Re: [asterisk-users] OT - Desktop SIP phone with OpenVPN client
Am 21.01.2013 14:21, schrieb Olivier: Hello, I've seen some desktop SIP phones (Snom, Yealink) intregrate a VPN (OpenVPN ?) client. Has someone experience to share about that particular feature ? Is this experience rather successful ? My underlying question is can one supervise and configure these desktop phones, in teleworking environment ? Is DHCP required ? With Snom phones, those need an underlying network connection (d'oh, you wouldn't guess :-). That can be configured just like you are used to do it with snom phones - DHCP, fixed IP, whichever you like. They also need a reachable NTP server. Then they will ( after booting) download the VPN config from your (hopefully protected) server and connect to the OpenVPN server. Address assignment on the VPN link is done by the OpenVPN internal mechanism. You will be able to reach the phone's web interface, afaik, both over its local address and the OpenVPN assigned one. Make sure to either have your PBX on the machine with the OpenVPN daemon or add appropriate route configuration to the OpenVPN client config. BR AMH -- _ -- 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] Recorded reminders
Am 13.01.2013 03:17, schrieb Adolphus Enaboifo: Hi List Members , its been about one months since I built my first Asterisk server. What I want to know is: are there ways to make Asterisk take recorded reminders. This is the scenario I have in mind. 1 You place a call to a specific extension say 350. 2 On recognizing the incoming extension the reminder application at extension 350 prompts you to enter a number say 1 to record a message to your profile as well as input the time when the application will call you to playback the message. 3 or enter another number say 0 to playback all your recorded reminders in your profile.with options to add to the list or delete from the list. and of course there will be a limit on the amount of messages per user. Please if such applications exist can you guys show me how to configure it. Hi Adolphus, this sounds like something that a little scripting + call files can do. There may be better ways (and others may point those out in a flash), but this is what springs to my mind: - Have one directory for recorded audio for each user. Name audio files for their target time, like 201301131500.wav - When someone calls 350 and presses 1, check the number of files in that directory. If more than MAXMSGS, deny. - If there is already a recording for the target time, deny. - Else prompt for target day (today, tomorrow...) and time. If a file for that date/time exists, deny. - Record file, move to the right directory - Create a call file on the same filesystem as the spool directory - touch it for the target date/time and move into call files directory You'd need some nice scripting later on to handle that outgoing call. If the messages is read (and possibly acknowledged by pressing 1 or the like), the sound file should be deleted. If either the call fails or the acknowledgement is not given, the sound file should be re-named to a new time (say, one hour later) and a new call file generated. A few things that also should be thought about: - To not have endless reminder calls over and over, you could have a failed delivery counter per user - once that reaches a certain threshold, say 5, the reminders can be emailed to the user and deleted from the spool. You can reset that counter if the counter file has a change date older than 2 days with a cronjob - so if no failed deliveries happen within 3 days or so, they will be activated again. Make sure the user is informed about this problem iff a file exists when he calls in to record a new message. - Do sane error checking. When the call file is fired and fails to find the wav file it expects, this should not trigger another call. Perhaps an email avoid endless loops. - It might be a good idea to have a variation in the touch to the call file such that the expected time is only precise in minutes. Like add a random number of seconds in the range (0...50, or even -180 to 180 if precision is not essential). Be sure to document that or users might complain that the telephone system clock is not space-age-precise (Lusers!) This should get around everyone wanting to be reminded of going home in time for the soccer match, and everyone typing in a reminder time of 1630. - You should monitor usage; there can be still quite a lot of calls to interesting times. Same problem that automatic window blinds have: If everyone sets the DCF controlled clock to open the shutters at 8:00 precisely, the start current of possibly many motors may be _noticeable_ for the power company. That is why those devices do not have and do not need clocks with ultra- high precision - some even vary the morning/evening action time by several minutes on purpose. BR AMH smime.p7s Description: S/MIME Kryptografische Unterschrift -- _ -- 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] Playing music through VoIP handsets while on hook
Am 11.01.2013 02:42, schrieb Christopher Harrington: Wow, that seems wildly bandwidth inefficient. Is it possible to do multicast VoIP? Snom phones[*] do support multicast streaming. You can setup an IP port combination that the phone will accept audio at; once stream data starts arriving, the phone will start playback. [*] and possibly others as well, but that is what I have on my desk. Reasonably multicast will be ignored during a call though. AFAIK Asterisk supports Page to multicast. VLC or the like may also be audio sources. BR AMH smime.p7s Description: S/MIME Kryptografische Unterschrift -- _ -- 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 send SMS to Gigaset phones ?
Hi Olivier, I remember having had a similar discussion a few years ago. I will paste my postings from around May 2007 further down. First, I did not try sending SMS over VOIP to the phone, just over Voip to an ATA and then over analogue line (or ISDN) to the phone. So I have no idea wether the new Gigaset VoIP phones will to 1200 baud mumbo SMS phone service over a Sip voice channel or if Gigaset invented something better by now. You will have to try yourself. As for Gigaset phones connected via (at least one cable of ;- ) landline, you can send SMS messages to those with smsq. In theory that should also work on other landline SMS capable phones. Am Montag, den 13.09.2010, 11:04 +0200 schrieb Olivier: Hi, Searching this list archives, I couldn't find a definitive answer to my question : how to send SMS to Gigaset phones ? My goal is to send Alert SMS such as This phone system will be stopped in 5mn for maintenance to every terminal (SIP phones and Gigaset DECT phones). (So at the moment, I'm not looking for way to send SMS from handsets). == Message 1 (from myself, 2007-May-22) The SMSq stuff is for landline-type SMS, like those that never became really popular here in Europe ;-) I do not know of any SIP hardphone that supports them, but regular analog and ISDN handsets behind a SIP-to-analog/ISDN gateway work for me. The point of this SMS transfer method is calling the destination handset with a certain callerid set (which differs between countries - whatever number the telco prefers to choose - this can also be configured in the phone). The phone will not ring but instead immediately answer the call and receive the short message at 1200bps whatever modem standard they chose to use. For sending SMS, the handset will call a similarly telco-provided number (premium-rate numbers here in Germany - maybe that is the reason for the lack of popularity of this service) and do that 1200bps talk. If you still think you can make use of it, make sure to call smsq with the user id that asterisk is running as. That _might_ already do the trick. If you do not get it running, ask again - I might have a working setup somewhere around ;-) Nevertheless, for me, landline SMS is a PITA. The only great thing is you can upload Ringtones to Siemens gigaset phones. == Message 2 (from myself, 2007-May-22) Just to get you started, try this: Find out which user asterisk runs as. Get a shell for that user. Run (all in one line) smsq --mt --oa=321 --mttx-callerid=01930101 --mttx-channel=SIP/abcde message text goes here where 321 will displayed as sender id on the handset, and 01930101 will have to replaced by the mobile center known to your phone, plus 1 at the end - the German T-Com seems to use 0193010, and this setting works for me. Further, SIP/abcde must be the channel that a SMS-capable handset is available on: If you have some ATA with a DECT handset connected, or similar, use the channel name exactly as you would in the Dial() command. First thing to find out is if this works. Be sure to have asterisk in extra-verbose running a console to see what happens. If the mobile handset rings (instead of getting the SMS) either the 01930101 number has not been set correctly or it probably is not compatible with Asterisk SMS. Once you get this far, you would need the other way round. When your mobile phone tries to _send_ a text message, it will go to 01930100 (sms center number plus 0). You will have to care for that in your extensions.conf, like this exten = 01930100,1,Wait(2) exten = 01930100,2,Answer() exten = 01930100,3,Wait(2) exten = 01930100,4,SMS(01930100,as) exten = 01930100,5,Wait(2) exten = 01930100,6,Hangup() In my experience those Wait(2) improve reliability over internet connections, they probably are superfluous if you have reliable low-latency LAN. For me, they made the difference between 10/100 and 95/100 successfuly sent messages. You will have to write your own scriptwork to play with the files that will be created from those commands. Their structure is simple, you will find out. Sending EMS (for ringtones and bitmaps) is a bit more complex, you will need the UDH flag for that. I think I documented that once on this ML but am not sure. However, it is possible with some Siemens Gigaset devices, and pictures or monophonic ringtones. == Message 3 (2006-Nov-12) can be found at http://www.mail-archive.com/asterisk-...@lists.digium.com/msg24205.html with an example of how to send an EMS (message with picture attached). This worked with both monochrome pictures and single-track MIDI ringtones on my Gigaset S1 back then. Never got around to sending multi-track ringtones though. == Best regards Anselm -- _ -- 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
Re: [asterisk-users] is possible to sen sms with asterisk in Spain?
Am Donnerstag, den 09.07.2009, 11:26 +0200 schrieb ESGLinux: Hi all, I´m a beginner with asterisk and I want to know if with asterisk I can send sms to a mobile, I´m on Spain, and I don´t know this can be a problem (with the operators...) Hi, the SMS code in Asterisk is - afaik - only for the landline type of SMS. It can behave as landline-SMS capable phone (like some of the Siemens Gigaset DECT devices, for example) and talk to a landline-SMS center that will for a certain charge forward short messages to mobile phones. It can also behave as landline-SMS center and talk to appropriate phones. As a background info, landline phones can recognize that a landline SMS center is calling them by caller ID (which must be programmed, many phones ship with the local companies' numbers preprogrammed) and will not ring the bell but silently answer the line. The message transfer works with 1200 baud modem-like analogue audio (even if the phone is an ISDN device) - you can watch the actual message bytes on the Asterisk CLI if you turn on debug, in some kind of simple protocol and some 8bit-to-7bit mapping. It cannot directly talk to mobile phones: short messages are transmitted out-of-band in the GSM networks, and the mobile operators will not allow you direct access there. After all, short messages make a hefty percentage of their income at a minimum percentage of infrastructure usage. The situation in Germany (and to my knowledge, in several other European states) is that you can connect to a premium-rate landline-SMS center and hand them a short message for relaying. As that is bound to cost hardly less than using a mobile phone directly, it is not at all interesting for me (ymmv). I prefer using one of those web-interface-to-sms providers (mine can be used with wget from scripts etc) and pay between 3 and 12 cents per message, depending on destination country and quality of service selection. They have been reliable for quite some time now, and I remember that landline-SMS was a little too fiddly for my taste. Regards Anselm ___ -- 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] London DDI test request
Am Freitag, den 27.03.2009, 16:35 + schrieb Phil Reynolds: Quoting Chris Bagnall li...@minotaur.cc: Thins number is wrong - it has too many digits - should only be eight after the 20. (possible you put a surplus 3 in?) Good guess, indeed +44 20 3393 7389 has an answering machine as announced (and can be reached from my telco, obviously). I feel some pity for the poor owner of the other number (well, minus the last digit) - he probably pulled the phone cord from the wall already. BR Anselm ___ -- 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] automatic call pickup
Am Dienstag, den 07.10.2008, 01:42 -0700 schrieb Vieri: Hi, Currently I'm using Asterisk 1.2 and 1.4 in different setups. When a user wants to pick up a call within his/her pickup group, *8 must be dialed (or whatever you define in features.conf). [...] I was thinking of configuring some sort of auto speed dial of the pickup code (*8) whenever the user picks the phone up but it seems that these phones don't support that. Hi Vieri, regarding your combination of analog phones and ATAs I would look for the auto-dial functionality in the ATA. I am pretty sure I saw it in one web-interface or the other, but surely not all vendors implement that kind of functionality. In your place I would also think about using a one-press pickup code, like #. I know this code is often in use for transfer or the like, but if pickup is the 95%+ action then transfer doing *# instead of # (or whatever) might be reasonable. This would reduce pickup to lifting the handset and pressing the bottom left-most key, which can be done without looking at the keypad. One last idea: Perhaps your multi port ATA supports different kind of ring codes (once short, twice short, no idea whatever) one of which will _not_ ring the phones (which could interpret that signal meant as a short ring as line noise or the like). Perhaps they even support silent ringing, not sending the ring signal at all, but nevertheless answering the line if hook-up happens. Best regards Anselm smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
Am Freitag, den 29.08.2008, 09:16 -0700 schrieb Ira: At 05:48 AM 8/29/2008, you wrote: (so since they still liked the Snoms otherwise, my solution is to get them to dial a star at the end of a number to select their 'home' account, otherwise it goes out on their work account and the dialplan fixes it up) We have 5 outgoing numbers we want to use selectively and we just dial 1,2,3,4 or 6 first which picks the proper rules. If you forget it asks which line you want to use and 1 is the line we want guests to use so it all works fine if you don't know what's going on. Hi, on German landlines the user can pick a carrier on a per-call basis (at least if it is a T-Com line). The national dialplan assigns 010NX and 0100NX prefixes for that purpose. I believe similar access codes are in operation in US (10-10-XXX?), and possibly your system blocks those for similar reasons as we do (they are useless on VoIP lines here). So those codes can be used for selecting a non-default line, where I use 01099 to send out a call with my mobile phone Caller ID, 01090 for anonymous no-callerid calling and 010[123][123] to select between three providers with up to three outgoing Caller ID numbers each (and a few others in the 010[4-8]X range). Not pre-pending such a number will make a reasonable default choice here depending on the phone used. If the percentage of non-default line calls is fair below 20% such a long prefix is still better than dialing a digit before each number (with the additional risk of forgetting that, or dialing that digit when calling from other places :-) This assures that guests have no trouble using the telephone, as they just dial the phone number as they would from their own telephone at home. If I want them to not have my caller ID on the callee display (no call back, privacy, whatever) I simply tell them to dial the 01090 first - they will assume I want them to use a certain carrier, and not need any additional instructions. Call-By-Call, being the German name for it, is widely known. Another aspect of carrier selection codes opposed to switching lines on the phone is that it is independant of the phone hardware at hand, be it an analogue + adapter, ISDN + adapter, DECT, SIP hardphone, software... Of course it is not necessary to allow all prefixes from all phones, or have the same meaning of a prefix on all phones (01099 here would be an example that sends a different Caller ID from different phones, depending of the mobile phone number of the person usually using that phone). Best regards, Anselm smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail
Am Dienstag, den 19.08.2008, 02:53 + schrieb Miguel Otamendi: Please, I need help. I have problem witch voicemail. -- Executing [EMAIL PROTECTED]:3] VoiceMail(Zap/4-1, s) in new stack [Aug 17 21:33:46] WARNING[11864]: app_voicemail.c:3061 leave_voicemail: No entry in voicemail config file for 's' -- Executing [EMAIL PROTECTED]:4] Hangup(Zap/4-1, ) in new stack == Spawn extension (Incoming, s, 4) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' Hi Miguel, please see http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail for details about the VoiceMail command. What seems to happen in your setup is that the call runs into the s extension, and then VoiceMail() is called. As you do not specify a voicemail box number, s is taken as a box number, which is probably not what you want. Check extensions.conf and alter the VoiceMail command like VoiceMail(1) instead of VoiceMail(), and define a mailbox number 1 in voicemail.conf (or any number you like, of course). You possibly can also define a mailbox number s in voicemail.conf, but that will run you into trouble if you want to listen to messages from abroad, as s is hard to enter by DTMF touchpad ;-) Not sure if that box s works at all though. The safe bet is to use numeric voicemail box numbers. Regards Anselm smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recordings...
Am Dienstag, den 22.07.2008, 14:53 -0500 schrieb Gregory Malsack: Hello, My boss is asking me to setup the asterisk server to record all calls. (Simple). However, he wants to be able to enter a key sequence during the call to stop the recording. Any ideas on how I would do that? Hi Gregory, I found something about recording at http://www.voip-info.org/wiki/view/Asterisk+config+features.conf (second example). If you combine that with a default_recording_enabled (Monitor() call before Dial(), I would expect), that could be used to turn _off_ recording by pressing a key. I would not know though how to protect against the external call party pressing the same key. Best regards Anselm smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] distintive ring
Am Dienstag, den 15.07.2008, 14:02 -0400 schrieb Fidel Garcia: Need to have a different TONE for any internal call (EXT OR TRANSFER) from an external (outside) call. Any suggestions? Fidel, I do not know what kind of tone you mean: The sound of a phone that signals a call coming from internal/external? The sound in the earpiece after you dialled while you wait for the other end to pick up? In the first case distinctive ring is probably the right term to search for. You will have to decide wether your phones are SIP or ZAP (or both, or different), because methods seem to differ. As a start reading point have a look at http://www.malico.com.tw/voip-info/wiki/view/Asterisk+SIP+channels.html The mailing list archives contain a lot of information *hint* Best regards Anselm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fring (softphone on mobile) and open vpn
Am Montag, den 14.07.2008, 09:45 -0700 schrieb bilal ghayyad: Hi All; Anyone can advise for a method to have open vpn client to be installed on the mobile, so it can open a vpn channel with Asterisk (I installed open vpn at it) from the mobile, and then I can let fring use the open vpn channel to do the communication? This I need it because I am facing a problem with NAT (when asterisk behind NAT), so voice does not run well, so I am trying to use VPN in that case, and I know that for voice open vpn is the best vpn method, but how to has such open vpn client on mobile to use it with fring? Hi Bilal, I take it your mobile phone runs some kind of Windows Mobile. As far as I know, there is no OpenVPN support for that operating system (at least back when I tried about a year ago I found nothing). I guess the CPU load of OpenVPN might prevent smooth phone calls anyway... You might try to solve your NAT problem instead. Best regards Anselm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR
Am Samstag, den 28.06.2008, 08:15 -0500 schrieb [EMAIL PROTECTED]: Hi List I have build an IVR on Asterisk from 1.2 to now 1.4.18 and has already processed more than 10million calls! I have one big challenge which is reporting... it is the requirement to have a web reporting module which should the following info based on selected time frame - Number of calls on specific branch- Done - Number of calls to branch 1 that came from branch 2 (this should be flexible) - talktime on specified branch (say how long caller listened to option 1 before choosing option 2 or hangup) On IVR, it is so important to understand how many callers select a specific branch and how long they spent on that branch. CDR stats can not provide these type of information and on trying freepbx, still can not go so detailed Dear Kili, in my opinion this is a good application for Database backends. You could, for example, write entries to a DB whenever someone presses a key (or is re-routed in the dialplan, which comes to a similar scheme). In data mining time some SQL logic can produce nearly any data you want, provided the input data is there. Millions of calls sounds a lot though, so be sure to have a reasonable database backend: The asterisk included one might be a bit on the small side here. This is just an idea, I did not implement anything the like (yet). BR Anselm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] breaking DNID into country code, area code, and local code
Am Sonntag, den 30.03.2008, 16:56 +0800 schrieb mark morreny:
Dear friends,
I am wondering if there is any efficient way of extract the country
code, area code, and local code into 3 different variables from one
DNID that can look like 001630233-4333 or 0086213345333?
International code can be 011, or 00.
National code can be 0 or 1
Country code can have 2 or 3 digits
Area code can have 2 or 3 digits
Local num can be 7-10 digits
Is there anyway to break this down efficiently in the dialplan or AGI?
