a virus? -- sorry -- ok i read MS CRM but... did you tried VTiger? www.vtiger.com the -- next -- release (5.1) will be integrated whit asterisk not only click to -- dial and -- popups on incoming calls a queue monitor system too. (Thanks to -- Wolfgang) -- -- I wasn't aware of VTiger. It looks pretty good. Do you know when 5.1 -- is -- supposed to be released? -- -- What version of asterisk is required for integration with VTiger? -- -- Thanks, -- -- -- -- http://www.critikart.net -- -- ___ -- -- 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] Fw: Re: mISDN BRI Asterisk 1.4
Have you got termination set correctly? I have a B410P working with 2 x NT and 2 x TE ports successfully. I had to turn the 100ohm termination on on the NT ports (even though I have them set as PTP in mISDN.conf). HTH -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Lee Wilson -- Sent: 21 January 2009 15:18 -- To: asterisk-users@lists.digium.com -- Subject: [asterisk-users] Fw: Re: mISDN BRI Asterisk 1.4 -- -- -- Its been a few days, I was wondering if anyone else has any ideas on -- how to get this to work? -- -- If not, could I ask a the direct question as to if anyone here has (or -- knows someone who has) successfully got something like a Cisco Router -- to successfully establish an ISDN data/Internet connection through the -- Asterisk PBX? -- -- From a telephony perspective it must be possible as our existing -- Nortel PBX does it, just don't know if Asterisk can. -- -- I feel that I've spent too much time that could have been better spend -- on this and am now at the point where I think the boss should just -- cough up and get a couple of ISDN BRI lines put in. -- -- Thanks again for everyones input so far. -- -- Lee -- -- --- On Fri, 16/1/09, Lee Wilson leef...@yahoo.co.uk wrote: -- -- From: Lee Wilson leef...@yahoo.co.uk -- Subject: Re: [asterisk-users] mISDN BRI Asterisk 1.4 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- asterisk-users@lists.digium.com -- Date: Friday, 16 January, 2009, 10:12 AM -- Hi Francesco, -- -- You were correct. I pulled the cable out before everyone -- got in this morning and it was a cross over. I've now -- connected a proper straight-through ISDN cable (don't -- know what the Nortel was using before) and L1 is now up on -- Asterisk: -- -- BEGIN STACK_LIST: -- * Port 1 Type NT Prot. PTP L2Link DOWN L1Link:UP -- Blocked:0 Debug:1 -- -- However I'm now seeing the following message as L2 is -- not coming up: -- -- P[ 1] !!! Could not Get the L2 up after 3 Attemps!!! -- -- I checked that misdn.conf was setup to use PTMP (despite -- was it is saying above): -- -- mISDNconf -- module poll=128 debug=1 -- timer=nohfcmulti/module -- module debug=1 -- options=0mISDN_dsp/module -- devnode user=root -- group=root -- mode=644mISDN/devnode -- card type=BN4S0 -- port mode=nt -- link=ptmp1/port -- port mode=nt -- link=ptmp2/port -- port mode=nt -- link=ptmp3/port -- port mode=nt -- link=ptmp4/port -- /card -- /mISDNconf -- -- Also, how can I definately tell it is running in PTMP mode? -- -- -- msisdnportinfo does not distinguish: -- Port 1: NT-mode BRI S/T interface port (for phones) --- Interface can be Poin-To-Point/Multipoint. -- -- Port 2: NT-mode BRI S/T interface port (for phones) --- Interface can be Poin-To-Point/Multipoint. -- -- -- And Asterisk says it is running in PTP as well when the -- module is loaded: -- *CLI module load chan_misdn.so -- mISDN_close: fid(21) isize(131072) inbuf(0xb7b12008) -- irp(0xb7b12008) iend(0xb7b12008) -- == Parsing '/etc/asterisk/misdn.conf': Found -- P[ 0] Got: 1ptp,2ptp from get_ports -- == Registered channel type 'mISDN' (Channel -- driver for mISDN Support (Bri/Pri)) -- == Registered application 'misdn_set_opt' -- == Registered application 'misdn_facility' -- == Registered application 'misdn_check_l2l1' -- P[ 0] -- mISDN Channel Driver Registered -- --Loaded chan_misdn.so = (Channel driver for mISDN -- Support (BRI/PRI)) -- -- I've got onto the Cisco router and found the BRI -- interface to be configured as below: -- -- interface BRI0 --no ip address --encapsulation ppp --dialer pool-member 1 --isdn switch-type basic-net3 --no cdp enable --ppp authentication chap callin -- end -- -- And the following events are being reported: -- -- Jan 16 10:03:06.099: ISDN BR0: Could not bring up interface -- Jan 16 10:03:06.103: BRI0: wait for isdn carrier timeout, -- call id=0x8004 -- -- The router also thinks that Layer 1 is still down: -- Global ISDN Switchtype = basic-net3 -- ISDN BRI0 interface -- dsl 0, interface ISDN Switchtype = basic-net3 -- Layer 1 Status: -- DEACTIVATED -- Layer 2 Status: -- Layer 2 NOT Activated -- Layer 3 Status: -- 0 Active Layer 3 Call(s) -- Active dsl 0 CCBs = 0 -- The Free Channel Mask: 0x8003 -- -- Looks like it may be worth speaking with OpenVox regarding -- exactly how there can is setup, only problem is I'm not -- sure what I should be asking them. -- -- Apologies for this long email. -- -- Regards -- -- Lee -- --
Re: [asterisk-users] RFC -- Improving the quality of the mailinglists
-- In many cases, this just isn't possible. While it would be nice to -- have all -- posts in the King's English, a great many users are in locales which -- don't King's English??? Anyway - to quote Ralph Wigham Me fail English? That's unpossible!. ___ -- 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] Incoming fax detection on mISDN hfcmulti B410Pcard
Put faxdetect = none in the misdn.conf and you'll be fine. -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Vieri -- Sent: 06 February 2009 12:40 -- To: asterisk-users@lists.digium.com -- Subject: Re: [asterisk-users] Incoming fax detection on mISDN hfcmulti -- B410Pcard -- -- -- --- On Thu, 2/5/09, Ex Vito ex.vitor...@gmail.com wrote: -- -- App nvfaxdetect() works fine for that purpose on both Zap -- and mISDN. --See http://www.voip-info.org/wiki-NVFaxDetect -- -- Thanks. I setup a system with both nvfaxdetect and the built-in fax -- detection because the built-in detection alone didn't always work, -- even with a wait(10). -- I hope they won't conflict but it's beem working fine for now. -- -- -- -- -- -- ___ -- -- 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] set caller id on outgoing calls through BRI ISDNlines
Use Set(CALLERID(num)=99) instead of using CALLERID(all). Remember to set this BEFORE you Dial. -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Vieri -- Sent: 06 February 2009 12:36 -- To: asterisk-users@lists.digium.com -- Subject: [asterisk-users] set caller id on outgoing calls through BRI -- ISDNlines -- -- I'm trying to set caller ids on outgoing calls. -- I have a quad BRI B410P card connected to my telephony provider. -- I know the list of DID numbers the provider assigned to my company. -- -- If I don't set the caller id then the callee always sees the same -- top-level number. -- -- If I set the caller id to a specific DID number we own, the callee -- keeps seeing the top-level number, as though the CALLERID function -- had no effect. -- -- When dialing out on the BRI lines I see this in *CLI: -- -- -- Executing Set(SIP/4053-b2ba5a28, CALLERID(all)=My Name -- 99) in new stack -- -- where 99 is a valid DID number. -- -- My telephony provider swears that if the BRI lines are connected to -- our PBX then we can set whichever caller id we want. -- -- How can I be sure that they are right? -- What can I do to see if Asterisk is failing to set the caller id and -- why? -- -- An misdn debug doesn't seem to reveal any errors setting the caller -- id: -- -- P[ 0] -- Group Call group: ISDN_GROUP -- P[ 1] Group [ISDN_GROUP] Port [1] -- P[ 1] portup:1 -- P[ 0] -- * NEW CHANNEL dad:656660499 oad:(null) -- P[ 1] * Queuing chan 0xf651398 -- P[ 1] read_config: Getting Config -- P[ 1] config_jb: Called -- P[ 1] -- * CallGrp: PickupGrp: -- P[ 1] -- TON: Unknown -- P[ 1] -- LTON: Unknown -- P[ 1] -- CTON: Unknown -- P[ 1] * CALL: g:ISDN_GROUP/66 -- P[ 1] -- * dad:66 tech:mISDN/1-u23701 ctx:from-pstn-deviate- -- custom -- P[ 1] -- * adding2newbc ext 66 -- P[ 1] -- * adding2newbc callerid 99 -- P[ 1] -- pres: -1 screen: -1 -- P[ 1] -- pres: 0 -- P[ 1] -- PRES: Allowed (0x0) -- P[ 1] -- SCREEN: Unscreened (0x0) -- P[ 1] NO OPTS GIVEN -- P[ 1] I SEND:SETUP oad:99 dad:66 pid:2626 -- P[ 1] -- bc_state:BCHAN_CLEANED -- P[ 1] -- channel:0 mode:TE cause:16 ocause:16 rad: cad: -- P[ 1] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 -- P[ 1] -- caps:Speech pi:0 keypad: sending_complete:0 -- P[ 1] -- screen:0 -- pres:0 -- P[ 1] -- addr:0 l3id:95bfc b_stid:0 layer_id:50010180 -- P[ 1] -- facility:FAC_NONE out_facility:FAC_NONE -- P[ 1] -- urate:0 rate:16 mode:0 user1:0 -- P[ 1] -- bc:8212590 h:0 sh:0 -- P[ 1] find_free_chan: req_chan:0 -- P[ 1] -- found chan: 1 -- P[ 1] set_chan_in_stack: 1 -- P[ 1] -- found channel: 1 -- P[ 1] -- new_l3id 95bfd -- P[ 1] -- * SEND: State Dialing pid:2626 -- -- Called g:ISDN_GROUP/66 -- P[ 1] Sending msg, prim:30580 addr:41000104 dinfo:95bfd -- P[ 1] handle_frm: frm-addr:42000103 frm-prim:30282 -- P[ 1] set_channel: bc-channel:1 channel:1 -- P[ 1] $$$ Setting up bc with stid :10010100 -- P[ 1] setup_bc: with dsp -- P[ 1] -- Channel is 1 -- P[ 1] -- TRANSPARENT Mode -- P[ 1] $$$ Bchan Activated addr 50010102 -- P[ 1] BC_STATE_CHANGE: l3id:95bfd from:BCHAN_CLEANED -- to:BCHAN_ACTIVATED -- P[ 1] lib Got Prim: Addr 42000103 prim 30282 dinfo 95bfd -- P[ 1] I IND :PROCEEDING oad:99 dad:66 pid:2626 -- state:CALLING -- P[ 1] -- channel:1 mode:TE cause:16 ocause:16 rad: cad: -- P[ 1] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 -- P[ 1] -- caps:Speech pi:0 keypad: sending_complete:0 -- P[ 1] -- screen:0 -- pres:0 -- P[ 1] -- addr:50010102 l3id:95bfd b_stid:10010100 layer_id:50010180 -- P[ 1] -- facility:FAC_NONE out_facility:FAC_NONE -- P[ 1] -- urate:0 rate:16 mode:0 user1:0 -- P[ 1] -- bc:8212590 h:0 sh:0 -- P[ 1] -- bc_state:BCHAN_ACTIVATED -- P[ 1] Freeing Msg on prim:30282 -- -- mISDN/1-u23701 is proceeding passing it to SIP/4053-b23d5580 -- -- The callee 66 is expecting to see our DID 99 on -- his/her phone screen but is seeing our top-level number, say, -- 00. -- -- Help appreciated. -- -- Vieri -- -- -- -- -- -- ___ -- -- 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] set caller id on outgoing calls through BRIISDNlines
