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] 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] 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] 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 ;tag=2612560371 > To: ;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] 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] 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=105596&view=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
[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] 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
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/sip01&SIP/sip02,60) exten => 004311234567,2,Hangup() [otherprovider-in] exten => 12,1,Dial(SIP/sip01&SIP/sip02,60) exten => 12,2,Hangup() >8 This should get
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] 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 ${CALLERID(name)}. Some hardware phones (I assume, the better ones ;-) display both; my Allnet for example seems to only display the name, but store the number for the "call back" list. My Fritz!Boxen seem to forward both name and number to ISDN devices on the internal S0-bus, just not many ISDN phones can actually display text "numbers". Let your asterisk have an ast database, looking like callerid/420123456789 => "Doe, John Q." callerid/492240224922 => "Mustermann, Dr. Peter" Then you could expand your dialplan logic a little. If you have a line exten => 12345,4,Dial(SIP/phone1,60) or whatever that looks like in your SIP-incoming context, insert those lines before it [and change the "4", "5", "6", "7"s ;-) ] exten => 12345,4,Set(CALLERID(name)=${DB(callerid/${CALLERID(num)})}) exten => 12345,5,GotoIf($["${CALLERID(name)}" = ""]?6:7) exten => 12345,6,Set(CALLERID(name)="-- ${CALLERID(num)}") exten => 12345,7,Dial(SIP/phone1,60) Line 6 treats the case that the number is not in your database and sets the callerid-name to "-- NUMBER_OF_CALLER" You can manually add data to the astdb from the asterisk CLI with database set callerid 420456789 "Silly, Roger M." You should check that both your SIP providers provide incoming CLI in the international formatting, without country prefix or "+". In my experience some SIP providers send numbers like 492240224922, others send +49... or 0049..., some send national format 02240... for all national calls, some even omit the leading "0" there, and some just change the behaviour depending from which network (T-Com landline, Arcor landline, T-Mobile cell phone, O2 cellphone, foreign callers...) the call originates. If you have more than two providers, this can be a PITA - you will need some dialplan logic to sanitize the callerid in those cases, and sometimes you are just left for guessing, for example when the provider signals calls from T-Mobile as 16177554224 and calls from Boston, MA, USA the very same. Germany does not have fixed-length numbers, even in the mobile phone networks the length differs, and the number given might be valid for both circumstances. > * The Ringtone is different for special call numbers If your phone supports that, yes, you can do it. The common method for this seems to be sending an additional header. There will be docs on "SIPAddHeader(blah)" or similar on www.voip-info.org, and you might want to also use a database here to find out wether special ringtones are to be activated or not. > * it is displayed, in which sipgate the call came from You could use the CALLERID(name) field for that, by adding the provider short name in front of the caller's name, like exten => 12345,4,Set(CALLERID(name)=at-${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
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/officinademarta&SIP/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] 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] 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=9&id=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/sip501&SIP/sip501&SIP/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] 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] 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] 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] 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] 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] 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] 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] 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] 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 ;-) 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) 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] 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] 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] 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] 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] [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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] Asterisk registering problem