I think it can not be done efficiently, reliably, and for international
numbers.
The first problem would be to create the uniform international number in
+(X[XX])[YYY] format.
For example consider the number
01149228730
This might very well be a valid Sheffield, UK, number (no idea if it is,
and I will not call to find out :-) of area code 0114 and local
seven-digit number 9228730.
If dialled from US it will connect you to the University switchboard in
Bonn, Germany. (I had to find a really short number to fit the
seven-digit dialplan of Sheffield).
The problem is that some countries have 011 being (part of) a valid
area code, banning it as identification for this is an international
number dialled from North America. Vice versa some countries seem to
have valid uses for 00 that mean different things than international
dialling. I think it was used for operator in Spain back when they
had 07 for international dialling, and had been in some area codes in
Russia until they decided to migrate from 8~10 to 00 for international
dialling until 2010.
So getting your numbers standardized to + C[CC] A[] SSS[SS]
may already break on those problems.
Sorry, but you are not all happy either once you have that standardized
form. US is easy with the fixed +C AAA SSS form, and some countries
are similarly easy as they have fixed-length area codes (France, AFAIK)
or no area codes at all (Denmark). UK has two (London 20, Coventry 24
and a few others) up to four (afaik) area code digits, which possibly
can be recignized by logic, as +44 2 always is two-digit, and +44 1x1
and +44 11x are always three-digit - I do not know if that is valid
universally though. Any logic breaks when it comes to German area
codes, where +49 x0 may or may not be a valid area (30 - Berlin, but
5031 - Wunstorf, and 209 - Gelsenkirchen), and area codes range from
two to five digits, with a few three-digit subsribers nearly anywhere,
but up to nine digit subscriber numbers in Berlin.
For some countries information may even be hard to get - although you
probably will not receive many calls from Benin, Ethiopia or Mongolia,
and if you do indeed, you will have no trouble getting their local
telephone system explained.
Once you have your numbers standardized ($NUMBER in +xxx form) you
could of course query a database, looking for ${NUMBER:1:7} down to
${NUMBER:1:2}, such that if applicable the Country-/-Area can be
returned as string, as a fall back the country only, and if nothing
helps, the number can be discarded as invalid (assuming you have a
complete list of country prefixes).
I think you will not find anything much simpler that which can handle
the structure of phone numbers, as that is for historical and political
reason rather messy ;-)
BR
Anselm
___
-- 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] Langugae issue
Am Sonntag, den 30.03.2008, 09:54 -0400 schrieb Mike Trest - Personal: Ayman, One solution is to write an AGI scrip to parse the number and read back in Arabic semantic order. for the last two digits and for certain special numbers like 11 , 100 , 1000, ... .I must bring out my old Arabic language books to do this myself, but if you will share the language files with the asterisk group, then I will make an example AGI for you that we can share with the list. If you are agreeable, let us continue EMAIL messages privately until we have something working that we can share with the list. ..mike.. Dear Mike, for me it seems that this is what say.conf is good for: http://svn.digium.com/view/asterisk/branches/1.6.0/configs/say.conf.sample?revision=105596view=markup (which seems to be considered the new format). Perhaps it would be better to implement Arabic there than by means of an AGI script. Be sure to check with the developers wether this will be relevant for Asterisk 1.4 or if you need to go with 1.6 SVN to benefit. Best regards Anselm ___ -- 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] Multiple sites, same extension
Am Donnerstag, den 20.03.2008, 08:59 -0600 schrieb Aaron Fransen:
Holy Mackeral. Ignore that last message. I still do NOT know how to
route calls with the same extension being used in two locations,
however the issue I've resolved is getting Cisco CallManager and
Asterisk talking together properly.
I've tried a dialing plan like:
exten = _8101,1,Dial(SIP/${EXTEN:4},,r)
to no avail.
Hi Aaron,
for my personal taste your Dial() command is lacking a SIP domain (or IP
address). Consider location A (Asterisk 10.1.1.1, prefix 8101) and
location B (Asterisk 10.2.2.2, prefix 8202), where users at B want to
dial 81012000 for extension 2000 at location A.
In that case, your Dial command looks like
Dial(SIP/2000,,r), which looks pretty much useless, unless one of B's
local (see, local to B, not A!) SIP peers has a [2000] stanza in
sip.conf, and even then you would not call peer 2000 at A, but at local
(B).
If you replace your command with
Dial(SIP/${EXTEN:[EMAIL PROTECTED],,r)
the world looks completely different.
At least I hope so...
BR HTH
Anselm
___
-- 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] Is Asterisk ready for Prime-Time?
Am Donnerstag, den 20.03.2008, 16:59 +0200 schrieb Tzafrir Cohen: And what happens if at the time of the shutdown there was a ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- ROTFL Trafrir, you made my day. (BTW: I think that is why restart when convenient exists) Anselm ___ -- 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] Desperately need help with Asterisk setup
Am Montag, den 17.03.2008, 15:08 +0800 schrieb Pete Kay: Hi, I am new to Asterisk and I am having a setup problem that I am trying to resolved for the last couple days without any success. I am pretty much desperated on this issue and I don't know why. Can someone please kindly help me to troubleshoot this? I can't hear any audio from Asterisk when running Playback or VoiceMail tests. Dear Pete, my first idea would be that something with your codecs is borken (TM). I personally use a setup quite similar to yours, with the one visible difference that I also allow the gsm codec, owing to the fact that at least my home-recorded prompts are gsm only. I _guess_ asterisk could or should handle format conversion from audio files automagically, but for making sure, please try adding gsm, at least for now. You might also want to setup the [sipclient] stanza in sip.conf such that nat is set to no, although I do not see why that should break things. Especially as Echo works. The externip is set to your current external IP, right? (Knowing full well that some DSL lines get a new IP as often as 6 times a day, or as a P2P bandwidth countermeasure down to five minute intervals at certain restrictive providers once your fair use volume is used up). Again this should not be the culprit... Poking with a stick in the swamps, but perhaps hitting the bug :-P BR Anselm ___ -- 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] Desperately need help with Asterisk setup
Am Montag, den 17.03.2008, 21:38 +0800 schrieb Pete Kay: Hi, Here is the SIP debug output for the playback test. Thank you so much for your help. Hi Pete, [Mar 18 05:33:08] -- Executing [EMAIL PROTECTED]:1] Answer(SIP/2000-081e0738, ) in new stack [Mar 18 05:33:08] Audio is at 192.168.1.101 port 10028 [Mar 18 05:33:08] Adding codec 0x4 (ulaw) to SDP [Mar 18 05:33:08] Adding codec 0x8 (alaw) to SDP [Mar 18 05:33:08] Adding non-codec 0x1 (telephone-event) to SDP I do not see gsm here. Any reason not to allow that codec? Or did I miss something? You wrote you enabled it, so it should be here IMO. --- Transmitting (NAT) to 192.168.1.102:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK793126083;received=192.168.1.102;rport=5060 From: 2001 sip:[EMAIL PROTECTED];tag=2612560371 To: sip:[EMAIL PROTECTED];tag=as0ca1ddb0 Call-ID: [EMAIL PROTECTED] CSeq: 20 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Accept: application/sdp Content-Length: 0 404 does not sound good. Please, look which sound files exist on your system (e.g. what does find /usr/share/asterisk -file vm-goodbye* say?) Another point: Which client do you use, is it Wengo or is it Xlite? Or both? In that case: Any differences? BR Anselm ___ -- 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] Weird NAT issue ...
Am Montag, den 17.03.2008, 13:59 + schrieb Alan Williamson: Afternoon one and all. I am having some interesting fun with our Asterisk setup. We have two CISCO handsets (7960) sitting on the same network (NAT). Each phone can successfully originate calls. Each phone can be called successfully from outside Each phone can be directly called by other extensions OUTSIDE the network HOWEVER -- when those 2 phones try to call each other; the connection is made, but no voice is heard. Any advice as to where i need to look? Hi Alan, my guess is this has to do with the Audio path. As long as audio only traverses the NAT router on the Cisco site, that device seems to handle data paths quite well (you probably enabled different SIP ports for those two devices? At least that helped me to a stable reachable phone, which would just not work with more than one SIP 5060 phone behind a single NAT). The tricky part seems to be the turnaround. One of the ciscos tries to send audio data to the external ip address of the nat router, for the other phone, and this might be something that the router does not handle. You could try to disallow direct audio between those two cisco phones by forcing Astrisk to stay in the audio path (e.g. let all audio packets go to asterisk, turnaround there and go to the other phone). This is surely not optimal in bandwidth terms etc., but may solve such NAT issues. You can force Asterisk to stay in the audio path by specifying a Dial option that requires Asterisk participation: Then it will not allow direct connection automatically. Options requiring key presses (allow * transfer or something, see Asterisk docs) should do. Somehow the reinvite could have to do with that as well, but don't ask me there :-) BR, Anselm ___ -- 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] NIN Ghosts music (free download) safe for MOH?
Am Montag, den 10.03.2008, 02:59 -0500 schrieb John Faubion: But, just to clarify, please remember that using music as MoH is considered a public performance, and if the pieces in question do not include a buyout license *for the performance Ok now I am curious, if a radio is playing in a store, a restaurant or at the beach, wouldn't that be considered a public performance? And even though the radio station has already paid the license fee, does this mean that the person who owns the radio is also subject to these fees? I know of several key systems with FM radio cards providing MoH and I've often wondered about the ramifications of that setup and the music industry. Good morning, the legal situation probably differs between countries. In Germany, you are required to register with the GEMA if you intend to play music in public if the artist is a GEMA customer. If you _only_ play free music, the law does not require you to register afaik, but in doubt you will have to prove that you did not play GEMA music (which is ridiculous when you think about it, but you do not want to fight against that machine). A party where two guests do not know each other's names may be considered public, even if only ten or twenty people are there. A class room, a barber shop, a supermarket or having a barbecue on the beach are surely public. The fees due will be calculated in regard to the area where the event takes place, because that limits the _maximum_ audience. Ain't it nice. (No idea though how exactly the area for music on hold is calculated - have a look at their tariffs jungle at http://www.gema.de/musiknutzer/abspielen-auffuehren/tarife-im-ueberblick/ ). I am not a lawyer, and am still lucky to not have to do with those music industry guys (and who is the pirate here...). BR Anselm ___ -- 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 as SMSC to GSM-Phones
Am Mittwoch, den 27.02.2008, 16:30 +0100 schrieb Hans-Peter Straub: Hello all, i today have searched on the internet about a solution to let asterisk act as a SMSC (Short MessageServiceCentre) to deliver SMSes directly to GSM Phones. I only have found some cases with use of an extern SMSC (i.e. by the Mobile Net Provider) Is there a possibillity to do that, or ist asterisk only able to send SMS to analog phones? Hi Hans-Peter, the asterisk implementation of SMS means the landline type SMS. A short message is sent to the phone by calling the number in question, giving a certain caller ID. Phones will usually recognize several caller numbers as SMSC, for example 0193010 in Germany. Whenever an apparent SMSC calls, a landline phone will answer the line and have a short chitchat in 1200 baud modem lingo, exchanging a recognise sequence, the data per-se and some final status message. Mobile phone SMS work in q completely different way. AFAIK (and I am even less an expert there) SMS are sent in frames otherwise unused, _not_ in the voice channel. I was told to imagine SMS transmission like UDP packets on a carrier usually running TCP (voice channels), being a by-product that the phone companuies earn a golden a** with. Landline SMS service and mobile phone GSM SMS service are completely different things. The seemless manner in which messages pass to and fro is in reality a job for gateways that speak both protocols. Germans seem to be lucky that it works across (nearly?) all fixed and mobile networks; I remember reading somewhere that most countries have limited interchange. You could use one of those landline SMS gateways, call it and hand in your text message. The downside of course is that you have to pay real money for it, usually about 19c/message iirc. An alternative for sending (lots of) SMS may be a web-based service. I personally use a carrier that relays mails to SMS; mails sent to [EMAIL PROTECTED] with the message as mail subject and my customer code in the mail body are relayed to a mobile phone. This works quite well for my purposes; I cannot even tell you the name of the provider because 'it just works', SMS are usually sent by a script so I do not ever enter the domainname by hand :-) Depending on the realiabilty and speed of transportation that you require prices may vary; I think I pay about 4 cent per message (unreliable, but works 99% in my experience) up to 9 cent (ultra reliable, less than 5 seconds usually). I always use the cheap service - works for me. Contact me off-list if you need a pointer there. BR Anselm (leaving for the local Linux User Group meeting :-) ___ -- 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] Matching + characters in dial plan
Am Donnerstag, den 07.02.2008, 00:18 + schrieb Ed W:
Can someone please explain how to match a + character in a dial plan (so
that I can swap it for the 00 country escape code).
In Europe at least the + is a common shortcut for the international
prefix (which is 00 in my country). However, my trunk chokes on the +
character and all my speed-dials are setup with a + at the start of
them... Trying to fix the phone rather than the addressbook...
You should get away with
exten = _+[1-9].,1,Goto(00${EXTEN:1},1)
If you had any special use for triple-0 numbers (as we do), you should
afaik also be able to use
exten = _+.,1,Goto(00${EXTEN:1},1)
We do not allow +0 numbers though because that would contradict the
meaning of a 000 number in our setup. Generally +AABBBCCC is dialled
as 00AABBBCCC, as international phone call, through our outward phone
provider without them noticing any weird + signs.
BR
Anselm
___
-- 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] Two Asterisks behind NAT and need to link them using IAX trunk
Am Freitag, den 18.01.2008, 04:21 -0800 schrieb bilal ghayyad: Hi; Via OpenVPN or port forwarding is known for me, but via SSH is new for me, how I can do it and what is the difference by SSH and OpenVPN? In principle both use a packet stream (SSH is TCP, OpenVPN is TCP or UDP) for encapsulating IP packets. The main difference is that SSH port forwarding forwards the packet data, but not the header: The packet is stripped at side A and a seemingly different TCP connection is established on side B. This also implies the main limitation of SSH, that it is restricted to tunneling TCP (afaik). OpenVPN in contrast takes entire IP packets, applies routing and tunnels the entire packet through. You can tunnel any IP traffic through OpenVPN, and the remote side IP address will persist. (You can even tunnel IPX or Appletalk, if using the BRIDGE mode with virtual TAP interfaces). Basically OpenVPN appears to the tunnel endpoint as a virtual wire that behaves like an ethernet port. OpenVPN is far more flexible when it comes to network restrictions. On the other hand the SSH main idea is not VPN but secure shell access :) For VoIP I'd imagine SSH is quite impractical, if usable at all. Most likely the TCP-only restriction will make life difficult. SIP over OpenVPN works - I used it to tunnel from a trip to California to my Asterisk back home in Germany. The voice quality was a bit poor, but this might also relate to the WLAN and the multi-hop-internet route in between. Speaking generally, of course an aditional layer (which both OpenVPN and SSH introduce) does not improve the signal path quality, or latency, or everything. I have read recommendations to use OpenVPN in UDP mode to reduce packetizing problems which would result in choppy sound as well. No comparison numbers available here though. BR Anselm ___ -- 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] OT - Is handover included in DECT GAP ?
Am Donnerstag, den 10.01.2008, 12:31 +0100 schrieb Michiel van Baak: On 11:22, Thu 10 Jan 08, Olivier wrote: Hi, Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support roaming and handover and are these functions transparent for handset (then, these functions are implemented in DECT base stations) ? Roaming/handover functionality is implemented in DECT-GAP. I dont know if all handsets allow it, but I think they do. AFAIK roaming in the sense of using another than the default base station is supported in even the cheapest handsets that are sold as GAP. This does work nicely with different vendors' base stations (as it should), although in that case only basic functionality can be achieved (like CALLER ID display, Call-on-Hold seems to work) - the phone book feature, internal calls to other handsets and base station configuration mostly do not work. I have not yet seen a DECT/GAP phone that supports roaming while a call is in progress, this would need the appropriate logic in the base stations. I know such hardware exists (Kirk!?), but if not advertised, the base stations most certainly do not have this feature. I expect that this function requires additional support in the handset as well, so using those 20$-handsets on your multi-k-$ roaming support base station will probably not help... BR Anselm ___ -- 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] RANT (was Re: Which IP Phone is really the best?)
Am Sonntag, den 06.01.2008, 21:05 -0600 schrieb Alejandro Kauffmann: As long as this is an official rant thread Good to know no new phones have hit the market since the last time this question was asked and answered. It's also good to know opinions about specific products don't change over time. It's great to know that no bugs have been found in any of the best phone's firmware that might drop them in the ranking since the last time this question was asked. And the OP would have had proper use for an answer of the form I preffered the ACME 4711 until they burliwooped the blobber function in their latest firmware? As one of the more often mentioned sources on asking questions, http://www.catb.org/~esr/faqs/smart-questions.html , suggest, the OP could at least have let everyone known that he found and read the previous threads about the best phone. Asking something like === I am searching for an IP phone for a customer's system. It should - have decent speakerphone - integrated phone directory - second Ethernet port for PC I read about the ACME 511, the Smon 370 and the Ccorsa 480j, although the ACME seems a bit on the expensive side. All of them seem fit for the job. Any other suggestions? How did your setups work out, what where the most common complaints about those phones? And how about manufacturer support / firmware releases? === would have triggered less of a rant. I wonder if this rant thread would exist if the OP had not mentioned he was preparing a quote for a customer? I guess so. There is a visible resistance against supplying service for free when the OP just hands the answer through to the customer and earns $$ with it - in my opinion a reasonable point. Nevertheless most writers here seem to be willing to at least help the people find an answer - not perfectly prepare one for them, but give them the few and necessary hints (just as in the above mentioned document). Counting all in, atmosphere is rather friendly on the mailing list - a few things like the response of Check your extensions.conf for the empty mail a few days back are the necessary part of sarcasm that I personally do not count as unfriendly as such. Enough cents for now, BR Anselm ___ --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] how to block spammer calls
Am Samstag, den 05.01.2008, 11:58 +0530 schrieb ram: Hi I understand what you are saying. so once we see he is not input the pin more than 2times he will be blocked for hour ( i will run cron job, after one hour release them) is this a good idea. Hi Ram, I do not think that is a good idea. 2 tries are not much on the one hand, and on the other hand, your competitors probably know how to fake CALLERID, so once they find out their calls are not answered anymore, they can just set another CALLERID and dial in again and again. If they really want you to pay for useless minutes, the only thing you could do against it (if you do not want to block everyone) is requiring your customers to register the phone number from which they will dial in, and throw away (by not answering) any other calls. Using cronjobs is possibly a bad idea because you create load spikes, if e.g. 5000 asterisk -rx commands are issued within a few seconds. A better way to implement it would be storing the last unsuccessful authentication system time and wrong PIN count for each CALLERID, and block the ID if a count of =3 happened less than 1800 seconds ago or similar. This blocking would need appropriate dialplan logic. I think there is soem material about astdb, time and blocking in the examples section on www.voip-info.org; if you cannot build something on your own, (as mentioned) you might want to pay someone some bucks for implementing it. In Germany I _think_ calling 0800 numbers for abuse can be sued against, on the grounds of tampering with phone infrastructure. If the same number calls in more than 100 times a day or so, you could probably ask the number provider to close the caller's account (and if they will not, you can still sue). If the person calling your 0800 is a competitor, there is a law called UWG here (law against unfair competition): It probably allows you to sue them for compensation of minutes and blocked lines, but you would need to ask a lawyer for details - and any other country will see a completely different solution anyway. BR Anselm ___ --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] iP0020 Phone busy signal all the time.