You're quite right. We'll need to see your misdn.conf file to check the settings. -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Vieri -- Sent: 06 February 2009 13:49 -- To: asterisk-users@lists.digium.com -- Subject: Re: [asterisk-users] set caller id on outgoing calls through -- BRIISDNlines -- -- Thanks but it still doesn't work. -- I did: -- -- -- Executing Set(SIP/4053-b23c5280, CALLERID(num)=99) -- in new stack -- -- before Dial(), of course. -- -- I've read somewhere that the misdn debug message: -- -- -- P[ 1] -- TON: Unknown -- -- may mean that the carrier did not recognize the caller id I set. Is -- this true? -- -- -- --- On Fri, 2/6/09, Andrew Thomas a...@datavox.co.uk wrote: -- -- Use Set(CALLERID(num)=99) instead of using -- CALLERID(all). -- -- Remember to set this BEFORE you Dial. -- -- -- -- -Original Message- -- -- From: asterisk-users-boun...@lists.digium.com -- [mailto:asterisk-users- -- -- boun...@lists.digium.com] On Behalf Of Vieri -- -- Sent: 06 February 2009 12:36 -- -- To: asterisk-users@lists.digium.com -- -- Subject: [asterisk-users] set caller id on -- outgoing calls through -- BRI -- -- ISDNlines -- -- -- -- I'm trying to set caller ids on outgoing -- calls. -- -- I have a quad BRI B410P card connected to my -- telephony provider. -- -- I know the list of DID numbers the provider -- assigned to my -- company. -- -- -- -- If I don't set the caller id then the -- callee always sees the same -- -- top-level number. -- -- -- -- If I set the caller id to a specific DID number -- we own, the callee -- -- keeps seeing the top-level number, -- as though the CALLERID -- function -- -- had no effect. -- -- -- -- When dialing out on the BRI lines I see this in -- *CLI: -- -- -- -- -- Executing -- Set(SIP/4053-b2ba5a28, -- CALLERID(all)=My Name -- -- 99) in new stack -- -- -- -- where 99 is a valid DID number. -- -- -- -- My telephony provider swears that if the BRI -- lines are connected -- to -- -- our PBX then we can set whichever caller id we -- want. -- -- -- -- How can I be sure that they are right? -- -- What can I do to see if Asterisk is failing to -- set the caller id -- and -- -- why? -- -- -- -- An misdn debug doesn't seem to reveal any -- errors setting the -- caller -- -- id: -- -- -- -- P[ 0] -- Group Call group: ISDN_GROUP -- -- P[ 1] Group [ISDN_GROUP] Port [1] -- -- P[ 1] portup:1 -- -- P[ 0] -- * NEW CHANNEL dad:656660499 -- oad:(null) -- -- P[ 1] * Queuing chan 0xf651398 -- -- P[ 1] read_config: Getting Config -- -- P[ 1] config_jb: Called -- -- P[ 1] -- * CallGrp: PickupGrp: -- -- P[ 1] -- TON: Unknown -- -- P[ 1] -- LTON: Unknown -- -- P[ 1] -- CTON: Unknown -- -- P[ 1] * CALL: g:ISDN_GROUP/66 -- -- P[ 1] -- * dad:66 -- tech:mISDN/1-u23701 -- ctx:from-pstn-deviate- -- -- custom -- -- P[ 1] -- * adding2newbc ext 66 -- -- P[ 1] -- * adding2newbc callerid -- 99 -- -- P[ 1] -- pres: -1 screen: -1 -- -- P[ 1] -- pres: 0 -- -- P[ 1] -- PRES: Allowed (0x0) -- -- P[ 1] -- SCREEN: Unscreened (0x0) -- -- P[ 1] NO OPTS GIVEN -- -- P[ 1] I SEND:SETUP oad:99 -- dad:66 pid:2626 -- -- P[ 1] -- bc_state:BCHAN_CLEANED -- -- P[ 1] -- channel:0 mode:TE cause:16 -- ocause:16 rad: cad: -- -- P[ 1] -- info_dad: onumplan:0 dnumplan:0 -- rnumplan:0 cpnnumplan:0 -- -- P[ 1] -- caps:Speech pi:0 keypad: -- sending_complete:0 -- -- P[ 1] -- screen:0 -- pres:0 -- -- P[ 1] -- addr:0 l3id:95bfc b_stid:0 -- layer_id:50010180 -- -- P[ 1] -- facility:FAC_NONE -- out_facility:FAC_NONE -- -- P[ 1] -- urate:0 rate:16 mode:0 user1:0 -- -- P[ 1] -- bc:8212590 h:0 sh:0 -- -- P[ 1] find_free_chan: req_chan:0 -- -- P[ 1] -- found chan: 1 -- -- P[ 1] set_chan_in_stack: 1 -- -- P[ 1] -- found channel: 1 -- -- P[ 1] -- new_l3id 95bfd -- -- P[ 1] -- * SEND: State Dialing pid:2626 -- -- -- Called g:ISDN_GROUP/66 -- -- P[ 1] Sending msg, prim:30580 addr:41000104 -- dinfo:95bfd -- -- P[ 1] handle_frm: frm-addr:42000103 -- frm-prim:30282 -- -- P[ 1] set_channel: bc-channel:1 channel:1 -- -- P[ 1] $$$ Setting up bc with stid :10010100 -- -- P[ 1] setup_bc: with dsp -- -- P[ 1] -- Channel is 1 -- -- P[ 1] -- TRANSPARENT Mode -- -- P[ 1] $$$ Bchan Activated addr 50010102 -- -- P[ 1] BC_STATE_CHANGE: l3id:95bfd -- from:BCHAN_CLEANED -- -- to:BCHAN_ACTIVATED -- -- P[ 1] lib Got Prim: Addr 42000103 prim 30282 -- dinfo 95bfd -- -- P[ 1] I IND :PROCEEDING oad:99 -- dad:66 pid:2626
[asterisk-users] InUseRinging
Hello, I'm just wondering if anyone has fixed the 'InUseRinging' problem. * v1.4.23.1 Ta ___ -- 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] InUseRinging
core show hints is showing extens as InUseRinging, which in turn causes the wrong 'hint' state to be shown and hang-up etc. which means BLF's actually show the wrong condition(s) http://bugs.digium.com/view.php?id=13238 gives more information (only seems to relate to v1.6.x.x though). I have this problem on v1.4.23.1 (doesn't seem to appear on 1.4.22). This is happening even with call-limit set to 2 (so they can transfer). I suppose I could always go back to 1.4.22 if needed (rather not though - as it is a live site). -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Philipp Kempgen -- Sent: 09 February 2009 11:50 -- To: Asterisk Users -- Subject: Re: [asterisk-users] InUseRinging -- -- Andrew Thomas schrieb: -- I'm just wondering if anyone has fixed the 'InUseRinging' problem. -- -- * v1.4.23.1 -- -- What's the InUseRinging problem? -- -- -- Philipp Kempgen -- -- -- -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de -- Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de -- Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- -- -- -- ___ -- -- 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] Asterisk AGX addons compile issues
svn co https://agx-ast-addons.svn.sourceforge.net/svnroot/agx-ast-addons agx-ast-addons ./build_sh from the trunk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 10 February 2009 18:35 To: mich...@networkstuff.co.nz; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk AGX addons compile issues 2008/12/18 Michael mich...@networkstuff.co.nz Has anyone seen this before, and know what is happening? u...@host:~/asterisk/agx-ast-addons# ./build.sh -- Configuring done -- Generating done -- Build files have been written to: /root/asterisk/agx-ast-addons [ 11%] Building C object CMakeFiles/app_devstate.dir/app_devstate.o Linking C shared module dist/app_devstate.so [ 11%] Built target app_devstate [ 22%] Building C object CMakeFiles/app_nv_backgrounddetect.dir/app_nv_backgrounddetect.o Linking C shared module dist/app_nv_backgrounddetect.so [ 22%] Built target app_nv_backgrounddetect [ 33%] Building C object CMakeFiles/app_nv_faxdetect.dir/app_nv_faxdetect.o Linking C shared module dist/app_nv_faxdetect.so [ 33%] Built target app_nv_faxdetect [ 44%] Building C object CMakeFiles/app_pickup2.dir/app_pickup2.o Linking C shared module dist/app_pickup2.so [ 44%] Built target app_pickup2 [ 55%] Building C object CMakeFiles/app_rxfax.dir/app_rxfax.o cc1: warnings being treated as errors /root/asterisk/agx-ast-addons/app_rxfax.c: In function 'phase_e_handler': /root/asterisk/agx-ast-addons/app_rxfax.c:126: warning: implicit declaration of function 't30_get_local_ident' /root/asterisk/agx-ast-addons/app_rxfax.c:127: warning: implicit declaration of function 't30_get_far_ident' /root/asterisk/agx-ast-addons/app_rxfax.c: In function 'rxfax_exec': /root/asterisk/agx-ast-addons/app_rxfax.c:380: warning: implicit declaration of function 't30_set_local_ident' /root/asterisk/agx-ast-addons/app_rxfax.c:383: warning: implicit declaration of function 't30_set_header_info' /root/asterisk/agx-ast-addons/app_rxfax.c:385: warning: passing argument 2 of 't30_set_phase_b_handler' from incompatible pointer type /root/asterisk/agx-ast-addons/app_rxfax.c:386: warning: passing argument 2 of 't30_set_phase_d_handler' from incompatible pointer type make[2]: *** [CMakeFiles/app_rxfax.dir/app_rxfax.o] Error 1 make[1]: *** [CMakeFiles/app_rxfax.dir/all] Error 2 make: *** [all] Error 2 u...@host:~/asterisk/agx-ast-addons# ___ -- 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 Hi, I think the trick is to download trunk version from svn (see voip-info.org for instrcution). Regards ___ -- 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] DTMF tones mid conversation
Hi helpers, I seem to have a problem of intermittent DTMF tones being played during a conversation. Eg: Extn 100 takes an inbound call and all is fine. Except, at an undetermined time the person on extn 100 will here a DTMF tone for no apparent reason (it's not the caller pressing buttons). The caller doesn't hear the tone - only the called person. The call itself progresses normally. I am using PRI -- * -- various SIP hard phones. But I have also heard it on BRI as well. Any ideas where to start looking to cure this? Cheers Andy ___ -- 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] DTMF tones mid conversation
Hi Francois, I am using the latest *, dahdi/zaptel and libpri (1.4-current). This happens with both Zaptel and Dahdi and various versions of * (1.4.22.1 and 1.4.23). So, even the latest 'stable' would seem to have a problem. Cheers Andy -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of F6HQZ -- Sent: 11 February 2009 16:49 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] DTMF tones mid conversation -- -- Hi men, -- -- Resolved for one of my customers by upgrading Asterisk/Libpri/Zaptel. -- I don't remember what wer the versions, sorry. -- Check and advise us the results, please. -- -- Best Regards, -- Francois -- -- No virus found in this outgoing message. -- Checked by AVG - www.avg.com -- Version: 8.0.233 / Virus Database: 270.10.19/1941 - Release Date: -- 02/09/09 06:50:00 -- -- -- ___ -- -- 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] AGI pdf book