Am Samstag, den 02.06.2007, 11:34 +0200 schrieb [EMAIL PROTECTED]: > Hi, > > Problem is: > I have a Dell 1950 server with 6 NIC's ( 1 for Voice / Asterisk rest of > them for other functions). > > The Voice LAN is on the 172.16.3.0 (255.255.0.0) subnet. One the other > NICS there are different but also Class B like 172.15.1.x and so on. > No problem at all i think. Nothing to do with Asterisk in special, but you should be aware that the range 172.16.0.0/16 to 172.31.0.0/16 (that means: 16 networks of 172.16.x.x ~ 16k addresses each) is reserved for "private use", so you use them correctly. AFAIK 172.15. is a regular network that might be assigned to someone like Google for use as webserver ip addresses. Without good reason (or it being assigned to you) you should NOT use it. Besides, giving 172.16.3.0 (255.255.0.0) is pointless, the "3" is in the "host number" part of the address. Perhaps you should check the subnet mask on your server as well as in dhcpd.conf! > In sip.conf bindaddres is 0.0.0.0 And spelt corrreclty, I suppose ;-) You can find out which processes listen on which IP addresses by using the netstat command, for example # netstat -lnp on my Linux box will list the SMTP server listening only on one address, but the SSH server listening on 0.0.0.0 port 22. Sorry for the comments not being specific to Asterisk. I notice lots of people get confused by networks that are not Class-C (aka 255.255.255.0 or /24), so I wanted to point that out. Best regards, Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending multiline SMS
Am Donnerstag, den 07.06.2007, 01:15 +0200 schrieb Patrick Zwahlen: > Hi everyone, > > How do you send multiline SMSs using smsq or .call files ? > > smsq --motx-channel="mISDN/g:bri/" 078 "line1 line2" > > How can I have a carriage return between line1 and line2 ? I have tried > the regular \n and \r. No success. > > Thanks for any help. - Patrick - I am not sure about sending texts on command-line (did not work for me either, although it probably should), but if you write those to a file instead, it will work: echo -en "This is a\nmultiline message\n\nCool, eh?" >/tmp/sms1 smsq --mt --oa=12345 --mttx-callerid=01930101 --mttx-channel="SIP/sip503" --ud-file="/tmp/sms1" did the trick for me. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing the Caller ID
Am Dienstag, den 12.06.2007, 09:57 -0400 schrieb Shad Mortazavi: > Dear Group, > > I have a scenario where I would like to change the caller ID based on > the number dialled; > > For example; > > ;Outbound UK and London Calls > exten=>_8.,1,Set(CALLERIDNAME=0207100) > exten=>_8.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r) > > exten=>_80039.,1,Set(CALLERIDNAME=0039024070) > exten=>_8.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r) > > exten=>_80034.,1,Set(CALLERIDNAME=003491187) > exten=>_8.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r) > > exten=>_80049.,1,Set(CALLERIDNAME=0049891214) > exten=>_8.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r) > > So for anything outside the Madrid, Milan, Munich extensions I would > like to use the generic UK number or have a pieced of logic that goes; > > exten=>_800.,1,Set(CALLERIDNAME=440207100) > exten=>_800.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r) > > Unfortunately this does not work, every time I dial I get > CALLERIDNAME=0207100. > > What am I missing? a/ Your provider needs to support setting other CALLERID values than the numbers he assigned to you, and even allow for international numbers. As an example, sipgate.de allows to set any german number as CALLERID, but no international phone numbers will work. Providers often suppress wrongly set CALLERIDs, replacing with the "default" CALLERID assigned to the phone line / account. b/ Besides 440207... probably is wrong, should be 44207... c/ You have several _8.,2, lines (identical), one is enough. And then, I do not know wether that pattern matching will work correctly ;-) BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX client USB phone
Am Samstag, den 23.06.2007, 09:52 -0300 schrieb Ronaldo Z. Afonso: > Hi all, > > Does anybody know any USB phone that I can use as an IAX Client? The USB "phones" I saw on the market just behave like an additional sound card, with some control buttons perhaps, and those will not work without a software (like twinkle, x-lite...). The devices that work without PC software usually come with a network plug instead of USB - the point is that they work without PC, so why use a USB connection and depend of a PC in that respect? In my experience (and I only had two cheapo Ebay USB-"phones") sound works without problems _but_ the keys might work or not, seems to be a bit of luck involved. One of those two phones had the number keys working, but neither "hangup" nor "dial" would do anything, the other one would not work at all - except the sound, which was OK. 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] callback and bridge problem
Am Donnerstag, den 28.06.2007, 07:07 +0200 schrieb Adam KOSA: > Hi guys, > > sorry for the long e-mail, i'm only trying to give as much information > as i think is relevant to my problem (console log, sip.conf and > extension.conf parts). I've sent this e-mail a couple of days ago, but > it bounced back today. > > i've been practicing with callback for a while, but i'm at a dead end. > I hope somebody can help me to move on. > > i have troubles getting two calls bridged together. Scenario is the > following: Just a short hint (and a wild guess). Try keeping Asterisk in the voice path, by using Dial with the appropriate parameters (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial), like "H" - the necessary DTMF detection hinders Asterisk from bridging the calls externally. In my experience SIP providers do not like having two calls from the same IP address bridged to each other, because that somehow smells of "smartass customers" ;-) As long as Asterisk stays in the voice path, no-one external will know that both calls are in fact one. 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] 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