Am Samstag, den 05.01.2008, 06:40 -0400 schrieb William Herrera: Hello to you all. Just got my first iP0020 phone and no matter what I do to it when I try to call I get a busy signal even though Asterisk and the phone web gui shows that the phone is “registered”. Has any body had any similar experience with this type of phone (or nay phone)? Hi William, the busy signal can have several meanings: - phone malconfigured - number invalid - number you called is unreachable My first guess would be that your asterisk drops the phone into an empty context (or similar) and such there are just no valid numbers to dial. Logically a step for debugging would be to enter the asterisk console asterisk -r activate logging of as much info as necessary set verbose 10 and try to call from the VoIP phones. What does the asterisk console show? BTW I saw hard phones that needed first dialing, then lifting the cradle, and others that worked the other way around. BR Anselm ___ --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] how to block spammer calls
Am Samstag, den 05.01.2008, 13:31 +0200 schrieb Tzafrir Cohen: On Sat, Jan 05, 2008 at 11:54:41AM +0100, Anselm Martin Hoffmeister wrote: Using cronjobs is possibly a bad idea because you create load spikes, if e.g. 5000 asterisk -rx commands are issued within a few seconds. Why would you do that? If you want to write 5000 commands, write them directly to the socket or use the manager interface. I posted one a simple script for writing batch commands to the asterisk.ctl socket using socat. You have to strip the '\n' in the end of each command, and write every command in a separate write (I used a 'sleep 0.001' for that). Hi Tzafrir, good to know, and thanks for pointing this out to me. Until now I always got around large command lists, ran just the occasional cronjob for the odd task. Besides the possible effects of running a large command list (which may be neglible, I admit), you gain another load spike by those users who know they can retry calling after the next full hour. So the first few minutes of each hour might be loaded heavier than the later parts. I do not like the idea of opening the gates at a fixed time. Releasing a lock for a certain CALLERID after a given time (in minutes), not a given point of time (as clock time) seems much more elegant to me. BR Anselm ___ --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 on Hp servers
Am Samstag, den 05.01.2008, 19:50 + schrieb Gres +: please can anyone help me knowing if i can install Linux and Asterisk on HP servers Gres, you will have to find out if _YOU_ can do that. Generally speaking it is very well possible. For a quick start, you might want to try an asterisk-centric distribution that makes starting with Linux and Asterisk quite a bit easier than e.g. LFS, Debian, or Gentoo might. Anselm ___ --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] automatic call marking an extension
Dear Rickygm, Am Donnerstag, den 03.01.2008, 20:19 -0600 schrieb troxlinux: hello list, happy new year to all, also to digium for their great work with asterisk . I want to make an automatic call marking an extension from my dial plan , an example that when marking the extension 100, tell me it records their message, mark the hour of their automatic call and at the end it marks the extension of the automatic call as I can make that, do they give me some ideas? If I understand your mail correctly, you want to schedule calls by placing a phone call: 1/ Call e.g. 870 on an internal phone 2/ voice prompt please select internal extension = type 215 3/ voice prompt please enter time for the scheduled call = 0815 4/ voice prompt please enter destination number = 02212045447# 5/ thanks. Your call has been scheduled This (first) part can easily be done, you could use Read(), like in exten = 870,1,Answer() exten = 870,2,Playback(scheduler-select-ext) exten = 870,3,Read(digits||3||5) exten = 870,4,Playback(scheduler-enter-time) exten = 870,5,Read(time||4||5) exten = 870,6,Playback(scheduler-enter-destination) exten = 870,7,Read(dest||20||30) exten = 870,8,AGI(scheduler) exten = 870,9,Playback(scheduler-thanks) exten = 870,10,Hangup() The second part would be writing an AGI that creates a .call file - there is quite some documentation available, also in the list archives. Create a file, enter the information as needed, touch it to the planned call time and move the file into the outgoing directory (probably a subdir of /var/spool/asterisk). As I come to think about it, writing the number entry part into the AGI gives greater flexibility with input validation etc. - consider that as well. The third part is getting the .call contents right - you need to introduce a call recording statement somewhere, probably in the outbound leg right before the Dial() (in the context used by the file, inside extensions.conf). Try getting it right without call recording first, and then add that. This is a mostly separate topic though, and has also been mentioned in the mailing list once or twice... I hope you got an idea how to find more information. It is not difficult to get such a thing working, just needs some fiddling and a little experience in those things asterisk. BR Anselm ___ --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] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
Am Samstag, den 15.12.2007, 16:55 -0800 schrieb Philip Prindeville: I've got the following set up: Someone calls into my PBX on a single number (via SIP trunk from my carrier), and the get a voice menu of extensions. On one of the extensions, it rings a bunch of internal SIP hardphones, plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN gateway. The issue is that my cellphone shows my PBX's number, not the original calling number. This topic has been covered in length. In most cases it seems to end at the fact that providers correct caller-ids they get from the calling party: If you send any number which is assigned to the PRI (or SIP trunk), that is fine; if you send another number, it will be changed to the (first) number of the PRI/trunk. Few providers allow for foreign caller ids to be sent over their equipment - in some countries this is even illegal. For example, one of my providers (German) allows to set any CALLERID, but their documentation warns to not do stupid tricks, as calls can be tracked and using malicious information will be prosecuted. This feature is to be used only for sending _my_ cell phone number etc. BR Anselm ___ --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] Caller ID Issue
Am Mittwoch, den 12.12.2007, 09:14 -0500 schrieb Lutgring, Sam:
I have a strange issue with CLID that I would appreciate if someone
could point me in the right direction. When a call comes in (either
from another SIP user on the same Asterisk box or from the ISDN PRI)
the Caller ID Name is displayed correctly, but the Caller ID Number
seems to be empty. My Grandstream phone is setting the Caller ID
number to the registered account name while SJ Phone soft client shows
the Caller ID number as empty.
Any suggestions would be greatly appreciated.
Hi Sam,
some phones seem to hate phone numbers with strange characters in them;
those might be spaces, + signs, - dashes etc. and refuse to display
anything at all. Perhaps the information is there, but it is in some way
or another taken as invalid.
You could see what Asterisk thinks those variables are. A
NOOP(CALLER-ID-Info: ${CALLERID(num)} / ${CALLERID(name)})
in the dialplan, together with CLI and set verbose 10 should show you
lots of information.
BR
Anselm
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with the ring timeout in dial command for local extensions
Am Freitag, den 07.12.2007, 17:53 -0300 schrieb [EMAIL PROTECTED]: Hi all, I don't know if this is the right list to ask, since I'm using Trixbox version 1.0.0.28, that has asterisk 1.2.17. I'm trying to configure the ring timeout value for my local extensions (when dialing from one to another), and the dial command simply ignores my values... I have one extension 0017 in my box, so I used the command Dial(SIP/0017|100|rTtWw) to dial to it. The call gets completed without a problem, but it only rings for 30 seconds, when it should ring for a 100 seconds. I'm pretty sure this is my mistake here, but I didn't find a solution. I also tried changing the value directly in trixbox web interface that says Number of seconds to ring phones before sending callers to voicemail and nothing happens. I know that trixbox does weird things to my configuration files, but I edited extenions.conf, since it does not get messed up by trixbox. It might be related to the phones you use; some models seem to have an internal timeout, after which the call will be rejected - this forces the Dial() to terminate. You might try to find information about that: It might be configurable in the phone. I am not well informed about trixbox, but I assume you can get an asterisk console by asterisk -r. Use that and verbose 10 to retrieve the events that occur when the timeout falls. That might help you find the culprit. Regards Anselm ___ --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 existing extensions.conf macros, and co-habitation
Am Freitag, den 30.11.2007, 15:08 -0800 schrieb Philip Prindeville:
bump...
Philip Prindeville wrote:
I'm trying to set up my extensions.conf file using some of the existing
macros like stdexten, etc. while at the same time having two logically
separate virtual PBX's (with no default context) and two trunks coming
into separate contexts, i.e. one for residence and one for my at-home
business.
I noticed, however, that macro-stdexten depends on the default context:
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well)
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20) ; Ring
the interface, 20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1); Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send
to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press
#, return to start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send
to voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press #,
return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat
anything else as no answer
exten = a,1,VoicemailMain(${ARG1})
The issue is that I have, per virtual pbx (i.e. home or business), two
contexts
that these get used from. The internal-xyzzy and incoming-xyzzy
contexts (one
for each pbx, ie. xyzzy is home or else it's office).
I was wondering if there wasn't a more flexible solution to this issue, than
hard-coding a Goto(default,s,1) into them (I have no default context,
because it
would be meaningless).
Perhaps using Gosub and Return. Or do I need to hack the macro, and
pass in a
3rd argument (bletch)?
I am not a macro guy, but I see three possible ways of operation:
1. extend the macro to have a third parameter, which would be the
Context the macro is called from (and have
exten = s-BUSY,2,GOTO(${ARG3},s,1)
2. use a global variable for the same purpose
3. Check wether the ${CONTEXT} variable is still set to the calling
context in a Macro (no idea if it is, worth a NOOP, right?)
HTH
Anselm
___
--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] Recommendations for 100 Wifi SIP phone setup
Am Montag, den 26.11.2007, 22:39 -0500 schrieb [EMAIL PROTECTED]: On Nov 26, 2007 9:52 AM, Alberto Pastore [EMAIL PROTECTED] wrote: I also found the Pirelli DP-L10 dual phone to be an excellent sip client with good roaming support and discrete battery saving capability. (Used in a 14-cell wifi network with 40 cellphones). I don't know what to say I have not used the Pirelli phone but at the same time it is the same ODM as most of the Linksys and D-Link phone and I have not been too pleased with those. They work. They roam ok but they also lock up every so often and the call quality isnt the best. You can tell the G729 codec is very taxing on the device it can take 2 sec for the phone to respond to a keypress. Hello *, I have the Pirelli phone (there are two actually, I have the bar-type one, I think it's L10) in daily use, both GSM (O2 Germany) and WLAN (registered to Asterisk of course - behind OpenWRT boxes, FritzBoxes, D-Link APs whatever is there). I had the latest firmware in August, did not check back since. In my opinion, this phone is not ready for production use for regular users. It works really nice in the short run, but a few things make it unacceptable or at least lack for my approval as a well-done product: - Connection loss on DHCP expiry (twice yet) - Relatively poor Wifi signal strength, compared to other Wifi devices - frequent lockups, which require battery removal and clock reprogramming: - if you power on the phone while it is on charger power - if you receive a WIFI call while you have a WLAN call, and the WIFI call is from the same contact in the phone book - plugging in, unplugging, plugging in headset fast in a row - no SIP voicemail support, GSM voicebox only (pressing 1) - slow user interface It further lacks - one-touch silent mode - you can only kind of emulate that - proper headset support - any key accept call does _not_ work, although it is a separate choice from accept key accept call. Auto-accept works OK though. Vibra seems to _not_ work once a headset is plugged in... or at least not always. - one-digit press in main menu to open the menu: It works in the submenues, but you always need to navigate the main menu with the arrow controls - quick mode switch WLAN on/off - if you are out of home range, WLAN seems to suck lots of battery. - display of CALLERID(name), currently only CALLERID(num) is displayed I would also like a modus which is if no known network is in range, connect to any unencrypted network you can get that seems to have network connectivity. This should of course be optional. That said, for my personal uses it is OK, lightyears in front of the two UTStarcom phones I also had in daily use. Well, while they worked anyway. Both the good WPA support (basically broken in the UTS) and the GSM function make me like it. The phone book (multiple entries) is great, although it would be nice to see from which of the numbers listed the call is coming (call Sam back on his mobile, or is he at home?) I believe most of the problems I see could be solved in software quite easily, but until they are, I would not give it to my users, rather I would go with DECT, Siemens Gigaset ISDN, on FritzBoxen internal S0 bus, because this combination I know works absolutely perfect, as good as ISDN, and that means a lot. I have been using the Pirelli for five or six months, and keep using it because the GSM/WLAN combo is just the killer app on it. It sucks a bit more than my regular mobile phone (which I sometimes carry instead, I have a Dual-SIM contract), but for me all mobiles suck. I have thin fingers, but using mobile keypads always makes me feel like having jelly sticks on my hands. That is why I love the BudgeTone 100 phone :-P Best regards, Anselm P.S.: Your fingers are too fat to dial - please mash the keys for your free dialing wand -- phone announcement in Simpsons King Size Homer ___ --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] Can Asterisk act like an ISP dialin server to data callls from Sipura 3000 or other ATA connected devices ?
Am Dienstag, den 27.11.2007, 18:00 +0100 schrieb Robert Rozman: Hi, I have an older phone with touch screen from Philips. It have it connected to Sipura 3000 FXS port and majority of features work ok. But phone also has touchscreen and web browser that I'd love to use for accessing my local web pages. But the phone only allows me to setup ISP phone number (username and password) and it wants to call it to get to Internet. Since it is connected to Sipura3000, call can come to Asterisk and I'd love to somehow fool that device and connect it to local web pages ? I guess I could somehow mimic ISP internet calling feature on local Asterisk server, but have no clue even where to start searching ... Any advice ? Hi Robert, I researched for something similar about a year ago, and came up with nothing really worth the work. If you can, try to get another ATA that has a real, old-fashioned serial modem plugged into it, and limit that modem to 9600. I think more than that will not work reliably, but you could of course try. The only working implementation of software emulating a modem in conjunction with asterisk I have seen is fax-related, and even there I read from several people that anything better than 9600 is hardly ever achieved. The code there is cranked into fax-use though, not modem use, which would require the PPP bytestream to be off-handed instead of fax parsing. Perhaps iaxmodem would do that No idea. I'd be interested in how you get that working, if you do indeed. BR Anselm ___ --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] Aastra 480i CT - No Incoming Calls
Am Mittwoch, den 21.11.2007, 13:47 -0500 schrieb [EMAIL PROTECTED]: I just bought an Aastra 480i CT for a client who needed cordless capabilities in their office. I'm trying to set up the base station and cordless handset in my office first. I'm able to connect the phone to my Asterisk box and make outgoing calls from either the base station or the handset - to extensions within my office as well as numbers outside the network. But I can't receive calls on either the base station or the handset. All of the calls go strait to voice mail. I've never had this problem with the phones I use in my office - Linksys SPA942. What am I doing wrong? Hi Danny, I do not know what you are doing wrong, but you could check the following: - Is Do-not-disturb possibly activated on the phone? (just checking) - What does sip show peers say (in on-hook mode, is the phone correctly registered to asterisk?) - What does the CLI show when you call the phone from another extensions, with verbose somewhere around 10? - Might NAT have to do with it? Something special with the network socket you use for the phone? - You have a recent firmware on it, I guess. Better check that anyway. Best regards Anselm ___ --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] Gigaset S450ip and simultaneous calls
Am Montag, den 19.11.2007, 13:45 +0100 schrieb Olivier: Hi, My Gigaset S450ip allows 2 simulatneous calls when each incoming call are targeted to different phones. When both calls target the same extension, the second one is forwarded to voicemail. I couldn't check yet SIP messages but has anyone met this limitation (one simultaneous call per phone) ? Hi Olivier, please read the manual. The german version (available online on gigaset.de) states on p. 33 that this feature can be activated or deactivated via phone menu. If you enable call waiting, incoming second calls will give the caller the usual ring sound, and the phone in use will have a sound and optical message that a call is waiting. If it is deactivated (which will be the case at your setup), calls will be rejected, and with many SIP providers this means they go to voicemail. The german menu item is called Netzdienste - VoIP - Anklopfen, which will probably be like Network services - VoIP - Call waiting in the English version. HTH Anselm ___ --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] [Fwd: Re: VoiceMail hangup]
Am Dienstag, den 13.11.2007, 09:29 -0500 schrieb Il Neofita:
Hi
I have the same problem
On Nov 13, 2007 9:10 AM, marcotasto [EMAIL PROTECTED] wrote:
Hi Neofita, Doug and All.
I think I've the same problem but I don't know if it's related to the bug
suggested below.
I try to explain my behavior:
- I dial the voicemail extension.
- I hear: You have 1 new message. Press 1 for new messages, press 2 for...
or # to exit (I listen the complete message or most part of it)
- I press 1
- I can hear the first recorded message.
But, if:
- I dial the voicemail extension.
- I hear you have 1 new message. Press 1... 1 pressed (without waiting
for the message playing)
- Asterisk hangups.
I'm not always able to replicate the problem but, as Il Neofita, I'm
using the italian prompts... could be a problem related to that?
Bye and regards
Marco Signorini.
Marco, Il Neofita,
it seems you exactly found that bug. May I suggest a workaround idea:
After the dialplan call to VoiceMail() for leaving the message, call an
AGI script that checks the related directory, especially the last
message. If it is less than 45 bytes, remove it. That AGI need not be
too complicated, might be a bash script like
#!/bin/bash
for A in /var/spool/asterix/voicemail/default/* ; do
for B in ${A}/*.wav ; do
SIZE=`ls -l --color=never ${B} | awk {print \$5; }`
if [[ $SIZE -le 44 ]] ; then
rm -f $B
fi
done
done
Caveat emptor, just a quick idea.
Trying another file format for voicemail recording might also be an
option, as this seems to relate to the wav header somehow. You might
choose alaw, ulaw, perhaps gsm or speex... Give it a try, and report
back if that helps. The voip-info.org wiki page about voicemail.conf
should tell you how to exactly set that up.
BR
Anselm
___
--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] VoiceMail hangup
Am Montag, den 12.11.2007, 15:14 -0500 schrieb Il Neofita: Hi additional information if I am going to wait at least 3 seconds after the voicemail starts to give me the instruction I am able to listen my messages. But why I need to wait? On Nov 12, 2007 2:28 PM, Il Neofita [EMAIL PROTECTED] wrote: Hi, with some messages the voicemailmain after give me the information about the call (Days, hours and minutes) it finish. Whant can I check for solve this problem? Read voicemail.conf. Look for minmessage setting - it will remove messages that are shorter than the given number of seconds. See http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail See http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf BR Anselm ___ --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] Recording just first part of call?
Am Dienstag, den 06.11.2007, 11:49 + schrieb Tony Mountifield:
I know that I can record the contents of a call by calling Monitor()
or MixMonitor() from the dialplan just before invoking Dial().