Thanks for this Jared (look - back on topic!). I've just ordered the print and downloaded the pdf. It does look very good (the bits I've managed to read so far). I'll give everyone my humble and worthless opinion of it when I get to read it some more. Andy -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Jared Smith -- Sent: 18 February 2009 15:57 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] AGI pdf book -- -- On Wed, 2009-02-18 at 14:36 +0200, michel freiha wrote: -- the asterisk book from oreilly does not make full description to the -- AGI scripting... -- -- You're right.. the O'Reilly book doesn't make a full and complete -- description of AGI programming. (It was better than anything else -- written at the time, but it's nowhere near perfect.) -- -- If you have specific suggestions on what more you'd like to see -- covered -- in the AGI chapter, I'm certainly open to feedback. -- -- I suggest please if someone advice to me a free PDF book just -- dedicated for AGI and nothing else -- -- The only book I'm aware of that covers AGI and only AGI is the AGI -- book -- written by Nir Simionovich. It's not free, but I hear that it's the -- best book in the world on the subject of AGI programming, and I'm -- looking forward to reading it myself. More info at -- http://www.packtpub.com/asterisk-gateway-interface-programming/book -- -- -- -- -- -- Jared Smith -- Digium, Inc. | Training Manager -- -- -- -- -- ___ -- -- 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] Fax detection on SIP channel
Have a look for agx-ast-addons and spandsp. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert McGilvray Sent: 06 March 2009 01:05 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Fax detection on SIP channel Is there a built-in way of detecting fax tones, or a switch to T.38 on a SIP channel? I need to periodically check some efax servers for availability and figured the best way to ensure they are operational is to check for tones. I've looked into Nvdetect but the company seems to have gone out of business and I don't want to be stuck with a solution that won't make it through an upgrade of asterisk. My ITSP supports T.38 and should send a Re-Invite so is there a way to just set a channel variable when it sees this and use that as an indicator of success? I have a sangoma card in the machine but it doesn't have any T1/E1 connections, so unless I'm mistaken I can't use the fax detection in zaptel. Bob This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. (W1) ___ -- 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] DAHDI and B410P (BRI)
Hi all, I am having trouble setting the signalling method for the B410P using DAHDI. Asterisk complains that it has never heard of 'bri_cpe' or 'bri_net' - but it doesn't mind having 'pri_cpe' etc. ERROR[4294]: chan_dahdi.c:11327 process_dahdi: Unknown signalling method 'bri_net' Dahdi - dahdi-linux-complete-2.1.0.4+2.1.0.2 Asterisk - 1.4.23.1 Libpri - 1.4.9 I have set the spans up with no problems (well, dahdi_cfg doesn't complain) - it's just my chan_dahdi.conf file I need to fix now. Thanks Andy ___ -- 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] DAHDI and B410P (BRI)
I have LibPri installed and working (.../wPRI). So, if I understand Tzafrir correctly - DAHDI support for the B410P isn't available in 1.4 at all. Looks like I'm going back to mISDN. Cheers Andy -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Jose Luis Villalon -- Sent: 09 March 2009 18:07 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] DAHDI and B410P (BRI) -- -- Hi -- -- What it's the result of execute -- -- strings /usr/lib/asterisk/modules/chan_dahdi.so | grep '^DAHDI -- Telephony' -- -- It's LibPri install before of Dahdi package? -- -- JL. -- -- El Lun, 9 de Marzo de 2009, 6:36 pm, Andrew Thomas escribió: -- Hi all, -- -- -- I am having trouble setting the signalling method for the B410P -- using -- DAHDI. Asterisk complains that it has never heard of 'bri_cpe' or -- 'bri_net' - but it doesn't mind having 'pri_cpe' etc. -- -- -- ERROR[4294]: chan_dahdi.c:11327 process_dahdi: Unknown signalling -- method -- 'bri_net' -- -- -- Dahdi - dahdi-linux-complete-2.1.0.4+2.1.0.2 -- Asterisk - 1.4.23.1 -- Libpri - 1.4.9 -- -- -- I have set the spans up with no problems (well, dahdi_cfg doesn't -- complain) - it's just my chan_dahdi.conf file I need to fix now. -- -- Thanks -- Andy -- -- -- -- ___ -- -- 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 ___ -- 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] MoH - always starting from the beginning
You could always run a shoutcast server and stream from that. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: 09 March 2009 19:02 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] MoH - always starting from the beginning Hi, I have a customer running a 120 second long WAV file on their MoH. The problem is that it's always starting from the beginning, so people being put on hold, talked to, put on hold again, etc always hear the first 10-15 seconds. Is there a way to have Asterisk MoH remember where it left off? Or at the very least just play the same stream to all people using the same MoH class, so that it just plays like a CD and the person hears wherever the stream is at at a given moment? Regards, Mike ___ -- 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] Update chan_dahdi.conf doc in voip-info.org
Don't forget to mention that the BRI signalling method doesn't work in 1.4 (and probably 1.2) ;). Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 10 March 2009 12:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Update chan_dahdi.conf doc in voip-info.org Hi, It seems BRI signalling settings are missing from http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf I would like to add those parameters : bri_cpe_ptmp bri_cpe bri_net Is this http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf tied to a specific Asterisk version ? Can I edit this ? Regards ___ -- 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] configuring channels for dahdi
Post up your chan_dahdi.conf and we'll fix it :) Hint - you are missing : 'signalling = fxo_ks' and 'signalling = fxs_ks' from it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aqua Man Sent: 10 March 2009 14:21 To: asterisk-users@lists.digium.com Subject: [asterisk-users] configuring channels for dahdi after installing asterisk 1.4.23.1 and dahdi-linux-2.1.0.4 and at CLI module load chan_dahdi.so receive the following: signalling must be specified before any channels are. CLI Warning [4663]: chan_dahdi.c:11627 process_dahdi: Ignoring signalling Error[4663]: chan_dahdi.c:10946 build_channels: Unable to reconfigure channel '1' Error[4663]: chan_dahdi.c:11970 reload: Reload of chan_dahdi.so is unsuccessful! NOTICE[4641]: loader.c:580 ast_module_reload: The module 'chan_dahdi.so' was not properly initialized. Before reloading the module, you must run 'module load chan_dahdi.so' and fix whatever is preventing the module from being initialized. dahdi_cfg -vvv dahdi version: 2.1.0.4 Echo Canceller(s): mg2 Configuration Channel map: channel 01: fxo kewlstart (Default) (Echo Canceler: mg2) (Slaves:01) channel 04: fxs kewlstart (Default) (Echo Canceler: mg2) (Slaves:04) 2 channels to configure. setting echocan for channel 1 to mg2 setting echocan for channel 4 to mg2 sudo cat /proc/dahdi/* Span 1: wctdm/4 'wildcard tdm400p rev I board 5' (master) 1 wctdm/4/0 fxoks (ec: mg2) 2 wctdm/4/1 3 wctdm/4/2 4 wctdm/4/3/ fxsks (ec:mg2) dahdi_hardware -v pci::00:06.0 wctdm+ e159:0001 wildcard tdm400p rev I ls -l /usr/lib/asterisk/modules/chan_dahdiso -rwx-xr-x 1 root root /usr/lib/asterisk/modules/chan_dahdi.so signalling must be specified before any channels are error[4641] core show channels location State Application(Data) 0 active channels 0 active calls any advice? There is no dialtone for the tdm400p card but the lights are activated on the card for ports 1 4 Windows Live(tm) Groups: Create an online spot for your favorite groups to meet. Check it out. ___ -- 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] AGX Asterisk Addon - Can't find app_fax.c withspandsp-0.0.4
You now need to compile and install SpanDSP-0.0.6pre3 at least (AGX has been changed). After you've done that - try AGX again. HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 11 March 2009 06:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] AGX Asterisk Addon - Can't find app_fax.c withspandsp-0.0.4 Hi, I've installed spandsp-0.0.4pre16 With this: cd ~ svn co https://agx-ast-addons.svn.sourceforge.net/svnroot/agx-ast-addons agx-ast-addons cd agx-ast-addons/trunk ./build.sh I've got this: CMake Error in spandsp-0.0.4/CMakeLists.txt: Cannot find source file app_fax.c. Tried extensions .c .C .c++ .cc .cpp .cxx .m .M .mm .h .hh .h++ .hm .hpp .hxx .in .txx This is coherent with : A agx-ast-addons/trunk/spandsp-0.0.4 A agx-ast-addons/trunk/spandsp-0.0.4/app_rxfax.c A agx-ast-addons/trunk/spandsp-0.0.4/app_txfax.c A agx-ast-addons/trunk/spandsp-0.0.4/CMakeLists.txt A agx-ast-addons/trunk/spandsp-0.0.4/README My CMake knowledge is too short to propose a workaround. Maybe, this could come from a change in agx-ast-addons/trunk/spandsp-0.0.4/CMakeLists.txt as the content of this file is: project (app-fax-span4) # -- # Target # we use MODULE cause it build a shared object module # -- ADD_LIBRARY(app_fax MODULE app_fax.c) # # We remove the lib prefix from the libmodule.so filename # SET_TARGET_PROPERTIES(app_fax PROPERTIES PREFIX ) # # We add library dependencies to use those modules # TARGET_LINK_LIBRARIES(app_fax spandsp tiff) # # override default INSTALL rules # INSTALL(TARGETS app_fax DESTINATION lib/asterisk/modules) Could you help ? Regards ___ -- 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] DAHDI and B410P (BRI)