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 ) 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] 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] 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
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] 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] 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] 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. CounterPath's X-Lite 3.0 is the market's leading free SIP based softphone available for download. . The first link in the google search list for "phoner" immediately led me to the phoner homepage, - 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. 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] 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] 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] 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] 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-514&SIP/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] 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] 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] 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] 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] 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] 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 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] 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, 1&1 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] 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] 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] 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+fax&btnI=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] 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=First<180> > 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=Second<181> > 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] 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] 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] wip5000 crash AP
Am Montag, den 27.11.2006, 16:41 +0200 schrieb Altus Snyman: > Good day all > They all connect to the 4 Senao Long range AP’s 11mb > > They all have the same ssi but 2 runs on channel 11 and 2 on channel 1 > BUT..for some reason each now and the the AP’s will crash, you can > find a signal when you scan, and you can ping it, the only way to get > it back up is to pull the power in and out This seems to me as not directly Asterisk related... I do not know the Senao devices, but I had several network devices (switches, access points, media converters, routers,...) chicken out from time to time, and it basically fell back on a single problem - heat. (Or the device operating system having broken down - D-Link wireless devices are good at that - firmware updates helped) Check wether the device operates in a place without air flow - and change that situation if possible. Something notably above 40°C/100°F will cause trouble in the long run. If you want a proper scientific analysis, try hammering the APs with half a dozen laptops sending files to and fro, and no phones in use at the same time. I guess it will cause the same problem: Once the CPU in that device is stressed, it runs at full wattage, and eventually all those Watts will be dissipated as heat. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] registration ip address
Am Dienstag, den 28.11.2006, 06:46 -0800 schrieb Khaled: > What is the variable like $peerip to get the registered ip address > for a peer You can use ${DB(SIP/Registry/sip507)} where "sip507" is the section name as well as username from my sip.conf- no idea which of both to use, try it out. This value contains the client IP, port number, and more information. Consider CUT() for obtaining the IP, something like exten=>123,45,Set(IPADDR=${CUT(${DB(SIP/Registry/sip504)},:,1)}) BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cmd Record doesn't resume Dialplan if phone Hangs-Up.
Am Dienstag, den 28.11.2006, 17:23 +0200 schrieb Jean-Marc Salsa: > Hi, > > I have tried to use the Record Command in Asterisk, > > Here is the configuration : > exten => record,1,Answer > ... > exten => > record,n,Record(/var/spool/asterisk/record/${CALLFILENAME}:WAV) > exten => record,n,Playback(vm-goodbye) > exten => record,n,system(/usr/local/bin/send-recording.pl --to > ${EMAILADDR} --file /var/spool/asterisk/record/${CALLFILENAME}.WAV) > exten => record,n,Hangup > > If I hung up the phone during recording, > then Message is well there, but Asterisk does not continue its way to > the system command to send me the file. > > If I change the record command to detect a 2 sec silence. > Then, Asterisk hangs up correctly and DO send the file ... > > Has anyone noticed something similar ? > Did I do something wrong ? > Can somebody help me ? For a first workaround, use the hash key "#" to stop recording; afterwards, asterisk will continue in the dialplan. The moment that you hangup your phone, your dialplan will be aborted. You might as well see wether the "h" extension could help. Read the fine manuals at http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension and http://www.voip-info.org/wiki/view/Asterisk+standard+extensions for more information ;) Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] direct IP calling with extension