I have a potential customer who wants only the first minute of each
call recorded (for identification purposes, without the storage overhead
of keeping the complete call).
Can anyone here think of the easiest way to do this? The only possibilities
I can think of are:
a) Add a new option to Monitor() or MixMonitor() to stop recording after
a specified length of time.
b) Record the whole call and post-process the recording file to discard
all except the required first part.
The asterisk manager API seems to offer a StopMonitor command, which
is basically the same as the StopMonitor() extensions.conf command,
afaik.
A quick ugly hack (and well, I did not have my coffee yet, so caveat
emptor):
Before calling the Monitor() in extensions.conf, call an AGI that kind
of starts a timer. This AGI would have to know about the Channel used
(you surely figure how to do that, I am to lazy to look it up right
now).
Something like
8
#!/usr/bin/php -q
GLOBAL $stdin, $stdout;
ob_implicit_flush(false);
set_time_limit(30);
error_reporting(0);
$stdin = fopen( 'php://stdin', 'r' );
$stdout = fopen( 'php://stdout', 'w' );
while ( !feof($stdin) )
{
$temp = fgets( $stdin );
$temp = str_replace( \n, , $temp );
$s = explode( :, $temp );
$agivar[$s[0]] = trim( $s[1] );
if ( ( $temp == ) || ($temp == \n) )
{
break;
}
}
$channel = $agivar[agi_channel];
system (screen -d -m /usr/local/bin/stop-recording .$channel);
exit(0);
8
The script at /usr/local/bin/stop-recording could be a bash script:
8
#!/bin/bash
sleep 60
# Before Stopping the monitor, you want to make sure that
# about 60 seconds went past
# Perhaps add some leeway if the other party answered
# after ring no. 5 or so
# The following should all be on one line, but emails tend to break...
( echo -e Action: login\nUsername: foo\nSecret: bar\nEvents: off\n\n ;
sleep 1 ; echo -e Action: StopMonitor\nChannel: $1\n\n ; sleep 1 ) |
netcat localhost 5038 /dev/null 2/dev/null
8
You would want to add a check that the original call is the one to be
StopMonitored() - e.g. if the caller hangs up and redials within a few
seconds, the second call would possibly be terminated. You could manage
this by writing the channel to a temporary file in the AGI, removing
the file after call termination. The Bash script would then read the
channel from the file, or just silently terminate if the file is not
there.
This is just an idea. It needs some tweaking here and there, and there
probably are way more elegant methods for solving the task... :-)
BR
Anselm
___
--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] issues with downloads.digium.com
Am Freitag, den 02.11.2007, 12:12 +0200 schrieb Atis Lezdins: On 11/2/07, Michiel van Baak [EMAIL PROTECTED] wrote: On 09:38, Fri 02 Nov 07, Tony Mountifield wrote: Does anyone from Digium want to comment on why this Eloqua stuff has been used, instead of just allowing Apache to serve the directory tree directly? And whether this decision might be reconsidered? I think it's some sort of tool to track downloads and stuff so the marketing people at digium can use that info for their campaigns and stuff. I dont like it neither, but I dont think they will reconsider. They closed the FTP downloads so everything has to go thru this www tool. That makes it pretty obvious they will stay with it. I wonder - why they can't get all the info from logs. They can even put .htaccess to route all downloads trough PHP that will log whatever else it needs.. I'm just annoyingly copying URL and deleting first part of it - it's simpler that to quote whole URL.. And without quoting - bash can't correctly interpret the ? stuff.. When you use wget to retrieve data from the web, you will have to quote the address if it contains more than one CGI parameter (and the character between parameters). Bash, as some other shells, reads the ampersand as run this command in background, even if it is amidst other character data. ? could cause havoc with file name completion, but it mostly will not - who really keeps files with names like http://www.digium.com/downloads/index.php...; ? ;-) For those who regularly download files, they probably could write some small bash script that mimicks a browser in downloading the file. I personally do not like forced download redirections. At least on sf.net there is always a direct link which can be copy-pasted to a console - this is especially important when you need to download the file to a ssh-administered server, but the browser runs on your desktop machine. BR Anselm ___ --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] Mobile phone codecs ...
Am Mittwoch, den 31.10.2007, 16:47 + schrieb Gordon Henderson: Not strictly asterisk related, however... No GSM! How odd is that, given that it's a GSM mobile phone... Maybe the GSM codec is implanted to the GSM chip and that one does alaw, ulaw... Anyway, my quest for the ultimate one handset solution is getting closer. If Wi-Fi weren't so rubbish and my house not made of Dartmoor Granite it might have half a chance of working outside the room with the access point, however ... That is one of the two points I like at the American style wood and wallpaper houses (the other being that construction is cheap and easy, in comparison). Living in a concrete house is not all the best thing as well though. My Pirelli dual-mode phone loses WLAN link just outside my flat door in the hall way, one concrete wall and about five meters from the Access Point. What luck they only used drywall inside the appartment. Anyone tried the Plantronics Voyager 510 bluetooth headsets which regsiters to both a mobile phone and their own base unit (which presumably has a USB sound device) I had a Plantronics device here that connected to a phone-line-tap base station or to my mobile via bluetooth. I did not buy it though because it only worked with my Sony T610 (stone-age old, about 2003), not with my O2 xda. I sold one plantronics 510 to a customer who uses it with his Nokia Esomething, and really likes it. AFAIK the USB device that comes with it is a bluetooth dongle, not a virtual audio device, but that might be different between versions of that device, and my customer definitely does not use it. I noticed with my plantronics device back then that you needed to re-pair it (whohoo, never noticed that similarity repair to re-pair) to whatever device you want to use it with, and that sucked because it took half a minute and some interaction with the mobile or base station. as in: https://www.ukheadsets.co.uk/thc-plantronics-voyager-510-usb-dongle-rn-422-action-show_detail-show_products_mode-cat_click I'm not a fan of soft-phones, and not sure I want to have a borg implant on when I'm not driving, but ... resistance is futile :-# Best regards, Anselm ___ --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 multi Bindport
Am Montag, den 29.10.2007, 15:54 + schrieb Gordon Henderson: On Mon, 29 Oct 2007, Abdul wrote: Hi, Is it possible to have multi listening bindport in asterisk? Now days mostly ISPs are Blocking the standard 5060 port so we want to keep option if 5060 is blocked we can ask our customers to use another port. Really? What country?? What ISP? This isn't my experience in the UK .. (yet???) Hi Gordon, I have heard of SIP/VoIP port blocking in certain Asian regions. I think in India the phone market regulations are in favour of their local Ma Bell company who wants to sell minutes, not transport cheap VoIP packets over DSL. I think I read about one of the states of the Arabic peninsula that their jurisdiction forbade any kind of communication that might be considered encrypted or untraceable. Tracing SIP is considered more difficult than wiretapping an analogue line copper pair, figures. I have been told by a friend of mine whose husband-to-be is in Shanghai for a few weeks that VoIP is not restricted there - contrary to the common assumption that the Chinese digital wall is airtight. There might exist restricions in Internet access in rural areas, or for locals (opposed to foreign tourists and workers). There are other regions and legislatures that might prefer strong control of international communications (not necessarily those called Axis of Evil). When I was a child, most of the letters I got from my eastern aunt were inspected, and older locals know of line noises from technologically outdated wiretapping equipment used by the Stasi- might be legends though. I once visited her, crossing the Iron curtain was an intimidating experience for a young boy, even with his father at his side (although other things of the then-East German Republic stuck more in my mind). I am quite glad we can mostly say publically what we think appropriate nowadays. Locals of those countries concerned will know better than me, possibly they are not interested in Asterisk though because of the obvious (legally or technically mandated) uselessness. You might check where those people asking about OpenVPN/Sip combination are from ;-) fiction If and once the more restrictive politicians take control and realize their personal idea of 1984, you surely will also notice the telecommunications regulations, that according to MINITRUTH, will have been there all the time. I am positive they will also cover the airstrip one region of Oceania, so don't run, they will come for you. (Orwell's 1984 was one of my final exam topics at high school) /fiction Coming back to reality I wish you a nice evening. Best regards Anselm *Wait, there's someone on the door, I !%$§)(A/SCNR!)(/§ CARRIER LOST ___ --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] G.729 codec between avaya and asterisk
Am Dienstag, den 23.10.2007, 22:21 -0700 schrieb satish patel: there is no special requiremnt to use g.729 but day to day my sip client incressing thats why some time i got breaking voice or voice quality not much better i think in LAN there is lots of brodcat on lan If your LAN is congested and a lot of single packet delay happens, you should improve the LAN. You cannot run a LAN at 99% saturation with VoIP, it will just not work, with packet drop rates and delays making phone calls more of a earth-to-moon radio experience (Houston *crackle* *crackle* have *crackle* problem). If _all_ that traffic is VoIP, G729 might help a bit, but I would not expect it to get around all your bandwidth problems. Try to improve the network first. One interesting aspect of g729 might be that your sip client phones that live behind a DSL line might profit from the smaller bandwidth requirement on their side. if i purches g.729 transcoder license for asterisk to convert g.729 to g.711 then it will work or not I _think_ it will work (btw this is, as of some website I found, the main revenue stream of Digium, so they will be interested in having it working). Others with real-world experience could tell you. but why i need codec on trunk Codec stands for coding-decoding (or something similar). If you imagine the original signal as voice and sound, meaning variations in air pressure around the membrane of the telephone handpiece microphone, then every digital representation is a kind of coding. This even refers to 8-bit-wave, which is the most obvious way of encoding: It merely writes down the voltage level at the microphone input in the range -128 to +127 (IIRC, correct me if I am wrong). Accordingly, 16-bit wave has the higher precision of -32768 to +32767. G711 is - again, if I remember correctly - an adaptation of these bytes to a logarithmic scales, bearing in mind the idea that small changes in the higher ranges are treated differently from small changes in the near-0-region. Something like the fiction bytestream value 0 1 2 3 representing the scale 0 4 6 7 of microphone values, instead of linear data. Please research this yourself if you are interested in details. G711 is the standard (and usually, the only available) codec for ISDN/T1/E1... Europeans and US Americans established two different kinds of G711 (µ-law and a-law) which seem to be functionally similar. BR Anselm ___ --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] Voicemail playback on iPhone
Am Mittwoch, den 24.10.2007, 11:19 -0500 schrieb Peder @ NetworkOblivion: This is semi-related, but I have a Tmobile MDA and I couldn't play the files either. The issue was not a codec issue, it was an email encoding issue. If I sent the message to an email account and it was then downloaded to my desktop via outlook and then forwarded on to my phone, I can listen to them. If I just send it direct to the phone, I see the attachment and it opens in media player, but it won't play. I don't know if you are having codec issues or email encoding issues, but it is a place to look. Incidentally, if someone knows how to get around the download email and then forward issue that I am having, I would like to hear it. Peder, you might want to start a new thread on this: If it really troubles you odds are others also have that problem. For a start you could investigate the difference between mails sent from the Comedian versus mail sent from Outlook (probably the latter's headers look as if they were meant to be funny... this would be the first time that I see Outlook produce mails more compatible than another mailer program :-/ ) The hint might be in different places: The exact settings of the MIME/multipart stuff might be the hinge point. IIRC you can use an external script to mail-forward new voice messages. You could try some mime-capable mailer to do that for you, perhaps they get it working. I also own an MDA (clone, some Korean HTC iirc, but the company logo is nowhere to see, just the network provider logo was there until it rubbed off in everyday wear and tear). As I do not use it to read mail I do not know wether this problem could be repeated here. Perhaps you could give a guide how to reproduce it? (I _do_ use Squirrelmail on that device to access my courier imap server holding voice mails - but that will not count for this problem). Best regards Anselm ___ --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] G.729 codec between avaya and asterisk
Am Dienstag, den 23.10.2007, 02:56 -0700 schrieb satish patel: Dear all i have asterisk connected with avaya through E1 back-2-back now when i configure my sip client with g.729 codec then i m not able to put call from asterisk to avaya and when i user g.711 it is working fine so i dont know why i need G.729 on E1 Trunk it is TDM technologies then why my call fail in g.729 case Hi Satish, Neither do I know why you _need_ G.729. Are there any specific reasons why you do not want to use G711 in the sip client, which is working fine? (Nota bene: there are some more codecs supported by asterisk, some of which may be also supported by your sip phone) Your E1 trunk obviously is G711-only - this is to be expected, because the G711 wave samples are those which go over the wire (as time-division multiplexed bitstream). Together with the information from http://www.voip-info.org/wiki-Asterisk+G.729+Licensing namely ** G.729 requires a license per channel unless it is used ** in pass-thru mode. which exactly matches your setup (by the way that was the first google match for g729 asterisk) we can guess that you did not buy the license which would be necessary for asterisk to transcode G729/G711. [sip_phone]--[asterisk]-E1[Avaya][analog_phone] Asterisk sip client configure with g.711 alaw/ulaw Avaya phone client configure g.711 alaw/ulaw suggest how do it implement g.729 on this case what change i have to done on both part Avaya / E1 stays as is, sip client stays as is, your credit card data is transferred to digium, and their license goes into the appropriate file on your asterisk machine hard drive. Others may have real world experience with those steps, but that is what I read on this mailing list. YMMV, Anselm ___ --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] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)
Am Samstag, den 20.10.2007, 22:58 -0700 schrieb Philip Prindeville: Erik Anderson wrote: On 10/20/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: If you are trying to use non-complied (XML) profiles... don't even bother wasting your time. Why is that? I'm using the xml-style config and they're working just fine. I'd like to be able to templatize a server, add a bunch of new handsets into sip.conf and extensions.conf, and then plug the phones into a network and have some DHCP and/or TFTP glue logic that sees the DHCP or TFTP request, and from it generates a boot file (an .XML file) and a response parameter list for DHCP... populates a file into the /tftpboot/ directory, etc. How viable is this? The problem there is that you have a very small windows. AFAIK there are no tftp servers that can generate files on-the-fly, so your script would have to generate the XML within less than a second, reliably, and do all the necessary asterisk changes within another second or two, and I doubt this will be possible _that_ quick. Of course you can use ISC dhcpd for tailoring answers to your needs (dynamic setting of config file etc), but IMO this will only work well if the phones support http config download, because that gives you a much better hook to put your script, and you can hold back the file until all the asterisk changes are done, and finally return the XML (or whatever). BR Anselm ___ --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] parse error in GosubIf
Am Mittwoch, den 17.10.2007, 21:57 +0200 schrieb Michael Iedema:
Greetings everyone,
today I spent the last part of my day trying to find a parse error
inside this snip:
http://pastebin.ca/740081
If there's anyone who can shed some light on why my GosubIf condition
is throwing a parse error, I'd greatly appreciate your insight. This
was really frustrating and is probably a stupid mistake.
Try changing the relevant line
exten = h,n,GosubIf($[${SENDNOTIFICATIONS} = 1]?notify,1)
to
exten = h,n,GosubIf($[${SENDNOTIFICATIONS} = 1]?notify,1)
BR
Anselm
___
--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] About .call files when the congestion is on myside
Am Montag, den 15.10.2007, 16:38 +0300 schrieb Cosmin Prund: Behalf Of Anselm Martin Hoffmeister wrote: Subject: Re: [asterisk-users] About .call files when the congestion is on myside Am Montag, den 15.10.2007, 11:26 +0300 schrieb Cosmin Prund: Hello everyone. I’m working on an application that needs to automatically send faxes. To send the faxes I create .call files but the .call files mostly fail because my lines are always congested within business hours! Is there any trick I can use to give the end user a better chance at actually receiving the faxes? Are you aware of the MaxRetries, RetryTime and WaitTime in your call-files? You can set quite large numbers, e.g. a RetryTime of 15 minutes and a MaxRetries of 32 would try for up to 8 hours. Note though that any answered call will stop the retry cycle. This is embarassing for Zap channels that cannot detect remote ringing / remote busy reliably. As you use ISDN this should not be a problem *IF* an unanswered call stops the retry cycle then it's true, I can simply ask for lots of retries. I assumed an unanswered call would NOT stop the retry cycle so I was afraid to set a large value here. I'll have to test what happens if the called line doesn't pick up the phone. An unanswered call should just initiate another Wait, followed by a retry. Unanswered means as much as unsuccessful, for the purpose of a call file is to dial out and get whatever done. If you want unanswered calls to be successful (which does not make much sense to me, because the fax has not been delivered), you probably need scripts that do the management for you. If you want to do this (looping) use MaxRetries = 0. I do not understand why having the remote side connecting to a local extension that does faxing would not work. Or is it that the CAPI FAX stuff will only work on unAnswer()ed channels? It's the CAPI stuff not wanting to send over a non-CAPI channel. And it somehow makes sense, because the CAPI stuff uses the DSP's in my ISDN card, so it can't work unless it's on a CAPI channel. Also I expected the capi application to see through the Local channel and notice it really is an CAPI channel! Sure. you will have to use CAPI for the outgoing leg of the call file, I think. If you need more specific handling than any successful call stopping to retry, you will need some kind of queue that faxes will be rescheduled from _until_ they either are delivered or at least once the number has been called but not been answered (if this really is what you want). BR Anselm ___ --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] About .call files when the congestion is on my side
Am Montag, den 15.10.2007, 11:26 +0300 schrieb Cosmin Prund: Hello everyone. I’m working on an application that needs to automatically send faxes. To send the faxes I create .call files but the .call files mostly fail because my lines are always congested within business hours! Is there any trick I can use to give the end user a better chance at actually receiving the faxes? Are you aware of the MaxRetries, RetryTime and WaitTime in your call-files? You can set quite large numbers, e.g. a RetryTime of 15 minutes and a MaxRetries of 32 would try for up to 8 hours. Note though that any answered call will stop the retry cycle. This is embarassing for Zap channels that cannot detect remote ringing / remote busy reliably. As you use ISDN this should not be a problem I already tried using the local channel for dialing (so I can put in there a loop that waits for a line to be available) but this doesn’t work because I’m sending faxes using chan_capi’s capicommand(sendfax) – and that command requires an chan_capi channel, it doesn’t like the “local” channel. Besides, looping in the dialplan would probably interfere with the “Wait” option in the .call file so that’s a really bad solution. If you want to do this (looping) use MaxRetries = 0. I do not understand why having the remote side connecting to a local extension that does faxing would not work. Or is it that the CAPI FAX stuff will only work on unAnswer()ed channels? Can you provide an example .call file? Regards, Anselm ___ --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] file.c: File digits/ett does not exist in any format
Am Samstag, den 13.10.2007, 15:01 +0200 schrieb Turbo Fredriksson: I'm using Swedish on version 1.4.13. The full part of the log is: [Oct 13 12:51:16] WARNING[7810] file.c: File digits/ett does not exist in any format [Oct 13 12:51:16] WARNING[7810] file.c: Unable to open digits/ett (format 0x8 (alaw)): No such file or directory The word 'ett' means 'one'. We have two words for one: 'en' and 'ett'. Any idea how to fix/solve this problem - can't listen to my voicemail if I only have one message? It works if I have 1 message... The easiest solution for the moment is to create a file named ett.gsm in the appropriate directory- that should be /usr/share/asterisk/sounds/se/ in your case. You can either create an empty file (with touch, for example) which will of course result in no number to be read out, which could be annoying. You could also copy the file en.gsm which should exist there over to ett.gsm - wrong reading will result, but I guess people understand what is meant, like they would understand you have _an_ messages instead of one message (details of swedish numbers are a mistery to me ;-) As soon as _any_ file can be played the voicemail will work. You could also record a file yourself - if you do not mind having a single digit read by a different person than the surrounding text. Or you could try and find the file ett.gsm somewhere; I have no idea why it does not exist, but it probably should. BR Anselm ___ --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 behind Multi-NAT question
Am Dienstag, den 09.10.2007, 19:50 +0100 schrieb WipeOut: Hi, Ok.. I know dual NAT is a problem for SIP.. ie. UA - NAT - Internet - NAT - Asterisk What about Multi-NAT where a dedicated public IP is mapped to the private IP of the asterisk box.. ie UA - NAT - Internet - Multi-NAT - Asterisk http://www.draytek.co.uk/support/kb_vigor_multinat.html Anyone tried it? My experience with SIP, Asterisk and more than one NAT in the path is not a good one. For example, several of my SIP hardphones refused to work behind a dual-NAT Phone (10.10.0.201) - NAT - internal net (192.168.174.0/24) - NAT - Internet - Asterisk where everything else worked as usual. Admittedly multiple NATs are not necessarily a good idea to have, but that was a customer's network, not mine ;-) Also quite regular setups like Phone - NAT - Internet - NAT - Asterisk and 2 Phones - NAT - Internet - Asterisk without NAT (One of those phones calling the other). might work - or just be a source of trouble. This also seems to depend on the cooperation of the NAT device; some work better than others. IAX seems to handle NAT issues much better, in my experience, but I did never have an IAX hardphone. BR Anselm ___ --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] Click to Talk Web Applications with Asterisk
Am Dienstag, den 09.10.2007, 14:23 -0500 schrieb Ricardo Melendez: Hi, I would like to develop a “click to talk” app to interface with asterisk, anyone know about some SDK/frameworks to implement this. I have not ever used such an application, but there are several solutions commercially available. If your intention is getting a solution, you might consider spending money. If your intention is learning, the better - but sorry, I cannot give adequate pointers there. I remember there were open source puzzles parts that could be mended to something like a web click-to-call app, might be the term jiaxclient relates to that. Do not count to much of that, my brain is getting old. I do not want to advertise a specific solution, but you could search the mailing list archives - click to call might be a subject worth reading. You could also look for something like IAX Client JAVA. I bet there is also some information to be found on voip-info.org. I think at least one vendor offers free trial versions so you could at least test wether the concept is viable, and then decide to either spend money or time on the project. I hope you did not trigger one of those Hey, I have a solution for you, hey, this is a non-commercial-list, go die flamewar - we had enough of those ;-) Best regards, Anselm ___ --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] Voice server
Am Montag, den 08.10.2007, 11:07 +0200 schrieb Vincent:
Hello
Now that I received an OpenVox PCI card
(www.openvox.com.cn/products_detail.php?genre_id=9id=28), I'm ready
to try and set up a voice server with Asterisk.