That's at least 2 of us then Paul ;). -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Paul Hales -- Sent: 11 March 2009 00:04 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] DAHDI and B410P (BRI) -- -- -- I wish it was available too - I have just had to back dahdi out of a -- system and revert to misdn after a whole day of testing. -- -- PaulH -- -- -- Andrew Thomas wrote: -- I have LibPri installed and working (.../wPRI). -- -- So, if I understand Tzafrir correctly - DAHDI support for the B410P -- isn't available in 1.4 at all. -- -- Looks like I'm going back to mISDN. -- -- Cheers -- Andy -- -- -- -- -- -Original Message- -- -- From: asterisk-users-boun...@lists.digium.com -- [mailto:asterisk-users- -- -- boun...@lists.digium.com] On Behalf Of Jose Luis Villalon -- -- Sent: 09 March 2009 18:07 -- -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- -- Subject: Re: [asterisk-users] DAHDI and B410P (BRI) -- -- -- -- Hi -- -- -- -- What it's the result of execute -- -- -- -- strings /usr/lib/asterisk/modules/chan_dahdi.so | grep '^DAHDI -- -- Telephony' -- -- -- -- It's LibPri install before of Dahdi package? -- -- -- -- JL. -- -- -- -- El Lun, 9 de Marzo de 2009, 6:36 pm, Andrew Thomas escribió: -- -- Hi all, -- -- -- -- -- -- I am having trouble setting the signalling method for the -- B410P -- -- using -- -- DAHDI. Asterisk complains that it has never heard of -- 'bri_cpe' or -- -- 'bri_net' - but it doesn't mind having 'pri_cpe' etc. -- -- -- -- -- -- ERROR[4294]: chan_dahdi.c:11327 process_dahdi: Unknown -- signalling -- -- method -- -- 'bri_net' -- -- -- -- -- -- Dahdi - dahdi-linux-complete-2.1.0.4+2.1.0.2 -- -- Asterisk - 1.4.23.1 -- -- Libpri - 1.4.9 -- -- -- -- -- -- I have set the spans up with no problems (well, dahdi_cfg -- doesn't -- -- complain) - it's just my chan_dahdi.conf file I need to fix -- now. -- -- -- -- Thanks -- -- Andy -- -- -- -- -- -- -- -- ___ -- -- -- 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 -- -- ___ -- -- 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 ___ -- 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] UK ISDN-30 and ANI
Please explain (in English) what you mean by ANI. Thanks -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith -- Sent: 12 March 2009 10:21 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: [asterisk-users] UK ISDN-30 and ANI -- -- Has anyone in the UK got ANI to work on an inbound call ? -- -- Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 -- -- Julian -- -- __ -- This email has been scanned by the MessageLabs Email Security System. -- For more information please visit http://www.messagelabs.com/email -- __ -- -- ___ -- -- 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] UK ISDN-30 and ANI
I think I understand what you mean now. The biggest difference between CLI and ANI is that ANI can't be blocked/withheld (like you can with CLI by using 141). It also uses different signalling. This is mainly used by law enforcement agencies to trace calls etc. So, you want the number - regardless of what the user dials. I presume you are some sort of 'carrier' then. You'll be lucky to get the information otherwise as it throws up all sorts of privacy laws (ie. you have to have a damn good reason for wanting it). BT are the main people to ask I suppose (unless your calls go through another main carrier). I'm not even sure if ANI signalling is implemented in Asterisk - one for the config file writers ;). Cheers -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith -- Sent: 13 March 2009 10:43 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] UK ISDN-30 and ANI -- -- David Quinton wrote: -- On Thu, 12 Mar 2009 10:21:06 +, Julian Lyndon-Smith -- aster...@dotr.com wrote: -- -- -- Has anyone in the UK got ANI to work on an inbound call ? -- -- Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 -- -- -- -- AFAIK (and our E1 doesn't go to * box) -- a) you mean CLI -- -- a) No I don't. CLI is different to ANI -- b) you have to pay BT extra for Calling Line Identity Presentation -- GBP7.50 / qtr on our last bill -- -- See a). We already have CLI. I need ANI ;) -- HTH -- -- -- ___ -- -- 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 -- -- -- -- -- __ -- This email has been scanned by the MessageLabs Email Security System. -- For more information please visit http://www.messagelabs.com/email -- __ -- -- ___ -- -- 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] PBX to gate interface
There are various ways of doing this. You could use an analogue port/ATA and connect any good old fashioned intercom to it (Pantel are a good make). You can now get SIP intercom systems as well. I haven't tried on of these - but they look good (and can contain a camera as well if needed). HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Mason (Lists) Sent: 17 March 2009 13:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PBX to gate interface Has anyone found a good wayt o do a gate intercom using Asterisk? I am looking at a Xorcom PBX with programmable contact, so I have no issue with opening the gate, but the interface at the gate is a bit tricky. I thought about a weather proof housing containing a phone but it seems a bit tacky. I also looked at a handsfree erather proof phone, but at $600 it is a bit steep. Any solutions that have been implemented successfully? -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] PBX to gate interface
Have a look at http://www.northsupply.co.uk/ (under Door Access Systems). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Mason (Lists) Sent: 18 March 2009 11:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PBX to gate interface How does a Push-to-talk intercom interface with Asterisk? Andrew Thomas wrote: There are various ways of doing this. You could use an analogue port/ATA and connect any good old fashioned intercom to it (Pantel are a good make). You can now get SIP intercom systems as well. I haven't tried on of these - but they look good (and can contain a camera as well if needed). HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Mason (Lists) Sent: 17 March 2009 13:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PBX to gate interface Has anyone found a good wayt o do a gate intercom using Asterisk? I am looking at a Xorcom PBX with programmable contact, so I have no issue with opening the gate, but the interface at the gate is a bit tricky. I thought about a weather proof housing containing a phone but it seems a bit tacky. I also looked at a handsfree erather proof phone, but at $600 it is a bit steep. Any solutions that have been implemented successfully? -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] DTMF tones mid conversation
Just to add P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk P[ 1] Sending :160 bytes 2 MISDN P[ 0] misdn_jb_fill: written:160 | Buffer status:256 p:861fee0 P[ 0] misdn_jb_empty: read:128 | Buffer status:128 p:861fee0 P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk P[ 1] PH_CONTROL: channel:1 oad2:07n dad0:820055 P[ 1] -- DTMF TONE: 7 P[ 0] get_index: event not found! P[ 1] I IND :DTMF_TONE oad:07n dad:820055 pid:64 state:CONNECTED P[ 1] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:820055 P[ 1] -- info_dad: onumplan:2 dnumplan:0 rnumplan: cpnnumplan:0 P[ 1] -- caps:Speech pi:0 keypad: sending_complete:1 P[ 1] -- screen:0 -- pres:0 P[ 1] -- addr:50010102 l3id:2000b b_stid:10010100 layer_id:50010180 P[ 1] -- facility:Fac_None out_facility:Fac_None P[ 1] -- urate:0 rate:16 mode:0 user1:0 P[ 1] -- bc:81aecfc h:0 sh:0 P[ 1] -- bc_state:BCHAN_ACTIVATED P[ 1] -- DTMF:7 P[ 1] Sending :160 bytes 2 MISDN P[ 0] misdn_jb_fill: written:160 | Buffer status:288 p:861fee0 P[ 1] * IND : Indication [20] from s P[ 1] -- * Unknown Indication:20 pid:64 P[ 1] * IND : Indication [20] from s P[ 1] -- * Unknown Indication:20 pid:64 P[ 0] misdn_jb_empty: read:128 | Buffer status:160 p:861fee0 P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk P[ 0] misdn_jb_empty: read:128 | Buffer status:32 p:861fee0 P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk mISDN is the culprit (as some have already concluded). That DTMF TONE: 7 was produced in mid call. Has anyone come up with a plan to fix this yet? Ta -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of stoffell Sent: 26 February 2009 19:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF tones mid conversation On Thu, Feb 26, 2009 at 6:08 PM, Simon Dixey simon_...@hotmail.co.uk wrote: I wonder if anyone is able to offer any [polite ;-)] words of wisdom?? I can be polite, I'm not sure about the wisdom .. :-) DTMF threshold in misdn-init look high doesn't it... I'm not entirely sure what it should be set to, to be honest.. (min-max values for tuning); have read max value is 100, but others suggest it'll go higher - but what exactly is it tuning the sensitivity value of specifically? (i.e. what is the threshold value). DTMF detection works well for *genuine* DTMF digits dialled over the ISDN trunk, but mISDN/Asterisk still recognises them incorrectly at times during calls (to/from cell phones). I'm using almost the same setup but I'm in Belgium. Also a b410p. The dtmf seems to get triggered more by some calls then other. It also depends on the voice. (higher tones trigger dtmf more easily) My dtmfthreshold is set to 100. Guess it's the (in)sane default ? :-) Oh, are you having random crashes on your mISDN setup too ? I've also seen other posts refer to settings in Dahdi.conf (such as relaxdtmf) - surely Dahdi doesn't have anything to do with this if I'm using chan_misdn?? Correct. Only has anything to do with it if you're using chan_dahdi. I can't reproduce the same behaviour on dahdi. Is anyone able to confirm exactly whether mISDN's hardware DSP and driver is responsible for detecting DTMF, or whether it's Asterisk analysing the inbound audio? Scanning the README.misdn (sourced separately) the chan_misdn driver readme comments a feature as DTMF Detection in HW+mISDNdsp (much better than asterisks internal!) - so surely DTMF is recognised and passed on by mISDN to Asterisk. The fact that the log messages prefixed by P[ 1] are mISDN - I think I've answered my own question there... Yes, it's mISDN that detects the dtmf. Prior to going down the mISDN route, I looked at Dahdi as the Dahdi configs mention native Dahdi B410P support. But, the conclusion I came to (although what I read didn't make it clear me) is that the readme was referring to Dahdi B410P support in Ast 1.6, not 1.4. That sound right? Dahdi readme: Right again. i've been experimenting with Dahdi's b410p for a while now, it's only available in asterisk 1.6, with the latest libpri and dahdi releases. It's much cleaner, imho, but I'm having issues with receiving faxes when using dahdi, so I'm stuck with mISDN for the moment :-) Enough reading.. if you're still awake! Any help would be very much appreciated. Nice to see someone else is using the same setup. I was beginning to think that people with BRI stopped using asterisk in Europe :-) Keep in touch or post to the mailing list if you have any further news/experiences.. cheers, stoffell ___ -- 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] Help: RED alarm on Wildcard TE122 card