Am Freitag, den 01.12.2006, 13:44 -0500 schrieb Jerry Geis: > All, > > If I have video phones behind an asterisk server (with 2 network cards) > and all the phones have extensions. Internally everything works great. > > Now for people that want to call my video phones external to my office > is there a way to do that? On the extenal persons phone enter an IP/EXTEN > where IP is my server and not the phone? Can that work? > > Would I have to have PUBLIC IP address for every phone NAT'ed through my > server to make the call? AFAIR, define a "context" in the global section of sip.conf. Any incoming SIP connections that are not identified to belong to any other context (registration) will come thru that extensions.conf context. Inside, just "forward" through to your proper local extensions: [locals] exten => 200,1,Dial(SIP/myphone1) ... [sipfromoutside] exten => johndoe,1,Goto(locals,200,1) ... Then direct IP calling as [EMAIL PROTECTED] should work. To get calling at a hostname working as well, you will need a few records in your DNS setup. example.com. IN A 123.45.67.89 sip.example.com. IN A 123.45.67.89 example.com. IN NAPTR 60 50 "s" "SIP+D2U" "" _sip.udp.example.com. _sip._udp.example.com. IN SRV 10 10 5060 sip.example.com. Which will allow for [EMAIL PROTECTED] I suspect the first "example.com" line is not necessary (such that you can host your domain on a different server than that which runs Asterisk), but I did not test. HTH Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID Rewrite
Am Freitag, den 01.12.2006, 14:26 + schrieb David Bath: > So onto the problem… I’m trying to write a quick on-liner which will > fix up incoming UK format numbers (0 + 10digits) into (44 + 9 digits). > I got as far as this: > > > > exten => ,1,Set(foo=${IF(REGEX("^0[1-9][0-9]\{9 > \}$" {CALLERID(number)})?Set(CALLERID(number)=44 > ${CALLERID(number):1})}) I would try something like exten => _0[1-9]X.,1,Goto(0044${EXTEN:1},1) (All numbers beginning "null" "not-null" will be rewritten to 0044 plus the number without the leading zero) Hth Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Caller ID Rewrite
Am Freitag, den 01.12.2006, 20:03 + schrieb David Bath: > Hi Anselm, > > Thanks for the help... > > I'm slightly confused as to your response. > > Wouldn't that look for a /dialled/ number in the format _0 try > and jump to another extension 0044 with priority 1? > > If so, that's not what I'm trying to achieve. Sorry, my brain is in need for a weekend off work. I obviously understood your question wrong. My fault. > I've made a bit more progress... and my current diaplan entry looks like > this: > > exten=>123456,1,Set(${IF(REGEX("^0[1-9][0-9]\{9\}$" > {CALLERID(number)})?CALLERID(number)=44${CALLERID(number):1})}) > exten => 123456,2,Goto(sipinternal,101,1) > > BUT! There's a very odd problem (and I'm sure it's my fault..) the > second callerid function is not being evaluated... I _think_ the IF is a string evaluation, so the format should be like SET MYVARIABLE = [IF condition? value1 : value2] (see http://www.voip-info.org/wiki/index.php?page=Asterisk+func+if ) Try exten=>123456,1,Set(CALLERID(number)= ${IF(REGEX("^0[1-9][0-9]\{9\}$" ${CALLERID(number)})? 44${CALLERID(number):1}:${CALLERID(number)})}) (two linebreaks to be removed) HTH, Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] direct IP calling with extension
Am Freitag, den 01.12.2006, 15:27 -0500 schrieb Jerry Geis: > >Am Freitag, den 01.12.2006, 13:44 -0500 schrieb Jerry Geis: > THanks, this seems to almost get me there... Once I call into the server > and goes to my locals I no longer get Video. > > When I call the extension directly I get video no problem. > > When I first call the server at my IP address then it routes to my local > I no longer get video. > > Any ideas why that might be? I suspect that is caused by asterisk not knowing about your video codecs (I have never done video over IP myself, so just a guess). If your devices talk to each other, # device <=> device they can use all the codecs both of them know As soon as asterisk is involved, # device <=> asterisk <=> device the codec negotiation goes through asterisk. So asterisk must at least _know_ the codec that those devices want to use. I think I saw some info about that in the www.voip-wiki.org BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Caller ID Rewrite
Am Freitag, den 01.12.2006, 20:41 + schrieb David Bath: > Hi, > > Thanks for quick response. > > I changed it as you suggested, but it has the same effect: > > In the console I get: > > --Executing > Set("SIP/604625-b79140a8",CALLERID(number)=44${CALLERID(number)}") in > new stack > > It's running the IF code correctly, but in the "true" it's just not > evaluating the variable... Well, perhaps the IF hinders evaluation from happening? It is by far not as elegant, but you could try exten=>123456,1,GotoIf($[${REGEX("^0..)} = 1]?2:3) exten=>123456,2,Set(CALLERID(num)=44${CALLERID(num):1}) exten=>123456,3,ContinueYourDialplanHere Btw. it should be CALLERID(num), not CALLERID(number), right? BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users