We need the following features:
1. When customers call in, they should hear a voice menu asking them
which software they're calling about
2. Next, they should be able to leave a voice message to explain what
their problem is
3. Next, Asterisk should send an e-mail to an alias that includes all
the people involved with the software
4. Finally, anyone involved should be able to listen to the voice
message and call the customer back.
I guess Asterisk can do all this easily. As for listening to voice
mail, some of the people are off-site so I guess I should either set
them up with SIP phones to let them connect to Asterisk through the
Net, or have Asterisk save messages as WAV files and upload them to a
web server so people can just click on the file to listen to the
message.
Has someone done such project and could give me some tips?
Asterisk can do all of that. Something along the lines of
exten = s,1,Answer()
exten = s,n(selector),Read(SELECTION|please-select-software|1)
; please-select-software should contain something like
; Welcome. Please select software. Press 1 for
; BongoSoft Exploder, 2 for ...
exten = s,n,GotoIf($[${SELECTION} = 1]?bongosoft)
exten = s,n,GotoIf($[${SELECTION} = 2]?product2)
exten = s,n,GotoIf($[${SELECTION} = *]?operator)
exten = s,n,Playback(sorry-please-try-again)
; sorry-please-try-again should contain something like
; sorry, we could not understand that. You will have
; to press a digit for us to help you
exten = s,n,Goto(selector)
exten = s,n(bongosoft),VoiceMail(1)
exten = s,n,Playback(thank-you)
exten = s,n,Hangup()
exten = s,n(product2),VoiceMail(2)
exten = s,n,Playback(thank-you)
exten = s,n,Hangup()
exten = s,n(operator),Dial(SIP/sip501SIP/sip501SIP/sip503,60)
exten = s,n,VoiceMail(0)
exten = s,n,Hangup()
in your extensions.conf, and the rest (sip.conf / zap*.conf /
voicemail.conf) fairly standard setup.
Just to give you an idea - you can also do multiple-stage IVRs, and
easily too.
The recording of messages could be done by means of the asterisk voice
mail system (which allows tons of options) - probably could do exactly
what you want, and attach voice mail to e-mails if needed, or via
scripting, upload them to a web server. You could even listen to
messages via a good old telephone ;-)
But there is still a question: Why do you want to do this? I just have
doubt that your customers really like to call your hotline just to leave
a message - if the system does not even offer the slightest chance of
ever speaking to a real human anyway. Would they not better talk to a
human, which could allow for interaction, questions back to the user
etc.? You could use queues for load (call) balancing with a flexible
number of available employees - and still redirect them to voicemail if
their call could not be handled within 60 seconds or the night
mode (no one in the office) is active.
With fast internet connections nowadays even remote agents could use SIP
(soft-)phones to take calls being millions of meters away from your
asterisk box.
HTH,
Anselm
___
--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 signalling, SIP, and Background()
Am Mittwoch, den 26.09.2007, 11:08 +0200 schrieb Bastian Friedrich: Hi, I am currently setting up a voice mail/IVR machine with asterisk (1.4.10 at the moment). During testing and evaluation, all was fine; in the - slightly different - production environment, the IVR contexts do not react sensibly. The environment is: POTS -- (ISDN) -- PBX -- (SIP) -- Asterisk with the Asterisk registering with our local PBX. When a user reaches the Asterisk machine via this path, key presses are ignored during the Background() function. My debugging possibilities have been a little restricted, unfortunately (I'm working on that), but as a wild guess, I suppose we might have the following problem: When a call is processed as a SIP call, in-band DTMF signalling does not trigger an event in Asterisk; our PBX possibly/probably does not create a SIP event for DTMF signalling. Would you think that this may be the reason for our experienced problems? Asterisk knows of three different ways for DTMF signalling, in-band being only one of those. There are also rfc2833 and info (SIP INFO) signalling. You could try and set the dtmfmode= parameter in sip.conf to one of those. voip-info.org has some info about it. On the other hand it might be the case that your SIP PBX does _not_ generate SIP INFO or RFC messages but the DTMF signal is poor, not allowing reasonable operation. I had that one with a SIP provider once, effectively meaning I could not remote-control the voicebox. Viel Erfolg, Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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
Re: [asterisk-users] Completing my Configuration
Am Dienstag, den 25.09.2007, 07:37 +0200 schrieb Guenther Sohler:
Hallo Group,
I have basically set up a small asterisk system,
which ahs 4 peers:
* registers at 2 Sipgates
* 2 hardware phones connected to it
Both Hardware phones can phone outwards(cheaper sipgate is selected with
dialplan)
Calls from both sipgates make my hardware phones ring
But here comes the challenges:
Is it possible to configure asterisk in such a way that in the phone:
* there are names instead of numbers in my hardware phone displayed
Depends on the hardware phones. In theory, with each SIP call connecting
to the phone, both a name and a number can be transferred. AFAIK sipgate
defaults to setting both to the usual callerID. That is exactly the
reason why you can set the variables ${CALLERID(num)} and
${CALLERID(name)}.
Some hardware phones (I assume, the better ones ;-) display both; my
Allnet for example seems to only display the name, but store the number
for the call back list. My Fritz!Boxen seem to forward both name and
number to ISDN devices on the internal S0-bus, just not many ISDN phones
can actually display text numbers.
Let your asterisk have an ast database, looking like
callerid/420123456789 = Doe, John Q.
callerid/492240224922 = Mustermann, Dr. Peter
Then you could expand your dialplan logic a little. If you have a line
exten = 12345,4,Dial(SIP/phone1,60)
or whatever that looks like in your SIP-incoming context, insert those
lines before it [and change the 4, 5, 6, 7s ;-) ]
exten = 12345,4,Set(CALLERID(name)=${DB(callerid/${CALLERID(num)})})
exten = 12345,5,GotoIf($[${CALLERID(name)} = ]?6:7)
exten = 12345,6,Set(CALLERID(name)=-- ${CALLERID(num)})
exten = 12345,7,Dial(SIP/phone1,60)
Line 6 treats the case that the number is not in your database and sets
the callerid-name to -- NUMBER_OF_CALLER
You can manually add data to the astdb from the asterisk CLI with
database set callerid 420456789 Silly, Roger M.
You should check that both your SIP providers provide incoming CLI in
the international formatting, without country prefix or +. In my
experience some SIP providers send numbers like
492240224922, others send +49... or 0049..., some send national format
02240... for all national calls, some even omit the leading 0 there,
and some just change the behaviour depending from which network (T-Com
landline, Arcor landline, T-Mobile cell phone, O2 cellphone, foreign
callers...) the call originates. If you have more than two providers,
this can be a PITA - you will need some dialplan logic to sanitize the
callerid in those cases, and sometimes you are just left for guessing,
for example when the provider signals calls from T-Mobile as 16177554224
and calls from Boston, MA, USA the very same. Germany does not have
fixed-length numbers, even in the mobile phone networks the length
differs, and the number given might be valid for both circumstances.
/rant
* The Ringtone is different for special call numbers
If your phone supports that, yes, you can do it. The common method for
this seems to be sending an additional header. There will be docs on
SIPAddHeader(blah) or similar on www.voip-info.org, and you might want
to also use a database here to find out wether special ringtones are to
be activated or not.
* it is displayed, in which sipgate the call came from
You could use the CALLERID(name) field for that, by adding the provider
short name in front of the caller's name, like
exten = 12345,4,Set(CALLERID(name)=at-${DB(callerid/${CALLERID(num)})})
for calls via the at provider - or whatever seems stylish enough.
I personally have a logic that makes use of the dial-around prefix in
use here in Germany: From a regular T-Com landline you can select the
provider that will carry the next call by dialling 010[1-9]X or 0100XX.
Those prefixes of course do not work on SIP provider lines, and my
asterisk does not have landlines connected. So I use those for my own
purposes, e.g. selecting the SIP account that the call may go out
through. Dialplan logic detects 010XX (100 possible accounts are
enough, I just ignore 0100XX as additional number field here) and
selects the outgoing provider accordingly.
If I wished to have the incoming line signalled to me, I would prefix
the incoming CALLERID(num) with the provider code. Callbacks would go
through the same line - nice bonus. Most of my phones do not handle text
and number simultaneous display in a reasonable way, so I do not rely on
the text.
* using an extension in my call number redirects the call just to one
sip phone ?
AFAIK you could only do this by Answer()ing the line (at which point the
caller starts paying the connection) and asking the caller to input an
extension. (Hint: Read()). I personally do not like this solution at
all, because that is what DID and number block allocation were invented
for. You can get a number block with SIP from some providers. Or you
just get yourself another private phone number ;-)
BR,
Anselm
Re: [asterisk-users] Hola Jonathan, a ver si tre suena...
Am Dienstag, den 25.09.2007, 11:01 + schrieb dadsadsadf dsadasdsa: Hola Jonathan Te cuento un pokillo lo q intento hacer por si me puedes orientar en algo o de algun sitio donde pueda mirar Existe una especificación de Microsoft de lo que llaman Dual-Forking, que básicamente consiste en poder usar tanto el teléfono como el propio PC como dispositivo de comunicaciones, según convenga. De esta manera, por ejemplo, se puede usar estando en la oficina el teléfono de nuestra IP PBX para hacer y recibir llamadas, y sin embargo si se está de viaje, es posible usar el propio PC para iniciar y recibir llamadas a teléfonos IP de una empresa o incluso a RTC a través de su infraestructura de Voz IP. Para gente móvil o en general para perfiles de cliente que se inclinen por una solución softphone, este desarrollo es clave. Para el Dual-Forking, no se necesita tomar algo de Microsoft. Situaciones en que una llamada puede ser recibida como en tu oficina como en softphone de tu ordenador en tu casa si, como en algun telefono real que podria ser situado en todo el mundo - parallel call lo llaman algunos registradores SIP por aqui - es realizado simplementissimo en Asterisk. Por ejemplo, se puede poner en su extensions.conf exten = 201,1,Dial(SIP/officinademartaSIP/martaslaptop,60) y se puede añadir mas telefonos, si SIP si IAX o ZAP - se pone un ampersand entre esos y asterisk prueba llamar todos en mismo momento. Mejor, no hay problema si un de esos no es conectado en ese momento - los otros telefonos van a functionar normalmente. Claro tambien es posible solo llamar al un telefono de que el usador ha puesto la ultima llamada (puede memorar eso en la AstDB, por ejemplo), o miles otras situactiones. Yo tengo un telefono movil que ofrece connexion GSM y WLAN/SIP, que normalmente tomo cuando dejo de casa, y (vale, mas o menos... ;-) dos telefonos fijos conectados a mi asterisk. A vezes (viajando, por ejemplo) tambien tengo un softphone in mi laptop. Tengo que decir que todos son SIP/UDP, pero no puedo imaginar que la software de MS ofrece cosas que no se puede realizar en asterisk. Si es posible para ti, podria ser mejor continuar en ingles - hace algunos años desde aprendio español en el insti secundar (disculpe lo que resulta :-), y la asterisk-users es normalmente usado en ingles, asi puedes recibir mas mensajes de mas gente. Saludo Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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
Re: [asterisk-users] Newcomer Question
Am Donnerstag, den 20.09.2007, 08:30 +0200 schrieb Guenther Sohler: Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. Hi Guenther, this place probably is the right one. Welcome! I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at both providers(numbers) at the same time. And If I call someone, calls to austria shall use sipgate, whereas calls to czech shall use mujtelefon. This is possible, and it does not require too difficult steps. First question though is wether your server has an external IP (e.g. does the internet routing) or there is a router in between (you wrote the server does NAT, but I already saw double- and even triple-NAT configurations - I have to mention that). Both will work, but _not_ having NAT in between might be one trouble source less - so if you run Asterisk on a machine with a globally valid and routable IP, you are better off. Your firewall should accept incoming TCP on port 5060 and incoming UDP on all the ports RTP uses (like 1 to 2) - I rarely bother firewalling incoming UDP packets on high ports, but you should check that. If your phone works behind the router, the UDP requirement probably is already sorted. Basically, you will have to edit a few configuration files. I will give some examples based on one of my asterisk configs, but you really should read about those files and check wether everything is OK - I will try to adapt to your situation, but do not blame me if I mistype or just mis-think something. In sip.conf, you will need to list the providers and the phones you are going to use. I assume you will have your allnet and perhaps a few softphones - you will probably want more than one phone some day ;-) 8 sip.conf (with example data indicated) [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm musicclass=default language=en ; well, no idea if there are czech audio files readily available. ; I personally use language=de, of course. dtmfmode=rfc2833 sipdebug=no register = 1234567:[EMAIL PROTECTED]:5060/004311234567 ; put your sip id (1234567), password (4321) and your ; phone number (004311234567) here register = 123321321:[EMAIL PROTECTED]:5060/12 [sipgateat] host=sipgate.at secret=4321 username=1234567 fromuser=1234567 fromdomain=sipgate.at srvlookup=yes context=sipgateat-in canreinvite=no nat=no ; perhaps this needs to be set to yes ; insecure=very ; perhaps this needs to be activated - try it. type=friend qualify=yes [otherprovider] host=otherprovider.example.org secret=abcd username=123321321 fromuser=123321321 fromdomain=otherprovider.example.org srvlookup=yes context=otherprovider-in canreinvite=no nat=no type=friend qualify=yes ; stanza for SIP clients [sip01] mailbox=01 callerid=11 type=friend username=sip01 secret=LaBananaLoca ; replace with the secret for your telephone, username should ; always be the same as the [stanza] name to avoid trouble context=sipclient host=dynamic nat=yes [sip02] mailbox=01 callerid=12 type=friend username=sip01 secret=AyayayDiosMio context=sipclient host=dynamic nat=yes 8 so much for the sip.conf. This allows for two accounts with providers, and two SIP phones (wether hard- or softphone does not matter, of course :-) You will also need to setup an extensions.conf, somehow like this 8 extensions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no ;; all of those have been like this in my conf for ages, I do not ;; even know what exactly those are good for. ; context where sipclient outgoing calls are handled [sipclient] ; let 11 and 12 be internal numbers exten = 11,1,Dial(SIP/sip01,60) exten = 11,2,Hangup() exten = 12,1,Dial(SIP/sip02,60) exten = 12,2,Hangup() ; Outward calls. If a country prefix is present _and_ it is Austria, ; use sipgate.at exten = _0043.,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _0043.,2,Hangup() ; Outward calls with country prefix for Czech Republic go through ; your other provider exten = _00420.,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _00420.,2,Hangup() ; All other non-international calls go through otherprovider - ; three digit minimum here, shorter numbers treated as internal exten = _0[1-9].,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _0[1-9].,2,Hangup() exten = _[1-9][0-9].,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _[1-9][0-9].,2,Hangup ; add stuff for voicemail call-in here ; context for incoming calls through sipgate [sipgateat-in] exten = 004311234567,1,Dial(SIP/sip01SIP/sip02,60) exten = 004311234567,2,Hangup() [otherprovider-in] exten = 12,1,Dial(SIP/sip01SIP/sip02,60) exten = 12,2,Hangup() 8 This should get you started. This is a very rough example, and I
Re: [asterisk-users] Problems sending more than 2 SMS with asterisk / smsq
Am Mittwoch, den 19.09.2007, 15:25 +0200 schrieb Christoph Adomeit: Hi there, I experience the same problem here with asterisk 1.2.24 on an E1 Line, only 2 of 3 sms are sent, this happens always and is reproducable. Did someone find out more about the problem ? Especially I do not see how I could add a wait to the dialplan as somebody suggested because there seems no dialplan invoked when I send sms. I use: smsq -d 017xxx -m TEST1 --motx-channel=Zap/g1/0193010 (Germen Telekom Message Center) How could I invoke smsq differently to use an own context of the dialplan ? I think I have been in error there. The wait occurs on incoming calls, like my Gigaset calling in to Asterisk, between answering the line: exten = _0193010.,1,Answer() exten = _0193010.,2,Wait(2) exten = _0193010.,3,SMS(blahfasel) Same for _incoming_ messages from the Telco SMSC. I do not immediately understand where I could have inserted a Wait() for outgoing SMS, especially as that SMS() seems to open the line itself. I would have to investigate, but do not have the equipment right here at the moment, sorry. BR Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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
Re: [asterisk-users] VoIP Provider for business
Am Mittwoch, den 19.09.2007, 11:02 +0530 schrieb Jim Boykin: Can someone suggests a good and resonable cost voip provider with business unlimited plan in USA and allows simultaneous outgoing calling. My experience with business unlimited is that they very well know which customer uses more than his share of minutes. Providers that buy minutes in millions probably get good prizes, but they calculate for an average call volume. If you are far above profitability - and you seem to exactly plan that - you will not stay their customer for long. IMO you would better find a VoIP provider with good minute rates - if you can afford it, service level agreements, and good customer management. This might not even be more expensive in the long run, as cheap stays cheap: Problems with a cheapo provider will cost YOUR money. YMMV, of course, and quality can not be always expressed in numbers as easy as (call minute/price) quantity. BR Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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
[asterisk-users] (Getting OT) Re: Call Center SoftPhone with Auto Answer