This sounds like you have pri_net instead of pri_cpe in Zapata.conf. When inserting the cable going into TE122 into an ISDN phone, the phone works perfectly. Any suggestions would be greatly appreciated :-) ___ -- 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] Grandstream 2010 and blinky lights
The quick answer is 'no'. It is not currently possible to monitor 'hints' for Agents - as an Agent never actually dials out (the device does). Even exten = 1234,hint,Agent/1234 won't work - as the 'core show hints' will show the agent as 'notinuse' when they can be. There are ways around it (I used a mixture of php and mysql) - but even these are not ideal (especially if you have a large dial plan). Clue : exten 1234,hint,SIP/ABC works - you just need to change the ABC bit every time an agent logs in our out. This then gives you the lovely job of lighting any MWI lamps for that user as well. Oh the joys of Asterisk and hotdesking! HTH Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: 02 July 2009 17:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Grandstream 2010 and blinky lights I am using 1.4, and have the above device, and it worked really well with monitoring 18 hints aka devices. Now, I've moved us to a hotdesking paradigm where the user is the extension not the device. IOW if I dial 1234, I will get user 1234 (who happens to log on to device ABC today, and DEF tomorrow). Can I make the GXP monitor user 1234, not extension 1234 ? Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] Grandstream 2010 and blinky lights
That's exactly the way I do it as well :D -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: 06 July 2009 11:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Grandstream 2010 and blinky lights Thanks for the info. We've managed to achieve or goal using 1.4 and a few hacks. 1) When the agent logs in / logs out, we rewrite the part of the dialplan for the hints and reload the dialplan 10 seconds after the *last* login / logout 2) For the MWI, we give each phone a fake voicemail (let's say _0001_). When an agent logs in, we link /var/spool/asterisk/voicemail/_0001_ to /var/spool/asterisk/voicemail/[mailbox] (where [mailbox] is the mailbox of the agent) and when they log out, we remove /var/spool/asterisk/voicemail/_0001_ This seems to work - the MWI lights up / off depending on the new vm within a couple of seconds 3) When checking for voicemail, each phone is configured to dial - the dialplan then checks the callerid (set by #1) and gets the mailbox for the agent. As I said, a bit of a hack, but it works for me ;) I know that this won't work for 1.6, but we are coming up with an alternative plan using Minivm Julian Andrew Thomas wrote: The quick answer is 'no'. It is not currently possible to monitor 'hints' for Agents - as an Agent never actually dials out (the device does). Even exten = 1234,hint,Agent/1234 won't work - as the 'core show hints' will show the agent as 'notinuse' when they can be. There are ways around it (I used a mixture of php and mysql) - but even these are not ideal (especially if you have a large dial plan). Clue : exten 1234,hint,SIP/ABC works - you just need to change the ABC bit every time an agent logs in our out. This then gives you the lovely job of lighting any MWI lamps for that user as well. Oh the joys of Asterisk and hotdesking! HTH Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: 02 July 2009 17:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Grandstream 2010 and blinky lights I am using 1.4, and have the above device, and it worked really well with monitoring 18 hints aka devices. Now, I've moved us to a hotdesking paradigm where the user is the extension not the device. IOW if I dial 1234, I will get user 1234 (who happens to log on to device ABC today, and DEF tomorrow). Can I make the GXP monitor user 1234, not extension 1234 ? Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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 ___ -- 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] Grandstream 2010 and blinky lights
Because DEVSTATE is for custom hints - and have you tried to set one every time a phone rings/is answered? This was thought about - but the logic in the dialplan would be a nightmare. Anyway, doing it the way I do it works for me (and others) as my dialplan contains nothing but 'include' and 'switch' statements now (so it reloads fast). Thanks for the reply though :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: 08 July 2009 09:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: aster...@dotr.com Subject: Re: [asterisk-users] Grandstream 2010 and blinky lights On 8/7/09 8:52 PM, Andrew Thomas wrote: That's exactly the way I do it as well :D -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: 06 July 2009 11:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Grandstream 2010 and blinky lights Thanks for the info. We've managed to achieve or goal using 1.4 and a few hacks. Why don't you just use func_devstate which was backported to 1.4? That way you can just set a DB variable on login/logout. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] DEVICE_STATE() and Asterisk 1.6.0.10
Why are you putting semi-colons at the end of every line? The dialplan
isn't written in PHP ;).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry L.
Kline
Sent: 15 July 2009 23:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Mark Michelson wrote:
You need to set a call-limit for the SIP peer. Device state
calculation for a
SIP peer is predicated on both the call-limit and busylevel. Let's say
that you
were to have a call-limit of 2, but no busylevel set. These are the
device
states reported for the peer based on the number of calls currently
handled:
Hi Mark. Thanks for your explanation of these parameters.
I should have posted my configurations. I double-checked the contents
of sip.conf and I have this. The 'subscribecontext' was added for
testing, per the other reply I got for my question.
;
; Settings common to all devices on our system
;
[basic-options](!)
type=friend
host=dynamic
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
qualify=yes
;
; Standard desksets here
;
[lan-deskset](!,basic-options)
context=sip-deskset
notifyringing = yes
notifyhold = yes
limitonpeers = yes
call-limit=99
[6668](lan-deskset)
secret=mysecret
callerid=Matts SIP 6668
username=Barry's IP450
call-limit=32
busylevel=1
subscribecontext=hint-context
My hint-context is:
[hint-context]
exten = 6668,hint,SIP/6668;
I'm still not getting anything other than NOT_INUSE from DEVICE_STATE.
Here is the CLI output:
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:1]
NoOp(SIP/-0955ecc8, SIP/6668 has state NOT_INUSE) in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:2]
NoOp(SIP/-0955ecc8, SIP/ has state NOT_INUSE) in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:3]
ExecIf(SIP/-0955ecc8, 0?Busy(10)) in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:4]
Dial(SIP/-0955ecc8, SIP/6668) in new stack
And here is sip show inuse:
corp-asterisk*CLI sip show inuse
* User name In use Limit
6668 1 32
6667 0 99
1 99
* Peer name In use Limit
6668 1/1/0 32
6667 0/0/0 99
0/0/0 99
For completeness, here is the dialplan that's producing this:
exten = 6668,1,NoOp(SIP/${EXTEN} has state
${DEVICE_STATE(SIP/${EXTEN})});
exten = 6668,n,NoOp(SIP/ has state ${DEVICE_STATE(SIP/)});
exten =
6668,n,ExecIf($[${DEVICE_STATE(SIP/${EXTEN})}=BUSY]?Busy(10));
exten = 6668,n,Dial(SIP/${EXTEN});
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Version: GnuPG v1.4.5 (GNU/Linux)
iD8DBQFKXlwaCFu3bIiwtTARAkRpAJ4+2WF9qrIwrC3Kdpwd0YAOm/5S1wCfUR1T
CtI9kZNQYpW2Sv6uFNud7Jo=
=9Zp/
-END PGP SIGNATURE-
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Re: [asterisk-users] AGI to announce temperature from weather.com XMLfile
I have just the thing in PHP. Drop me a personal e-mail and I'll whiz it over. Andrew Thomas Technical Services Manager a...@datavox.co.uk DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Trevor Hammonds Sent: 16 July 2009 12:50 To: 'Asterisk Users List' Subject: [asterisk-users] AGI to announce temperature from weather.com XMLfile 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 ___ -- 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] AGI to announce temperature from weather.com XML file
It appears I opened some flood gates when I offered my 'alternative' version. So, rather than send hundreds of e-mails out - here's the link : http://www.dv-ip.com//downloads/files/misc/weather.txt Any questions - just 'yell'. Andrew Thomas Technical Services Manager a...@datavox.co.uk DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen Sent: 17 July 2009 18:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AGI to announce temperature from weather.com XML file Trevor Hammonds wrote: 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. Since your problem came up on the VoIP Users Conference today, it ended up being the basis for a blog post I wrote today. The blog post (which may solve your problem) is available here: http://leifmadsen.wordpress.com/2009/07/17/howto-read-a-value-from-a-fil e-and-say-it-back/ Let me know if that works for you -- just respond on the comments section since I don't always check this users list. Note: I haven't actually tested the dialplan yet, so if someone can test it for errors, let me know if you run into any, and I'll update the blog post with any that may be found. Thanks! Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- 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] sip configuration masking the peers
'host=dynamic' is your problem - as this allows any IP address to register as
that friend - assuming they know the password/username combination.