Am Dienstag, den 18.09.2007, 17:33 -0400 schrieb James FitzGibbon: On 9/18/07, David Gomillion [EMAIL PROTECTED] wrote: I've stayed out of this thread for a long time, and really didn't read the past comments, so if I'm repeating someone, I'm sorry. I've been thinking this for a while, and just have to say it. If you feel like you have to keep people from turning off the auto-answer feature on a softphone, you don't need a new softphone. You need new people. Yes, but have you ever drawn up a budget for a full-blown meatware(tm) upgrade? I do not know American work law, but if you tell your people to NOT turn off auto answer, and they do for having a break, would that not count as work refusal? As long as they get all the breaks they are supposed to take, of course. If for example a cashier in a supermarket here in Germany would just leave her position for a few minutes for a smoke, small-talk or whatever, outside her assigned break times, she could afaik get a written warning, and at the second occasion the full wad of papers (aka been fired). On the other hand, if you count only times while they are on-a-call, with appropriate logging software, adding a few seconds per-call for overhead, as their worktime, they pretty soon will keep auto answer on to get the required number of work minutes during their shift, I would expect. But this is not as much a technical problem as a social one: If your agents are unmotivated, they might spend time talking off-business to any caller/callee on the phone that seems to be interested in small-talk, and _that_ you could hardly find out technically. So you might get an upgrade without paying for the deinstallation of the previous meatware, but the installation process of course has costs. BTW putting too much pressure on your agents might do bad things to their effiency, motivation, even mental health. Getting the balance between control and good atmosphere right is not easy, and something that cannot be generalized but must be tailored to the situation. The value of a human asset (imagine me vomiting my way through those words) can materialize in the number of sales, calls, ... and also in the customer experience he creates, which is hard to be counted in numbers. For example, I recently bought some music instrument and accessories at a phone-order company. The people there were relaxed, friendly, helpful and made the effort of giving me competent, quick information that I needed. All contact with them was extremely positive. As I needed some more stuff that I knew was a bit cheaper at another store (which only deals with customers in a matter-of-fact way), I decided to honour that effort. I also recommended the company to friends, which they probably will never know about and as such cannot count in as a bonus for their sales personell. *Just my loose change. Man, there were lots of coins in that purse.* Makes Vista look like a picnic. IMO Vista is an apple short of one ;-) To get the Asterisk relevant topics: You could - count on-call minutes to rate agent performance - track off-call intervals on a certain line and so track the turned-off auto answer - do some social engineering or policy work to get this sorted in a non-technical way (work contract terms, etc) - pay some softphone manufacturer to implement needed changes BTW what would hinder your agents from shutting down the softphone app when they do not want to answer calls? What would hinder them from just not talking to the caller when they do not want to? Best regards, Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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
Re: [asterisk-users] Filesharing + video + voice supported Soft phone
Am Montag, den 17.09.2007, 05:09 -0700 schrieb satish patel: Dear all I have setup of asterisk 1.4.11 Now i want soft phone which one support file sharring + video + voice call with asterisk SIP is there any soft phone which support this all feature ?? Yes, there is such a soft phone. with asterisk Probably yes. Guess how you could find such a software. You might search google with the following search term: sip-softphone file-transfer video The very first result for me is Messenger - SIP Softphone - Soft Client File transfer. IP Telephone, Do not disturb, busy available status. softclient conferencing, Custom available/away status. video telephony ... www.eyeball.com/products/messenger.html - 22k - Cached - Similar pages and from the decription on the page there, this software does what you want. That was not too difficult, was it? If you wanted to find out if someone on this list uses such a software, and what experience they have, your question probably would have looked differently. Nevertheless, voip-info lists some softphones that are compatible with asterisk, http://www.voip-info.org/wiki-VOIP+Phones some even have information and user experience stories. Eyeball messenger is also listed there, although without further information. Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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
Re: [asterisk-users] Softphone RTP Session Start-up Delay
Am Montag, den 17.09.2007, 15:50 -0400 schrieb [EMAIL PROTECTED]: Hello, I have a small LAN network where I am running a Jain-Sip softphone on two user pc's. These softphones are connected through Asterisk(Trixbox). Although the phones do work in providing an audio conversation, there is a long delay(about 20 seconds) in the initial RTP session setup. I have tried a few values for the buffer length including setting it to zero. I assumed this would drastically reduce the delay but there was no change. I also tried a number of values for the minimum threshold and this did not change the amount of delay either. Would anyone have an idea of why this delay is occurring and possibly how to reduce it? Hello Denis, delays in that magnitude (20 seconds or about) may be related to DNS issues - like trying to resolve a hostname, or trying to find a hostname for an IP address. You could try to add all relevant IPs to the /etc/hosts file (or C:\windows\system32\drivers\etc\hosts), like 192.168.0.2 host2 and see wether that helps. Regards, Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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
Re: [asterisk-users] alphabetical extension patterns
Am Dienstag, den 11.09.2007, 17:11 +0530 schrieb Benjamin Jacob:
Thanks Anselm. This does clears a few things for me.
Tho, I couldnt find the patterns you mentioned in the docs(do point me
to the location if you know of it).
I started on
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
Patterns have to begin with _, meaning it is a pattern. A . stands
for one or more characters, so I only allow three-and-more character
SIP phone numbers like [EMAIL PROTECTED], but not [EMAIL PROTECTED] This
is deliberate: I rather not have catchall-type phone numbers, I already
get enough mail spam on the few catchall-addresses I have (well, for
historical reasons - I once was small and stupid ;)
About multiple domains, that is my target for sure.
I think the domain(in sip.conf) thing should come into help here,
where I associate a domain name to a context. I did try it once,
worked fine for a couple of test domains. But it seems I can't
associate one domain name to multple contexts. Am I correct?
You can specify one context for every domain your asterisk supports. On
one of my machines, a sip.conf might look like
8 sip.conf
[general]
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
domain=main.example.com,sip-in-examplecom
domain=private.example.org,sip-in-privateexampleorg
domain=customer.example.net,sip-in-customerexamplenet
8
So calls coming in for [EMAIL PROTECTED] are going through the
sip.conf context sip-in-examplecom.
In extensions.conf, I would configure like this:
8 extensions.conf
[sip-in-domains]
exten=_...,1,Set(A=${DB(callroute/names/[EMAIL PROTECTED])})
exten=_...,2,GotoIf($[A = A${A}]?900)
exten=_...,3,Goto(localdialplan,${A},1)
[sip-in-examplecom]
exten=_...,1,Set(DOMAIN=example.com)
exten=_...,2,Goto(sip-in-domains,${EXTEN},1)
[sip-in-privateexampleorg]
exten=_...,1,Set(DOMAIN=private.example.org)
exten=_...,2,Goto(sip-in-domains,${EXTEN},1)
[sip-in-customerexamplenet]
exten=_...,1,Set(DOMAIN=customer.example.net)
exten=_...,2,Goto(sip-in-domains,${EXTEN},1)
8
This would require database entries for users like
callroute/names/[EMAIL PROTECTED] = 201
callroute/names/[EMAIL PROTECTED] = 661
You can also have several domains map to the same users, e.g. you want
example.com and main.example.com to be equivalent, so you just add
another domain line to sip.conf, like
domain=example.com,sip-in-examplecom
You should be able to get around this multiple-context setup by using
the variable ${SIPDOMAIN} and only one context, but this somehow did not
work for me, so I came up with this solution. Play around, see if you
get it running. For me, it has been like this for a while, and then, I
try to avoid changing a running system. You could, for example, set all
your domains to
domain=example.net,sip-in-domains
and use
exten=_...,1,Set(A=${DB(callroute/names/[EMAIL PROTECTED])})
which _should_ work just as well.
You probably already found out that SRV records should be set for the
domains that asterisk is going to handle, let me give an example:
[EMAIL PROTECTED]:~$ dig @localhost example.org any
; (1 server found)
;; global options: printcmd
;; Got answer:
;; -HEADER- opcode: QUERY, status: NOERROR, id: 52979
;; flags: qr aa rd; QUERY: 1, ANSWER: 6, AUTHORITY: 0, ADDITIONAL: 1
;; WARNING: recursion requested but not available
;; QUESTION SECTION:
;example.org.IN ANY
;; ANSWER SECTION:
example-org. 604800 IN SOA ns1.example.net. root.example.org.
2007060504 21600 3600 1209600 21600
example.org. 604800 IN TXT v=spf1 mx a:mxs.example.org -all
example.org. 604800 IN MX 10 example.org.
example.org. 604800 IN A 81.12.999.999
example.org. 604800 IN NS ns1.example.net.
example.org. 604800 IN NS al25b.xi.yu.fiber.example.com.
example.org. 604800 IN NAPTR 60 50 s SIP+D2U
_sip._udp.example.org.
;; ADDITIONAL SECTION:
ns1.example.net.604800 IN A 81.12.999.999
;; Query time: 5 msec
;; SERVER: 127.0.0.1#53(127.0.0.1)
;; WHEN: Sat Sep 15 11:38:14 2007
;; MSG SIZE rcvd: 269
Where
_sip._udp.example.org. 604800 IN SRV 10 10 5060 example.org.
This is a setup with all web, mail and sip running on the same machine
(IP addresses and domains changed, of course) - but you should be able
to move things around so that those services actually can be run on
different machines.
Anything other to be done on Asterisk to support multiple domains?
Well, I think that is about enough ;-)
BR
Anselm
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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
Re: [asterisk-users] Prevent multiple sip registrations
Am Dienstag, den 11.09.2007, 19:09 +0500 schrieb Rizwan Hisham: The whole point of doing this is because if the user gives away his username/password to his friends or relative and allows them to use his account, that way we r gona have a lot more traffic in our asterisk server. Also we charge our users a fix amount of money every month for their account so if any user gives out his username and password then his account is more likely to do 2 to 3 times the calls as compared to aan account which is used by only one user. So ultimately we lose money. Dear Rizwan, imagine one of your customers uses asterisk. His asterisk server registers to your server, and he manages his own local dialplan to have 250 SIP devices using the one SIP account. (I think Asterisk can be told to send a UserAgent ID other than the default Asterisk whatever - you will not easily find out *reliably* wether someone is an Asterisk user or not) Are you screwed? Well, probably. You cannot outsmart some people if you give them the liberty to play tricks on you. If you want to go secure, buy the hardware they are going to use, register all the SIP stuff into that hardware and make sure it cannot be read-out easily (most SIP phones will not allow to read the password that was previously entered, although some web-interfaces still contain the old password in the HTML page source). Your customers will hate you... My personal approach would be to not bother with registrations but log the IP addresses from which their phones register. If - over a busy telephone day - the log shows a pattern like 123.45.67.89 - 11:15h 131.66.14.56 - 11:27h 123.45.67.89 - 11:58h 131.66.14.56 - 12:44h 123.45.67.89 - 14:05h 131.66.14.56 - 14:09h 123.45.67.89 - 14:32h then you could still call the user and tell him to buy another account - your contracts probably explicitely restrict usage to a single person, right? Even more, your contracts _could_ contain clauses like for private users only, and the option for immediate termination on your part if any doubts on that arise (users tend to hate those statements as well). Anyone having more than 400 outgoing minutes in more than 50 calls (insert other numbers to your liking) on a day, or more than 7000 outgoing hours in more than 1000 calls in a month might attract your special attention. You could have some log analysis to find power users. Just an idea popping up: AFAIK you _can_ restrict asterisk SIP easily to not more than one concurrent call for any account - and you probably should with your business model. How about, once they trigger a certain number of minutes threshold on their account (perhaps 2000 minutes during the last 7*24 hours), preceding any outgoing call they make with a short announcement like *bling* your_telco_name Please be aware this account is for private use only. Call customer service to get more information *blong*? At least this would sever re-selling of your services - and legitimate users would in 99.99% of cases never hear that announcement. I know some SIP providers always send out CALLERID, not to be suppressed, so those flat tarrifs are also less interesting for resale. Some customers (like me) prefer being able to set that CALLERID, on the other hand. And I surely do not abuse the tariffs I contracted for. Whatever your system looks like in the end, that would of course be interesting to me. On the other hand I can only advise you to not publish the exact numbers, triggers and restrictions - for obvious reasons. Finally it all boils down to you offer a flat fee, you suffer. Try to attract customers that use less minutes than you calculated your tariff for. Try make it attractive for the use it is intended for, and less attractive for (irregular) power-users, re-sellers or call-center-like businesses. Try to not irritate your users by unpopular, stupid restrictions. If the world were just a better place, sometimes... Just my 3 pence, Anselm (just being returned from holidays in Kent, still in relaxed mode) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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
Re: [asterisk-users] Strange Behaviour
Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita: Hi, my ATA has two phones attached and the possibility to set different accounts. I put two account of my asterisk server, however, it is able to call only with the second one in order to the sip.conf and the first it gives me 403. And idea how to solve it? Well, it seems there are differences between those accounts then. You might want to post your sip.conf, and -if that is possible- the ATA conf file; or at least a writedown of the configuration there. If those are not the source of trouble, _I_ probably would switch the accounts in the ATA (port A versus port B) and try if the problem sticks with the port or with the account. I would also google if there are known problems with my ATA, look if a newer firmware is available, if there are informative messages that are worth a verbatim quote, and get another bottle of beer to keep the sunday relaxation at a proper level. BR Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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
Re: [asterisk-users] Strange Behaviour
Am Montag, den 10.09.2007, 05:14 +0200 schrieb Il Neofita: On 9/9/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita: Well, it seems there are differences between those accounts then. You might want to post your sip.conf, and -if that is possible- the ATA conf file; or at least a writedown of the configuration there. First of all, thank you for you reply The ATA is the Fritz!Box and I tried with different FW version but I have the same behaviour I have been using FritzBoxes for quite a while, and have not found such strange bugs - except after a Firmware Upgrade. It seems after some upgrades you need to do a factory reset (via the web interface) and enter your data again, else they behave stupidly. this is part of the sip.conf [180] type=peer username=180 secret=aa callerid=First180 canreinvite = yes host = dynamic dtmfmode = rfc2833 qualify = yes nat = yes context = mycont disallow = all allow = g726 allow = g723 allow = ulaw allow = alaw allow = g729 allow = gsm [181] type=peer username=181 secret=bb callerid=Second181 canreinvite = yes host = dynamic dtmfmode = rfc2833 qualify = yes nat = yes context = mycont disallow = all allow = g726 allow = g723 allow = ulaw allow = alaw allow = g729 allow = gsm Looks pretty OK to me. Just a stupid idea: Do you have a [general] section before those two? And then, I use type=friend, not type=peer, that _might_ make a difference in how asterisk matches sip.conf contexts to registered clients. 8 From my sip.conf: [sip501] mailbox=01 callerid=501 type=friend username=sip501 secret=lk1j2eu89 context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm allow=ulaw [sip502] mailbox=02 callerid=502 type=friend username=sip502 secret=1092jd0 context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm allow=ulaw =8 Note: Those two accounts belong to the same FritzBox. I tried to switch the account for the two ports but what it is important is only the order in the sip.conf That made me think about that friend/peer thingy. I found some information in german and I do not know it The FritzBoxes are popular here in Germany - no wonder, being a German manufactured product and being given away for (nearly) free with any 2-year DSL contract... I like them nevertheless :) BR, HTH Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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
Re: [asterisk-users] FAX machine connect with audiocode SIP device
Am Mittwoch, den 05.09.2007, 22:58 -0700 schrieb satish patel: Dear all I have FAX machine connected with audiocode SIP device i am trying to send fax and when negosiation going on and i start send fax button then my after half page it got stuck in fax machine so is there any codec problem i am useing ulaw/alaw is it fine or not anybody have idea about sending fax with SIP connected device Satish, you already asked twice about fax and asterisk. As far as I can see, no-one answered those questions. Think why that may be: - Because asterisk and fax have been debated often enough? - Because people expect from you to use google instead of pester the mailing list with questions already answered on the web? - Because your mails do not leave the impression that you really tried to achieve things by yourself _first_ and then come answering with a visible amount of experience (indicated by what you tried, why that did not help, and so on)? For your viewing pleasure there are texts about posting questions in mailing lists, like http://www.eyrie.org/~eagle/faqs/questions.html http://perl.plover.com/Questions.html I try not to be overly sarcastic or malevolent, but I could not resist to write this mail. Hope it helps. Anselm PS: Try http://www.google.com/search?q=asterisk+sip+faxbtnI=go ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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
Re: [asterisk-users] alphabetical extension patterns
Am Donnerstag, den 06.09.2007, 10:16 +0530 schrieb Benjamin Jacob:
Hello ppl,
Any way to specify alphabetical exten patterns in the dialplans on Asterisk?
All my users would have alpha/numerical ids. I don't want to add a line
for every user in my dialplans.
I searched around, but couldn't get anything useful. Any way to get
around this?
As from the docs, you can use letters in brackets, like
exten = _[ABC][DEF].,.
From my config I will give you an example of using names for extensions.
In my case, this is only used for incoming external SIP calls, so that
the extensions on my asterisk can be dialled as sip:[EMAIL PROTECTED]
from the internet.