Why not simply have group 1 as 'secret=pass123' and group2 as 'secret=pass456'?
Just don't tell group 1 uses the password for group 2 - and vice-versa!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of peace keeper
Sent: 22 July 2009 09:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] sip configuration masking the peers
Hi all,
I need to specify two groups of peers who are on two sub networks, the
case is as follows:
two groups of users (that are supposed to use the X-lite) group1 and group2,
each group is on a sub network net1, and net2, respectively, each group has
its own dial plan defined in the extension.conf,
we have defined the peers in the sip.conf for both groups, and successfully
made a call between two peers from the groups, but the idea is we need to
prevent users from network1 to register as peers of group1,
I suppose this would be a configuration solution, but I am afraid that do know
what are the right needed configurations:
here is definition of two peers each from different group:
[1010]
type=friend
host=dynamic
context=group1
secret=pass
host=dynamic
callerid=TestAccount1010
vm Extension=test 1010
mailbox=1...@default
nat=yes
[2003]
type=friend
context=group2
secret=pass
host=dynamic
callerid=Account2003
vm Extension=test 2003
mailbox=2...@default
nat=yes
each of group1 and group2 context are defined in the extension configuration as
follows :
exten = _2XXX,1,Dial(SIP/${EXTEN})
exten = _2XXX,n,Playback(unavailable)
exten = _2XXX,n,Hangup()
exten = _1XXX,1,Dial(SIP/${EXTEN})
exten = _1XXX,n,Playback(unavailable)
exten = _1XXX,n,Hangup()
in order the both groups can talk to each other,
currentlly users in network1 can register as peer 2003 which is supposed to be
allowed just for users from network2 , although this registration is supposed
to be failed, any suggestions plz!!
hope I made the scenario clear , any help would appreciated.
Thanks in advance.
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Re: [asterisk-users] Music on hold based on user
I do this using the setvar facility in sip.conf.
eg. setvar=MOH=music1
Then in the dialplan (extensions.conf) all you need to do is
'Set(CHANNEL(musicclass)=${MOH})'
Remember, setvar in sip.conf makes that variable a global variable.
Andrew Thomas
Technical Services Manager
Juan C. Crespo R. wrote:
Hi
Guys I wonder if its possible to set a different MoH based on
groups, I mean if one of the Admin group put on hold the call play
music 1, if another from Technical Support put on hold the call play
music 3, something like this
Admin - Music1
Contrallors - Music 2
Technical Support - Music 3
Thanks
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Re: [asterisk-users] Music on hold based on user
You can use 'Set(CHANNEL(musicclass)=${MOH})' anywhere in your dialplan - so
you can set it at any stage of an inbound or outbound call (as long as it is
before the Dial/Queue command).
Eg:
[inbound]
exten = _X.,1,Set(CHANNEL(musicclass)=${MOH})
exten = _X.,n,Dial(whomever-you-want)
[outbound]
exten = _X.,1,Set(CHANNEL(musicclass)=${MOH})
exten = _X.,n,Dial(where-ever-you-want)
Then, when 'whomever-you-want' puts the call on hold - they get
'whomever-you-want's MOH.
Simples :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen
Sent: 24 July 2009 14:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Music on hold based on user
Andrew Thomas schrieb:
I do this using the setvar facility in sip.conf.
eg. setvar=MOH=music1
Then in the dialplan (extensions.conf) all you need to do is
'Set(CHANNEL(musicclass)=${MOH})'
Juan C. Crespo R. wrote:
Guys I wonder if its possible to set a different MoH based on
groups, I mean if one of the Admin group put on hold the call play
music 1, if another from Technical Support put on hold the call play
music 3, something like this
Admin - Music1
Contrallors - Music 2
Technical Support - Music 3
The way I understood the OP was that he wants different MoH classes
depending on the callee (not depending on the caller).
Philipp Kempgen
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Re: [asterisk-users] Possibly I don't understand sip peers
[peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 read what you've put!!! The 'allow' should be 'permit' as Jared already told you (and he should know what he's talking about). insecure=port,invite -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Ferrell Sent: 29 July 2009 23:34 To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Possibly I don't understand sip peers Jared Smith wrote: On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote: I have a carrier who tells me he will be sending me traffic from a wide range of IP addresses. so I set up a realtime peer as follows: [peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 insecure=port,invite Yes, he's really claiming to originate from any of the IP in the block When I leave the host blank, we reject calls with a 404. shouldn't I be able to put in a kind of wildcard for his IP block or am I just being silly? If not, what am I doing wrong? I think you've got your syntax wrong there... permit and deny statements are used to create Access Control Lists and to limit the IP address ranges. The allow and disallow statements are to allow or disallow various codecs. They way you've specified it above, you're allowing a codec called xxx.xxx.xxx.0/255.255.255.0, which probably isn't what you want. I have the codec permissions in the columns allow and disallow. Those seem to work ok. it's permit/deny/mask I seem to be having a problem with. Like I say, I don't think I understand their use or perhaps they don't work in realtime ___ -- 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 ___ -- 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] context does not work
Underscore won't help as that's for pattern matching. Under the sip conf, have you tried adding 'fromuser=8001187e0' to the [8001187e0] bit? I have this in my Sipgate setup and it works. Worth a try. Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Plattes Sent: 10 August 2009 11:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] context does not work Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- 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 ___ -- 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] context does not work
V1.6.1.0 [9290740] type = peer username = 9290740 fromuser = 9290740 secret = you-wish! host = sipgate.co.uk fromdomain = sipgate.co.uk insecure = port,invite context = inbound caninvite = no canreinvite = no nat = yes disallow = all allow = ulaw allow = alaw dtmfmode = info qualify = 5000 That works for me. Any inbound call to my 9290740 number goes to my inbound context and does what it should. PS - Don't forget to do a 'sip reload' when you change the sip.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Plattes Sent: 10 August 2009 13:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] context does not work Hi Andrew, it didn't help. Which version of Asterisk do you use? Thanks On Mon, Aug 10, 2009 at 1:55 PM, Andrew Thomasa...@datavox.co.uk wrote: Underscore won't help as that's for pattern matching. Under the sip conf, have you tried adding 'fromuser=8001187e0' to the [8001187e0] bit? I have this in my Sipgate setup and it works. Worth a try. Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Plattes Sent: 10 August 2009 11:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] context does not work Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- 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 ___ -- 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 -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- 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 ___ -- 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] stutter playback
This sounds more like the alarm system putting pulses/tones on the line (maybe the alarm has a dialler/anti-cut-line-detection? So, as the alarm is adding stuff AFTER the asterisk box - I doubt you will see anything on the PC itself. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Samad Sent: 22 August 2009 04:48 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] stutter playback On Fri, Aug 21, 2009 at 08:53:23AM -0400, Steve Totaro wrote: On Fri, Aug 21, 2009 at 8:39 AM, Alex Samad a...@samad.com.au wrote: Hi I had a working system, until recently - its asterisk 1.6.1 from debian - not the lastest as the last doesn't seem to work. but somebody who rang me said my voice mail announcement was all stuttery. so i dialed my voicemail box and its really stuttery... so I have done a reboot and its just as bad, now I am not sure what to check to try and get this working again . Alex I would check cpu, diskpace, memory, I/O, network wasn't that, I have a alarm system on the backup pstn line, seems like there is something wrong there, cause when I remove the alarm system from the equation everything seems okay, so I am guessing it was causing some problem on my tdm410 card. strange thing is i did not see any spikes on io , cpu, network... Alex -- Think of it! With VLSI we can pack 100 ENIACs in 1 sq. cm.! ___ -- 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] Prevent Agent Login from a second extension
The only way around the 'auto-logout' problem I found was to call a script when agents login. This script checks to see if they are already logged in or not - then, if they are, it does whatever I want (I manually log them off the other phone first - you could play a message instead). HTH Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shanavaz E A Sent: 02 September 2009 07:27 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Prevent Agent Login from a second extension Hi friends, Is there any way to prevent an Agent from logging in from a second extension if he is already logged on from an extension. Right now, the scenario is if he login from a second extension, asterisk will automatically log him off from first extension. What I need is that asterisk should tell him that he is already logged on from an extension and should prevent him from logging in again from another extn. The problem with existing scenario is that, I am not getting CDR record for the automatic log out event. I need this for evaluation purposes. I am using asterisk 1.2.30. I have 1.4 also but that also is having the same behavior. Thanks in advance for any help. Regards Shanavaz. ___ -- 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] onnecting two asterisk using B410p BRI cards