Regular internal extensions are defined in my context [localdialplan],
my Asterisk DB contains several lines like
callroute/names/anselm = 201
callroute/names/flo = 212
8=== extensions.conf
;* Look up exten in database
exten = _...,5,Set(A=${DB(callroute/names/${EXTEN})})
exten = _...,6,GotoIf($[A = A${A}]?900)
exten = _...,7,Goto(localdialplan,${A},1)
exten = _...,900,Congestion()
===8
(you'd need a bit more intelligence for more than one domain, but I
guess that is not what you think of right now)
HTH
Anselm
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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
Re: [asterisk-users] asterisk voicemail to email and relaying
Am Donnerstag, den 06.09.2007, 02:07 -0600 schrieb Al lists: Hi list, I'm trying to get some ideas on this subject. Normally astersik sends emails with voicemail attached trough local MTA. As far as i know there is no way for asterisk to authenticate to an external mailserver to relay these emails. Well, these days every provider has some sort of spam blocking, to add to that usually users of asterisk are behid a dynamic IP with no PTR and list grows depending on what target mail server requirements are. Base on these facts i came to conclusion of setting up local MTA to relay emails trough another mail server (another mail server beeing their ISP mail server), i dont have very good results with sendmail/procmail and SASL, its inconsitance, works with some provider not all... I was wonderin what do you guys use for your asterisk boxes? I have good experience with exim4, the default config needs some tweaking (at least under Debian) for SSL and AUTH stuff, but that is fairly documented and not difficult to setup. I only have one upstream provider, a so-called smarthost, so I need not fear it will break with any other mail host. YMMV. Of course running exim4 only for mail-forwarding is a bit like hunting sparrows with cannons (or whatever the equivalent english phrase is :-) but then, it gets the job done, and without any mail in the queue its memory footprint and cpu usage are neglible. BR Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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
Re: [asterisk-users] unnumbered priorities
Am Sonntag, den 02.09.2007, 23:25 -0700 schrieb fateme fatah: Hi: When should we use unnumbered priorities(n) in extensions.What is the different between these 2 forms of extensions.conf? and ,Are both true? extensions.conf: form1: [Conferencerooms] exten = 333,1,Answer exten = 333,n,meetme(8000|cim) exten = 333,n,playback(vm-goodbye) exten = 333,n,hangup form2: [Conferencerooms] exten = 333,1,Answer exten = 333,2,meetme(8000|cim) exten = 333,3,playback(vm-goodbye) exten = 333,4,hangup On one of my Asterisk 1.2 setups, I have a dialplan that mixes both - so they can coexist in the same extensions.conf. The difference is that with the n type extensions, you can easily insert a line or three without renumbering lots of lines - and searching for all those GOTOs that also need a new line number. Renumbering error-prone. An advantage of numbering is that the line order is not important, because of course Asterisk would select by number, not order - and possibly (although I did not investigate this) including _parts_ of an extension from another context might work better. All my new extensions use the n style, but I am not going to rewrite the older parts of the dialplan soon. BR Anselm ___ --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] voip provider settings problem, please help
Am Montag, den 27.08.2007, 08:55 -0400 schrieb Jody Gugelhupf: hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when picked up, but after a while it stops working, incoming calls for this provider are not shown anymore in the CLI, but from other providers it always works, but the phone is ringingn nevertheless when calling my skypho account...when i then turn off the pbx and restart after sumthing like 2 hours my skypho account is working fine again, the incmiong calls are shown in the asterisk CLI, but after, i don't know let's say an hour or so it again stops working, incoming calls for my skypho account can not be seen in the asterisk CLI, then if i turn off the pbx for an hour or so it works again, so i thought it must be a setting issue, maybe something with the register? althought it always shows it registered when i use 'sip show registry' someone has an idea what i have to set or do to have it working permanently? what could be the problem here? i got no clue whatsoever and i have been using asterisk only since half a year, please help me, i'm totaly desperate, thx in advance jody :) Jody, you could post the relevant parts of your sip.conf here. For me (with a similar problem) introducing qualify=yes to the provider context in sip.conf solved the problem about 99.9% of the time; about three times a week I am off for less than 5 minutes at one particular providers - others work fine (I have a cronjob checking asterisk -rx sip show registry | grep 022396whatever which reports if status is NOT Registered - it does not do anything if the peer is not registered except sending me a notifier mail, so I have some kind of tracking). I am not familiar with italian voiceone though. Best, Anselm ___ --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] Setting caller ID on outgoing calls.
Am Montag, den 20.08.2007, 13:57 -0400 schrieb Joe acquisto: Excuse me if I recently posted on this, but I cannot find it, in my, or the list archives. Is it possible, when transferring a call, to set the user ID to that of the outgoing number instead of the incoming number? I believe the answer is (was) yes. I seem to remember there is an option to the Dial command, possibly o. Check on the voip-info.org wiki. Maybe this is not true for 1.4 anymore, no idea. There seems to be a difference between transfer with or without speaking to the callee first, and most probably transfers made from the phone features (i.e. transfer button on some phones) will not allow to send the original caller ID. New twist, does it matter what the destination media is? Meaning, the call would be coming in on a T1, going out on a T1, but ending on a POTS line (which supports caller ID). Not the media(SIP/T1) is the problem for outgoing caller ID, but the provider/carrier. For SIP, Zap... devices connected to your asterisk as their server, you can of course send any callerid you want. As soon as you have to hand over the call to any provider, be it SIP, T1, ISDN, IAX,... you have to check wether they allow to set any ID, to only set your own IDs (which are assigned to the outgoing line as incoming numbers) or if they allow doing any CALLERID changing at all. Some providers do not even allow to block the own number, outgoing (for example, afaik, 11 SIP in Germany), others allow to set only numbers assigned to the phone line in question or block calls (without special agreements this seems to be the standard setup for PRI, ISDN and analogue lines in Germany) - others allow to send any number if it is valid (where valid means the provider's idea of a phone number, it seems) - sipgate.de seems to allow to set nearly anything starting with +49. From what I learned about North American whereabouts the situation seems to be similar - business providers seem to be willing to offer more options to their customers, check with your providers. BR HTH Anselm ___ --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] How strip +1 from caller id on inbound call
Am Dienstag, den 14.08.2007, 09:06 -0500 schrieb Brandon Kruse:
I just use
exten = +12564286115,1,Goto(${EXTEN:1})
exten = 12564286115,1,noop(It worked.)
I believe that should work
That rewrites the callee number, not the CALLERID, so no, it would not
work for Todd's original problem.
BR
Anselm
___
--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] 99 bottles of beer
Am Donnerstag, den 16.08.2007, 12:08 +0100 schrieb Gordon Henderson: On Thu, 16 Aug 2007, Diego Iastrubni wrote: DUD! THIS KICKS ASS! (I know I am getting into trouble, but hey! it's already in our PBX!) Heh... Well I updated it and added some lyrics (and the guys from the website have said they'd put it up!) So if you want to hear a (rather odd!) mix of me Allison, then dial +44 1364 698 225. I started it at 3 as you don't want to hang about all day, I'm sure :) Somehow it kicked me out after the second time two is mentioned... Whatever, now I know what a All-Europe-Landline flatrate is good for ;-) BR Anselm ___ --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] CallerID Error causes problems for Polycom phones
Am Mittwoch, den 15.08.2007, 10:14 -0400 schrieb Lee Jenkins: Hi everyone, I have been dealing with a certain issue with a particular customer site for months now. The problem occurs when there is an error with caller id as shown in the following: WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error on channel 'Zap/3-1' When this happens, it appears that the call still goes through as I can see the caller still navigating through the systems menus and dialplan by watching the CLI. The problem however is manifested with polycom 301's that are setup with the system. When a call comes in after receiving that particular caller id error, the polycoms, which are on a group ring by the way, will all ring but you cannot pickup the call. The Answer|Reject soft buttons display, but only the reject button works. Pressing the Answer button or picking up the handset does nothing. To me this looks like a firmware problem in your phones. Perhaps a firmware update could fix this. However - as it looks to me - the firmware chokes on some CALLERID strings, not on others. What is the caller id that is displayed in the error case? Perhaps you could get around by having a dialplan hook that rewrites the callerid to 000 if that invalid callerid comes in. Maybe those phones just choke on CALLERIDs with empty num or name With your test .call file that reproduces the problem, if you insert a line in your dialplan before the Dial() happens, that reads Set(CALLERID(all)=000) does that help? Does Set(CALLERID(num)=000) alone help, does Set(CALLERID(name)=000) ? BR Anselm ___ --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] How strip +1 from caller id on inbound call
Am Sonntag, den 12.08.2007, 21:16 -0400 schrieb C F:
you can do like this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than grab the last 10 digits of the CIDNUM
exten =
_X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):$[${LEN(${CALLERID(num)})}-10]});this
grabs the last 10 digits of CALLERID(num) and sets it to CALLERID(num)
exten = _X.,n,Return()
Argh! You do not ever get international calls, do you? (Well, Canada
does not count here for obvious reasons)
The clean solution to the question
I get some calls with a leading +1. If that is the case, how do I
strip that off?
is of course
If the CALLERID(num) starts +1, re-set it to the same value, offset 2:
...
exten = _X.,n,GoSubIf($[${CALLERID(num):0:2} = +1]?strip1)
...
exten = _X.,n(strip1),Set(CALLERID(num)=${CALLERID(num):2})
exten = _X.,n,Return()
Which leaves international calls for themselves. Of course you still
could replace the leading + for all other numbers by 011, if you
like.
Your code would probably handle
+12125551212
correctly, would work OK with
+495924236
(which might or might not be one of the old, short numbers still present
in some places in Germany), leaving it intact, but not with
+4916177554224
which would be remapped to a Boston MA number (actually a Cingular cell
phone number) instead of mapping it to a german mobile phone.
Variable handling (offset et al) is documented on
http://www.voip-info.org/wiki/view/Asterisk+variables
BR
Anselm
___
--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] Dialplan loop
Am Donnerstag, den 09.08.2007, 20:12 -0500 schrieb David Bandel:
Folks,
I'm trying to implement a simple loop in a dialplan. The object is to
set a counter, run through some IVR options, increment the counter,
return to the start, then finally fall through to an operator or
voicemail.
exten = s,n,Set(loop = 0)
...
exten = s,n,Set(loop = $[${loop} + 1])
The above loop increment doesn't work. The error message is:
WARNING[14490]: ast_expr2.fl:398 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '+', expecting $end; Input:
+ 1
^
WARNING[14490]: ast_expr2.fl:402 ast_yyerror: If you have questions,
please refer to doc/channelvariables.txt in the asterisk source.
Try removing extra space characters around the =. Very similar example
from my dialplan
exten = _2XX,n,Set(I=1)
...
exten = _2XX,n,Set(EXTR=$[${I} + 1])
Works fine. Also assigning a variable a new value based on the old value
works OK here (although not calculated, but concatenated):
exten = _2XX,n,Set(D=${D}SIP/sip501)
I am using Asterisk 1.2 here, but I remember similar errors with stray
characters.
BR
Anselm
___
--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] Free sitting
Am Freitag, den 10.08.2007, 09:02 +0200 schrieb Olivier: Hi, My question is more what should be done than how should it be done. I could say : If you were a teacher, teaching and preparing your courses once a week (as you can't be called while teaching, can you ?) Well, yes. It always depends ;-) In an English or Arts course you could probably answer the phone to internal calls - those calling you will know you are in class and keep it as short as possible and just call instead of knocking on the door, which probably disturbs pretty much to the same amount. Getting external calls should then be turned off, or silent-ringer with a display showing external call and the send to voicemail button available. I assume that answering the phone while teaching the usage of circular saw and all those tools in a woodworks course or while teaching martial arts would be a bit too disturbing to make it happen ;-) would you prefer your phone system to log you in or out 1- automatically according a schedule stored somewhere, 2- whenever you turn your PC or or off, 3- when you dial something (for login) and logout) is done during nightimes, 4- when you dial something (for login and logout). 3/ and 4/ are compatible. You could further reason wether a user shall be logged out when the next one logs in. Logging the user out from a place when he logs in somewhere else is also reasonable (as you write below). Those two are even compatible with 2/ if only the login procedure shall login the phone, or only with 4/ if the logout is also coupled to the phone. My vote would go for the last one as it somehow keeps users responsible for themselves. A colleague prefers the third choice. Which would you pick ? If someone logs in from one place and logs in once again from somewhere else, then user previous log shall be replaced by the new one : incoming calls rings new phone. I'm wondering whether or not, 2 people could share the same phone but beside calling features, many supporting features such as MWI, BLF wouldn't it easily. Right. This depends on wether it will be a very seldom or a common case. Example a: There is a teachers' room where they usually sit in their non-teaching time and prepare lessons. Every place has (possibly a computer and a) phone. Example b: The same room has only one phone. Thinking about the computer coupling, that probably also depends on wether they regularly use the PC (all the time, part of the time, sometimes...) What do you think ? I would go for a combination of your 3/ and 4/ settings above. Allow them to logout, and if they do not, autologout after 3 hours or so (teachers probably not too often stay within the same room for more than three hours) or whenever they logout manually. You could combine that someone (you) is logged into this phone with a lamp on the phone (although you probably need a patch to asterisk to support non-regular presence/status settings) - perhaps making that lamp blink for 15 minutes before auto-logout, or depending on the number of states that the phone supports, signal message-waiting or one of about 1000 others things. You could also designate conference room phones such that multiple users can be logged in (without MWI and further features) while teacher's room phones and classroom phones could be strictly single-user and therefore offer extended features. Depending on the phone it can display both CALLERID(num) and CALLERID(name). You could tweak that to change CALLERID(name) to for Mr. Peters, for example, so that the display will tell both the caller number and the callee name. With 1000 more options of course. Users often lack the ability to know what they want and precisely be able to tell that. Asking them about their usage habits, with well formulated questions, might reveal which of the methods is best for your setting. I am not a teacher, but have lots of them in the family, so I know that between schools there are huge differences in work habits and so on. As an external consultant you will have to ask those who will (have to) use the system you design. A friend of mine says, Linux is all about choice. Same here for asterisk, and you are the one to choose. Best regards, Anselm ___ --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 Manager to Record Greetings
Am Freitag, den 10.08.2007, 11:26 -0500 schrieb Peder @ NetworkOblivion:
That's great, now say you have 5 or 6 AA's and each one has 10 different
parts that you want to record (thank you for calling... for Steve
press 1 for dave press 2). Rather than having to record a long
message, I want to break it into pieces so that if dave leaves, we can
just record that one chunk rather than the whole thing. I would need
lots of extensions pre-setup for each chunk. Not very efficient.
Gordon Henderson wrote:
On Fri, 10 Aug 2007, Peder @ NetworkOblivion wrote:
I am trying to use Asterisk Manager via php to record auto attendant
greetings and I just can't figure out how to do it. I've got the php
page working and I can click to call between two phones. However if I
click to call just a single phone and then try to enable monitor, when
I pick up the ringing phone, it just hangs up and doesn't record
anything. I'm sure I just don't know the appropriate syntax. Has
anybody done something like this? I can do the php stuff, I just need
the Asterisk Manager syntax.
I did something similar using multiple records in a row.
Something like
exten = 931,1,Answer()
exten = 931,2,Wait(2)
exten = 931,3,Set(E=1000)
exten = 931,4,Playback(beep)
exten = 931,5,Set(E=$[${E} + 1])
exten = 931,6,Record(/tmp/asterisk-recording-${E:1})
exten = 931,7,Playback(/tmp/asterisk-recording-${E:1})
exten = 931,8,Wait(2)
exten = 931,9,Goto(4)
This will loop: beep, record until # pressed, replay, wait, beep...
The files will be written with ascending numbers starting 001. Move
them to another place before doing the next recording session.
HTH
Anselm
___
--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] FSK callerid
Am Mittwoch, den 08.08.2007, 23:55 +0900 schrieb Balgansuren Batsukh: Hello, I installed Asterisk on Dell Precision workstation and configured with sample configuration. I have two TDM400 board and one with 4xFXO and second one 4xFXS module installed. I made call to telephone line connected to FXO port and never seen callerid on those lines. I tested cidsignalling and cidstart types and all doesn't work. Just a guess. Try a Wait(2) in the dialplan before Answer()ing the line (or doing anything else). The CID might be sent in or after the first ring... so if you immediately answer the line is already up and no CID can be read from it. BR Anselm ___ --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] Free sitting
Am Dienstag, den 07.08.2007, 07:47 +0200 schrieb Olivier: So no proper logoff between logins, right ? As I will apply free sitting in school environment, chances are phones would then remain logged-in several hours or days between another user logs in. My thoughts are focused on finding the right balance between cost control and ease of use requirements. Maybe, we should program something like 3 states logins : - normal status : user receives call or can call cheap destinations - enhanced status : user can call expensive destinations - logged off status : no incoming calls Downgrading from enhanced to normal status is automatic : if a teacher is working during off hours, he will still receive incoming calls even after being downgraded to normal status. To elevate to enhanced status, you just have to enter your PIN code. What do you think of this ? has anyone tried something approaching ? This somehow reminds me of how sudo works: For the first time you want to run a root command, you have to enter your password. After that, the password will stick (not be asked again) for a few minutes. You surely could put together something like that (time based): The first time you want to place an expensive call, enter your pin: The phone will be granted access for this call +15 minutes, and every next usage of the phone (incoming or outgoing) appends additional time. Same for follow-me function: Keep the person logged in for incoming calls for 90 minutes after the last time he used the phone, or until he logs out. I would probably implement in like that in an environment like a school office, where people share desks: They still _can_ logout, but there will not be much harm if they do not. An intelligent system could also couple the login to the logout of the previous teacher (if that is reasonable in that environment), and auto-login a teacher to the phone adjacent to the PC standing on the desk... BR Anselm ___ --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] TAE to RJ11 connector (hope not OT)
Am Montag, den 06.08.2007, 18:09 +0200 schrieb gincantalupo: Hi, I'm trying to use a Detewe TA 33-clip but there is no rj11 connector on it...only a TAE connector. I'd like to create an adapter so I need to know which TAE pins to connect to RJ 11 pins. Is there anybody who knows where I can find a schema of that adapter? Single connector pinout may help too. Have a look at http://de.wikipedia.org/wiki/TAE You need the La and Lb wires. They are usually top-left and middle-left, watching the device from the plug side. BR Anselm ___ --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] OT - Callto:// tags inside web pages
Am Dienstag, den 07.08.2007, 16:51 +0200 schrieb Olivier: Hi, Where can I find relevant information concerning callto:// tags ? Is it standardized or browser specific ? How within your browser, can you specify the software and parameters to used when clicking on such callto:// tags ? I couldn't find much googling or reading Preferences tab in Firefox. AFAIK for SIP the sip: protocol would be what you want, callto: is the Skype idea of phone URIs IIRC. In windows you can assign protocol handlers for protocols like sip:, some softphones will do that automatically. Works pretty much the same like file type associations. BR Anselm ___ --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] multiple pbxes, multiple domains, same user ids?
Am Mittwoch, den 01.08.2007, 16:32 +0530 schrieb Benjamin Jacob: Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the domain field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with the same specified in sip.conf). But, Asterisk maintains the registration of the latest registree!! thats really sad for me . Any work around for this one(multiple pbx)? I would be zapped and amazed if multiple pbx isn't possible in Asterisk. Help anyone? If multiple domains means you want to register SIP phones with the usernames sip501 at domain1 and sip501 at domain2, that in my experience will not work out this way, because for registered users only the peer name is relevant (corrections welcome, but it seems like that to me). What you could do of course is name the peers reasonably: customera-501, customerb-501 On the first thought, this is not as elegant, but on the other hand, if the phone displays the username, it is better than displaying sip-501. You would need to have some magic to distinguish between your domains in the dialplan. There is a static way of doing it (by setting the context=blah in the sip peers) or a dynamic way, by giving them all into the same context, and then do some Asterisk DB magic to make out which internal partner to reach if 581 is dialled, or which trunk line to use, or whom to bill calls to. This is absolutely possible, without the customers noticing. If you want to support incoming SIP as in sip:[EMAIL PROTECTED], for different domains, you can specify that in sip.conf. In my experience (again, I am ready to learn there are better ways) the best working thing is having a separate domain name for registrations (to get things easily separated), like register.yourcompany.domain, with a line domain=register.yourcompany.domain and for all further domains have separate contexts, like domain-examplecom and domain-exampleorg, looking like domain=example.com,domain-examplecom domain=example.org,domain-exampleorg and in extensions.conf, you could go like [domain-examplecom] exten = secretary,1,Dial(SIP/customera-505) exten = bigboss,1,Dial(SIP/customera-500) [domain-exampleorg] exten = secretary,1,Dial(SIP/customerb-555) exten = sales,1,Dial(SIP/customerb-514SIP/customerb-519) HTH Anselm ___ --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] 1and1 dedicated servers have been down for a few hours .
Am Dienstag, den 31.07.2007, 07:39 -0500 schrieb Asterisk guy: 1and1 dedicated server's service has been down for a few hours , unable to reach them by phone or email. do anyone know what is going on there ? There were rumours they had trouble with an outdated version of the web administration tools (Confixx?) which had a security flaw - and had not been updated by their customers. This security flaw has been used by hackers to gain access and do all sorts of evil things, so -afaik- some customer servers had to be shut down. I could imagine they are just buried in user help requests :-) BR Anselm ___ --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] software bloat - is this really useful to anyone?
Am Montag, den 30.07.2007, 14:29 -0700 schrieb Lee Howard: http://www.asterisk.org/node/48327 I mean, really... you're kidding me, right? It is not at all April 1st... however, I see the point in having a simple demo app. Wether you call it helloworld or hellomarc, the difference is not too large, right? For me having a working example codebase has been a helpful guide in programming modules for existing applications (not Asterisk yet, though). You carve out Marco and put in your own code, there you are. Besides that and an overly eager Mr. Smithers programmer, I do not see a point ;) BR Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound caller ID
Am Montag, den 30.07.2007, 05:24 -0700 schrieb Vieri: Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) 1) No 2) Depends. In some ISDN networks you can pay for an additional feature CLI not screened or similar, which means the number sent will not be corrected by the telco switching equipment if an invalid number is sent. AFAIK standard ISDN lines do not allow to send a number that is not connected to the line in question. 3) AFAIK same for ISDN PRIs. 4) Depends. Some allow sending any number, some always send your number and do not even allow to bar the number. sipgate.de for example allow to send any German regular number (or any number that looks like a valid number), but blocks special (0800, 0900, 112) numbers. BR Anselm ___ --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] What is the best softphone work with Asterisk
Am Mittwoch, den 25.07.2007, 12:13 -0700 schrieb bilal ghayyad: Hi BaharatSamaria; Thanks for your kindly email. Are (Xlite and phoner) IAX or SIP? From where I can download them (Xlite and phoner)? I googled for xlite. One of the first matches was a wiki page on voip-info.org, which in turn linked me to the X-Lite manufacturer's homepage. quote CounterPath's X-Lite 3.0 is the market's leading free SIP based softphone available for download. /quote. The first link in the google search list for phoner immediately led me to the phoner homepage, quote - VoIP support for SIP connections Phoner is freeware, so this program can be used and distributed without any restrictions. Distribution has to be free of charge. /quote I think you will have no trouble to find the URIs yourself, probably within about 30 seconds. In doubt you might consult http://www.googleguide.com/ to learn about google. Anselm ___ --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] SNOM vs. SNOM INDIA (was: phone directory with asterisk)
Am Dienstag, den 24.07.2007, 11:26 -0500 schrieb John Faubion: To prevent further missunderstanding please do not refer the SI-120 as a snom phone. If you need support please contact snom India. Tim, If it is sold by snom India, and one is to contact snom India, I can certainly see how one could infer that it is indeed a snom phone. John The trouble with this distinction seems to be that snom india sells phones as the snom 300 and similars that seem to be the same that people know as snom phones in Europe. The SI phones, as far as I can tell, are a separate line of products for entry-level users, probably rather cheap and probably not developed by the European snom AG. Details can be found on http://www.snomindia.com/ So there are phones that are both snom and snom india and others that are India-Only and sold by the model names SI-90 and SI-120. For simplicity sake, I myself will try to refer to the European snom phones as snom 300 series or similar, and -if applicable- call the snom India products (SI-90, SI-120) by their model names. This misunderstanding made me learn that there are not only highly sophisticated, want-to-have snom devices out there, but also phones that probably compete with the Budgetone 200 and Allnet ALL7950 (both of which I have and know their limits). To add something to the original topic though: The voice-/DTMF-based directory() app should work with any SIP (soft)phone or Zap devices. You will probably have no influence to the SI-90 phone display though for delivering a text-based phone number lookup feature. If someone wants this, he or she should be ready to buy one of the more expensive Snom 300 series devices. Best regards, Anselm ___ --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] phone directory with asterisk
Am Montag, den 23.07.2007, 06:44 -0700 schrieb satish patel: Dear all I have configure asterisk with 100 SIP PHONE ( SNOM ) but now thing is that my boss need phonebook feature find extention number by Pbook so i have read about it there is a feature in asterisk but it is with voicemail now i have IP SIP phone of SNOM so how to fine phone number by SIP phone ?? how to asterisk directory work ? As far as I know the popular asterisk phonebook solution (Directory) works by calling an extension and punching in the first letters of the name (calling me, punching 463... for Hoffmeister, for example) and makes use of the information in voicemail.conf. Some SNOM phones have a micro browser, it seems you can use it for phonebook display. Read (way down) http://www.voip-info.org/wiki/view/Asterisk+phone+snom and perhaps the manufacturer homepage for details. Not been there, not done that ;-) BR Anselm ___ --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] Upgrade Procedure
Am Montag, den 23.07.2007, 16:21 -0400 schrieb Michael J. Liberatore: I noticed in 1.4.x I can no longer use n+101 ? I use this all over my dial plan and wouldn't even know how to replace it. Like when trying to call out and a channel is busy, would I need to do all if then's??? How can I upgrade and keep n+101? Please read the documentation, for example at http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial (other commands can be found linked from http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of +application+commands ) There is an additional option you will have to set to the Dial() to restore the jump to n+101 behaviour, named j. So you would for example change exten=123,4,Dial(SIP/sip123,30,w) to exten=123,4,Dial(SIP/sip123,30,jw) Other commands may also feature such an option, if appropriate - should be found easily in voip-info. I _think_ there is also a kind of global option to restore the n+101 behaviour for the entire dialplan (instead of defaulting to setting variables), actually http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+General might be your best friend there. HTH Anselm ___ --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] Dialplan
Am Montag, den 23.07.2007, 14:33 -0400 schrieb Matt:
Hi,
What dialplan option do I need to send a call out like this:
NPA-NXX- local calls
1-NPA-NXX- - long distance
Won't 'national' send it out NPA-NXX- no matter if it's long
distance or not?
I do not understand your point here. If the user dials 1-212-5551212,
you could send out exactly that string, as in
exten = _1NX,1,Dial(SIP/[EMAIL PROTECTED])
and if she dials 617-1234567, similarly.
Or do you wish Asterisk to magically remove the leading 1, but only
for two or three area codes, because in that case the calls will be
charged as local calls? In that case, you might require your users to
_always_ dial the leading 1 and get away with something like
exten = _1617XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _1857XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _1NX,1,Dial(SIP/[EMAIL PROTECTED])
(assuming 617 and 857 are local area codes)
ymmv, and the documentation about pattern in dialplans
http://www.voip-info.org/wiki/index.php?page=Asterisk+config
+extensions.conf
should be the next text you read, probably.
If this is not what you want, please describe your idea.
BR
Anselm (who never owned a landline in the NANP...)
___
--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] USB Cordless
Am Montag, den 16.07.2007, 09:44 -0500 schrieb Jeremy Mann:
Does anyone know if X-Ten or SJPhone support multiple cordless
handsets for multiple lines? I have an office with multiple roaming
users(nurses) that are in and out. I’d like to provide them
telephones, and my idea is to have a PC sitting in a corner somewhere
running a softphone client. When a nurse comes in she just picks up
any available handset(anywhere from 2-5 per office) and starts
calling. Each handset would be labeled with their extension so that
if any inbound calls came to them they’d be able to let the
receptionist know their extension.
Any ideas?
I personally would prefer giving them real phones, be that a
combination DECT/ATA or WLAN phones. If you shop around, DECT/ATA will
probably be the less steep pricing. In the long run this would probably
be easier to keep running. If you talk 4 phones, you might calculate 2
ATAs of 2 ports each, plus 4 DECT thingies, summing up about 2*75€ +
4*25€, which means the rather cheap devices, which expectedly will
nevertheless look better than the wireless USB things. And then, get a
PC and 4 wireless sound devices for 250 bucks...
But I do not want to lack answering your question: I know for sure that
some softphones allow to select a certain sound channel / sound
controller. Take the Linux softphone ekiga as example. If you run
several of those with different configuration files and on different
port numbers (this will most probably be possible, although it might
turn out a nightmare to configure), that might get the job done.
I do not know wether there is a softphone that uses different sound
channels for different lines, but I doubt it - it would be rather
counterintuitive to have a single phone software, but several
handsets; rather several instances would fulfill the multiple phone on
one desk computer screen analogy.
You might also have some trouble with the keys on your wireless phones
in a multiple-softphone scenario - depending on how the OS handles
those, they might be handed to the window in focus. I have plainly no
idea how this could possibly work with a phone hardware - softphone
mapping without royally screwing up.
Also, is it possible to transfer a call directly to someone’s VM(if
they are out of the office) bypassing their extension? If so, could
someone post the asterisk logic behind the extension setup? I don’t
want anything too complex(like setting the DND or phone to busy).
I want to describe a scenario, and you can decide wether that is too
complex ;-)
Let us assume your asterisk has two internal number plan ranges
available for the project, being 23XX and 4XXX. Let us further assume
that all the ATAs live in the 23XX range and will be called out of the
context [internal], like
[internal]
exten = 2300,1,Dial(SIP/device2300,60)
exten = 2300,2,Hangup()
exten = 2301,1,Dial(SIP/device2301,60)
...
(or, if your devices are named reasonably in sip.conf, you might get
away with)
exten = _23XX,1,Dial(SIP/device${EXTEN},60)
exten = _23XX,2,Hangup()
So those numbers end up calling a specific DECT phone, but you would not
know which nurse to reach on which phone, unless she told you beforehand
that she just picked up phone 56 resulting in phone number 2356.
To get around that, every nurse gets assigned a personal number from the
4XXX range that will follow her or go to voicemail. You could make use
of the Asterisk Database, like this:
[internal]
exten = _4XXX,1,Set(CURRENTPHONE=${DB(nurse/${EXTEN})})
exten = _4XXX,2,GotoIf($[${CURRENTPHONE:1} = ]?4)
exten = _4XXX,3,Dial(SIP/device${CURRENTPHONE},60)
exten = _4XXX,4,VoiceMail(${EXTEN})
exten = _4XXX,5,Hangup
So if the nurse is not logged in the call will go to voicemail
immediately.
Instead of calling the receptionist Hi Linda, I'm on phone 56 today
she would keep her 4113 for all times.
The reason I chose two-digit DECT phone numbers and three-digit nurse
numbers is that there are usually more nurses than phones :-) Anyway
a somehow competent receptionist would be able to deal with a static
personell number list better than dynamic phone numbers changing twice
every day.
Of course the nurse would need to tell the phone system where she
currently is, like by picking a phone and dialling her own code number,
plus *1 (provided CALLERID is working correctly) - or her own number
plus *0 to log off. Mind, you could also have an IVR available (on 777
or whatever internal number suits you) that greets the caller, asks for
the nurse's number and her PIN and wether she is coming or going.
[internal]
exten = _4XXX*1,1,GotoIf($[${CALLERID(num):0:2} = 23]?2:100)
exten = _4XXX*1,2,Set(DB(nurse/${EXTEN:0:4})=${CALLERID(num)})
exten = _4XXX*1,3,Playback(nurse-registered-thank-you)
exten = _4XXX*1,4,Hangup
exten = _4XXX*1,100,Playback(not-possible-from-this-phone)
exten = _4XXX*1,101,Hangup
exten = _4XXX*0,1,Set(DB(nurse/${EXTEN:0:4})=0)
exten = _4XXX*0,2,Playback(thanks-have-a-nice-time-we-will-miss-you)
Re: [asterisk-users] improved SMS?
Am Donnerstag, den 12.07.2007, 16:57 -0700 schrieb Russ McBride: Newbie question(s): From what I can determine it sounds like the SMS messaging isn't as robust as it could be (?). I'm wondering if there's active work on that right now or if it's more of an issue about PSTN carrier that one would be using who would be responsible for passing the messages into the PLMN. Background-- I'm looking into the possibility of setting up an emergency messaging system here at the University that would send out voice, SMS, and emails. Any input relevant to that goal would probably be appreciated. Hi Russ, my personal experience with short messages is that the system sometimes chews on them for minutes, sometimes several hours, even inside one mobile network, from cell phone to cell phone. This surely screws using it as a primary tier emergency system, but as a backup after e-mail and automated phone-out that could be OK. Sending from web-interfaces or via Uwhatever-that-protocol-is-called will not improve the overall performance. Considering all options to send out SMs: - Asterisk, SMS() app to a landline SMS gateway - Web interface with script/wget - Uwhatever-modem-dialup the second seems the easiest to use to me, and in my experience the first tends to choke on some messages, be it 1 in 100 - still not 100% perfect. The web interface method surely is by far cheaper than the other two, at least here in Germany, where #1 will be charged as a call-to-cellphone, first minute, about 17 cent, and #3 if available for the network you want to use will be similar. With the web interface approach you also get rid of the problem of number portability: The #3 approach will only deliver the message if you connect to the provider that the number currently is contracted to, while #2 will not care about that (#1 should also work). You see I tend to prefer the web-based thing. If you intend to send emergeny SMs, please try and find a trustworthy supplier. In the European price scale the cheapest readily found providers will charge about 3 cent per message, but if you go for the 10 cent providers you will find higher reliability (without routing messages to Germany through a Romanian mobile network to save money). Those messages directly inserted into the destination network are sold as provider messages here, opposed to cheapest or economy or whatever euphemism for crap they invent. Do not mistake me though: For fun messages they use to be good enough. If you talk about emergency, a few cent probably will not make a huge difference though, and time might be an issue. BR Anselm ___ --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] Problems sending more than 2 SMS with asterisk / smsq
Am Montag, den 09.07.2007, 17:21 +0200 schrieb Matthias Huber: When i send more than one messages shortly after the other, my log (/var/spool/asterisk/sms ) looks like this and only two of four messages arrive. What am i doing wrong ? I am using an AVM B1 PCI with chan-capi and 1.4.4. and also, when sending with smsq -x only two of the messages are handled. (i thought, asterisk itself handles the queues ? ) Here the log: 2007-07-09T15:04:14 YOM04 0 - 0172xxx test11 2007-07-09T15:04:15 ?OM05 0 - 0172xxx test12 2007-07-09T15:07:51 YOM06 0 - 0172xxx test13 2007-07-09T15:07:53 ?OM07 0 - 0172xxx test14 sorry - i am a total newbie at asterisk. My experience with sending several subsequent short messages is that this might run you into a timing issue. Whyever, some calls will not successfully transmit the first two packets of the SMS handshake, resulting in a non-delivery. This can be seen on the CLI, so perhaps your problem shows up there as well: Try asterisk -r CLI set verbose 10 (keep CLI open) and send those messages. I would expect those failing messages to show a different pattern. I got this failing probability _way_ down by using an additional Wait(1) or Wait(2) in the dialplan where the SMS sending happens, after bringing up the line and before sending the SMS proper. HTH Anselm ___ --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] awful list delays: 4 days!
Am Freitag, den 29.06.2007, 14:23 -0600 schrieb Anthony Francis: Andres Paglayan wrote: On Jun 29, 2007, at 12:50 PM, Lenz wrote: Hello list, I am getting the list with days of delay, take for example this message: As you can see, the message was posted on June 25th and was sent to my email on June 29th! am I the only one who is getting such an awful message turn-around time? l. I'll let you know next week, ;^) ROFL, yeah its you. I see posts within a few hours. This one just arrived here. From the mail headers: Delivery-date: Tue, 10 Jul 2007 08:21:49 +0200 Received: from lists.digium.com ([216.207.245.17]) by server2.hoffmeister.be with esmtps (TLS-1.0:RSA_AES_256_CBC_SHA1:32) (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I896X-0001I0-U1 for anselm (a)hoffmeister-online.de; Tue, 10 Jul 2007 08:21:49 +0200 Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I4N0F-0002TB-Hz; Fri, 29 Jun 2007 15:23:39 -0500 Received: from exprod8mx3.postini.com ([64.18.3.103] helo=psmtp.com) by lists.digium.com with smtp (Exim 4.63) (envelope-from anthonyf (a)rockynet.com) id 1I4N01-0002SD-UZ for asterisk-users@lists.digium.com; Fri, 29 Jun 2007 15:23:26 -0500 So it seems to be trouble between lists.digium.com and my mailserver. Judging from what I know about other people's trouble with mail delays, probably the earlier. This becomes rather unnerving, as a regular discussion cannot take place. 11 days delays is just incredible (but some messages take only 5 days ;-/ ) Perhaps someone at the server management team knows something about all this, I have forwarded this mail over there. Thanks for input how to get around this. I do not assume it is a problem on my part, but if it is, I would like to know. Anselm ___ --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] List delays
Am Mittwoch, den 04.07.2007, 11:00 -0400 schrieb Noah Miller: Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev list seems fine! I'm getting new messages within a matter of minutes. I dunno. As this topic is mentioned, I have similar problems. I often get messages in the wrong order, like this one: I got the reply (from Noah), but the original message will probably arrive tomorrow or so, if at all. No, it is not my spamassassin eating those - that will be invoked _after_ asterisk-users mails already sorted into the proper folder for the rest of incoming mails. This is extremely annoying when discussion thread view stops working. With a volume like that of asterisk-users, discussion threading is a feature worth using, but it breaks when the original message comes long after the reply to it. For some reasons, two mails I sent seem not to have gone through. Or will do so some day now... BR Anselm ___ --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