...and did you switch the termination dip switches over (on the NT ports of the B410P card)? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of voip crazy Sent: 17 August 2009 07:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] onnecting two asterisk using B410p BRI cards I just plug the junper in NT mode with no success. VoipCrazy 2009/8/15 Paul Hales pdha...@optusnet.com.au: Use a standard network cable - but you have to activate the 'terminate' jumper on the NT end. - Also, the new BRI stuff in dahdi is much easier to work with than misdn. PaulH voip crazy wrote: Hello all, I'm trying to conect two asterisk servers using two B410p Digium cards. One card on each server. I just setting up the first BRI port on server A as nt_ptp and the first BRI port on server B as te_ptp. I use an ethernet wire to connect the first port of server A (nt_ptp) with the first port on server B (te_ptp) but the port light cotinues blinking on red on both sides once the cable was pluged. Then I use an isdn crossover wire with this king of schema and the lights get blinking red again. Tx+ 3 --+ +- 3 . X Rx+ 4 --+ +- 4 . Tx- 5 --+ +--5 . X Rx- 6 --+ +--6 In both servers when I do in asterisk CLI misdn shos stacks, the port one on each machine shows Server A: BEGIN STACK_LIST: * Port 1 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 Server B: BEGIN STACK_LIST: * Port 1 Type NT Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 Which kind of cable should I use? Why both in ports L1Link is failed? How could I solve that? Any clue will be welcomed. Thanks in advance. VoipCrazy. ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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] Ringing for incoming call
exten = did,1,Answer exten = did,n,Playtones(ring) exten = did,n,Wait(10) exten = did,n,StopPlaytones() exten = did,n,BackGround(sound file) did = the DID number as presented and note the '1' before Answer. This works for me. exten = 820055,1,Answer() exten = 820055,n,PlayTones(ring) exten = 820055,n,Wait(5) exten = 820055,n,StopPlayTones() exten = 820055,n,[do something interesting from now on] That's my DID (820055) being answered first and then waiting for 5 seconds. I use it for fax detect this way. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Smither Sent: 18 December 2009 23:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ringing for incoming call Dear All, I am using Asterisk 1.4 on CentOS 5. I have an incoming DID provided by Vitelity. When the number is called it goes to my Asterisk box. The protocol is SIP. This all works just fine if I answer the call and begin a playback. I want to let the number ring for a few seconds before it is answered, and would like the caller to hear it ringing. I have tried: ... exten = s,n,Answer exten = s,n,Playtones(ring) exten = s,n,Wait(10) exten = s,n,StopPlaytones() exten = s,n,BackGround(sound file) ... also ... exten = s,n,Answer exten = s,n,Ringing() exten = s,n,Wait(10) exten = s,n,BackGround(sound file) ... I have also tried moving the Answer app to right before the BackGround app. In all cases when I call the number I never hear it ringing. After the 10 second delay, the BackGround app does run. Connecting to the CLI does not give me any useful information - for example the Ringing app is shown to run, but the caller does not hear it. Any suggestions? Many thanks! -- Bob Smither smit...@c-c-i.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 -- _ -- 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 50-limit blf
Hello all, Just wondering if anyone ever solved the Aastra 50-BLF limit when used with Asterisk (any flavour)? I know it's not strictly and Asterisk question - but I'm sure there's plenty of you out there using Aastra's on the end. Cheers, Andrew Thomas dCAP #1473 -- _ -- 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] 'dirty' upgrade of 1.4
Apologies if this has been asked before. Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1? Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the source files for 1.4.34 over the top of the existing 1.4.24.1 files. Obviously, I will need to keep my config files (and sound files etc) - so I'll back them up first. Also, will I need to stop * to perform this routine - or can I just 'upgrade' and then do a * 'restart'? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'dirty' upgrade of 1.4
Hi Danny, I understand (and welcome) the separate src directories. This would allow me to 'revert' should I feel the need (assuming I can just re-compile over each one). I just need to know if I can re-compile over the existing first. Thanks for your reply. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 26 July 2010 14:15 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 'dirty' upgrade of 1.4 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- Subject: [asterisk-users] 'dirty' upgrade of 1.4 Apologies if this has been asked before. Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1? Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the source files for 1.4.34 over the top of the existing 1.4.24.1 files. Also, will I need to stop * to perform this routine - or can I just 'upgrade' and then do a * 'restart'? Question 1 - unless you are un-tarring to a specific directory, you would have /usr/local/src/asterisk-1.4.24.1 and /usr/local/src/asterisk-1.4.34 segregated source trees. Question 2 - you don't have to stop asterisk, but you should (best practice?) since installing a new release usually involves removing/replacing the .so files in /usr/lib/asterisk/modules. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'dirty' upgrade of 1.4
Thanks to everyone who replied. This is great news ;). I'll get the thing upgraded tonight (when it's quiet). Thanks again. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner Sent: 26 July 2010 16:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 'dirty' upgrade of 1.4 When you run make, it compiles the binaries in the src directory. Once it is done compiling stop asterisk. Running make install will copy the compiled binaries into their respective folders on your system. Then just start asterisk. If you need to revert, stop asterisk, run make install in the old src directory, then start asterisk. Ryan On Mon, Jul 26, 2010 at 9:45 AM, Andrew Thomas a...@datavox.co.uk wrote: Hi Danny, I understand (and welcome) the separate src directories. This would allow me to 'revert' should I feel the need (assuming I can just re-compile over each one). I just need to know if I can re-compile over the existing first. Thanks for your reply. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 26 July 2010 14:15 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 'dirty' upgrade of 1.4 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- Subject: [asterisk-users] 'dirty' upgrade of 1.4 Apologies if this has been asked before. Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1? Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the source files for 1.4.34 over the top of the existing 1.4.24.1 files. Also, will I need to stop * to perform this routine - or can I just 'upgrade' and then do a * 'restart'? Question 1 - unless you are un-tarring to a specific directory, you would have /usr/local/src/asterisk-1.4.24.1 and /usr/local/src/asterisk-1.4.34 segregated source trees. Question 2 - you don't have to stop asterisk, but you should (best practice?) since installing a new release usually involves removing/replacing the .so files in /usr/lib/asterisk/modules. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Upgrade from 1.4 to 1.6 : problems with realtimemysql
This is a problem with extconfig.conf - not your res_ or cdr_ ones. In your case - extconfig.conf probably contained something like 'sippeers = mysql,MyDBase,sippeers'. The 'problem' is that the middle parameter is no longer for the database name - it is for the context in res_mysql.conf. So, the above now becomes 'sippeers = mysql,general,sippeers'. Give that a go... Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: 08 September 2010 15:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtimemysql Hello, in asterisk 1.4.30 all realtime configurations go well. In asterisk 1.6.2.11 the following appears on CLI : [Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql: MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf) [Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql: MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf) res_mysql.conf : [general] dbhost = 127.0.0.1 dbname = MyDBase dbuser = asterisk dbpass = mysecret dbport = 3306 dbsock = /tmp/mysql.sock requirements=warn ; or createclose or createchar What do I need to change to be conform asterisk 1.6 ?! Reloading, restarting asterisk and restarting my CentOS-server all doesn't help. Jonas. If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] High volume BLF - Suggestions?
As a side note to this - do NOT try and use Aastra's - as they tend to crash after 50 BLF's! Also, could you please send me (perhaps off-list to a...@datavox.co.uk) your Yealink T28 findings - as I am a beta tester for them? Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 13 September 2010 11:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] High volume BLF - Suggestions? 2010/9/13 Steve Davies davies...@gmail.com Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite low-end hardware, but we cannot find any handsets that can cope with it longer term. Our test involves about 10 BLF-NOTIFY messages per second to each handset with a 5-second pause every 5 seconds. This will either crash or render unusable all of the following combinations: snom360 + 1 x sidecar As Snom phones have a parameter to express a time period during which BLF's SUBSCRIBE messages are spread and sent to Asterisk, I thought Snom phones would handle this load more easily. Yealink T28 + 1 x sidecar Yealink T28 + 2 x sidecar Cisco SPA504g + 1 x sidecar Cisco SPA504g + 2 x sidecar Cisco SPA525g + 1 x sidecar (reboots often) Cisco SPA525g + 2 x sidecar (reboots quickly) Aastra 55i + non-LCD sidecar Did not try Polycom as they do not do directed pickup and only small sidecars. Linksys SPA962 with one sidecar is OK but is discontinued hardware. Help? Thanks, Steve -- _ -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime semi-colon
Hi list, Does anyone know how to send * a semi-colon from a realtime database. I know that * uses the semi-colon as a 'seperator' - but I need to be able to use one in a command. I know I can use \; in the non-realtime configs, but this doesn't work in realtime. Cheers, Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime semi-colon
I'd forgot about doing it that way (I use that for $).
Thanks for the memory jog :)
Cheers
Andy
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 16 September 2010 13:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime semi-colon
On 16 Sep 2010, at 12:56, Andrew Thomas wrote:
Does anyone know how to send * a semi-colon from a realtime database.
I know that * uses the semi-colon as a 'seperator' - but I need to be
able to use one in a command. I know I can use \; in the non-realtime
configs, but this doesn't work in realtime.
in /etc/asterisk/extensions.conf
[globals]
SEMICOLON=\;
Then use ${SEMICOLON} in realitime Hacky, but it's what I'm using at
the moment..
S
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Re: [asterisk-users] Not able to join conference
What happens if you put in a 'room' number? Eg: exten = 8080,3,MeetMe(500|MDci) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: 17 September 2010 14:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Not able to join conference Hi All, We are running to a weird problem, we're using asterisk 1.2 as a production server (I'm wiling to move very soon to more recent version) and our problem is when somebody try to join a conference he's told that he's the only one in the conference but in fact there is some 3 or 5 or whatever people in that same conference, after several tries he can/cannot enter the conference and meet with the people already in, here is the lines corresponding to conf in the dialplan, that would be a big help if you guys can help diagnose the issue. exten = 8080,1,Answer exten = 8080,2,Wait,1 exten = 8080,3,MeetMe(|MDci) If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Not able to join conference
I was wondering what happened if YOU put that number in. Does it put everyone in to the same conference? That would, at least, prove that the MeetMe app was working as it should (unless you've tried this already). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: 20 September 2010 14:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Not able to join conference it's going to put you in conf no 500 without prompting you to enter a conference number I guess, but i don't it's going to solve my issue. actually I'm atill wondering is there a way to debug just Meetme app output or the only way is turn the whole debug thing on? On Mon, Sep 20, 2010 at 4:11 AM, Andrew Thomas a...@datavox.co.uk wrote: What happens if you put in a 'room' number? Eg: exten = 8080,3,MeetMe(500|MDci) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: 17 September 2010 14:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Not able to join conference Hi All, We are running to a weird problem, we're using asterisk 1.2 as a production server (I'm wiling to move very soon to more recent version) and our problem is when somebody try to join a conference he's told that he's the only one in the conference but in fact there is some 3 or 5 or whatever people in that same conference, after several tries he can/cannot enter the conference and meet with the people already in, here is the lines corresponding to conf in the dialplan, that would be a big help if you guys can help diagnose the issue. exten = 8080,1,Answer exten = 8080,2,Wait,1 exten = 8080,3,MeetMe(|MDci) If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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 -- Abdullah -- _ -- 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] Weird Behavior with DAHDI
Downgrade your LibPri instead (1.4.10.2 is fine). See https://issues.asterisk.org/view.php?id=17270 for more info. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias Sent: 29 September 2010 13:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Weird Behavior with DAHDI Hello, I'm experiencing some weird problems on my server: - 1) The following messages are filling up my logs: [Sep 29 08:24:59] WARNING[7077]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! [Sep 29 08:24:59] WARNING[7078]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 171 as D-channel anyway! [Sep 29 08:24:59] WARNING[7075]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 78 as D-channel anyway! [Sep 29 08:24:59] WARNING[7079]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 202 as D-channel anyway! [Sep 29 08:24:59] WARNING[7073]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! [Sep 29 08:24:59] WARNING[7080]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 233 as D-channel anyway! [Sep 29 08:24:59] WARNING[7074]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 47 as D-channel anyway! In the Asterisk CLI, i'm watching these messages constantly 2) I've plugged in a real E1 PRI ISDN: r...@sangoma-testing:/usr/src# asterisk -rx 'pri show spans' PRI span 1/0: Provisioned, In Alarm, Down, Active PRI span 2/0: Provisioned, In Alarm, Down, Active PRI span 3/0: Provisioned, In Alarm, Down, Active PRI span 4/0: Provisioned, Up, Active PRI span 5/0: Provisioned, In Alarm, Down, Active PRI span 6/0: Provisioned, In Alarm, Down, Active PRI span 7/0: Provisioned, In Alarm, Down, Active PRI span 8/0: Provisioned, In Alarm, Down, Active Seems to be OK! but i can't make a call: -- Executing [691918...@pbx1:1] Dial(SIP/xtravoip200-021a47e0, DAHDI/g4/691918892|30|m) in new stack [Sep 29 08:29:51] WARNING[7338]: channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 [Sep 29 08:29:51] WARNING[7338]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [691918...@pbx1:2] Hangup(SIP/xtravoip200-021a47e0, ) in new stack == Spawn extension (pbx1, 691918892, 2) exited non-zero on 'SIP/xtravoip200-021a47e0' What is happening? Could you let me know how to debug or to understand the output from pri intense debug span 4? TEI: 0 State 7(Multi-frame established) V(A)=30, V(S)=30, V(R)=30 K=7, RC=0, l3initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ 00 01 01 3d ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 030 P/F: 1 0 bytes of data -- Starting T200 timer Sangoma-Testing*CLI TEI: 0 State 8(Timer recovery) V(A)=30, V(S)=30, V(R)=30 K=7, RC=0, l3initiated=0, reject_except=0, ack_pend=0 T200_id=1, N200=3, T203_id=0 [ 02 01 01 3d ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 030 P/F: 1 0 bytes of data TEI: 0 State 8(Timer recovery) V(A)=30, V(S)=30, V(R)=30 K=7, RC=0, l3initiated=0, reject_except=0, ack_pend=0 T200_id=1, N200=3, T203_id=0 [ 02 01 01 3d ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 030 P/F: 1 0 bytes of data -- Got ACK for N(S)=30 to (but not including) N(S)=30 Done handling message for SAPI/TEI=0/0 Sangoma-Testing*CLI TEI: 0 State 8(Timer recovery) V(A)=30, V(S)=30, V(R)=30 K=7, RC=0, l3initiated=0, reject_except=0, ack_pend=0 T200_id=1, N200=3, T203_id=0 [ 00 01 01 3d ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 030 P/F: 1 0 bytes of data -- Got ACK for N(S)=30 to (but not including) N(S)=30 -- Stopping T200 timer -- Starting T203 timer What should i check on the above span debug? what's important there? the timers? the ACK? SAPI? TEI? is there any place to learn how to understand this output? Hope you can help me Verions of my server: libpri version: 1.4.11.4 Asterisk 1.4.24.1 DAHDI Version: 2.4.0 WANPIPE Release: 3.5.15.4 Please don't ask me to upgrade my Asterisk Version, the idea is to test this environment Best Regards! If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential
Re: [asterisk-users] DAHDI FXO port only recognizes the S extension?
The cause is bad programming. You can't go from an 's' to an '_X.' the way you tried. exten =s,1,Answer() exten =s,n,Wait(1) exten =s,n,Dial(DAHDI/3) exten =s,n,Hangup Is correct (that's why it works). What is it you are trying to achieve? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Songtao Yu Sent: 29 September 2010 10:56 To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI FXO port only recognizes the S extension? Hi All, When I tried to write my dial plan as below for my FXO port, which connects one PSTN line: [from-pstn] exten =s,1,Answer() exten =s,n,Wait(1) exten =_X.,1,Dial(DAHDI/1) exten =_X.,n,Hangup I got the following message: Connected to Asterisk 1.6.2.13 currently running on fax (pid = 8154) Verbosity was 0 and is now 4 -- Starting simple switch on 'DAHDI/1-1' -- Executing [...@from-pstn:1] Answer(DAHDI/1-1, ) in new stack -- Executing [...@from-pstn:2] Wait(DAHDI/1-1, 1) in new stack -- Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN' -- Hungup 'DAHDI/1-1' But if I changed the _X. to S extension, I can get the whole thing to work well: [from-pstn] exten =s,1,Answer() exten =s,n,Wait(1) exten =s,n,Dial(DAHDI/3) exten =s,n,Hangup Would you please let me which casuses this issue? Thanks, Songtao Yu If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Go from *100* to just 100
${EXTEN:1:3}
http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/
asterisk-CHP-5-SECT-3.html#asterisk-CHP-5-SECT-3.6.3
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 30 September 2010 08:31
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Go from *100* to just 100
Hello list,
how can I go from *100* to 100 ?
I know I can do something like ${EXTEN:1} but that way I only get rid of
just one *.
Kind regards,
Jonas.
If you have received this communication in error we would appreciate
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of this message, and any attachments, are the property of DataVox,
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If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
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Re: [asterisk-users] debian/dahdi/zaphfc - Unable to receive TEI fromnetwork!
What happens if you change to: signalling=bri_cpe_ptp -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Sent: 01 October 2010 11:37 To: asterisk-users@lists.digium.com Subject: [asterisk-users] debian/dahdi/zaphfc - Unable to receive TEI fromnetwork! Hello, snip # cat /etc/asterisk/chan_dahdi.conf [trunkgroups] [channels] language=fr switchtype=euroisdn ... group=1 signalling=bri_cpe_ptmp context=from-isdn channel = 1-2 snip If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] How to test BRI lines energy saving mode ?
The D-channel isn't actually 'dropped' - it is put in to a 'power-save' state. See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to 'Activation / Deactivation' for more information. Anyway - this is a known 'problem' - https://issues.asterisk.org/view.php?id=17270 As there is no fix for the above - then I doubt * will be able to emulate the NT's function. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: 07 October 2010 01:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to test BRI lines energy saving mode ? Olivier wrote: Hello, If my understanding is correct, these days it seems that many ISDN BRI lines are configured in energy saving mode in which signalling D-channel is dropped until a new call comes in. Is it possible to replicate this behaviour with Asterisk (when Asterisk is in NT mode and is seen as a public ISDN by another PBX, for instance) ? If not, would you it would be a useful addition to Asterisk ? Regards Energy saving??? I don't think so. If the D channel is down, how would I make an outgoing phone call? Something in this mode or your explanation just does not sound right... Lyle Giese LCR Computer Services, Inc. -- _ -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] How to test BRI lines energy saving mode ?
Well, to go slightly O/T: If you read the issue tracker for 17270 - it appears to be a LibPri 'fault'. So I would say that the main work would need to be in LibPri Q:is this how DAHDI talks to the ISDN?. Maybe someone who knows LibPri and DAHDI better can explain how the two combine... Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 07 October 2010 11:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to test BRI lines energy saving mode ? 2010/10/7 Andrew Thomas a...@datavox.co.uk The D-channel isn't actually 'dropped' - it is put in to a 'power-save' state. See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to 'Activation / Deactivation' for more information. Anyway - this is a known 'problem' - https://issues.asterisk.org/view.php?id=17270 As there is no fix for the above - then I doubt * will be able to emulate the NT's function. Thanks for these interesting links ! So this Activation/Desactivation feature seems to be missing in Asterisk. Would you say it should be implemented in libpri, in dahdi, or both ? If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] How to hangup all channels
2 ways:
Read http://www.voip-info.org/wiki/view/Asterisk+AGI
or in PHP - system (asterisk -rx 'core restart now' /dev/null);
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe D'alessio
Sent: 29 November 2010 14:47
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to hangup all channels
Thank you, i want to follow your idea, how i can send and receive data from/to
Command Line in PHP Script?
Thank you in advance
Date: Sat, 27 Nov 2010 08:45:47 -0800
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to hangup all channels
On Sat, 27 Nov 2010, Giuseppe D'alessio wrote:
Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels.
1) sudo /etc/init.d/asterisk restart
2) Write a script to do asterisk -r -x 'core show channels', parse the
output and do asterisk -r -x 'channel request hangup ${CHANNEL_NAME} for
each channel.
3) Write a script to do #2 using AMI.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
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Re: [asterisk-users] How to hangup all channels
Re-top-posting... I was merely pointing out that AGI exists (teach a man to fish...)! Sorry for not being as perfect as you... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: 29 November 2010 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to hangup all channels Un-top-posting... From: Giuseppe D'alessio Thank you, i want to follow your idea, how i can send and receive data from/to Command Line in PHP Script? On Mon, 29 Nov 2010, Andrew Thomas wrote: Read http://www.voip-info.org/wiki/view/Asterisk+AGI An AGI is executed in the context of a channel. Are you suggesting the OP write an AGI so he can call into his system to ask it to hang up all channels? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Push central phone book to phones
For the Yealink - you can use a 'remote' XML file. The XML is stored on a web server and is retrieved by the phone every time you press the phones 'key'. This has the advantage of not needing the directory to be pushed to the handset - and the handset always gets the latest version. Of course, the XML file needs to be kept up to date every time someone's name/extn changes. HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: 03 December 2010 13:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Push central phone book to phones On 12/02/2010 04:31 PM, Ishfaq Malik wrote: On Thu, 2010-12-02 at 16:01 +0100, Jonas Kellens wrote: On 12/02/2010 03:47 PM, Ishfaq Malik wrote: On Thu, 2010-12-02 at 15:19 +0100, Jonas Kellens wrote: Hello, I have Snom, Cisco, Grandstream YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. -- With Snom phones (and also Yealink I think) you can use centralised LDAP directories on a server This is a public server on the internet. I don't think I can use LDAP to push then ? Kind regards, Jonas. If you can set up and administer LDAP on the server you will be able to use it on the Snom (and maybe Yealink) phones. I can use different Organizational Units for different phone books ?? Kind regards, Jonas. -- _ -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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
